summaryrefslogtreecommitdiffstats
path: root/kernel/include/sound
diff options
context:
space:
mode:
authorYunhong Jiang <yunhong.jiang@intel.com>2015-08-04 12:17:53 -0700
committerYunhong Jiang <yunhong.jiang@intel.com>2015-08-04 15:44:42 -0700
commit9ca8dbcc65cfc63d6f5ef3312a33184e1d726e00 (patch)
tree1c9cafbcd35f783a87880a10f85d1a060db1a563 /kernel/include/sound
parent98260f3884f4a202f9ca5eabed40b1354c489b29 (diff)
Add the rt linux 4.1.3-rt3 as base
Import the rt linux 4.1.3-rt3 as OPNFV kvm base. It's from git://git.kernel.org/pub/scm/linux/kernel/git/rt/linux-rt-devel.git linux-4.1.y-rt and the base is: commit 0917f823c59692d751951bf5ea699a2d1e2f26a2 Author: Sebastian Andrzej Siewior <bigeasy@linutronix.de> Date: Sat Jul 25 12:13:34 2015 +0200 Prepare v4.1.3-rt3 Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de> We lose all the git history this way and it's not good. We should apply another opnfv project repo in future. Change-Id: I87543d81c9df70d99c5001fbdf646b202c19f423 Signed-off-by: Yunhong Jiang <yunhong.jiang@intel.com>
Diffstat (limited to 'kernel/include/sound')
-rw-r--r--kernel/include/sound/ac97_codec.h664
-rw-r--r--kernel/include/sound/aci.h90
-rw-r--r--kernel/include/sound/ad1816a.h181
-rw-r--r--kernel/include/sound/ad1843.h46
-rw-r--r--kernel/include/sound/adau1373.h34
-rw-r--r--kernel/include/sound/aess.h53
-rw-r--r--kernel/include/sound/ak4113.h330
-rw-r--r--kernel/include/sound/ak4114.h212
-rw-r--r--kernel/include/sound/ak4117.h189
-rw-r--r--kernel/include/sound/ak4531_codec.h85
-rw-r--r--kernel/include/sound/ak4641.h26
-rw-r--r--kernel/include/sound/ak4xxx-adda.h99
-rw-r--r--kernel/include/sound/alc5623.h15
-rw-r--r--kernel/include/sound/asequencer.h86
-rw-r--r--kernel/include/sound/asound.h40
-rw-r--r--kernel/include/sound/asoundef.h325
-rw-r--r--kernel/include/sound/atmel-abdac.h23
-rw-r--r--kernel/include/sound/atmel-ac97c.h38
-rw-r--r--kernel/include/sound/compress_driver.h185
-rw-r--r--kernel/include/sound/control.h259
-rw-r--r--kernel/include/sound/core.h463
-rw-r--r--kernel/include/sound/cs4231-regs.h187
-rw-r--r--kernel/include/sound/cs4271.h40
-rw-r--r--kernel/include/sound/cs42l52.h32
-rw-r--r--kernel/include/sound/cs42l56.h48
-rw-r--r--kernel/include/sound/cs42l73.h22
-rw-r--r--kernel/include/sound/cs8403.h257
-rw-r--r--kernel/include/sound/cs8427.h202
-rw-r--r--kernel/include/sound/da7213.h52
-rw-r--r--kernel/include/sound/da9055.h33
-rw-r--r--kernel/include/sound/designware_i2s.h69
-rw-r--r--kernel/include/sound/dmaengine_pcm.h157
-rw-r--r--kernel/include/sound/emu10k1.h1906
-rw-r--r--kernel/include/sound/emu10k1_synth.h39
-rw-r--r--kernel/include/sound/emu8000.h121
-rw-r--r--kernel/include/sound/emu8000_reg.h207
-rw-r--r--kernel/include/sound/emux_legacy.h146
-rw-r--r--kernel/include/sound/emux_synth.h244
-rw-r--r--kernel/include/sound/es1688.h122
-rw-r--r--kernel/include/sound/gus.h631
-rw-r--r--kernel/include/sound/hda_hwdep.h44
-rw-r--r--kernel/include/sound/hda_regmap.h219
-rw-r--r--kernel/include/sound/hda_verbs.h554
-rw-r--r--kernel/include/sound/hdaudio.h247
-rw-r--r--kernel/include/sound/hwdep.h82
-rw-r--r--kernel/include/sound/i2c.h104
-rw-r--r--kernel/include/sound/info.h202
-rw-r--r--kernel/include/sound/initval.h104
-rw-r--r--kernel/include/sound/jack.h121
-rw-r--r--kernel/include/sound/l3.h18
-rw-r--r--kernel/include/sound/max9768.h24
-rw-r--r--kernel/include/sound/max98088.h50
-rw-r--r--kernel/include/sound/max98090.h29
-rw-r--r--kernel/include/sound/max98095.h66
-rw-r--r--kernel/include/sound/memalloc.h157
-rw-r--r--kernel/include/sound/minors.h112
-rw-r--r--kernel/include/sound/mixer_oss.h81
-rw-r--r--kernel/include/sound/mpu401.h138
-rw-r--r--kernel/include/sound/omap-hdmi-audio.h43
-rw-r--r--kernel/include/sound/omap-pcm.h30
-rw-r--r--kernel/include/sound/opl3.h393
-rw-r--r--kernel/include/sound/opl4.h32
-rw-r--r--kernel/include/sound/pcm-indirect.h177
-rw-r--r--kernel/include/sound/pcm.h1397
-rw-r--r--kernel/include/sound/pcm_oss.h89
-rw-r--r--kernel/include/sound/pcm_params.h376
-rw-r--r--kernel/include/sound/pt2258.h37
-rw-r--r--kernel/include/sound/pxa2xx-lib.h38
-rw-r--r--kernel/include/sound/rawmidi.h189
-rw-r--r--kernel/include/sound/rcar_snd.h104
-rw-r--r--kernel/include/sound/rt286.h19
-rw-r--r--kernel/include/sound/rt5640.h26
-rw-r--r--kernel/include/sound/rt5645.h32
-rw-r--r--kernel/include/sound/rt5651.h21
-rw-r--r--kernel/include/sound/rt5670.h28
-rw-r--r--kernel/include/sound/rt5677.h45
-rw-r--r--kernel/include/sound/s3c24xx_uda134x.h14
-rw-r--r--kernel/include/sound/sb.h375
-rw-r--r--kernel/include/sound/sb16_csp.h90
-rw-r--r--kernel/include/sound/seq_device.h96
-rw-r--r--kernel/include/sound/seq_kernel.h109
-rw-r--r--kernel/include/sound/seq_midi_emul.h197
-rw-r--r--kernel/include/sound/seq_midi_event.h54
-rw-r--r--kernel/include/sound/seq_oss.h96
-rw-r--r--kernel/include/sound/seq_oss_legacy.h31
-rw-r--r--kernel/include/sound/seq_virmidi.h81
-rw-r--r--kernel/include/sound/sh_dac_audio.h21
-rw-r--r--kernel/include/sound/sh_fsi.h35
-rw-r--r--kernel/include/sound/simple_card.h36
-rw-r--r--kernel/include/sound/snd_wavefront.h142
-rw-r--r--kernel/include/sound/soc-dai.h316
-rw-r--r--kernel/include/sound/soc-dapm.h626
-rw-r--r--kernel/include/sound/soc-dpcm.h162
-rw-r--r--kernel/include/sound/soc.h1562
-rw-r--r--kernel/include/sound/soundfont.h129
-rw-r--r--kernel/include/sound/spear_dma.h34
-rw-r--r--kernel/include/sound/spear_spdif.h29
-rw-r--r--kernel/include/sound/sta32x.h43
-rw-r--r--kernel/include/sound/sta350.h57
-rw-r--r--kernel/include/sound/tas2552-plat.h25
-rw-r--r--kernel/include/sound/tas5086.h7
-rw-r--r--kernel/include/sound/tea6330t.h31
-rw-r--r--kernel/include/sound/timer.h143
-rw-r--r--kernel/include/sound/tlv.h101
-rw-r--r--kernel/include/sound/tlv320aic32x4.h32
-rw-r--r--kernel/include/sound/tlv320aic3x.h68
-rw-r--r--kernel/include/sound/tlv320dac33-plat.h24
-rw-r--r--kernel/include/sound/tpa6130a2-plat.h30
-rw-r--r--kernel/include/sound/uda134x.h27
-rw-r--r--kernel/include/sound/uda1380.h22
-rw-r--r--kernel/include/sound/util_mem.h64
-rw-r--r--kernel/include/sound/vx_core.h548
-rw-r--r--kernel/include/sound/wavefront.h695
-rw-r--r--kernel/include/sound/wm0010.h27
-rw-r--r--kernel/include/sound/wm1250-ev1.h27
-rw-r--r--kernel/include/sound/wm2000.h23
-rw-r--r--kernel/include/sound/wm2200.h61
-rw-r--r--kernel/include/sound/wm5100.h59
-rw-r--r--kernel/include/sound/wm8903.h266
-rw-r--r--kernel/include/sound/wm8904.h163
-rw-r--r--kernel/include/sound/wm8955.h26
-rw-r--r--kernel/include/sound/wm8960.h24
-rw-r--r--kernel/include/sound/wm8962.h61
-rw-r--r--kernel/include/sound/wm8993.h48
-rw-r--r--kernel/include/sound/wm8996.h55
-rw-r--r--kernel/include/sound/wm9081.h28
-rw-r--r--kernel/include/sound/wm9090.h28
-rw-r--r--kernel/include/sound/wss.h235
128 files changed, 20669 insertions, 0 deletions
diff --git a/kernel/include/sound/ac97_codec.h b/kernel/include/sound/ac97_codec.h
new file mode 100644
index 000000000..0e9d75b49
--- /dev/null
+++ b/kernel/include/sound/ac97_codec.h
@@ -0,0 +1,664 @@
+#ifndef __SOUND_AC97_CODEC_H
+#define __SOUND_AC97_CODEC_H
+
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.1
+ * by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/bitops.h>
+#include <linux/device.h>
+#include <linux/workqueue.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/info.h>
+
+/* maximum number of devices on the AC97 bus */
+#define AC97_BUS_MAX_DEVICES 4
+
+/*
+ * AC'97 codec registers
+ */
+
+#define AC97_RESET 0x00 /* Reset */
+#define AC97_MASTER 0x02 /* Master Volume */
+#define AC97_HEADPHONE 0x04 /* Headphone Volume (optional) */
+#define AC97_MASTER_MONO 0x06 /* Master Volume Mono (optional) */
+#define AC97_MASTER_TONE 0x08 /* Master Tone (Bass & Treble) (optional) */
+#define AC97_PC_BEEP 0x0a /* PC Beep Volume (optinal) */
+#define AC97_PHONE 0x0c /* Phone Volume (optional) */
+#define AC97_MIC 0x0e /* MIC Volume */
+#define AC97_LINE 0x10 /* Line In Volume */
+#define AC97_CD 0x12 /* CD Volume */
+#define AC97_VIDEO 0x14 /* Video Volume (optional) */
+#define AC97_AUX 0x16 /* AUX Volume (optional) */
+#define AC97_PCM 0x18 /* PCM Volume */
+#define AC97_REC_SEL 0x1a /* Record Select */
+#define AC97_REC_GAIN 0x1c /* Record Gain */
+#define AC97_REC_GAIN_MIC 0x1e /* Record Gain MIC (optional) */
+#define AC97_GENERAL_PURPOSE 0x20 /* General Purpose (optional) */
+#define AC97_3D_CONTROL 0x22 /* 3D Control (optional) */
+#define AC97_INT_PAGING 0x24 /* Audio Interrupt & Paging (AC'97 2.3) */
+#define AC97_POWERDOWN 0x26 /* Powerdown control / status */
+/* range 0x28-0x3a - AUDIO AC'97 2.0 extensions */
+#define AC97_EXTENDED_ID 0x28 /* Extended Audio ID */
+#define AC97_EXTENDED_STATUS 0x2a /* Extended Audio Status and Control */
+#define AC97_PCM_FRONT_DAC_RATE 0x2c /* PCM Front DAC Rate */
+#define AC97_PCM_SURR_DAC_RATE 0x2e /* PCM Surround DAC Rate */
+#define AC97_PCM_LFE_DAC_RATE 0x30 /* PCM LFE DAC Rate */
+#define AC97_PCM_LR_ADC_RATE 0x32 /* PCM LR ADC Rate */
+#define AC97_PCM_MIC_ADC_RATE 0x34 /* PCM MIC ADC Rate */
+#define AC97_CENTER_LFE_MASTER 0x36 /* Center + LFE Master Volume */
+#define AC97_SURROUND_MASTER 0x38 /* Surround (Rear) Master Volume */
+#define AC97_SPDIF 0x3a /* S/PDIF control */
+/* range 0x3c-0x58 - MODEM */
+#define AC97_EXTENDED_MID 0x3c /* Extended Modem ID */
+#define AC97_EXTENDED_MSTATUS 0x3e /* Extended Modem Status and Control */
+#define AC97_LINE1_RATE 0x40 /* Line1 DAC/ADC Rate */
+#define AC97_LINE2_RATE 0x42 /* Line2 DAC/ADC Rate */
+#define AC97_HANDSET_RATE 0x44 /* Handset DAC/ADC Rate */
+#define AC97_LINE1_LEVEL 0x46 /* Line1 DAC/ADC Level */
+#define AC97_LINE2_LEVEL 0x48 /* Line2 DAC/ADC Level */
+#define AC97_HANDSET_LEVEL 0x4a /* Handset DAC/ADC Level */
+#define AC97_GPIO_CFG 0x4c /* GPIO Configuration */
+#define AC97_GPIO_POLARITY 0x4e /* GPIO Pin Polarity/Type, 0=low, 1=high active */
+#define AC97_GPIO_STICKY 0x50 /* GPIO Pin Sticky, 0=not, 1=sticky */
+#define AC97_GPIO_WAKEUP 0x52 /* GPIO Pin Wakeup, 0=no int, 1=yes int */
+#define AC97_GPIO_STATUS 0x54 /* GPIO Pin Status, slot 12 */
+#define AC97_MISC_AFE 0x56 /* Miscellaneous Modem AFE Status and Control */
+/* range 0x5a-0x7b - Vendor Specific */
+#define AC97_VENDOR_ID1 0x7c /* Vendor ID1 */
+#define AC97_VENDOR_ID2 0x7e /* Vendor ID2 / revision */
+/* range 0x60-0x6f (page 1) - extended codec registers */
+#define AC97_CODEC_CLASS_REV 0x60 /* Codec Class/Revision */
+#define AC97_PCI_SVID 0x62 /* PCI Subsystem Vendor ID */
+#define AC97_PCI_SID 0x64 /* PCI Subsystem ID */
+#define AC97_FUNC_SELECT 0x66 /* Function Select */
+#define AC97_FUNC_INFO 0x68 /* Function Information */
+#define AC97_SENSE_INFO 0x6a /* Sense Details */
+
+/* volume controls */
+#define AC97_MUTE_MASK_MONO 0x8000
+#define AC97_MUTE_MASK_STEREO 0x8080
+
+/* slot allocation */
+#define AC97_SLOT_TAG 0
+#define AC97_SLOT_CMD_ADDR 1
+#define AC97_SLOT_CMD_DATA 2
+#define AC97_SLOT_PCM_LEFT 3
+#define AC97_SLOT_PCM_RIGHT 4
+#define AC97_SLOT_MODEM_LINE1 5
+#define AC97_SLOT_PCM_CENTER 6
+#define AC97_SLOT_MIC 6 /* input */
+#define AC97_SLOT_SPDIF_LEFT1 6
+#define AC97_SLOT_PCM_SLEFT 7 /* surround left */
+#define AC97_SLOT_PCM_LEFT_0 7 /* double rate operation */
+#define AC97_SLOT_SPDIF_LEFT 7
+#define AC97_SLOT_PCM_SRIGHT 8 /* surround right */
+#define AC97_SLOT_PCM_RIGHT_0 8 /* double rate operation */
+#define AC97_SLOT_SPDIF_RIGHT 8
+#define AC97_SLOT_LFE 9
+#define AC97_SLOT_SPDIF_RIGHT1 9
+#define AC97_SLOT_MODEM_LINE2 10
+#define AC97_SLOT_PCM_LEFT_1 10 /* double rate operation */
+#define AC97_SLOT_SPDIF_LEFT2 10
+#define AC97_SLOT_HANDSET 11 /* output */
+#define AC97_SLOT_PCM_RIGHT_1 11 /* double rate operation */
+#define AC97_SLOT_SPDIF_RIGHT2 11
+#define AC97_SLOT_MODEM_GPIO 12 /* modem GPIO */
+#define AC97_SLOT_PCM_CENTER_1 12 /* double rate operation */
+
+/* basic capabilities (reset register) */
+#define AC97_BC_DEDICATED_MIC 0x0001 /* Dedicated Mic PCM In Channel */
+#define AC97_BC_RESERVED1 0x0002 /* Reserved (was Modem Line Codec support) */
+#define AC97_BC_BASS_TREBLE 0x0004 /* Bass & Treble Control */
+#define AC97_BC_SIM_STEREO 0x0008 /* Simulated stereo */
+#define AC97_BC_HEADPHONE 0x0010 /* Headphone Out Support */
+#define AC97_BC_LOUDNESS 0x0020 /* Loudness (bass boost) Support */
+#define AC97_BC_16BIT_DAC 0x0000 /* 16-bit DAC resolution */
+#define AC97_BC_18BIT_DAC 0x0040 /* 18-bit DAC resolution */
+#define AC97_BC_20BIT_DAC 0x0080 /* 20-bit DAC resolution */
+#define AC97_BC_DAC_MASK 0x00c0
+#define AC97_BC_16BIT_ADC 0x0000 /* 16-bit ADC resolution */
+#define AC97_BC_18BIT_ADC 0x0100 /* 18-bit ADC resolution */
+#define AC97_BC_20BIT_ADC 0x0200 /* 20-bit ADC resolution */
+#define AC97_BC_ADC_MASK 0x0300
+#define AC97_BC_3D_TECH_ID_MASK 0x7c00 /* Per-vendor ID of 3D enhancement */
+
+/* general purpose */
+#define AC97_GP_DRSS_MASK 0x0c00 /* double rate slot select */
+#define AC97_GP_DRSS_1011 0x0000 /* LR(C) 10+11(+12) */
+#define AC97_GP_DRSS_78 0x0400 /* LR 7+8 */
+
+/* powerdown bits */
+#define AC97_PD_ADC_STATUS 0x0001 /* ADC status (RO) */
+#define AC97_PD_DAC_STATUS 0x0002 /* DAC status (RO) */
+#define AC97_PD_MIXER_STATUS 0x0004 /* Analog mixer status (RO) */
+#define AC97_PD_VREF_STATUS 0x0008 /* Vref status (RO) */
+#define AC97_PD_PR0 0x0100 /* Power down PCM ADCs and input MUX */
+#define AC97_PD_PR1 0x0200 /* Power down PCM front DAC */
+#define AC97_PD_PR2 0x0400 /* Power down Mixer (Vref still on) */
+#define AC97_PD_PR3 0x0800 /* Power down Mixer (Vref off) */
+#define AC97_PD_PR4 0x1000 /* Power down AC-Link */
+#define AC97_PD_PR5 0x2000 /* Disable internal clock usage */
+#define AC97_PD_PR6 0x4000 /* Headphone amplifier */
+#define AC97_PD_EAPD 0x8000 /* External Amplifer Power Down (EAPD) */
+
+/* extended audio ID bit defines */
+#define AC97_EI_VRA 0x0001 /* Variable bit rate supported */
+#define AC97_EI_DRA 0x0002 /* Double rate supported */
+#define AC97_EI_SPDIF 0x0004 /* S/PDIF out supported */
+#define AC97_EI_VRM 0x0008 /* Variable bit rate supported for MIC */
+#define AC97_EI_DACS_SLOT_MASK 0x0030 /* DACs slot assignment */
+#define AC97_EI_DACS_SLOT_SHIFT 4
+#define AC97_EI_CDAC 0x0040 /* PCM Center DAC available */
+#define AC97_EI_SDAC 0x0080 /* PCM Surround DACs available */
+#define AC97_EI_LDAC 0x0100 /* PCM LFE DAC available */
+#define AC97_EI_AMAP 0x0200 /* indicates optional slot/DAC mapping based on codec ID */
+#define AC97_EI_REV_MASK 0x0c00 /* AC'97 revision mask */
+#define AC97_EI_REV_22 0x0400 /* AC'97 revision 2.2 */
+#define AC97_EI_REV_23 0x0800 /* AC'97 revision 2.3 */
+#define AC97_EI_REV_SHIFT 10
+#define AC97_EI_ADDR_MASK 0xc000 /* physical codec ID (address) */
+#define AC97_EI_ADDR_SHIFT 14
+
+/* extended audio status and control bit defines */
+#define AC97_EA_VRA 0x0001 /* Variable bit rate enable bit */
+#define AC97_EA_DRA 0x0002 /* Double-rate audio enable bit */
+#define AC97_EA_SPDIF 0x0004 /* S/PDIF out enable bit */
+#define AC97_EA_VRM 0x0008 /* Variable bit rate for MIC enable bit */
+#define AC97_EA_SPSA_SLOT_MASK 0x0030 /* Mask for slot assignment bits */
+#define AC97_EA_SPSA_SLOT_SHIFT 4
+#define AC97_EA_SPSA_3_4 0x0000 /* Slot assigned to 3 & 4 */
+#define AC97_EA_SPSA_7_8 0x0010 /* Slot assigned to 7 & 8 */
+#define AC97_EA_SPSA_6_9 0x0020 /* Slot assigned to 6 & 9 */
+#define AC97_EA_SPSA_10_11 0x0030 /* Slot assigned to 10 & 11 */
+#define AC97_EA_CDAC 0x0040 /* PCM Center DAC is ready (Read only) */
+#define AC97_EA_SDAC 0x0080 /* PCM Surround DACs are ready (Read only) */
+#define AC97_EA_LDAC 0x0100 /* PCM LFE DAC is ready (Read only) */
+#define AC97_EA_MDAC 0x0200 /* MIC ADC is ready (Read only) */
+#define AC97_EA_SPCV 0x0400 /* S/PDIF configuration valid (Read only) */
+#define AC97_EA_PRI 0x0800 /* Turns the PCM Center DAC off */
+#define AC97_EA_PRJ 0x1000 /* Turns the PCM Surround DACs off */
+#define AC97_EA_PRK 0x2000 /* Turns the PCM LFE DAC off */
+#define AC97_EA_PRL 0x4000 /* Turns the MIC ADC off */
+
+/* S/PDIF control bit defines */
+#define AC97_SC_PRO 0x0001 /* Professional status */
+#define AC97_SC_NAUDIO 0x0002 /* Non audio stream */
+#define AC97_SC_COPY 0x0004 /* Copyright status */
+#define AC97_SC_PRE 0x0008 /* Preemphasis status */
+#define AC97_SC_CC_MASK 0x07f0 /* Category Code mask */
+#define AC97_SC_CC_SHIFT 4
+#define AC97_SC_L 0x0800 /* Generation Level status */
+#define AC97_SC_SPSR_MASK 0x3000 /* S/PDIF Sample Rate bits */
+#define AC97_SC_SPSR_SHIFT 12
+#define AC97_SC_SPSR_44K 0x0000 /* Use 44.1kHz Sample rate */
+#define AC97_SC_SPSR_48K 0x2000 /* Use 48kHz Sample rate */
+#define AC97_SC_SPSR_32K 0x3000 /* Use 32kHz Sample rate */
+#define AC97_SC_DRS 0x4000 /* Double Rate S/PDIF */
+#define AC97_SC_V 0x8000 /* Validity status */
+
+/* Interrupt and Paging bit defines (AC'97 2.3) */
+#define AC97_PAGE_MASK 0x000f /* Page Selector */
+#define AC97_PAGE_VENDOR 0 /* Vendor-specific registers */
+#define AC97_PAGE_1 1 /* Extended Codec Registers page 1 */
+#define AC97_INT_ENABLE 0x0800 /* Interrupt Enable */
+#define AC97_INT_SENSE 0x1000 /* Sense Cycle */
+#define AC97_INT_CAUSE_SENSE 0x2000 /* Sense Cycle Completed (RO) */
+#define AC97_INT_CAUSE_GPIO 0x4000 /* GPIO bits changed (RO) */
+#define AC97_INT_STATUS 0x8000 /* Interrupt Status */
+
+/* extended modem ID bit defines */
+#define AC97_MEI_LINE1 0x0001 /* Line1 present */
+#define AC97_MEI_LINE2 0x0002 /* Line2 present */
+#define AC97_MEI_HANDSET 0x0004 /* Handset present */
+#define AC97_MEI_CID1 0x0008 /* caller ID decode for Line1 is supported */
+#define AC97_MEI_CID2 0x0010 /* caller ID decode for Line2 is supported */
+#define AC97_MEI_ADDR_MASK 0xc000 /* physical codec ID (address) */
+#define AC97_MEI_ADDR_SHIFT 14
+
+/* extended modem status and control bit defines */
+#define AC97_MEA_GPIO 0x0001 /* GPIO is ready (ro) */
+#define AC97_MEA_MREF 0x0002 /* Vref is up to nominal level (ro) */
+#define AC97_MEA_ADC1 0x0004 /* ADC1 operational (ro) */
+#define AC97_MEA_DAC1 0x0008 /* DAC1 operational (ro) */
+#define AC97_MEA_ADC2 0x0010 /* ADC2 operational (ro) */
+#define AC97_MEA_DAC2 0x0020 /* DAC2 operational (ro) */
+#define AC97_MEA_HADC 0x0040 /* HADC operational (ro) */
+#define AC97_MEA_HDAC 0x0080 /* HDAC operational (ro) */
+#define AC97_MEA_PRA 0x0100 /* GPIO power down (high) */
+#define AC97_MEA_PRB 0x0200 /* reserved */
+#define AC97_MEA_PRC 0x0400 /* ADC1 power down (high) */
+#define AC97_MEA_PRD 0x0800 /* DAC1 power down (high) */
+#define AC97_MEA_PRE 0x1000 /* ADC2 power down (high) */
+#define AC97_MEA_PRF 0x2000 /* DAC2 power down (high) */
+#define AC97_MEA_PRG 0x4000 /* HADC power down (high) */
+#define AC97_MEA_PRH 0x8000 /* HDAC power down (high) */
+
+/* modem gpio status defines */
+#define AC97_GPIO_LINE1_OH 0x0001 /* Off Hook Line1 */
+#define AC97_GPIO_LINE1_RI 0x0002 /* Ring Detect Line1 */
+#define AC97_GPIO_LINE1_CID 0x0004 /* Caller ID path enable Line1 */
+#define AC97_GPIO_LINE1_LCS 0x0008 /* Loop Current Sense Line1 */
+#define AC97_GPIO_LINE1_PULSE 0x0010 /* Opt./ Pulse Dial Line1 (out) */
+#define AC97_GPIO_LINE1_HL1R 0x0020 /* Opt./ Handset to Line1 relay control (out) */
+#define AC97_GPIO_LINE1_HOHD 0x0040 /* Opt./ Handset off hook detect Line1 (in) */
+#define AC97_GPIO_LINE12_AC 0x0080 /* Opt./ Int.bit 1 / Line1/2 AC (out) */
+#define AC97_GPIO_LINE12_DC 0x0100 /* Opt./ Int.bit 2 / Line1/2 DC (out) */
+#define AC97_GPIO_LINE12_RS 0x0200 /* Opt./ Int.bit 3 / Line1/2 RS (out) */
+#define AC97_GPIO_LINE2_OH 0x0400 /* Off Hook Line2 */
+#define AC97_GPIO_LINE2_RI 0x0800 /* Ring Detect Line2 */
+#define AC97_GPIO_LINE2_CID 0x1000 /* Caller ID path enable Line2 */
+#define AC97_GPIO_LINE2_LCS 0x2000 /* Loop Current Sense Line2 */
+#define AC97_GPIO_LINE2_PULSE 0x4000 /* Opt./ Pulse Dial Line2 (out) */
+#define AC97_GPIO_LINE2_HL1R 0x8000 /* Opt./ Handset to Line2 relay control (out) */
+
+/* specific - SigmaTel */
+#define AC97_SIGMATEL_OUTSEL 0x64 /* Output Select, STAC9758 */
+#define AC97_SIGMATEL_INSEL 0x66 /* Input Select, STAC9758 */
+#define AC97_SIGMATEL_IOMISC 0x68 /* STAC9758 */
+#define AC97_SIGMATEL_ANALOG 0x6c /* Analog Special */
+#define AC97_SIGMATEL_DAC2INVERT 0x6e
+#define AC97_SIGMATEL_BIAS1 0x70
+#define AC97_SIGMATEL_BIAS2 0x72
+#define AC97_SIGMATEL_VARIOUS 0x72 /* STAC9758 */
+#define AC97_SIGMATEL_MULTICHN 0x74 /* Multi-Channel programming */
+#define AC97_SIGMATEL_CIC1 0x76
+#define AC97_SIGMATEL_CIC2 0x78
+
+/* specific - Analog Devices */
+#define AC97_AD_TEST 0x5a /* test register */
+#define AC97_AD_TEST2 0x5c /* undocumented test register 2 */
+#define AC97_AD_HPFD_SHIFT 12 /* High Pass Filter Disable */
+#define AC97_AD_CODEC_CFG 0x70 /* codec configuration */
+#define AC97_AD_JACK_SPDIF 0x72 /* Jack Sense & S/PDIF */
+#define AC97_AD_SERIAL_CFG 0x74 /* Serial Configuration */
+#define AC97_AD_MISC 0x76 /* Misc Control Bits */
+#define AC97_AD_VREFD_SHIFT 2 /* V_REFOUT Disable (AD1888) */
+
+/* specific - Cirrus Logic */
+#define AC97_CSR_ACMODE 0x5e /* AC Mode Register */
+#define AC97_CSR_MISC_CRYSTAL 0x60 /* Misc Crystal Control */
+#define AC97_CSR_SPDIF 0x68 /* S/PDIF Register */
+#define AC97_CSR_SERIAL 0x6a /* Serial Port Control */
+#define AC97_CSR_SPECF_ADDR 0x6c /* Special Feature Address */
+#define AC97_CSR_SPECF_DATA 0x6e /* Special Feature Data */
+#define AC97_CSR_BDI_STATUS 0x7a /* BDI Status */
+
+/* specific - Conexant */
+#define AC97_CXR_AUDIO_MISC 0x5c
+#define AC97_CXR_SPDIFEN (1<<3)
+#define AC97_CXR_COPYRGT (1<<2)
+#define AC97_CXR_SPDIF_MASK (3<<0)
+#define AC97_CXR_SPDIF_PCM 0x0
+#define AC97_CXR_SPDIF_AC3 0x2
+
+/* specific - ALC */
+#define AC97_ALC650_SPDIF_INPUT_STATUS1 0x60
+/* S/PDIF input status 1 bit defines */
+#define AC97_ALC650_PRO 0x0001 /* Professional status */
+#define AC97_ALC650_NAUDIO 0x0002 /* Non audio stream */
+#define AC97_ALC650_COPY 0x0004 /* Copyright status */
+#define AC97_ALC650_PRE 0x0038 /* Preemphasis status */
+#define AC97_ALC650_PRE_SHIFT 3
+#define AC97_ALC650_MODE 0x00C0 /* Preemphasis status */
+#define AC97_ALC650_MODE_SHIFT 6
+#define AC97_ALC650_CC_MASK 0x7f00 /* Category Code mask */
+#define AC97_ALC650_CC_SHIFT 8
+#define AC97_ALC650_L 0x8000 /* Generation Level status */
+
+#define AC97_ALC650_SPDIF_INPUT_STATUS2 0x62
+/* S/PDIF input status 2 bit defines */
+#define AC97_ALC650_SOUCE_MASK 0x000f /* Source number */
+#define AC97_ALC650_CHANNEL_MASK 0x00f0 /* Channel number */
+#define AC97_ALC650_CHANNEL_SHIFT 4
+#define AC97_ALC650_SPSR_MASK 0x0f00 /* S/PDIF Sample Rate bits */
+#define AC97_ALC650_SPSR_SHIFT 8
+#define AC97_ALC650_SPSR_44K 0x0000 /* Use 44.1kHz Sample rate */
+#define AC97_ALC650_SPSR_48K 0x0200 /* Use 48kHz Sample rate */
+#define AC97_ALC650_SPSR_32K 0x0300 /* Use 32kHz Sample rate */
+#define AC97_ALC650_CLOCK_ACCURACY 0x3000 /* Clock accuracy */
+#define AC97_ALC650_CLOCK_SHIFT 12
+#define AC97_ALC650_CLOCK_LOCK 0x4000 /* Clock locked status */
+#define AC97_ALC650_V 0x8000 /* Validity status */
+
+#define AC97_ALC650_SURR_DAC_VOL 0x64
+#define AC97_ALC650_LFE_DAC_VOL 0x66
+#define AC97_ALC650_UNKNOWN1 0x68
+#define AC97_ALC650_MULTICH 0x6a
+#define AC97_ALC650_UNKNOWN2 0x6c
+#define AC97_ALC650_REVISION 0x6e
+#define AC97_ALC650_UNKNOWN3 0x70
+#define AC97_ALC650_UNKNOWN4 0x72
+#define AC97_ALC650_MISC 0x74
+#define AC97_ALC650_GPIO_SETUP 0x76
+#define AC97_ALC650_GPIO_STATUS 0x78
+#define AC97_ALC650_CLOCK 0x7a
+
+/* specific - Yamaha YMF7x3 */
+#define AC97_YMF7X3_DIT_CTRL 0x66 /* DIT Control (YMF743) / 2 (YMF753) */
+#define AC97_YMF7X3_3D_MODE_SEL 0x68 /* 3D Mode Select */
+
+/* specific - C-Media */
+#define AC97_CM9738_VENDOR_CTRL 0x5a
+#define AC97_CM9739_MULTI_CHAN 0x64
+#define AC97_CM9739_SPDIF_IN_STATUS 0x68 /* 32bit */
+#define AC97_CM9739_SPDIF_CTRL 0x6c
+
+/* specific - wolfson */
+#define AC97_WM97XX_FMIXER_VOL 0x72
+#define AC97_WM9704_RMIXER_VOL 0x74
+#define AC97_WM9704_TEST 0x5a
+#define AC97_WM9704_RPCM_VOL 0x70
+#define AC97_WM9711_OUT3VOL 0x16
+
+
+/* ac97->scaps */
+#define AC97_SCAP_AUDIO (1<<0) /* audio codec 97 */
+#define AC97_SCAP_MODEM (1<<1) /* modem codec 97 */
+#define AC97_SCAP_SURROUND_DAC (1<<2) /* surround L&R DACs are present */
+#define AC97_SCAP_CENTER_LFE_DAC (1<<3) /* center and LFE DACs are present */
+#define AC97_SCAP_SKIP_AUDIO (1<<4) /* skip audio part of codec */
+#define AC97_SCAP_SKIP_MODEM (1<<5) /* skip modem part of codec */
+#define AC97_SCAP_INDEP_SDIN (1<<6) /* independent SDIN */
+#define AC97_SCAP_INV_EAPD (1<<7) /* inverted EAPD */
+#define AC97_SCAP_DETECT_BY_VENDOR (1<<8) /* use vendor registers for read tests */
+#define AC97_SCAP_NO_SPDIF (1<<9) /* don't build SPDIF controls */
+#define AC97_SCAP_EAPD_LED (1<<10) /* EAPD as mute LED */
+#define AC97_SCAP_POWER_SAVE (1<<11) /* capable for aggressive power-saving */
+
+/* ac97->flags */
+#define AC97_HAS_PC_BEEP (1<<0) /* force PC Speaker usage */
+#define AC97_AD_MULTI (1<<1) /* Analog Devices - multi codecs */
+#define AC97_CS_SPDIF (1<<2) /* Cirrus Logic uses funky SPDIF */
+#define AC97_CX_SPDIF (1<<3) /* Conexant's spdif interface */
+#define AC97_STEREO_MUTES (1<<4) /* has stereo mute bits */
+#define AC97_DOUBLE_RATE (1<<5) /* supports double rate playback */
+#define AC97_HAS_NO_MASTER_VOL (1<<6) /* no Master volume */
+#define AC97_HAS_NO_PCM_VOL (1<<7) /* no PCM volume */
+#define AC97_DEFAULT_POWER_OFF (1<<8) /* no RESET write */
+#define AC97_MODEM_PATCH (1<<9) /* modem patch */
+#define AC97_HAS_NO_REC_GAIN (1<<10) /* no Record gain */
+#define AC97_HAS_NO_PHONE (1<<11) /* no PHONE volume */
+#define AC97_HAS_NO_PC_BEEP (1<<12) /* no PC Beep volume */
+#define AC97_HAS_NO_VIDEO (1<<13) /* no Video volume */
+#define AC97_HAS_NO_CD (1<<14) /* no CD volume */
+#define AC97_HAS_NO_MIC (1<<15) /* no MIC volume */
+#define AC97_HAS_NO_TONE (1<<16) /* no Tone volume */
+#define AC97_HAS_NO_STD_PCM (1<<17) /* no standard AC97 PCM volume and mute */
+#define AC97_HAS_NO_AUX (1<<18) /* no standard AC97 AUX volume and mute */
+#define AC97_HAS_8CH (1<<19) /* supports 8-channel output */
+
+/* rates indexes */
+#define AC97_RATES_FRONT_DAC 0
+#define AC97_RATES_SURR_DAC 1
+#define AC97_RATES_LFE_DAC 2
+#define AC97_RATES_ADC 3
+#define AC97_RATES_MIC_ADC 4
+#define AC97_RATES_SPDIF 5
+
+/*
+ *
+ */
+
+struct snd_ac97;
+struct snd_pcm_chmap;
+
+struct snd_ac97_build_ops {
+ int (*build_3d) (struct snd_ac97 *ac97);
+ int (*build_specific) (struct snd_ac97 *ac97);
+ int (*build_spdif) (struct snd_ac97 *ac97);
+ int (*build_post_spdif) (struct snd_ac97 *ac97);
+#ifdef CONFIG_PM
+ void (*suspend) (struct snd_ac97 *ac97);
+ void (*resume) (struct snd_ac97 *ac97);
+#endif
+ void (*update_jacks) (struct snd_ac97 *ac97); /* for jack-sharing */
+};
+
+struct snd_ac97_bus_ops {
+ void (*reset) (struct snd_ac97 *ac97);
+ void (*warm_reset)(struct snd_ac97 *ac97);
+ void (*write) (struct snd_ac97 *ac97, unsigned short reg, unsigned short val);
+ unsigned short (*read) (struct snd_ac97 *ac97, unsigned short reg);
+ void (*wait) (struct snd_ac97 *ac97);
+ void (*init) (struct snd_ac97 *ac97);
+};
+
+struct snd_ac97_bus {
+ /* -- lowlevel (hardware) driver specific -- */
+ struct snd_ac97_bus_ops *ops;
+ void *private_data;
+ void (*private_free) (struct snd_ac97_bus *bus);
+ /* --- */
+ struct snd_card *card;
+ unsigned short num; /* bus number */
+ unsigned short no_vra: 1, /* bridge doesn't support VRA */
+ dra: 1, /* bridge supports double rate */
+ isdin: 1;/* independent SDIN */
+ unsigned int clock; /* AC'97 base clock (usually 48000Hz) */
+ spinlock_t bus_lock; /* used mainly for slot allocation */
+ unsigned short used_slots[2][4]; /* actually used PCM slots */
+ unsigned short pcms_count; /* count of PCMs */
+ struct ac97_pcm *pcms;
+ struct snd_ac97 *codec[4];
+ struct snd_info_entry *proc;
+};
+
+/* static resolution table */
+struct snd_ac97_res_table {
+ unsigned short reg; /* register */
+ unsigned short bits; /* resolution bitmask */
+};
+
+struct snd_ac97_template {
+ void *private_data;
+ void (*private_free) (struct snd_ac97 *ac97);
+ struct pci_dev *pci; /* assigned PCI device - used for quirks */
+ unsigned short num; /* number of codec: 0 = primary, 1 = secondary */
+ unsigned short addr; /* physical address of codec [0-3] */
+ unsigned int scaps; /* driver capabilities */
+ const struct snd_ac97_res_table *res_table; /* static resolution */
+};
+
+struct snd_ac97 {
+ /* -- lowlevel (hardware) driver specific -- */
+ const struct snd_ac97_build_ops *build_ops;
+ void *private_data;
+ void (*private_free) (struct snd_ac97 *ac97);
+ /* --- */
+ struct snd_ac97_bus *bus;
+ struct pci_dev *pci; /* assigned PCI device - used for quirks */
+ struct snd_info_entry *proc;
+ struct snd_info_entry *proc_regs;
+ unsigned short subsystem_vendor;
+ unsigned short subsystem_device;
+ struct mutex reg_mutex;
+ struct mutex page_mutex; /* mutex for AD18xx multi-codecs and paging (2.3) */
+ unsigned short num; /* number of codec: 0 = primary, 1 = secondary */
+ unsigned short addr; /* physical address of codec [0-3] */
+ unsigned int id; /* identification of codec */
+ unsigned short caps; /* capabilities (register 0) */
+ unsigned short ext_id; /* extended feature identification (register 28) */
+ unsigned short ext_mid; /* extended modem ID (register 3C) */
+ const struct snd_ac97_res_table *res_table; /* static resolution */
+ unsigned int scaps; /* driver capabilities */
+ unsigned int flags; /* specific code */
+ unsigned int rates[6]; /* see AC97_RATES_* defines */
+ unsigned int spdif_status;
+ unsigned short regs[0x80]; /* register cache */
+ DECLARE_BITMAP(reg_accessed, 0x80); /* bit flags */
+ union { /* vendor specific code */
+ struct {
+ unsigned short unchained[3]; // 0 = C34, 1 = C79, 2 = C69
+ unsigned short chained[3]; // 0 = C34, 1 = C79, 2 = C69
+ unsigned short id[3]; // codec IDs (lower 16-bit word)
+ unsigned short pcmreg[3]; // PCM registers
+ unsigned short codec_cfg[3]; // CODEC_CFG bits
+ unsigned char swap_mic_linein; // AD1986/AD1986A only
+ unsigned char lo_as_master; /* LO as master */
+ } ad18xx;
+ unsigned int dev_flags; /* device specific */
+ } spec;
+ /* jack-sharing info */
+ unsigned char indep_surround;
+ unsigned char channel_mode;
+
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ unsigned int power_up; /* power states */
+ struct delayed_work power_work;
+#endif
+ struct device dev;
+
+ struct snd_pcm_chmap *chmaps[2]; /* channel-maps (optional) */
+};
+
+#define to_ac97_t(d) container_of(d, struct snd_ac97, dev)
+
+/* conditions */
+static inline int ac97_is_audio(struct snd_ac97 * ac97)
+{
+ return (ac97->scaps & AC97_SCAP_AUDIO);
+}
+static inline int ac97_is_modem(struct snd_ac97 * ac97)
+{
+ return (ac97->scaps & AC97_SCAP_MODEM);
+}
+static inline int ac97_is_rev22(struct snd_ac97 * ac97)
+{
+ return (ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_22;
+}
+static inline int ac97_can_amap(struct snd_ac97 * ac97)
+{
+ return (ac97->ext_id & AC97_EI_AMAP) != 0;
+}
+static inline int ac97_can_spdif(struct snd_ac97 * ac97)
+{
+ return (ac97->ext_id & AC97_EI_SPDIF) != 0;
+}
+
+/* functions */
+/* create new AC97 bus */
+int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops,
+ void *private_data, struct snd_ac97_bus **rbus);
+/* create mixer controls */
+int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
+ struct snd_ac97 **rac97);
+const char *snd_ac97_get_short_name(struct snd_ac97 *ac97);
+
+void snd_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
+unsigned short snd_ac97_read(struct snd_ac97 *ac97, unsigned short reg);
+void snd_ac97_write_cache(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
+int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
+int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup);
+#else
+static inline int snd_ac97_update_power(struct snd_ac97 *ac97, int reg,
+ int powerup)
+{
+ return 0;
+}
+#endif
+#ifdef CONFIG_PM
+void snd_ac97_suspend(struct snd_ac97 *ac97);
+void snd_ac97_resume(struct snd_ac97 *ac97);
+#endif
+
+/* quirk types */
+enum {
+ AC97_TUNE_DEFAULT = -1, /* use default from quirk list (not valid in list) */
+ AC97_TUNE_NONE = 0, /* nothing extra to do */
+ AC97_TUNE_HP_ONLY, /* headphone (true line-out) control as master only */
+ AC97_TUNE_SWAP_HP, /* swap headphone and master controls */
+ AC97_TUNE_SWAP_SURROUND, /* swap master and surround controls */
+ AC97_TUNE_AD_SHARING, /* for AD1985, turn on OMS bit and use headphone */
+ AC97_TUNE_ALC_JACK, /* for Realtek, enable JACK detection */
+ AC97_TUNE_INV_EAPD, /* inverted EAPD implementation */
+ AC97_TUNE_MUTE_LED, /* EAPD bit works as mute LED */
+ AC97_TUNE_HP_MUTE_LED, /* EAPD bit works as mute LED, use headphone control as master */
+};
+
+struct ac97_quirk {
+ unsigned short subvendor; /* PCI subsystem vendor id */
+ unsigned short subdevice; /* PCI subsystem device id */
+ unsigned short mask; /* device id bit mask, 0 = accept all */
+ unsigned int codec_id; /* codec id (if any), 0 = accept all */
+ const char *name; /* name shown as info */
+ int type; /* quirk type above */
+};
+
+int snd_ac97_tune_hardware(struct snd_ac97 *ac97,
+ const struct ac97_quirk *quirk,
+ const char *override);
+int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate);
+
+/*
+ * PCM allocation
+ */
+
+enum ac97_pcm_cfg {
+ AC97_PCM_CFG_FRONT = 2,
+ AC97_PCM_CFG_REAR = 10, /* alias surround */
+ AC97_PCM_CFG_LFE = 11, /* center + lfe */
+ AC97_PCM_CFG_40 = 4, /* front + rear */
+ AC97_PCM_CFG_51 = 6, /* front + rear + center/lfe */
+ AC97_PCM_CFG_SPDIF = 20
+};
+
+struct ac97_pcm {
+ struct snd_ac97_bus *bus;
+ unsigned int stream: 1, /* stream type: 1 = capture */
+ exclusive: 1, /* exclusive mode, don't override with other pcms */
+ copy_flag: 1, /* lowlevel driver must fill all entries */
+ spdif: 1; /* spdif pcm */
+ unsigned short aslots; /* active slots */
+ unsigned short cur_dbl; /* current double-rate state */
+ unsigned int rates; /* available rates */
+ struct {
+ unsigned short slots; /* driver input: requested AC97 slot numbers */
+ unsigned short rslots[4]; /* allocated slots per codecs */
+ unsigned char rate_table[4];
+ struct snd_ac97 *codec[4]; /* allocated codecs */
+ } r[2]; /* 0 = standard rates, 1 = double rates */
+ unsigned long private_value; /* used by the hardware driver */
+};
+
+int snd_ac97_pcm_assign(struct snd_ac97_bus *ac97,
+ unsigned short pcms_count,
+ const struct ac97_pcm *pcms);
+int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate,
+ enum ac97_pcm_cfg cfg, unsigned short slots);
+int snd_ac97_pcm_close(struct ac97_pcm *pcm);
+int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime);
+
+/* ad hoc AC97 device driver access */
+extern struct bus_type ac97_bus_type;
+
+/* AC97 platform_data adding function */
+static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data)
+{
+ ac97->dev.platform_data = data;
+}
+
+#endif /* __SOUND_AC97_CODEC_H */
diff --git a/kernel/include/sound/aci.h b/kernel/include/sound/aci.h
new file mode 100644
index 000000000..ee639d355
--- /dev/null
+++ b/kernel/include/sound/aci.h
@@ -0,0 +1,90 @@
+#ifndef _ACI_H_
+#define _ACI_H_
+
+#define ACI_REG_COMMAND 0 /* write register offset */
+#define ACI_REG_STATUS 1 /* read register offset */
+#define ACI_REG_BUSY 2 /* busy register offset */
+#define ACI_REG_RDS 2 /* PCM20: RDS register offset */
+#define ACI_MINTIME 500 /* ACI time out limit */
+
+#define ACI_SET_MUTE 0x0d
+#define ACI_SET_POWERAMP 0x0f
+#define ACI_SET_TUNERMUTE 0xa3
+#define ACI_SET_TUNERMONO 0xa4
+#define ACI_SET_IDE 0xd0
+#define ACI_SET_WSS 0xd1
+#define ACI_SET_SOLOMODE 0xd2
+#define ACI_SET_PREAMP 0x03
+#define ACI_GET_PREAMP 0x21
+#define ACI_WRITE_TUNE 0xa7
+#define ACI_READ_TUNERSTEREO 0xa8
+#define ACI_READ_TUNERSTATION 0xa9
+#define ACI_READ_VERSION 0xf1
+#define ACI_READ_IDCODE 0xf2
+#define ACI_INIT 0xff
+#define ACI_STATUS 0xf0
+#define ACI_S_GENERAL 0x00
+#define ACI_ERROR_OP 0xdf
+
+/* ACI Mixer */
+
+/* These are the values for the right channel GET registers.
+ Add an offset of 0x01 for the left channel register.
+ (left=right+0x01) */
+
+#define ACI_GET_MASTER 0x03
+#define ACI_GET_MIC 0x05
+#define ACI_GET_LINE 0x07
+#define ACI_GET_CD 0x09
+#define ACI_GET_SYNTH 0x0b
+#define ACI_GET_PCM 0x0d
+#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */
+#define ACI_GET_LINE2 0x12
+
+#define ACI_GET_EQ1 0x22 /* from Bass ... */
+#define ACI_GET_EQ2 0x24
+#define ACI_GET_EQ3 0x26
+#define ACI_GET_EQ4 0x28
+#define ACI_GET_EQ5 0x2a
+#define ACI_GET_EQ6 0x2c
+#define ACI_GET_EQ7 0x2e /* ... to Treble */
+
+/* And these are the values for the right channel SET registers.
+ For left channel access you have to add an offset of 0x08.
+ MASTER is an exception, which needs an offset of 0x01 */
+
+#define ACI_SET_MASTER 0x00
+#define ACI_SET_MIC 0x30
+#define ACI_SET_LINE 0x31
+#define ACI_SET_CD 0x34
+#define ACI_SET_SYNTH 0x33
+#define ACI_SET_PCM 0x32
+#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */
+#define ACI_SET_LINE2 0x36
+
+#define ACI_SET_EQ1 0x40 /* from Bass ... */
+#define ACI_SET_EQ2 0x41
+#define ACI_SET_EQ3 0x42
+#define ACI_SET_EQ4 0x43
+#define ACI_SET_EQ5 0x44
+#define ACI_SET_EQ6 0x45
+#define ACI_SET_EQ7 0x46 /* ... to Treble */
+
+struct snd_miro_aci {
+ unsigned long aci_port;
+ int aci_vendor;
+ int aci_product;
+ int aci_version;
+ int aci_amp;
+ int aci_preamp;
+ int aci_solomode;
+
+ struct mutex aci_mutex;
+};
+
+int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3);
+
+struct snd_miro_aci *snd_aci_get_aci(void);
+
+#endif /* _ACI_H_ */
+
diff --git a/kernel/include/sound/ad1816a.h b/kernel/include/sound/ad1816a.h
new file mode 100644
index 000000000..f2d3a6d07
--- /dev/null
+++ b/kernel/include/sound/ad1816a.h
@@ -0,0 +1,181 @@
+#ifndef __SOUND_AD1816A_H
+#define __SOUND_AD1816A_H
+
+/*
+ ad1816a.h - definitions for ADI SoundPort AD1816A chip.
+ Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio@libero.it>
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/timer.h>
+
+#define AD1816A_REG(r) (chip->port + r)
+
+#define AD1816A_CHIP_STATUS 0x00
+#define AD1816A_INDIR_ADDR 0x00
+#define AD1816A_INTERRUPT_STATUS 0x01
+#define AD1816A_INDIR_DATA_LOW 0x02
+#define AD1816A_INDIR_DATA_HIGH 0x03
+#define AD1816A_PIO_DEBUG 0x04
+#define AD1816A_PIO_STATUS 0x05
+#define AD1816A_PIO_DATA 0x06
+#define AD1816A_RESERVED_7 0x07
+#define AD1816A_PLAYBACK_CONFIG 0x08
+#define AD1816A_CAPTURE_CONFIG 0x09
+#define AD1816A_RESERVED_10 0x0a
+#define AD1816A_RESERVED_11 0x0b
+#define AD1816A_JOYSTICK_RAW_DATA 0x0c
+#define AD1816A_JOYSTICK_CTRL 0x0d
+#define AD1816A_JOY_POS_DATA_LOW 0x0e
+#define AD1816A_JOY_POS_DATA_HIGH 0x0f
+
+#define AD1816A_LOW_BYTE_TMP 0x00
+#define AD1816A_INTERRUPT_ENABLE 0x01
+#define AD1816A_EXTERNAL_CTRL 0x01
+#define AD1816A_PLAYBACK_SAMPLE_RATE 0x02
+#define AD1816A_CAPTURE_SAMPLE_RATE 0x03
+#define AD1816A_VOICE_ATT 0x04
+#define AD1816A_FM_ATT 0x05
+#define AD1816A_I2S_1_ATT 0x06
+#define AD1816A_I2S_0_ATT 0x07
+#define AD1816A_PLAYBACK_BASE_COUNT 0x08
+#define AD1816A_PLAYBACK_CURR_COUNT 0x09
+#define AD1816A_CAPTURE_BASE_COUNT 0x0a
+#define AD1816A_CAPTURE_CURR_COUNT 0x0b
+#define AD1816A_TIMER_BASE_COUNT 0x0c
+#define AD1816A_TIMER_CURR_COUNT 0x0d
+#define AD1816A_MASTER_ATT 0x0e
+#define AD1816A_CD_GAIN_ATT 0x0f
+#define AD1816A_SYNTH_GAIN_ATT 0x10
+#define AD1816A_VID_GAIN_ATT 0x11
+#define AD1816A_LINE_GAIN_ATT 0x12
+#define AD1816A_MIC_GAIN_ATT 0x13
+#define AD1816A_PHONE_IN_GAIN_ATT 0x13
+#define AD1816A_ADC_SOURCE_SEL 0x14
+#define AD1816A_ADC_PGA 0x14
+#define AD1816A_CHIP_CONFIG 0x20
+#define AD1816A_DSP_CONFIG 0x21
+#define AD1816A_FM_SAMPLE_RATE 0x22
+#define AD1816A_I2S_1_SAMPLE_RATE 0x23
+#define AD1816A_I2S_0_SAMPLE_RATE 0x24
+#define AD1816A_RESERVED_37 0x25
+#define AD1816A_PROGRAM_CLOCK_RATE 0x26
+#define AD1816A_3D_PHAT_CTRL 0x27
+#define AD1816A_PHONE_OUT_ATT 0x27
+#define AD1816A_RESERVED_40 0x28
+#define AD1816A_HW_VOL_BUT 0x29
+#define AD1816A_DSP_MAILBOX_0 0x2a
+#define AD1816A_DSP_MAILBOX_1 0x2b
+#define AD1816A_POWERDOWN_CTRL 0x2c
+#define AD1816A_TIMER_CTRL 0x2c
+#define AD1816A_VERSION_ID 0x2d
+#define AD1816A_RESERVED_46 0x2e
+
+#define AD1816A_READY 0x80
+
+#define AD1816A_PLAYBACK_IRQ_PENDING 0x80
+#define AD1816A_CAPTURE_IRQ_PENDING 0x40
+#define AD1816A_TIMER_IRQ_PENDING 0x20
+
+#define AD1816A_PLAYBACK_ENABLE 0x01
+#define AD1816A_PLAYBACK_PIO 0x02
+#define AD1816A_CAPTURE_ENABLE 0x01
+#define AD1816A_CAPTURE_PIO 0x02
+
+#define AD1816A_FMT_LINEAR_8 0x00
+#define AD1816A_FMT_ULAW_8 0x08
+#define AD1816A_FMT_LINEAR_16_LIT 0x10
+#define AD1816A_FMT_ALAW_8 0x18
+#define AD1816A_FMT_LINEAR_16_BIG 0x30
+#define AD1816A_FMT_ALL 0x38
+#define AD1816A_FMT_STEREO 0x04
+
+#define AD1816A_PLAYBACK_IRQ_ENABLE 0x8000
+#define AD1816A_CAPTURE_IRQ_ENABLE 0x4000
+#define AD1816A_TIMER_IRQ_ENABLE 0x2000
+#define AD1816A_TIMER_ENABLE 0x0080
+
+#define AD1816A_SRC_LINE 0x00
+#define AD1816A_SRC_OUT 0x10
+#define AD1816A_SRC_CD 0x20
+#define AD1816A_SRC_SYNTH 0x30
+#define AD1816A_SRC_VIDEO 0x40
+#define AD1816A_SRC_MIC 0x50
+#define AD1816A_SRC_MONO 0x50
+#define AD1816A_SRC_PHONE_IN 0x60
+#define AD1816A_SRC_MASK 0x70
+
+#define AD1816A_CAPTURE_NOT_EQUAL 0x1000
+#define AD1816A_WSS_ENABLE 0x8000
+
+struct snd_ad1816a {
+ unsigned long port;
+ struct resource *res_port;
+ int irq;
+ int dma1;
+ int dma2;
+
+ unsigned short hardware;
+ unsigned short version;
+
+ spinlock_t lock;
+
+ unsigned short mode;
+ unsigned int clock_freq;
+
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+ unsigned int p_dma_size;
+ unsigned int c_dma_size;
+
+ struct snd_timer *timer;
+#ifdef CONFIG_PM
+ unsigned short image[48];
+#endif
+};
+
+
+#define AD1816A_HW_AUTO 0
+#define AD1816A_HW_AD1816A 1
+#define AD1816A_HW_AD1815 2
+#define AD1816A_HW_AD18MAX10 3
+
+#define AD1816A_MODE_PLAYBACK 0x01
+#define AD1816A_MODE_CAPTURE 0x02
+#define AD1816A_MODE_TIMER 0x04
+#define AD1816A_MODE_OPEN (AD1816A_MODE_PLAYBACK | \
+ AD1816A_MODE_CAPTURE | \
+ AD1816A_MODE_TIMER)
+
+
+extern int snd_ad1816a_create(struct snd_card *card, unsigned long port,
+ int irq, int dma1, int dma2,
+ struct snd_ad1816a *chip);
+
+extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device);
+extern int snd_ad1816a_mixer(struct snd_ad1816a *chip);
+extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device);
+#ifdef CONFIG_PM
+extern void snd_ad1816a_suspend(struct snd_ad1816a *chip);
+extern void snd_ad1816a_resume(struct snd_ad1816a *chip);
+#endif
+
+#endif /* __SOUND_AD1816A_H */
diff --git a/kernel/include/sound/ad1843.h b/kernel/include/sound/ad1843.h
new file mode 100644
index 000000000..b236a9d1d
--- /dev/null
+++ b/kernel/include/sound/ad1843.h
@@ -0,0 +1,46 @@
+/*
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License. See the file "COPYING" in the main directory of this archive
+ * for more details.
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ * Copyright 2008 Thomas Bogendoerfer <tsbogend@franken.de>
+ */
+
+#ifndef __SOUND_AD1843_H
+#define __SOUND_AD1843_H
+
+struct snd_ad1843 {
+ void *chip;
+ int (*read)(void *chip, int reg);
+ int (*write)(void *chip, int reg, int val);
+};
+
+#define AD1843_GAIN_RECLEV 0
+#define AD1843_GAIN_LINE 1
+#define AD1843_GAIN_LINE_2 2
+#define AD1843_GAIN_MIC 3
+#define AD1843_GAIN_PCM_0 4
+#define AD1843_GAIN_PCM_1 5
+#define AD1843_GAIN_SIZE (AD1843_GAIN_PCM_1+1)
+
+int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id);
+int ad1843_get_gain(struct snd_ad1843 *ad1843, int id);
+int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval);
+int ad1843_get_recsrc(struct snd_ad1843 *ad1843);
+int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc);
+void ad1843_setup_dac(struct snd_ad1843 *ad1843,
+ unsigned int id,
+ unsigned int framerate,
+ snd_pcm_format_t fmt,
+ unsigned int channels);
+void ad1843_shutdown_dac(struct snd_ad1843 *ad1843,
+ unsigned int id);
+void ad1843_setup_adc(struct snd_ad1843 *ad1843,
+ unsigned int framerate,
+ snd_pcm_format_t fmt,
+ unsigned int channels);
+void ad1843_shutdown_adc(struct snd_ad1843 *ad1843);
+int ad1843_init(struct snd_ad1843 *ad1843);
+
+#endif /* __SOUND_AD1843_H */
diff --git a/kernel/include/sound/adau1373.h b/kernel/include/sound/adau1373.h
new file mode 100644
index 000000000..1b19c7666
--- /dev/null
+++ b/kernel/include/sound/adau1373.h
@@ -0,0 +1,34 @@
+/*
+ * Analog Devices ADAU1373 Audio Codec drive
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef __SOUND_ADAU1373_H__
+#define __SOUND_ADAU1373_H__
+
+enum adau1373_micbias_voltage {
+ ADAU1373_MICBIAS_2_9V = 0,
+ ADAU1373_MICBIAS_2_2V = 1,
+ ADAU1373_MICBIAS_2_6V = 2,
+ ADAU1373_MICBIAS_1_8V = 3,
+};
+
+#define ADAU1373_DRC_SIZE 13
+
+struct adau1373_platform_data {
+ bool input_differential[4];
+ bool lineout_differential;
+ bool lineout_ground_sense;
+
+ unsigned int num_drc;
+ uint8_t drc_setting[3][ADAU1373_DRC_SIZE];
+
+ enum adau1373_micbias_voltage micbias1;
+ enum adau1373_micbias_voltage micbias2;
+};
+
+#endif
diff --git a/kernel/include/sound/aess.h b/kernel/include/sound/aess.h
new file mode 100644
index 000000000..cee0d09fa
--- /dev/null
+++ b/kernel/include/sound/aess.h
@@ -0,0 +1,53 @@
+/*
+ * AESS IP block reset
+ *
+ * Copyright (C) 2012 Texas Instruments, Inc.
+ * Paul Walmsley
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any
+ * kind, whether express or implied; without even the implied warranty
+ * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+#ifndef __SOUND_AESS_H__
+#define __SOUND_AESS_H__
+
+#include <linux/kernel.h>
+#include <linux/io.h>
+
+/*
+ * AESS_AUTO_GATING_ENABLE_OFFSET: offset in bytes of the AESS IP
+ * block's AESS_AUTO_GATING_ENABLE__1 register from the IP block's
+ * base address
+ */
+#define AESS_AUTO_GATING_ENABLE_OFFSET 0x07c
+
+/* Register bitfields in the AESS_AUTO_GATING_ENABLE__1 register */
+#define AESS_AUTO_GATING_ENABLE_SHIFT 0
+
+/**
+ * aess_enable_autogating - enable AESS internal autogating
+ * @oh: struct omap_hwmod *
+ *
+ * Enable internal autogating on the AESS. This allows the AESS to
+ * indicate that it is idle to the OMAP PRCM. Returns 0.
+ */
+static inline void aess_enable_autogating(void __iomem *base)
+{
+ u32 v;
+
+ /* Set AESS_AUTO_GATING_ENABLE__1.ENABLE to allow idle entry */
+ v = 1 << AESS_AUTO_GATING_ENABLE_SHIFT;
+ writel(v, base + AESS_AUTO_GATING_ENABLE_OFFSET);
+}
+
+#endif /* __SOUND_AESS_H__ */
diff --git a/kernel/include/sound/ak4113.h b/kernel/include/sound/ak4113.h
new file mode 100644
index 000000000..58c145620
--- /dev/null
+++ b/kernel/include/sound/ak4113.h
@@ -0,0 +1,330 @@
+#ifndef __SOUND_AK4113_H
+#define __SOUND_AK4113_H
+
+/*
+ * Routines for Asahi Kasei AK4113
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Copyright (c) by Pavel Hofman <pavel.hofman@ivitera.com>,
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* AK4113 registers */
+/* power down */
+#define AK4113_REG_PWRDN 0x00
+/* format control */
+#define AK4113_REG_FORMAT 0x01
+/* input/output control */
+#define AK4113_REG_IO0 0x02
+/* input/output control */
+#define AK4113_REG_IO1 0x03
+/* interrupt0 mask */
+#define AK4113_REG_INT0_MASK 0x04
+/* interrupt1 mask */
+#define AK4113_REG_INT1_MASK 0x05
+/* DAT mask & DTS select */
+#define AK4113_REG_DATDTS 0x06
+/* receiver status 0 */
+#define AK4113_REG_RCS0 0x07
+/* receiver status 1 */
+#define AK4113_REG_RCS1 0x08
+/* receiver status 2 */
+#define AK4113_REG_RCS2 0x09
+/* RX channel status byte 0 */
+#define AK4113_REG_RXCSB0 0x0a
+/* RX channel status byte 1 */
+#define AK4113_REG_RXCSB1 0x0b
+/* RX channel status byte 2 */
+#define AK4113_REG_RXCSB2 0x0c
+/* RX channel status byte 3 */
+#define AK4113_REG_RXCSB3 0x0d
+/* RX channel status byte 4 */
+#define AK4113_REG_RXCSB4 0x0e
+/* burst preamble Pc byte 0 */
+#define AK4113_REG_Pc0 0x0f
+/* burst preamble Pc byte 1 */
+#define AK4113_REG_Pc1 0x10
+/* burst preamble Pd byte 0 */
+#define AK4113_REG_Pd0 0x11
+/* burst preamble Pd byte 1 */
+#define AK4113_REG_Pd1 0x12
+/* Q-subcode address + control */
+#define AK4113_REG_QSUB_ADDR 0x13
+/* Q-subcode track */
+#define AK4113_REG_QSUB_TRACK 0x14
+/* Q-subcode index */
+#define AK4113_REG_QSUB_INDEX 0x15
+/* Q-subcode minute */
+#define AK4113_REG_QSUB_MINUTE 0x16
+/* Q-subcode second */
+#define AK4113_REG_QSUB_SECOND 0x17
+/* Q-subcode frame */
+#define AK4113_REG_QSUB_FRAME 0x18
+/* Q-subcode zero */
+#define AK4113_REG_QSUB_ZERO 0x19
+/* Q-subcode absolute minute */
+#define AK4113_REG_QSUB_ABSMIN 0x1a
+/* Q-subcode absolute second */
+#define AK4113_REG_QSUB_ABSSEC 0x1b
+/* Q-subcode absolute frame */
+#define AK4113_REG_QSUB_ABSFRM 0x1c
+
+/* sizes */
+#define AK4113_REG_RXCSB_SIZE ((AK4113_REG_RXCSB4-AK4113_REG_RXCSB0)+1)
+#define AK4113_REG_QSUB_SIZE ((AK4113_REG_QSUB_ABSFRM-AK4113_REG_QSUB_ADDR)\
+ +1)
+
+#define AK4113_WRITABLE_REGS (AK4113_REG_DATDTS + 1)
+
+/* AK4113_REG_PWRDN bits */
+/* Channel Status Select */
+#define AK4113_CS12 (1<<7)
+/* Block Start & C/U Output Mode */
+#define AK4113_BCU (1<<6)
+/* Master Clock Operation Select */
+#define AK4113_CM1 (1<<5)
+/* Master Clock Operation Select */
+#define AK4113_CM0 (1<<4)
+/* Master Clock Frequency Select */
+#define AK4113_OCKS1 (1<<3)
+/* Master Clock Frequency Select */
+#define AK4113_OCKS0 (1<<2)
+/* 0 = power down, 1 = normal operation */
+#define AK4113_PWN (1<<1)
+/* 0 = reset & initialize (except thisregister), 1 = normal operation */
+#define AK4113_RST (1<<0)
+
+/* AK4113_REQ_FORMAT bits */
+/* V/TX Output select: 0 = Validity Flag Output, 1 = TX */
+#define AK4113_VTX (1<<7)
+/* Audio Data Control */
+#define AK4113_DIF2 (1<<6)
+/* Audio Data Control */
+#define AK4113_DIF1 (1<<5)
+/* Audio Data Control */
+#define AK4113_DIF0 (1<<4)
+/* Deemphasis Autodetect Enable (1 = enable) */
+#define AK4113_DEAU (1<<3)
+/* 32kHz-48kHz Deemphasis Control */
+#define AK4113_DEM1 (1<<2)
+/* 32kHz-48kHz Deemphasis Control */
+#define AK4113_DEM0 (1<<1)
+#define AK4113_DEM_OFF (AK4113_DEM0)
+#define AK4113_DEM_44KHZ (0)
+#define AK4113_DEM_48KHZ (AK4113_DEM1)
+#define AK4113_DEM_32KHZ (AK4113_DEM0|AK4113_DEM1)
+/* STDO: 16-bit, right justified */
+#define AK4113_DIF_16R (0)
+/* STDO: 18-bit, right justified */
+#define AK4113_DIF_18R (AK4113_DIF0)
+/* STDO: 20-bit, right justified */
+#define AK4113_DIF_20R (AK4113_DIF1)
+/* STDO: 24-bit, right justified */
+#define AK4113_DIF_24R (AK4113_DIF1|AK4113_DIF0)
+/* STDO: 24-bit, left justified */
+#define AK4113_DIF_24L (AK4113_DIF2)
+/* STDO: I2S */
+#define AK4113_DIF_24I2S (AK4113_DIF2|AK4113_DIF0)
+/* STDO: 24-bit, left justified; LRCLK, BICK = Input */
+#define AK4113_DIF_I24L (AK4113_DIF2|AK4113_DIF1)
+/* STDO: I2S; LRCLK, BICK = Input */
+#define AK4113_DIF_I24I2S (AK4113_DIF2|AK4113_DIF1|AK4113_DIF0)
+
+/* AK4113_REG_IO0 */
+/* XTL1=0,XTL0=0 -> 11.2896Mhz; XTL1=0,XTL0=1 -> 12.288Mhz */
+#define AK4113_XTL1 (1<<6)
+/* XTL1=1,XTL0=0 -> 24.576Mhz; XTL1=1,XTL0=1 -> use channel status */
+#define AK4113_XTL0 (1<<5)
+/* Block Start Signal Output: 0 = U-bit, 1 = C-bit (req. BCU = 1) */
+#define AK4113_UCE (1<<4)
+/* TX Output Enable (1 = enable) */
+#define AK4113_TXE (1<<3)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS2 (1<<2)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS1 (1<<1)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS0 (1<<0)
+/* 11.2896 MHz ref. Xtal freq. */
+#define AK4113_XTL_11_2896M (0)
+/* 12.288 MHz ref. Xtal freq. */
+#define AK4113_XTL_12_288M (AK4113_XTL0)
+/* 24.576 MHz ref. Xtal freq. */
+#define AK4113_XTL_24_576M (AK4113_XTL1)
+
+/* AK4113_REG_IO1 */
+/* Interrupt 0 pin Hold */
+#define AK4113_EFH1 (1<<7)
+/* Interrupt 0 pin Hold */
+#define AK4113_EFH0 (1<<6)
+#define AK4113_EFH_512LRCLK (0)
+#define AK4113_EFH_1024LRCLK (AK4113_EFH0)
+#define AK4113_EFH_2048LRCLK (AK4113_EFH1)
+#define AK4113_EFH_4096LRCLK (AK4113_EFH1|AK4113_EFH0)
+/* PLL Lock Time: 0 = 384/fs, 1 = 1/fs */
+#define AK4113_FAST (1<<5)
+/* MCKO2 Output Select: 0 = CMx/OCKSx, 1 = Xtal */
+#define AK4113_XMCK (1<<4)
+/* MCKO2 Output Freq. Select: 0 = x1, 1 = x0.5 (req. XMCK = 1) */
+#define AK4113_DIV (1<<3)
+/* Input Recovery Data Select */
+#define AK4113_IPS2 (1<<2)
+/* Input Recovery Data Select */
+#define AK4113_IPS1 (1<<1)
+/* Input Recovery Data Select */
+#define AK4113_IPS0 (1<<0)
+#define AK4113_IPS(x) ((x)&7)
+
+/* AK4113_REG_INT0_MASK && AK4113_REG_INT1_MASK*/
+/* mask enable for QINT bit */
+#define AK4113_MQI (1<<7)
+/* mask enable for AUTO bit */
+#define AK4113_MAUT (1<<6)
+/* mask enable for CINT bit */
+#define AK4113_MCIT (1<<5)
+/* mask enable for UNLOCK bit */
+#define AK4113_MULK (1<<4)
+/* mask enable for V bit */
+#define AK4113_V (1<<3)
+/* mask enable for STC bit */
+#define AK4113_STC (1<<2)
+/* mask enable for AUDN bit */
+#define AK4113_MAN (1<<1)
+/* mask enable for PAR bit */
+#define AK4113_MPR (1<<0)
+
+/* AK4113_REG_DATDTS */
+/* DAT Start ID Counter */
+#define AK4113_DCNT (1<<4)
+/* DTS-CD 16-bit Sync Word Detect */
+#define AK4113_DTS16 (1<<3)
+/* DTS-CD 14-bit Sync Word Detect */
+#define AK4113_DTS14 (1<<2)
+/* mask enable for DAT bit (if 1, no INT1 effect */
+#define AK4113_MDAT1 (1<<1)
+/* mask enable for DAT bit (if 1, no INT0 effect */
+#define AK4113_MDAT0 (1<<0)
+
+/* AK4113_REG_RCS0 */
+/* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
+#define AK4113_QINT (1<<7)
+/* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
+#define AK4113_AUTO (1<<6)
+/* channel status buffer interrupt, 0 = no change, 1 = change */
+#define AK4113_CINT (1<<5)
+/* PLL lock status, 0 = lock, 1 = unlock */
+#define AK4113_UNLCK (1<<4)
+/* Validity bit, 0 = valid, 1 = invalid */
+#define AK4113_V (1<<3)
+/* sampling frequency or Pre-emphasis change, 0 = no detect, 1 = detect */
+#define AK4113_STC (1<<2)
+/* audio bit output, 0 = audio, 1 = non-audio */
+#define AK4113_AUDION (1<<1)
+/* parity error or biphase error status, 0 = no error, 1 = error */
+#define AK4113_PAR (1<<0)
+
+/* AK4113_REG_RCS1 */
+/* sampling frequency detection */
+#define AK4113_FS3 (1<<7)
+#define AK4113_FS2 (1<<6)
+#define AK4113_FS1 (1<<5)
+#define AK4113_FS0 (1<<4)
+/* Pre-emphasis detect, 0 = OFF, 1 = ON */
+#define AK4113_PEM (1<<3)
+/* DAT Start ID Detect, 0 = no detect, 1 = detect */
+#define AK4113_DAT (1<<2)
+/* DTS-CD bit audio stream detect, 0 = no detect, 1 = detect */
+#define AK4113_DTSCD (1<<1)
+/* Non-PCM bit stream detection, 0 = no detect, 1 = detect */
+#define AK4113_NPCM (1<<0)
+#define AK4113_FS_8000HZ (AK4113_FS3|AK4113_FS0)
+#define AK4113_FS_11025HZ (AK4113_FS2|AK4113_FS0)
+#define AK4113_FS_16000HZ (AK4113_FS2|AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_22050HZ (AK4113_FS2)
+#define AK4113_FS_24000HZ (AK4113_FS2|AK4113_FS1)
+#define AK4113_FS_32000HZ (AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_44100HZ (0)
+#define AK4113_FS_48000HZ (AK4113_FS1)
+#define AK4113_FS_64000HZ (AK4113_FS3|AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_88200HZ (AK4113_FS3)
+#define AK4113_FS_96000HZ (AK4113_FS3|AK4113_FS1)
+#define AK4113_FS_176400HZ (AK4113_FS3|AK4113_FS2)
+#define AK4113_FS_192000HZ (AK4113_FS3|AK4113_FS2|AK4113_FS1)
+
+/* AK4113_REG_RCS2 */
+/* CRC for Q-subcode, 0 = no error, 1 = error */
+#define AK4113_QCRC (1<<1)
+/* CRC for channel status, 0 = no error, 1 = error */
+#define AK4113_CCRC (1<<0)
+
+/* flags for snd_ak4113_check_rate_and_errors() */
+#define AK4113_CHECK_NO_STAT (1<<0) /* no statistics */
+#define AK4113_CHECK_NO_RATE (1<<1) /* no rate check */
+
+#define AK4113_CONTROLS 13
+
+typedef void (ak4113_write_t)(void *private_data, unsigned char addr,
+ unsigned char data);
+typedef unsigned char (ak4113_read_t)(void *private_data, unsigned char addr);
+
+struct ak4113 {
+ struct snd_card *card;
+ ak4113_write_t *write;
+ ak4113_read_t *read;
+ void *private_data;
+ atomic_t wq_processing;
+ struct mutex reinit_mutex;
+ spinlock_t lock;
+ unsigned char regmap[AK4113_WRITABLE_REGS];
+ struct snd_kcontrol *kctls[AK4113_CONTROLS];
+ struct snd_pcm_substream *substream;
+ unsigned long parity_errors;
+ unsigned long v_bit_errors;
+ unsigned long qcrc_errors;
+ unsigned long ccrc_errors;
+ unsigned char rcs0;
+ unsigned char rcs1;
+ unsigned char rcs2;
+ struct delayed_work work;
+ unsigned int check_flags;
+ void *change_callback_private;
+ void (*change_callback)(struct ak4113 *ak4113, unsigned char c0,
+ unsigned char c1);
+};
+
+int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
+ ak4113_write_t *write,
+ const unsigned char *pgm,
+ void *private_data, struct ak4113 **r_ak4113);
+void snd_ak4113_reg_write(struct ak4113 *ak4113, unsigned char reg,
+ unsigned char mask, unsigned char val);
+void snd_ak4113_reinit(struct ak4113 *ak4113);
+int snd_ak4113_build(struct ak4113 *ak4113,
+ struct snd_pcm_substream *capture_substream);
+int snd_ak4113_external_rate(struct ak4113 *ak4113);
+int snd_ak4113_check_rate_and_errors(struct ak4113 *ak4113, unsigned int flags);
+
+#ifdef CONFIG_PM
+void snd_ak4113_suspend(struct ak4113 *chip);
+void snd_ak4113_resume(struct ak4113 *chip);
+#else
+static inline void snd_ak4113_suspend(struct ak4113 *chip) {}
+static inline void snd_ak4113_resume(struct ak4113 *chip) {}
+#endif
+
+#endif /* __SOUND_AK4113_H */
+
diff --git a/kernel/include/sound/ak4114.h b/kernel/include/sound/ak4114.h
new file mode 100644
index 000000000..b6feb7e22
--- /dev/null
+++ b/kernel/include/sound/ak4114.h
@@ -0,0 +1,212 @@
+#ifndef __SOUND_AK4114_H
+#define __SOUND_AK4114_H
+
+/*
+ * Routines for Asahi Kasei AK4114
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* AK4114 registers */
+#define AK4114_REG_PWRDN 0x00 /* power down */
+#define AK4114_REG_FORMAT 0x01 /* format control */
+#define AK4114_REG_IO0 0x02 /* input/output control */
+#define AK4114_REG_IO1 0x03 /* input/output control */
+#define AK4114_REG_INT0_MASK 0x04 /* interrupt0 mask */
+#define AK4114_REG_INT1_MASK 0x05 /* interrupt1 mask */
+#define AK4114_REG_RCS0 0x06 /* receiver status 0 */
+#define AK4114_REG_RCS1 0x07 /* receiver status 1 */
+#define AK4114_REG_RXCSB0 0x08 /* RX channel status byte 0 */
+#define AK4114_REG_RXCSB1 0x09 /* RX channel status byte 1 */
+#define AK4114_REG_RXCSB2 0x0a /* RX channel status byte 2 */
+#define AK4114_REG_RXCSB3 0x0b /* RX channel status byte 3 */
+#define AK4114_REG_RXCSB4 0x0c /* RX channel status byte 4 */
+#define AK4114_REG_TXCSB0 0x0d /* TX channel status byte 0 */
+#define AK4114_REG_TXCSB1 0x0e /* TX channel status byte 1 */
+#define AK4114_REG_TXCSB2 0x0f /* TX channel status byte 2 */
+#define AK4114_REG_TXCSB3 0x10 /* TX channel status byte 3 */
+#define AK4114_REG_TXCSB4 0x11 /* TX channel status byte 4 */
+#define AK4114_REG_Pc0 0x12 /* burst preamble Pc byte 0 */
+#define AK4114_REG_Pc1 0x13 /* burst preamble Pc byte 1 */
+#define AK4114_REG_Pd0 0x14 /* burst preamble Pd byte 0 */
+#define AK4114_REG_Pd1 0x15 /* burst preamble Pd byte 1 */
+#define AK4114_REG_QSUB_ADDR 0x16 /* Q-subcode address + control */
+#define AK4114_REG_QSUB_TRACK 0x17 /* Q-subcode track */
+#define AK4114_REG_QSUB_INDEX 0x18 /* Q-subcode index */
+#define AK4114_REG_QSUB_MINUTE 0x19 /* Q-subcode minute */
+#define AK4114_REG_QSUB_SECOND 0x1a /* Q-subcode second */
+#define AK4114_REG_QSUB_FRAME 0x1b /* Q-subcode frame */
+#define AK4114_REG_QSUB_ZERO 0x1c /* Q-subcode zero */
+#define AK4114_REG_QSUB_ABSMIN 0x1d /* Q-subcode absolute minute */
+#define AK4114_REG_QSUB_ABSSEC 0x1e /* Q-subcode absolute second */
+#define AK4114_REG_QSUB_ABSFRM 0x1f /* Q-subcode absolute frame */
+
+/* sizes */
+#define AK4114_REG_RXCSB_SIZE ((AK4114_REG_RXCSB4-AK4114_REG_RXCSB0)+1)
+#define AK4114_REG_TXCSB_SIZE ((AK4114_REG_TXCSB4-AK4114_REG_TXCSB0)+1)
+#define AK4114_REG_QSUB_SIZE ((AK4114_REG_QSUB_ABSFRM-AK4114_REG_QSUB_ADDR)+1)
+
+/* AK4117_REG_PWRDN bits */
+#define AK4114_CS12 (1<<7) /* Channel Status Select */
+#define AK4114_BCU (1<<6) /* Block Start & C/U Output Mode */
+#define AK4114_CM1 (1<<5) /* Master Clock Operation Select */
+#define AK4114_CM0 (1<<4) /* Master Clock Operation Select */
+#define AK4114_OCKS1 (1<<3) /* Master Clock Frequency Select */
+#define AK4114_OCKS0 (1<<2) /* Master Clock Frequency Select */
+#define AK4114_PWN (1<<1) /* 0 = power down, 1 = normal operation */
+#define AK4114_RST (1<<0) /* 0 = reset & initialize (except this register), 1 = normal operation */
+
+/* AK4114_REQ_FORMAT bits */
+#define AK4114_MONO (1<<7) /* Double Sampling Frequency Mode: 0 = stereo, 1 = mono */
+#define AK4114_DIF2 (1<<6) /* Audio Data Control */
+#define AK4114_DIF1 (1<<5) /* Audio Data Control */
+#define AK4114_DIF0 (1<<4) /* Audio Data Control */
+#define AK4114_DIF_16R (0) /* STDO: 16-bit, right justified */
+#define AK4114_DIF_18R (AK4114_DIF0) /* STDO: 18-bit, right justified */
+#define AK4114_DIF_20R (AK4114_DIF1) /* STDO: 20-bit, right justified */
+#define AK4114_DIF_24R (AK4114_DIF1|AK4114_DIF0) /* STDO: 24-bit, right justified */
+#define AK4114_DIF_24L (AK4114_DIF2) /* STDO: 24-bit, left justified */
+#define AK4114_DIF_24I2S (AK4114_DIF2|AK4114_DIF0) /* STDO: I2S */
+#define AK4114_DIF_I24L (AK4114_DIF2|AK4114_DIF1) /* STDO: 24-bit, left justified; LRCLK, BICK = Input */
+#define AK4114_DIF_I24I2S (AK4114_DIF2|AK4114_DIF1|AK4114_DIF0) /* STDO: I2S; LRCLK, BICK = Input */
+#define AK4114_DEAU (1<<3) /* Deemphasis Autodetect Enable (1 = enable) */
+#define AK4114_DEM1 (1<<2) /* 32kHz-48kHz Deemphasis Control */
+#define AK4114_DEM0 (1<<1) /* 32kHz-48kHz Deemphasis Control */
+#define AK4114_DEM_44KHZ (0)
+#define AK4114_DEM_48KHZ (AK4114_DEM1)
+#define AK4114_DEM_32KHZ (AK4114_DEM0|AK4114_DEM1)
+#define AK4114_DEM_96KHZ (AK4114_DEM1) /* DFS must be set */
+#define AK4114_DFS (1<<0) /* 96kHz Deemphasis Control */
+
+/* AK4114_REG_IO0 */
+#define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */
+#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */
+#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */
+#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */
+#define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */
+#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */
+#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */
+#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */
+
+/* AK4114_REG_IO1 */
+#define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */
+#define AK4114_EFH0 (1<<6) /* Interrupt 0 pin Hold */
+#define AK4114_EFH_512 (0)
+#define AK4114_EFH_1024 (AK4114_EFH0)
+#define AK4114_EFH_2048 (AK4114_EFH1)
+#define AK4114_EFH_4096 (AK4114_EFH1|AK4114_EFH0)
+#define AK4114_UDIT (1<<5) /* U-bit Control for DIT (0 = fixed '0', 1 = recovered) */
+#define AK4114_TLR (1<<4) /* Double Sampling Frequency Select for DIT (0 = L channel, 1 = R channel) */
+#define AK4114_DIT (1<<3) /* TX1 out: 0 = Through Data (RX data), 1 = Transmit Data (DAUX data) */
+#define AK4114_IPS2 (1<<2) /* Input Recovery Data Select */
+#define AK4114_IPS1 (1<<1) /* Input Recovery Data Select */
+#define AK4114_IPS0 (1<<0) /* Input Recovery Data Select */
+#define AK4114_IPS(x) ((x)&7)
+
+/* AK4114_REG_INT0_MASK && AK4114_REG_INT1_MASK*/
+#define AK4117_MQI (1<<7) /* mask enable for QINT bit */
+#define AK4117_MAT (1<<6) /* mask enable for AUTO bit */
+#define AK4117_MCI (1<<5) /* mask enable for CINT bit */
+#define AK4117_MUL (1<<4) /* mask enable for UNLOCK bit */
+#define AK4117_MDTS (1<<3) /* mask enable for DTSCD bit */
+#define AK4117_MPE (1<<2) /* mask enable for PEM bit */
+#define AK4117_MAN (1<<1) /* mask enable for AUDN bit */
+#define AK4117_MPR (1<<0) /* mask enable for PAR bit */
+
+/* AK4114_REG_RCS0 */
+#define AK4114_QINT (1<<7) /* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
+#define AK4114_AUTO (1<<6) /* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
+#define AK4114_CINT (1<<5) /* channel status buffer interrupt, 0 = no change, 1 = change */
+#define AK4114_UNLCK (1<<4) /* PLL lock status, 0 = lock, 1 = unlock */
+#define AK4114_DTSCD (1<<3) /* DTS-CD Detect, 0 = No detect, 1 = Detect */
+#define AK4114_PEM (1<<2) /* Pre-emphasis Detect, 0 = OFF, 1 = ON */
+#define AK4114_AUDION (1<<1) /* audio bit output, 0 = audio, 1 = non-audio */
+#define AK4114_PAR (1<<0) /* parity error or biphase error status, 0 = no error, 1 = error */
+
+/* AK4114_REG_RCS1 */
+#define AK4114_FS3 (1<<7) /* sampling frequency detection */
+#define AK4114_FS2 (1<<6)
+#define AK4114_FS1 (1<<5)
+#define AK4114_FS0 (1<<4)
+#define AK4114_FS_44100HZ (0)
+#define AK4114_FS_48000HZ (AK4114_FS1)
+#define AK4114_FS_32000HZ (AK4114_FS1|AK4114_FS0)
+#define AK4114_FS_88200HZ (AK4114_FS3)
+#define AK4114_FS_96000HZ (AK4114_FS3|AK4114_FS1)
+#define AK4114_FS_176400HZ (AK4114_FS3|AK4114_FS2)
+#define AK4114_FS_192000HZ (AK4114_FS3|AK4114_FS2|AK4114_FS1)
+#define AK4114_V (1<<3) /* Validity of Channel Status, 0 = Valid, 1 = Invalid */
+#define AK4114_QCRC (1<<1) /* CRC for Q-subcode, 0 = no error, 1 = error */
+#define AK4114_CCRC (1<<0) /* CRC for channel status, 0 = no error, 1 = error */
+
+/* flags for snd_ak4114_check_rate_and_errors() */
+#define AK4114_CHECK_NO_STAT (1<<0) /* no statistics */
+#define AK4114_CHECK_NO_RATE (1<<1) /* no rate check */
+
+#define AK4114_CONTROLS 15
+
+typedef void (ak4114_write_t)(void *private_data, unsigned char addr, unsigned char data);
+typedef unsigned char (ak4114_read_t)(void *private_data, unsigned char addr);
+
+struct ak4114 {
+ struct snd_card *card;
+ ak4114_write_t * write;
+ ak4114_read_t * read;
+ void * private_data;
+ atomic_t wq_processing;
+ struct mutex reinit_mutex;
+ spinlock_t lock;
+ unsigned char regmap[6];
+ unsigned char txcsb[5];
+ struct snd_kcontrol *kctls[AK4114_CONTROLS];
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+ unsigned long parity_errors;
+ unsigned long v_bit_errors;
+ unsigned long qcrc_errors;
+ unsigned long ccrc_errors;
+ unsigned char rcs0;
+ unsigned char rcs1;
+ struct delayed_work work;
+ unsigned int check_flags;
+ void *change_callback_private;
+ void (*change_callback)(struct ak4114 *ak4114, unsigned char c0, unsigned char c1);
+};
+
+int snd_ak4114_create(struct snd_card *card,
+ ak4114_read_t *read, ak4114_write_t *write,
+ const unsigned char pgm[6], const unsigned char txcsb[5],
+ void *private_data, struct ak4114 **r_ak4114);
+void snd_ak4114_reg_write(struct ak4114 *ak4114, unsigned char reg, unsigned char mask, unsigned char val);
+void snd_ak4114_reinit(struct ak4114 *ak4114);
+int snd_ak4114_build(struct ak4114 *ak4114,
+ struct snd_pcm_substream *playback_substream,
+ struct snd_pcm_substream *capture_substream);
+int snd_ak4114_external_rate(struct ak4114 *ak4114);
+int snd_ak4114_check_rate_and_errors(struct ak4114 *ak4114, unsigned int flags);
+
+#ifdef CONFIG_PM
+void snd_ak4114_suspend(struct ak4114 *chip);
+void snd_ak4114_resume(struct ak4114 *chip);
+#else
+static inline void snd_ak4114_suspend(struct ak4114 *chip) {}
+static inline void snd_ak4114_resume(struct ak4114 *chip) {}
+#endif
+
+#endif /* __SOUND_AK4114_H */
+
diff --git a/kernel/include/sound/ak4117.h b/kernel/include/sound/ak4117.h
new file mode 100644
index 000000000..1e8178171
--- /dev/null
+++ b/kernel/include/sound/ak4117.h
@@ -0,0 +1,189 @@
+#ifndef __SOUND_AK4117_H
+#define __SOUND_AK4117_H
+
+/*
+ * Routines for Asahi Kasei AK4117
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define AK4117_REG_PWRDN 0x00 /* power down */
+#define AK4117_REG_CLOCK 0x01 /* clock control */
+#define AK4117_REG_IO 0x02 /* input/output control */
+#define AK4117_REG_INT0_MASK 0x03 /* interrupt0 mask */
+#define AK4117_REG_INT1_MASK 0x04 /* interrupt1 mask */
+#define AK4117_REG_RCS0 0x05 /* receiver status 0 */
+#define AK4117_REG_RCS1 0x06 /* receiver status 1 */
+#define AK4117_REG_RCS2 0x07 /* receiver status 2 */
+#define AK4117_REG_RXCSB0 0x08 /* RX channel status byte 0 */
+#define AK4117_REG_RXCSB1 0x09 /* RX channel status byte 1 */
+#define AK4117_REG_RXCSB2 0x0a /* RX channel status byte 2 */
+#define AK4117_REG_RXCSB3 0x0b /* RX channel status byte 3 */
+#define AK4117_REG_RXCSB4 0x0c /* RX channel status byte 4 */
+#define AK4117_REG_Pc0 0x0d /* burst preamble Pc byte 0 */
+#define AK4117_REG_Pc1 0x0e /* burst preamble Pc byte 1 */
+#define AK4117_REG_Pd0 0x0f /* burst preamble Pd byte 0 */
+#define AK4117_REG_Pd1 0x10 /* burst preamble Pd byte 1 */
+#define AK4117_REG_QSUB_ADDR 0x11 /* Q-subcode address + control */
+#define AK4117_REG_QSUB_TRACK 0x12 /* Q-subcode track */
+#define AK4117_REG_QSUB_INDEX 0x13 /* Q-subcode index */
+#define AK4117_REG_QSUB_MINUTE 0x14 /* Q-subcode minute */
+#define AK4117_REG_QSUB_SECOND 0x15 /* Q-subcode second */
+#define AK4117_REG_QSUB_FRAME 0x16 /* Q-subcode frame */
+#define AK4117_REG_QSUB_ZERO 0x17 /* Q-subcode zero */
+#define AK4117_REG_QSUB_ABSMIN 0x18 /* Q-subcode absolute minute */
+#define AK4117_REG_QSUB_ABSSEC 0x19 /* Q-subcode absolute second */
+#define AK4117_REG_QSUB_ABSFRM 0x1a /* Q-subcode absolute frame */
+
+/* sizes */
+#define AK4117_REG_RXCSB_SIZE ((AK4117_REG_RXCSB4-AK4117_REG_RXCSB0)+1)
+#define AK4117_REG_QSUB_SIZE ((AK4117_REG_QSUB_ABSFRM-AK4117_REG_QSUB_ADDR)+1)
+
+/* AK4117_REG_PWRDN bits */
+#define AK4117_EXCT (1<<4) /* 0 = X'tal mode, 1 = external clock mode */
+#define AK4117_XTL1 (1<<3) /* XTL1=0,XTL0=0 -> 11.2896Mhz; XTL1=0,XTL0=1 -> 12.288Mhz */
+#define AK4117_XTL0 (1<<2) /* XTL1=1,XTL0=0 -> 24.576Mhz; XTL1=1,XTL0=1 -> use channel status */
+#define AK4117_XTL_11_2896M (0)
+#define AK4117_XTL_12_288M AK4117_XTL0
+#define AK4117_XTL_24_576M AK4117_XTL1
+#define AK4117_XTL_EXT (AK4117_XTL1|AK4117_XTL0)
+#define AK4117_PWN (1<<1) /* 0 = power down, 1 = normal operation */
+#define AK4117_RST (1<<0) /* 0 = reset & initialize (except this register), 1 = normal operation */
+
+/* AK4117_REQ_CLOCK bits */
+#define AK4117_LP (1<<7) /* 0 = normal mode, 1 = low power mode (Fs up to 48kHz only) */
+#define AK4117_PKCS1 (1<<6) /* master clock frequency at PLL mode (when LP == 0) */
+#define AK4117_PKCS0 (1<<5)
+#define AK4117_PKCS_512fs (0)
+#define AK4117_PKCS_256fs AK4117_PKCS0
+#define AK4117_PKCS_128fs AK4117_PKCS1
+#define AK4117_DIV (1<<4) /* 0 = MCKO == Fs, 1 = MCKO == Fs / 2; X'tal mode only */
+#define AK4117_XCKS1 (1<<3) /* master clock frequency at X'tal mode */
+#define AK4117_XCKS0 (1<<2)
+#define AK4117_XCKS_128fs (0)
+#define AK4117_XCKS_256fs AK4117_XCKS0
+#define AK4117_XCKS_512fs AK4117_XCKS1
+#define AK4117_XCKS_1024fs (AK4117_XCKS1|AK4117_XCKS0)
+#define AK4117_CM1 (1<<1) /* MCKO operation mode select */
+#define AK4117_CM0 (1<<0)
+#define AK4117_CM_PLL (0) /* use RX input as master clock */
+#define AK4117_CM_XTAL (AK4117_CM0) /* use X'tal as master clock */
+#define AK4117_CM_PLL_XTAL (AK4117_CM1) /* use Rx input but X'tal when PLL loses lock */
+#define AK4117_CM_MONITOR (AK4117_CM0|AK4117_CM1) /* use X'tal as master clock, but use PLL for monitoring */
+
+/* AK4117_REG_IO */
+#define AK4117_IPS (1<<7) /* Input Recovery Data Select, 0 = RX0, 1 = RX1 */
+#define AK4117_UOUTE (1<<6) /* U-bit output enable to UOUT, 0 = disable, 1 = enable */
+#define AK4117_CS12 (1<<5) /* channel status select, 0 = channel1, 1 = channel2 */
+#define AK4117_EFH2 (1<<4) /* INT0 pin hold count select */
+#define AK4117_EFH1 (1<<3)
+#define AK4117_EFH_512LRCLK (0)
+#define AK4117_EFH_1024LRCLK (AK4117_EFH1)
+#define AK4117_EFH_2048LRCLK (AK4117_EFH2)
+#define AK4117_EFH_4096LRCLK (AK4117_EFH1|AK4117_EFH2)
+#define AK4117_DIF2 (1<<2) /* audio data format control */
+#define AK4117_DIF1 (1<<1)
+#define AK4117_DIF0 (1<<0)
+#define AK4117_DIF_16R (0) /* STDO: 16-bit, right justified */
+#define AK4117_DIF_18R (AK4117_DIF0) /* STDO: 18-bit, right justified */
+#define AK4117_DIF_20R (AK4117_DIF1) /* STDO: 20-bit, right justified */
+#define AK4117_DIF_24R (AK4117_DIF1|AK4117_DIF0) /* STDO: 24-bit, right justified */
+#define AK4117_DIF_24L (AK4117_DIF2) /* STDO: 24-bit, left justified */
+#define AK4117_DIF_24I2S (AK4117_DIF2|AK4117_DIF0) /* STDO: I2S */
+
+/* AK4117_REG_INT0_MASK & AK4117_REG_INT1_MASK */
+#define AK4117_MULK (1<<7) /* mask enable for UNLOCK bit */
+#define AK4117_MPAR (1<<6) /* mask enable for PAR bit */
+#define AK4117_MAUTO (1<<5) /* mask enable for AUTO bit */
+#define AK4117_MV (1<<4) /* mask enable for V bit */
+#define AK4117_MAUD (1<<3) /* mask enable for AUDION bit */
+#define AK4117_MSTC (1<<2) /* mask enable for STC bit */
+#define AK4117_MCIT (1<<1) /* mask enable for CINT bit */
+#define AK4117_MQIT (1<<0) /* mask enable for QINT bit */
+
+/* AK4117_REG_RCS0 */
+#define AK4117_UNLCK (1<<7) /* PLL lock status, 0 = lock, 1 = unlock */
+#define AK4117_PAR (1<<6) /* parity error or biphase error status, 0 = no error, 1 = error */
+#define AK4117_AUTO (1<<5) /* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
+#define AK4117_V (1<<4) /* Validity bit, 0 = valid, 1 = invalid */
+#define AK4117_AUDION (1<<3) /* audio bit output, 0 = audio, 1 = non-audio */
+#define AK4117_STC (1<<2) /* sampling frequency or Pre-emphasis change, 0 = no detect, 1 = detect */
+#define AK4117_CINT (1<<1) /* channel status buffer interrupt, 0 = no change, 1 = change */
+#define AK4117_QINT (1<<0) /* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
+
+/* AK4117_REG_RCS1 */
+#define AK4117_DTSCD (1<<6) /* DTS-CD bit audio stream detect, 0 = no detect, 1 = detect */
+#define AK4117_NPCM (1<<5) /* Non-PCM bit stream detection, 0 = no detect, 1 = detect */
+#define AK4117_PEM (1<<4) /* Pre-emphasis detect, 0 = OFF, 1 = ON */
+#define AK4117_FS3 (1<<3) /* sampling frequency detection */
+#define AK4117_FS2 (1<<2)
+#define AK4117_FS1 (1<<1)
+#define AK4117_FS0 (1<<0)
+#define AK4117_FS_44100HZ (0)
+#define AK4117_FS_48000HZ (AK4117_FS1)
+#define AK4117_FS_32000HZ (AK4117_FS1|AK4117_FS0)
+#define AK4117_FS_88200HZ (AK4117_FS3)
+#define AK4117_FS_96000HZ (AK4117_FS3|AK4117_FS1)
+#define AK4117_FS_176400HZ (AK4117_FS3|AK4117_FS2)
+#define AK4117_FS_192000HZ (AK4117_FS3|AK4117_FS2|AK4117_FS1)
+
+/* AK4117_REG_RCS2 */
+#define AK4117_CCRC (1<<1) /* CRC for channel status, 0 = no error, 1 = error */
+#define AK4117_QCRC (1<<0) /* CRC for Q-subcode, 0 = no error, 1 = error */
+
+/* flags for snd_ak4117_check_rate_and_errors() */
+#define AK4117_CHECK_NO_STAT (1<<0) /* no statistics */
+#define AK4117_CHECK_NO_RATE (1<<1) /* no rate check */
+
+#define AK4117_CONTROLS 13
+
+typedef void (ak4117_write_t)(void *private_data, unsigned char addr, unsigned char data);
+typedef unsigned char (ak4117_read_t)(void *private_data, unsigned char addr);
+
+struct ak4117 {
+ struct snd_card *card;
+ ak4117_write_t * write;
+ ak4117_read_t * read;
+ void * private_data;
+ unsigned int init: 1;
+ spinlock_t lock;
+ unsigned char regmap[5];
+ struct snd_kcontrol *kctls[AK4117_CONTROLS];
+ struct snd_pcm_substream *substream;
+ unsigned long parity_errors;
+ unsigned long v_bit_errors;
+ unsigned long qcrc_errors;
+ unsigned long ccrc_errors;
+ unsigned char rcs0;
+ unsigned char rcs1;
+ unsigned char rcs2;
+ struct timer_list timer; /* statistic timer */
+ void *change_callback_private;
+ void (*change_callback)(struct ak4117 *ak4117, unsigned char c0, unsigned char c1);
+};
+
+int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t *write,
+ const unsigned char pgm[5], void *private_data, struct ak4117 **r_ak4117);
+void snd_ak4117_reg_write(struct ak4117 *ak4117, unsigned char reg, unsigned char mask, unsigned char val);
+void snd_ak4117_reinit(struct ak4117 *ak4117);
+int snd_ak4117_build(struct ak4117 *ak4117, struct snd_pcm_substream *capture_substream);
+int snd_ak4117_external_rate(struct ak4117 *ak4117);
+int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags);
+
+#endif /* __SOUND_AK4117_H */
+
diff --git a/kernel/include/sound/ak4531_codec.h b/kernel/include/sound/ak4531_codec.h
new file mode 100644
index 000000000..85ea86ea3
--- /dev/null
+++ b/kernel/include/sound/ak4531_codec.h
@@ -0,0 +1,85 @@
+#ifndef __SOUND_AK4531_CODEC_H
+#define __SOUND_AK4531_CODEC_H
+
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.1
+ * by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/info.h>
+#include <sound/control.h>
+
+/*
+ * ASAHI KASEI - AK4531 codec
+ * - not really AC'97 codec, but it uses very similar interface as AC'97
+ */
+
+/*
+ * AK4531 codec registers
+ */
+
+#define AK4531_LMASTER 0x00 /* master volume left */
+#define AK4531_RMASTER 0x01 /* master volume right */
+#define AK4531_LVOICE 0x02 /* channel volume left */
+#define AK4531_RVOICE 0x03 /* channel volume right */
+#define AK4531_LFM 0x04 /* FM volume left */
+#define AK4531_RFM 0x05 /* FM volume right */
+#define AK4531_LCD 0x06 /* CD volume left */
+#define AK4531_RCD 0x07 /* CD volume right */
+#define AK4531_LLINE 0x08 /* LINE volume left */
+#define AK4531_RLINE 0x09 /* LINE volume right */
+#define AK4531_LAUXA 0x0a /* AUXA volume left */
+#define AK4531_RAUXA 0x0b /* AUXA volume right */
+#define AK4531_MONO1 0x0c /* MONO1 volume left */
+#define AK4531_MONO2 0x0d /* MONO1 volume right */
+#define AK4531_MIC 0x0e /* MIC volume */
+#define AK4531_MONO_OUT 0x0f /* Mono-out volume */
+#define AK4531_OUT_SW1 0x10 /* Output mixer switch 1 */
+#define AK4531_OUT_SW2 0x11 /* Output mixer switch 2 */
+#define AK4531_LIN_SW1 0x12 /* Input left mixer switch 1 */
+#define AK4531_RIN_SW1 0x13 /* Input right mixer switch 1 */
+#define AK4531_LIN_SW2 0x14 /* Input left mixer switch 2 */
+#define AK4531_RIN_SW2 0x15 /* Input right mixer switch 2 */
+#define AK4531_RESET 0x16 /* Reset & power down */
+#define AK4531_CLOCK 0x17 /* Clock select */
+#define AK4531_AD_IN 0x18 /* AD input select */
+#define AK4531_MIC_GAIN 0x19 /* MIC amplified gain */
+
+struct snd_ak4531 {
+ void (*write) (struct snd_ak4531 *ak4531, unsigned short reg,
+ unsigned short val);
+ void *private_data;
+ void (*private_free) (struct snd_ak4531 *ak4531);
+ /* --- */
+ unsigned char regs[0x20];
+ struct mutex reg_mutex;
+};
+
+int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531,
+ struct snd_ak4531 **rak4531);
+
+#ifdef CONFIG_PM
+void snd_ak4531_suspend(struct snd_ak4531 *ak4531);
+void snd_ak4531_resume(struct snd_ak4531 *ak4531);
+#endif
+
+#endif /* __SOUND_AK4531_CODEC_H */
diff --git a/kernel/include/sound/ak4641.h b/kernel/include/sound/ak4641.h
new file mode 100644
index 000000000..96d1991c8
--- /dev/null
+++ b/kernel/include/sound/ak4641.h
@@ -0,0 +1,26 @@
+/*
+ * AK4641 ALSA SoC Codec driver
+ *
+ * Copyright 2009 Philipp Zabel
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __AK4641_H
+#define __AK4641_H
+
+/**
+ * struct ak4641_platform_data - platform specific AK4641 configuration
+ * @gpio_power: GPIO to control external power to AK4641
+ * @gpio_npdn: GPIO connected to AK4641 nPDN pin
+ *
+ * Both GPIO parameters are optional.
+ */
+struct ak4641_platform_data {
+ int gpio_power;
+ int gpio_npdn;
+};
+
+#endif /* __AK4641_H */
diff --git a/kernel/include/sound/ak4xxx-adda.h b/kernel/include/sound/ak4xxx-adda.h
new file mode 100644
index 000000000..030b87c2f
--- /dev/null
+++ b/kernel/include/sound/ak4xxx-adda.h
@@ -0,0 +1,99 @@
+#ifndef __SOUND_AK4XXX_ADDA_H
+#define __SOUND_AK4XXX_ADDA_H
+
+/*
+ * ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4381
+ * AD and DA converters
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifndef AK4XXX_MAX_CHIPS
+#define AK4XXX_MAX_CHIPS 4
+#endif
+
+struct snd_akm4xxx;
+
+struct snd_ak4xxx_ops {
+ void (*lock)(struct snd_akm4xxx *ak, int chip);
+ void (*unlock)(struct snd_akm4xxx *ak, int chip);
+ void (*write)(struct snd_akm4xxx *ak, int chip, unsigned char reg,
+ unsigned char val);
+ void (*set_rate_val)(struct snd_akm4xxx *ak, unsigned int rate);
+};
+
+#define AK4XXX_IMAGE_SIZE (AK4XXX_MAX_CHIPS * 16) /* 64 bytes */
+
+/* DAC label and channels */
+struct snd_akm4xxx_dac_channel {
+ char *name; /* mixer volume name */
+ unsigned int num_channels;
+ char *switch_name; /* mixer switch*/
+};
+
+/* ADC labels and channels */
+struct snd_akm4xxx_adc_channel {
+ char *name; /* capture gain volume label */
+ char *switch_name; /* capture switch */
+ unsigned int num_channels;
+ char *selector_name; /* capture source select label */
+ const char **input_names; /* capture source names (NULL terminated) */
+};
+
+struct snd_akm4xxx {
+ struct snd_card *card;
+ unsigned int num_adcs; /* AK4524 or AK4528 ADCs */
+ unsigned int num_dacs; /* AK4524 or AK4528 DACs */
+ unsigned char images[AK4XXX_IMAGE_SIZE]; /* saved register image */
+ unsigned char volumes[AK4XXX_IMAGE_SIZE]; /* saved volume values */
+ unsigned long private_value[AK4XXX_MAX_CHIPS]; /* helper for driver */
+ void *private_data[AK4XXX_MAX_CHIPS]; /* helper for driver */
+ /* template should fill the following fields */
+ unsigned int idx_offset; /* control index offset */
+ enum {
+ SND_AK4524, SND_AK4528, SND_AK4529,
+ SND_AK4355, SND_AK4358, SND_AK4381,
+ SND_AK5365, SND_AK4620,
+ } type;
+
+ /* (array) information of combined codecs */
+ const struct snd_akm4xxx_dac_channel *dac_info;
+ const struct snd_akm4xxx_adc_channel *adc_info;
+
+ struct snd_ak4xxx_ops ops;
+ unsigned int num_chips;
+ unsigned int total_regs;
+ const char *name;
+};
+
+void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg,
+ unsigned char val);
+void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state);
+void snd_akm4xxx_init(struct snd_akm4xxx *ak);
+int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak);
+
+#define snd_akm4xxx_get(ak,chip,reg) \
+ (ak)->images[(chip) * 16 + (reg)]
+#define snd_akm4xxx_set(ak,chip,reg,val) \
+ ((ak)->images[(chip) * 16 + (reg)] = (val))
+#define snd_akm4xxx_get_vol(ak,chip,reg) \
+ (ak)->volumes[(chip) * 16 + (reg)]
+#define snd_akm4xxx_set_vol(ak,chip,reg,val) \
+ ((ak)->volumes[(chip) * 16 + (reg)] = (val))
+
+#endif /* __SOUND_AK4XXX_ADDA_H */
diff --git a/kernel/include/sound/alc5623.h b/kernel/include/sound/alc5623.h
new file mode 100644
index 000000000..422c97d43
--- /dev/null
+++ b/kernel/include/sound/alc5623.h
@@ -0,0 +1,15 @@
+#ifndef _INCLUDE_SOUND_ALC5623_H
+#define _INCLUDE_SOUND_ALC5623_H
+struct alc5623_platform_data {
+ /* configure : */
+ /* Lineout/Speaker Amps Vmid ratio control */
+ /* enable/disable adc/dac high pass filters */
+ unsigned int add_ctrl;
+ /* configure : */
+ /* output to enable when jack is low */
+ /* output to enable when jack is high */
+ /* jack detect (gpio/nc/jack detect [12] */
+ unsigned int jack_det_ctrl;
+};
+#endif
+
diff --git a/kernel/include/sound/asequencer.h b/kernel/include/sound/asequencer.h
new file mode 100644
index 000000000..75935ce73
--- /dev/null
+++ b/kernel/include/sound/asequencer.h
@@ -0,0 +1,86 @@
+/*
+ * Main header file for the ALSA sequencer
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ * (c) 1998-1999 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SOUND_ASEQUENCER_H
+#define __SOUND_ASEQUENCER_H
+
+#include <linux/ioctl.h>
+#include <sound/asound.h>
+#include <uapi/sound/asequencer.h>
+
+/* helper macro */
+#define snd_seq_event_bounce_ext_data(ev) ((void*)((char *)(ev)->data.ext.ptr + sizeof(struct snd_seq_event_bounce)))
+
+/*
+ * type check macros
+ */
+/* result events: 0-4 */
+#define snd_seq_ev_is_result_type(ev) ((ev)->type < 5)
+/* channel specific events: 5-19 */
+#define snd_seq_ev_is_channel_type(ev) ((ev)->type >= 5 && (ev)->type < 20)
+/* note events: 5-9 */
+#define snd_seq_ev_is_note_type(ev) ((ev)->type >= 5 && (ev)->type < 10)
+/* control events: 10-19 */
+#define snd_seq_ev_is_control_type(ev) ((ev)->type >= 10 && (ev)->type < 20)
+/* queue control events: 30-39 */
+#define snd_seq_ev_is_queue_type(ev) ((ev)->type >= 30 && (ev)->type < 40)
+/* system status messages */
+#define snd_seq_ev_is_message_type(ev) ((ev)->type >= 60 && (ev)->type < 69)
+/* sample messages */
+#define snd_seq_ev_is_sample_type(ev) ((ev)->type >= 70 && (ev)->type < 79)
+/* user-defined messages */
+#define snd_seq_ev_is_user_type(ev) ((ev)->type >= 90 && (ev)->type < 99)
+/* fixed length events: 0-99 */
+#define snd_seq_ev_is_fixed_type(ev) ((ev)->type < 100)
+/* variable length events: 130-139 */
+#define snd_seq_ev_is_variable_type(ev) ((ev)->type >= 130 && (ev)->type < 140)
+/* reserved for kernel */
+#define snd_seq_ev_is_reserved(ev) ((ev)->type >= 150)
+
+/* direct dispatched events */
+#define snd_seq_ev_is_direct(ev) ((ev)->queue == SNDRV_SEQ_QUEUE_DIRECT)
+
+/*
+ * macros to check event flags
+ */
+/* prior events */
+#define snd_seq_ev_is_prior(ev) (((ev)->flags & SNDRV_SEQ_PRIORITY_MASK) == SNDRV_SEQ_PRIORITY_HIGH)
+
+/* event length type */
+#define snd_seq_ev_length_type(ev) ((ev)->flags & SNDRV_SEQ_EVENT_LENGTH_MASK)
+#define snd_seq_ev_is_fixed(ev) (snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_FIXED)
+#define snd_seq_ev_is_variable(ev) (snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
+#define snd_seq_ev_is_varusr(ev) (snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_VARUSR)
+
+/* time-stamp type */
+#define snd_seq_ev_timestamp_type(ev) ((ev)->flags & SNDRV_SEQ_TIME_STAMP_MASK)
+#define snd_seq_ev_is_tick(ev) (snd_seq_ev_timestamp_type(ev) == SNDRV_SEQ_TIME_STAMP_TICK)
+#define snd_seq_ev_is_real(ev) (snd_seq_ev_timestamp_type(ev) == SNDRV_SEQ_TIME_STAMP_REAL)
+
+/* time-mode type */
+#define snd_seq_ev_timemode_type(ev) ((ev)->flags & SNDRV_SEQ_TIME_MODE_MASK)
+#define snd_seq_ev_is_abstime(ev) (snd_seq_ev_timemode_type(ev) == SNDRV_SEQ_TIME_MODE_ABS)
+#define snd_seq_ev_is_reltime(ev) (snd_seq_ev_timemode_type(ev) == SNDRV_SEQ_TIME_MODE_REL)
+
+/* queue sync port */
+#define snd_seq_queue_sync_port(q) ((q) + 16)
+
+#endif /* __SOUND_ASEQUENCER_H */
diff --git a/kernel/include/sound/asound.h b/kernel/include/sound/asound.h
new file mode 100644
index 000000000..c2dff5369
--- /dev/null
+++ b/kernel/include/sound/asound.h
@@ -0,0 +1,40 @@
+/*
+ * Advanced Linux Sound Architecture - ALSA - Driver
+ * Copyright (c) 1994-2003 by Jaroslav Kysela <perex@perex.cz>,
+ * Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SOUND_ASOUND_H
+#define __SOUND_ASOUND_H
+
+#include <linux/ioctl.h>
+#include <linux/time.h>
+#include <asm/byteorder.h>
+
+#ifdef __LITTLE_ENDIAN
+#define SNDRV_LITTLE_ENDIAN
+#else
+#ifdef __BIG_ENDIAN
+#define SNDRV_BIG_ENDIAN
+#else
+#error "Unsupported endian..."
+#endif
+#endif
+
+#include <uapi/sound/asound.h>
+#endif /* __SOUND_ASOUND_H */
diff --git a/kernel/include/sound/asoundef.h b/kernel/include/sound/asoundef.h
new file mode 100644
index 000000000..bb05c02f8
--- /dev/null
+++ b/kernel/include/sound/asoundef.h
@@ -0,0 +1,325 @@
+#ifndef __SOUND_ASOUNDEF_H
+#define __SOUND_ASOUNDEF_H
+
+/*
+ * Advanced Linux Sound Architecture - ALSA - Driver
+ * Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/****************************************************************************
+ * *
+ * Digital audio interface *
+ * *
+ ****************************************************************************/
+
+/* AES/IEC958 channel status bits */
+#define IEC958_AES0_PROFESSIONAL (1<<0) /* 0 = consumer, 1 = professional */
+#define IEC958_AES0_NONAUDIO (1<<1) /* 0 = audio, 1 = non-audio */
+#define IEC958_AES0_PRO_EMPHASIS (7<<2) /* mask - emphasis */
+#define IEC958_AES0_PRO_EMPHASIS_NOTID (0<<2) /* emphasis not indicated */
+#define IEC958_AES0_PRO_EMPHASIS_NONE (1<<2) /* none emphasis */
+#define IEC958_AES0_PRO_EMPHASIS_5015 (3<<2) /* 50/15us emphasis */
+#define IEC958_AES0_PRO_EMPHASIS_CCITT (7<<2) /* CCITT J.17 emphasis */
+#define IEC958_AES0_PRO_FREQ_UNLOCKED (1<<5) /* source sample frequency: 0 = locked, 1 = unlocked */
+#define IEC958_AES0_PRO_FS (3<<6) /* mask - sample frequency */
+#define IEC958_AES0_PRO_FS_NOTID (0<<6) /* fs not indicated */
+#define IEC958_AES0_PRO_FS_44100 (1<<6) /* 44.1kHz */
+#define IEC958_AES0_PRO_FS_48000 (2<<6) /* 48kHz */
+#define IEC958_AES0_PRO_FS_32000 (3<<6) /* 32kHz */
+#define IEC958_AES0_CON_NOT_COPYRIGHT (1<<2) /* 0 = copyright, 1 = not copyright */
+#define IEC958_AES0_CON_EMPHASIS (7<<3) /* mask - emphasis */
+#define IEC958_AES0_CON_EMPHASIS_NONE (0<<3) /* none emphasis */
+#define IEC958_AES0_CON_EMPHASIS_5015 (1<<3) /* 50/15us emphasis */
+#define IEC958_AES0_CON_MODE (3<<6) /* mask - mode */
+#define IEC958_AES1_PRO_MODE (15<<0) /* mask - channel mode */
+#define IEC958_AES1_PRO_MODE_NOTID (0<<0) /* not indicated */
+#define IEC958_AES1_PRO_MODE_STEREOPHONIC (2<<0) /* stereophonic - ch A is left */
+#define IEC958_AES1_PRO_MODE_SINGLE (4<<0) /* single channel */
+#define IEC958_AES1_PRO_MODE_TWO (8<<0) /* two channels */
+#define IEC958_AES1_PRO_MODE_PRIMARY (12<<0) /* primary/secondary */
+#define IEC958_AES1_PRO_MODE_BYTE3 (15<<0) /* vector to byte 3 */
+#define IEC958_AES1_PRO_USERBITS (15<<4) /* mask - user bits */
+#define IEC958_AES1_PRO_USERBITS_NOTID (0<<4) /* not indicated */
+#define IEC958_AES1_PRO_USERBITS_192 (8<<4) /* 192-bit structure */
+#define IEC958_AES1_PRO_USERBITS_UDEF (12<<4) /* user defined application */
+#define IEC958_AES1_CON_CATEGORY 0x7f
+#define IEC958_AES1_CON_GENERAL 0x00
+#define IEC958_AES1_CON_LASEROPT_MASK 0x07
+#define IEC958_AES1_CON_LASEROPT_ID 0x01
+#define IEC958_AES1_CON_IEC908_CD (IEC958_AES1_CON_LASEROPT_ID|0x00)
+#define IEC958_AES1_CON_NON_IEC908_CD (IEC958_AES1_CON_LASEROPT_ID|0x08)
+#define IEC958_AES1_CON_MINI_DISC (IEC958_AES1_CON_LASEROPT_ID|0x48)
+#define IEC958_AES1_CON_DVD (IEC958_AES1_CON_LASEROPT_ID|0x18)
+#define IEC958_AES1_CON_LASTEROPT_OTHER (IEC958_AES1_CON_LASEROPT_ID|0x78)
+#define IEC958_AES1_CON_DIGDIGCONV_MASK 0x07
+#define IEC958_AES1_CON_DIGDIGCONV_ID 0x02
+#define IEC958_AES1_CON_PCM_CODER (IEC958_AES1_CON_DIGDIGCONV_ID|0x00)
+#define IEC958_AES1_CON_MIXER (IEC958_AES1_CON_DIGDIGCONV_ID|0x10)
+#define IEC958_AES1_CON_RATE_CONVERTER (IEC958_AES1_CON_DIGDIGCONV_ID|0x18)
+#define IEC958_AES1_CON_SAMPLER (IEC958_AES1_CON_DIGDIGCONV_ID|0x20)
+#define IEC958_AES1_CON_DSP (IEC958_AES1_CON_DIGDIGCONV_ID|0x28)
+#define IEC958_AES1_CON_DIGDIGCONV_OTHER (IEC958_AES1_CON_DIGDIGCONV_ID|0x78)
+#define IEC958_AES1_CON_MAGNETIC_MASK 0x07
+#define IEC958_AES1_CON_MAGNETIC_ID 0x03
+#define IEC958_AES1_CON_DAT (IEC958_AES1_CON_MAGNETIC_ID|0x00)
+#define IEC958_AES1_CON_VCR (IEC958_AES1_CON_MAGNETIC_ID|0x08)
+#define IEC958_AES1_CON_DCC (IEC958_AES1_CON_MAGNETIC_ID|0x40)
+#define IEC958_AES1_CON_MAGNETIC_DISC (IEC958_AES1_CON_MAGNETIC_ID|0x18)
+#define IEC958_AES1_CON_MAGNETIC_OTHER (IEC958_AES1_CON_MAGNETIC_ID|0x78)
+#define IEC958_AES1_CON_BROADCAST1_MASK 0x07
+#define IEC958_AES1_CON_BROADCAST1_ID 0x04
+#define IEC958_AES1_CON_DAB_JAPAN (IEC958_AES1_CON_BROADCAST1_ID|0x00)
+#define IEC958_AES1_CON_DAB_EUROPE (IEC958_AES1_CON_BROADCAST1_ID|0x08)
+#define IEC958_AES1_CON_DAB_USA (IEC958_AES1_CON_BROADCAST1_ID|0x60)
+#define IEC958_AES1_CON_SOFTWARE (IEC958_AES1_CON_BROADCAST1_ID|0x40)
+#define IEC958_AES1_CON_IEC62105 (IEC958_AES1_CON_BROADCAST1_ID|0x20)
+#define IEC958_AES1_CON_BROADCAST1_OTHER (IEC958_AES1_CON_BROADCAST1_ID|0x78)
+#define IEC958_AES1_CON_BROADCAST2_MASK 0x0f
+#define IEC958_AES1_CON_BROADCAST2_ID 0x0e
+#define IEC958_AES1_CON_MUSICAL_MASK 0x07
+#define IEC958_AES1_CON_MUSICAL_ID 0x05
+#define IEC958_AES1_CON_SYNTHESIZER (IEC958_AES1_CON_MUSICAL_ID|0x00)
+#define IEC958_AES1_CON_MICROPHONE (IEC958_AES1_CON_MUSICAL_ID|0x08)
+#define IEC958_AES1_CON_MUSICAL_OTHER (IEC958_AES1_CON_MUSICAL_ID|0x78)
+#define IEC958_AES1_CON_ADC_MASK 0x1f
+#define IEC958_AES1_CON_ADC_ID 0x06
+#define IEC958_AES1_CON_ADC (IEC958_AES1_CON_ADC_ID|0x00)
+#define IEC958_AES1_CON_ADC_OTHER (IEC958_AES1_CON_ADC_ID|0x60)
+#define IEC958_AES1_CON_ADC_COPYRIGHT_MASK 0x1f
+#define IEC958_AES1_CON_ADC_COPYRIGHT_ID 0x16
+#define IEC958_AES1_CON_ADC_COPYRIGHT (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x00)
+#define IEC958_AES1_CON_ADC_COPYRIGHT_OTHER (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x60)
+#define IEC958_AES1_CON_SOLIDMEM_MASK 0x0f
+#define IEC958_AES1_CON_SOLIDMEM_ID 0x08
+#define IEC958_AES1_CON_SOLIDMEM_DIGITAL_RECORDER_PLAYER (IEC958_AES1_CON_SOLIDMEM_ID|0x00)
+#define IEC958_AES1_CON_SOLIDMEM_OTHER (IEC958_AES1_CON_SOLIDMEM_ID|0x70)
+#define IEC958_AES1_CON_EXPERIMENTAL 0x40
+#define IEC958_AES1_CON_ORIGINAL (1<<7) /* this bits depends on the category code */
+#define IEC958_AES2_PRO_SBITS (7<<0) /* mask - sample bits */
+#define IEC958_AES2_PRO_SBITS_20 (2<<0) /* 20-bit - coordination */
+#define IEC958_AES2_PRO_SBITS_24 (4<<0) /* 24-bit - main audio */
+#define IEC958_AES2_PRO_SBITS_UDEF (6<<0) /* user defined application */
+#define IEC958_AES2_PRO_WORDLEN (7<<3) /* mask - source word length */
+#define IEC958_AES2_PRO_WORDLEN_NOTID (0<<3) /* not indicated */
+#define IEC958_AES2_PRO_WORDLEN_22_18 (2<<3) /* 22-bit or 18-bit */
+#define IEC958_AES2_PRO_WORDLEN_23_19 (4<<3) /* 23-bit or 19-bit */
+#define IEC958_AES2_PRO_WORDLEN_24_20 (5<<3) /* 24-bit or 20-bit */
+#define IEC958_AES2_PRO_WORDLEN_20_16 (6<<3) /* 20-bit or 16-bit */
+#define IEC958_AES2_CON_SOURCE (15<<0) /* mask - source number */
+#define IEC958_AES2_CON_SOURCE_UNSPEC (0<<0) /* unspecified */
+#define IEC958_AES2_CON_CHANNEL (15<<4) /* mask - channel number */
+#define IEC958_AES2_CON_CHANNEL_UNSPEC (0<<4) /* unspecified */
+#define IEC958_AES3_CON_FS (15<<0) /* mask - sample frequency */
+#define IEC958_AES3_CON_FS_44100 (0<<0) /* 44.1kHz */
+#define IEC958_AES3_CON_FS_NOTID (1<<0) /* non indicated */
+#define IEC958_AES3_CON_FS_48000 (2<<0) /* 48kHz */
+#define IEC958_AES3_CON_FS_32000 (3<<0) /* 32kHz */
+#define IEC958_AES3_CON_FS_22050 (4<<0) /* 22.05kHz */
+#define IEC958_AES3_CON_FS_24000 (6<<0) /* 24kHz */
+#define IEC958_AES3_CON_FS_88200 (8<<0) /* 88.2kHz */
+#define IEC958_AES3_CON_FS_768000 (9<<0) /* 768kHz */
+#define IEC958_AES3_CON_FS_96000 (10<<0) /* 96kHz */
+#define IEC958_AES3_CON_FS_176400 (12<<0) /* 176.4kHz */
+#define IEC958_AES3_CON_FS_192000 (14<<0) /* 192kHz */
+#define IEC958_AES3_CON_CLOCK (3<<4) /* mask - clock accuracy */
+#define IEC958_AES3_CON_CLOCK_1000PPM (0<<4) /* 1000 ppm */
+#define IEC958_AES3_CON_CLOCK_50PPM (1<<4) /* 50 ppm */
+#define IEC958_AES3_CON_CLOCK_VARIABLE (2<<4) /* variable pitch */
+#define IEC958_AES4_CON_MAX_WORDLEN_24 (1<<0) /* 0 = 20-bit, 1 = 24-bit */
+#define IEC958_AES4_CON_WORDLEN (7<<1) /* mask - sample word length */
+#define IEC958_AES4_CON_WORDLEN_NOTID (0<<1) /* not indicated */
+#define IEC958_AES4_CON_WORDLEN_20_16 (1<<1) /* 20-bit or 16-bit */
+#define IEC958_AES4_CON_WORDLEN_22_18 (2<<1) /* 22-bit or 18-bit */
+#define IEC958_AES4_CON_WORDLEN_23_19 (4<<1) /* 23-bit or 19-bit */
+#define IEC958_AES4_CON_WORDLEN_24_20 (5<<1) /* 24-bit or 20-bit */
+#define IEC958_AES4_CON_WORDLEN_21_17 (6<<1) /* 21-bit or 17-bit */
+#define IEC958_AES4_CON_ORIGFS (15<<4) /* mask - original sample frequency */
+#define IEC958_AES4_CON_ORIGFS_NOTID (0<<4) /* not indicated */
+#define IEC958_AES4_CON_ORIGFS_192000 (1<<4) /* 192kHz */
+#define IEC958_AES4_CON_ORIGFS_12000 (2<<4) /* 12kHz */
+#define IEC958_AES4_CON_ORIGFS_176400 (3<<4) /* 176.4kHz */
+#define IEC958_AES4_CON_ORIGFS_96000 (5<<4) /* 96kHz */
+#define IEC958_AES4_CON_ORIGFS_8000 (6<<4) /* 8kHz */
+#define IEC958_AES4_CON_ORIGFS_88200 (7<<4) /* 88.2kHz */
+#define IEC958_AES4_CON_ORIGFS_16000 (8<<4) /* 16kHz */
+#define IEC958_AES4_CON_ORIGFS_24000 (9<<4) /* 24kHz */
+#define IEC958_AES4_CON_ORIGFS_11025 (10<<4) /* 11.025kHz */
+#define IEC958_AES4_CON_ORIGFS_22050 (11<<4) /* 22.05kHz */
+#define IEC958_AES4_CON_ORIGFS_32000 (12<<4) /* 32kHz */
+#define IEC958_AES4_CON_ORIGFS_48000 (13<<4) /* 48kHz */
+#define IEC958_AES4_CON_ORIGFS_44100 (15<<4) /* 44.1kHz */
+#define IEC958_AES5_CON_CGMSA (3<<0) /* mask - CGMS-A */
+#define IEC958_AES5_CON_CGMSA_COPYFREELY (0<<0) /* copying is permitted without restriction */
+#define IEC958_AES5_CON_CGMSA_COPYONCE (1<<0) /* one generation of copies may be made */
+#define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0) /* condition not be used */
+#define IEC958_AES5_CON_CGMSA_COPYNEVER (3<<0) /* no copying is permitted */
+
+/****************************************************************************
+ * *
+ * CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort *
+ * *
+ ****************************************************************************/
+#define CEA861_AUDIO_INFOFRAME_DB1CC (7<<0) /* mask - channel count */
+#define CEA861_AUDIO_INFOFRAME_DB1CT (0xf<<4) /* mask - coding type */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM (0<<4) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_IEC60958 (1<<4) /* IEC-60958 L-PCM */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AC3 (2<<4) /* AC-3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG1 (3<<4) /* MPEG1 Layers 1 & 2 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MP3 (4<<4) /* MPEG1 Layer 3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG2_MULTICH (5<<4) /* MPEG2 Multichannel */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AAC (6<<4) /* AAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS (7<<4) /* DTS */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ATRAC (8<<4) /* ATRAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ONEBIT (9<<4) /* One Bit Audio */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DOLBY_DIG_PLUS (10<<4) /* Dolby Digital + */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS_HD (11<<4) /* DTS-HD */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MAT (12<<4) /* MAT (MLP) */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DST (13<<4) /* DST */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_WMA_PRO (14<<4) /* WMA Pro */
+#define CEA861_AUDIO_INFOFRAME_DB2SF (7<<2) /* mask - sample frequency */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM (0<<2) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_32000 (1<<2) /* 32kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_44100 (2<<2) /* 44.1kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_48000 (3<<2) /* 48kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_88200 (4<<2) /* 88.2kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_96000 (5<<2) /* 96kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_176400 (6<<2) /* 176.4kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_192000 (7<<2) /* 192kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SS (3<<0) /* mask - sample size */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM (0<<0) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_16BIT (1<<0) /* 16 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_20BIT (2<<0) /* 20 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_24BIT (3<<0) /* 24 bits */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH (1<<7) /* mask - inhibit downmixing */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED (0<<7) /* stereo downmix permitted */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED (1<<7) /* stereo downmis prohibited */
+#define CEA861_AUDIO_INFOFRAME_DB5_LSV (0xf<<3) /* mask - level-shift values */
+
+/*****************************************************************************
+ * *
+ * MIDI v1.0 interface *
+ * *
+ *****************************************************************************/
+
+#define MIDI_CHANNELS 16
+#define MIDI_GM_DRUM_CHANNEL (10-1)
+
+/*
+ * MIDI commands
+ */
+
+#define MIDI_CMD_NOTE_OFF 0x80
+#define MIDI_CMD_NOTE_ON 0x90
+#define MIDI_CMD_NOTE_PRESSURE 0xa0
+#define MIDI_CMD_CONTROL 0xb0
+#define MIDI_CMD_PGM_CHANGE 0xc0
+#define MIDI_CMD_CHANNEL_PRESSURE 0xd0
+#define MIDI_CMD_BENDER 0xe0
+
+#define MIDI_CMD_COMMON_SYSEX 0xf0
+#define MIDI_CMD_COMMON_MTC_QUARTER 0xf1
+#define MIDI_CMD_COMMON_SONG_POS 0xf2
+#define MIDI_CMD_COMMON_SONG_SELECT 0xf3
+#define MIDI_CMD_COMMON_TUNE_REQUEST 0xf6
+#define MIDI_CMD_COMMON_SYSEX_END 0xf7
+#define MIDI_CMD_COMMON_CLOCK 0xf8
+#define MIDI_CMD_COMMON_START 0xfa
+#define MIDI_CMD_COMMON_CONTINUE 0xfb
+#define MIDI_CMD_COMMON_STOP 0xfc
+#define MIDI_CMD_COMMON_SENSING 0xfe
+#define MIDI_CMD_COMMON_RESET 0xff
+
+/*
+ * MIDI controllers
+ */
+
+#define MIDI_CTL_MSB_BANK 0x00
+#define MIDI_CTL_MSB_MODWHEEL 0x01
+#define MIDI_CTL_MSB_BREATH 0x02
+#define MIDI_CTL_MSB_FOOT 0x04
+#define MIDI_CTL_MSB_PORTAMENTO_TIME 0x05
+#define MIDI_CTL_MSB_DATA_ENTRY 0x06
+#define MIDI_CTL_MSB_MAIN_VOLUME 0x07
+#define MIDI_CTL_MSB_BALANCE 0x08
+#define MIDI_CTL_MSB_PAN 0x0a
+#define MIDI_CTL_MSB_EXPRESSION 0x0b
+#define MIDI_CTL_MSB_EFFECT1 0x0c
+#define MIDI_CTL_MSB_EFFECT2 0x0d
+#define MIDI_CTL_MSB_GENERAL_PURPOSE1 0x10
+#define MIDI_CTL_MSB_GENERAL_PURPOSE2 0x11
+#define MIDI_CTL_MSB_GENERAL_PURPOSE3 0x12
+#define MIDI_CTL_MSB_GENERAL_PURPOSE4 0x13
+#define MIDI_CTL_LSB_BANK 0x20
+#define MIDI_CTL_LSB_MODWHEEL 0x21
+#define MIDI_CTL_LSB_BREATH 0x22
+#define MIDI_CTL_LSB_FOOT 0x24
+#define MIDI_CTL_LSB_PORTAMENTO_TIME 0x25
+#define MIDI_CTL_LSB_DATA_ENTRY 0x26
+#define MIDI_CTL_LSB_MAIN_VOLUME 0x27
+#define MIDI_CTL_LSB_BALANCE 0x28
+#define MIDI_CTL_LSB_PAN 0x2a
+#define MIDI_CTL_LSB_EXPRESSION 0x2b
+#define MIDI_CTL_LSB_EFFECT1 0x2c
+#define MIDI_CTL_LSB_EFFECT2 0x2d
+#define MIDI_CTL_LSB_GENERAL_PURPOSE1 0x30
+#define MIDI_CTL_LSB_GENERAL_PURPOSE2 0x31
+#define MIDI_CTL_LSB_GENERAL_PURPOSE3 0x32
+#define MIDI_CTL_LSB_GENERAL_PURPOSE4 0x33
+#define MIDI_CTL_SUSTAIN 0x40
+#define MIDI_CTL_PORTAMENTO 0x41
+#define MIDI_CTL_SOSTENUTO 0x42
+#define MIDI_CTL_SOFT_PEDAL 0x43
+#define MIDI_CTL_LEGATO_FOOTSWITCH 0x44
+#define MIDI_CTL_HOLD2 0x45
+#define MIDI_CTL_SC1_SOUND_VARIATION 0x46
+#define MIDI_CTL_SC2_TIMBRE 0x47
+#define MIDI_CTL_SC3_RELEASE_TIME 0x48
+#define MIDI_CTL_SC4_ATTACK_TIME 0x49
+#define MIDI_CTL_SC5_BRIGHTNESS 0x4a
+#define MIDI_CTL_SC6 0x4b
+#define MIDI_CTL_SC7 0x4c
+#define MIDI_CTL_SC8 0x4d
+#define MIDI_CTL_SC9 0x4e
+#define MIDI_CTL_SC10 0x4f
+#define MIDI_CTL_GENERAL_PURPOSE5 0x50
+#define MIDI_CTL_GENERAL_PURPOSE6 0x51
+#define MIDI_CTL_GENERAL_PURPOSE7 0x52
+#define MIDI_CTL_GENERAL_PURPOSE8 0x53
+#define MIDI_CTL_PORTAMENTO_CONTROL 0x54
+#define MIDI_CTL_E1_REVERB_DEPTH 0x5b
+#define MIDI_CTL_E2_TREMOLO_DEPTH 0x5c
+#define MIDI_CTL_E3_CHORUS_DEPTH 0x5d
+#define MIDI_CTL_E4_DETUNE_DEPTH 0x5e
+#define MIDI_CTL_E5_PHASER_DEPTH 0x5f
+#define MIDI_CTL_DATA_INCREMENT 0x60
+#define MIDI_CTL_DATA_DECREMENT 0x61
+#define MIDI_CTL_NONREG_PARM_NUM_LSB 0x62
+#define MIDI_CTL_NONREG_PARM_NUM_MSB 0x63
+#define MIDI_CTL_REGIST_PARM_NUM_LSB 0x64
+#define MIDI_CTL_REGIST_PARM_NUM_MSB 0x65
+#define MIDI_CTL_ALL_SOUNDS_OFF 0x78
+#define MIDI_CTL_RESET_CONTROLLERS 0x79
+#define MIDI_CTL_LOCAL_CONTROL_SWITCH 0x7a
+#define MIDI_CTL_ALL_NOTES_OFF 0x7b
+#define MIDI_CTL_OMNI_OFF 0x7c
+#define MIDI_CTL_OMNI_ON 0x7d
+#define MIDI_CTL_MONO1 0x7e
+#define MIDI_CTL_MONO2 0x7f
+
+#endif /* __SOUND_ASOUNDEF_H */
diff --git a/kernel/include/sound/atmel-abdac.h b/kernel/include/sound/atmel-abdac.h
new file mode 100644
index 000000000..a8f735d67
--- /dev/null
+++ b/kernel/include/sound/atmel-abdac.h
@@ -0,0 +1,23 @@
+/*
+ * Driver for the Atmel Audio Bitstream DAC (ABDAC)
+ *
+ * Copyright (C) 2009 Atmel Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+#ifndef __INCLUDE_SOUND_ATMEL_ABDAC_H
+#define __INCLUDE_SOUND_ATMEL_ABDAC_H
+
+#include <linux/platform_data/dma-dw.h>
+
+/**
+ * struct atmel_abdac_pdata - board specific ABDAC configuration
+ * @dws: DMA slave interface to use for sound playback.
+ */
+struct atmel_abdac_pdata {
+ struct dw_dma_slave dws;
+};
+
+#endif /* __INCLUDE_SOUND_ATMEL_ABDAC_H */
diff --git a/kernel/include/sound/atmel-ac97c.h b/kernel/include/sound/atmel-ac97c.h
new file mode 100644
index 000000000..f2a1cdc37
--- /dev/null
+++ b/kernel/include/sound/atmel-ac97c.h
@@ -0,0 +1,38 @@
+/*
+ * Driver for the Atmel AC97C controller
+ *
+ * Copyright (C) 2005-2009 Atmel Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+#ifndef __INCLUDE_SOUND_ATMEL_AC97C_H
+#define __INCLUDE_SOUND_ATMEL_AC97C_H
+
+#include <linux/platform_data/dma-dw.h>
+
+#define AC97C_CAPTURE 0x01
+#define AC97C_PLAYBACK 0x02
+#define AC97C_BOTH (AC97C_CAPTURE | AC97C_PLAYBACK)
+
+/**
+ * struct atmel_ac97c_pdata - board specific AC97C configuration
+ * @rx_dws: DMA slave interface to use for sound capture.
+ * @tx_dws: DMA slave interface to use for sound playback.
+ * @reset_pin: GPIO pin wired to the reset input on the external AC97 codec,
+ * optional to use, set to -ENODEV if not in use. AC97 layer will
+ * try to do a software reset of the external codec anyway.
+ *
+ * If the user do not want to use a DMA channel for playback or capture, i.e.
+ * only one feature is required on the board. The slave for playback or capture
+ * can be set to NULL. The AC97C driver will take use of this when setting up
+ * the sound streams.
+ */
+struct ac97c_platform_data {
+ struct dw_dma_slave rx_dws;
+ struct dw_dma_slave tx_dws;
+ int reset_pin;
+};
+
+#endif /* __INCLUDE_SOUND_ATMEL_AC97C_H */
diff --git a/kernel/include/sound/compress_driver.h b/kernel/include/sound/compress_driver.h
new file mode 100644
index 000000000..fa1d05512
--- /dev/null
+++ b/kernel/include/sound/compress_driver.h
@@ -0,0 +1,185 @@
+/*
+ * compress_driver.h - compress offload driver definations
+ *
+ * Copyright (C) 2011 Intel Corporation
+ * Authors: Vinod Koul <vinod.koul@linux.intel.com>
+ * Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+#ifndef __COMPRESS_DRIVER_H
+#define __COMPRESS_DRIVER_H
+
+#include <linux/types.h>
+#include <linux/sched.h>
+#include <sound/core.h>
+#include <sound/compress_offload.h>
+#include <sound/asound.h>
+#include <sound/pcm.h>
+
+struct snd_compr_ops;
+
+/**
+ * struct snd_compr_runtime: runtime stream description
+ * @state: stream state
+ * @ops: pointer to DSP callbacks
+ * @buffer: pointer to kernel buffer, valid only when not in mmap mode or
+ * DSP doesn't implement copy
+ * @buffer_size: size of the above buffer
+ * @fragment_size: size of buffer fragment in bytes
+ * @fragments: number of such fragments
+ * @total_bytes_available: cumulative number of bytes made available in
+ * the ring buffer
+ * @total_bytes_transferred: cumulative bytes transferred by offload DSP
+ * @sleep: poll sleep
+ * @private_data: driver private data pointer
+ */
+struct snd_compr_runtime {
+ snd_pcm_state_t state;
+ struct snd_compr_ops *ops;
+ void *buffer;
+ u64 buffer_size;
+ u32 fragment_size;
+ u32 fragments;
+ u64 total_bytes_available;
+ u64 total_bytes_transferred;
+ wait_queue_head_t sleep;
+ void *private_data;
+};
+
+/**
+ * struct snd_compr_stream: compressed stream
+ * @name: device name
+ * @ops: pointer to DSP callbacks
+ * @runtime: pointer to runtime structure
+ * @device: device pointer
+ * @direction: stream direction, playback/recording
+ * @metadata_set: metadata set flag, true when set
+ * @next_track: has userspace signal next track transition, true when set
+ * @private_data: pointer to DSP private data
+ */
+struct snd_compr_stream {
+ const char *name;
+ struct snd_compr_ops *ops;
+ struct snd_compr_runtime *runtime;
+ struct snd_compr *device;
+ enum snd_compr_direction direction;
+ bool metadata_set;
+ bool next_track;
+ void *private_data;
+};
+
+/**
+ * struct snd_compr_ops: compressed path DSP operations
+ * @open: Open the compressed stream
+ * This callback is mandatory and shall keep dsp ready to receive the stream
+ * parameter
+ * @free: Close the compressed stream, mandatory
+ * @set_params: Sets the compressed stream parameters, mandatory
+ * This can be called in during stream creation only to set codec params
+ * and the stream properties
+ * @get_params: retrieve the codec parameters, mandatory
+ * @set_metadata: Set the metadata values for a stream
+ * @get_metadata: retrieves the requested metadata values from stream
+ * @trigger: Trigger operations like start, pause, resume, drain, stop.
+ * This callback is mandatory
+ * @pointer: Retrieve current h/w pointer information. Mandatory
+ * @copy: Copy the compressed data to/from userspace, Optional
+ * Can't be implemented if DSP supports mmap
+ * @mmap: DSP mmap method to mmap DSP memory
+ * @ack: Ack for DSP when data is written to audio buffer, Optional
+ * Not valid if copy is implemented
+ * @get_caps: Retrieve DSP capabilities, mandatory
+ * @get_codec_caps: Retrieve capabilities for a specific codec, mandatory
+ */
+struct snd_compr_ops {
+ int (*open)(struct snd_compr_stream *stream);
+ int (*free)(struct snd_compr_stream *stream);
+ int (*set_params)(struct snd_compr_stream *stream,
+ struct snd_compr_params *params);
+ int (*get_params)(struct snd_compr_stream *stream,
+ struct snd_codec *params);
+ int (*set_metadata)(struct snd_compr_stream *stream,
+ struct snd_compr_metadata *metadata);
+ int (*get_metadata)(struct snd_compr_stream *stream,
+ struct snd_compr_metadata *metadata);
+ int (*trigger)(struct snd_compr_stream *stream, int cmd);
+ int (*pointer)(struct snd_compr_stream *stream,
+ struct snd_compr_tstamp *tstamp);
+ int (*copy)(struct snd_compr_stream *stream, char __user *buf,
+ size_t count);
+ int (*mmap)(struct snd_compr_stream *stream,
+ struct vm_area_struct *vma);
+ int (*ack)(struct snd_compr_stream *stream, size_t bytes);
+ int (*get_caps) (struct snd_compr_stream *stream,
+ struct snd_compr_caps *caps);
+ int (*get_codec_caps) (struct snd_compr_stream *stream,
+ struct snd_compr_codec_caps *codec);
+};
+
+/**
+ * struct snd_compr: Compressed device
+ * @name: DSP device name
+ * @dev: associated device instance
+ * @ops: pointer to DSP callbacks
+ * @private_data: pointer to DSP pvt data
+ * @card: sound card pointer
+ * @direction: Playback or capture direction
+ * @lock: device lock
+ * @device: device id
+ */
+struct snd_compr {
+ const char *name;
+ struct device dev;
+ struct snd_compr_ops *ops;
+ void *private_data;
+ struct snd_card *card;
+ unsigned int direction;
+ struct mutex lock;
+ int device;
+};
+
+/* compress device register APIs */
+int snd_compress_register(struct snd_compr *device);
+int snd_compress_deregister(struct snd_compr *device);
+int snd_compress_new(struct snd_card *card, int device,
+ int type, struct snd_compr *compr);
+
+/* dsp driver callback apis
+ * For playback: driver should call snd_compress_fragment_elapsed() to let the
+ * framework know that a fragment has been consumed from the ring buffer
+ *
+ * For recording: we want to know when a frame is available or when
+ * at least one frame is available so snd_compress_frame_elapsed()
+ * callback should be called when a encodeded frame is available
+ */
+static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream)
+{
+ wake_up(&stream->runtime->sleep);
+}
+
+static inline void snd_compr_drain_notify(struct snd_compr_stream *stream)
+{
+ if (snd_BUG_ON(!stream))
+ return;
+
+ stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+ wake_up(&stream->runtime->sleep);
+}
+
+#endif
diff --git a/kernel/include/sound/control.h b/kernel/include/sound/control.h
new file mode 100644
index 000000000..95aad6d3f
--- /dev/null
+++ b/kernel/include/sound/control.h
@@ -0,0 +1,259 @@
+#ifndef __SOUND_CONTROL_H
+#define __SOUND_CONTROL_H
+
+/*
+ * Header file for control interface
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/asound.h>
+
+#define snd_kcontrol_chip(kcontrol) ((kcontrol)->private_data)
+
+struct snd_kcontrol;
+typedef int (snd_kcontrol_info_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_info * uinfo);
+typedef int (snd_kcontrol_get_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
+typedef int (snd_kcontrol_put_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
+typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol,
+ int op_flag, /* SNDRV_CTL_TLV_OP_XXX */
+ unsigned int size,
+ unsigned int __user *tlv);
+
+enum {
+ SNDRV_CTL_TLV_OP_READ = 0,
+ SNDRV_CTL_TLV_OP_WRITE = 1,
+ SNDRV_CTL_TLV_OP_CMD = -1,
+};
+
+struct snd_kcontrol_new {
+ snd_ctl_elem_iface_t iface; /* interface identifier */
+ unsigned int device; /* device/client number */
+ unsigned int subdevice; /* subdevice (substream) number */
+ const unsigned char *name; /* ASCII name of item */
+ unsigned int index; /* index of item */
+ unsigned int access; /* access rights */
+ unsigned int count; /* count of same elements */
+ snd_kcontrol_info_t *info;
+ snd_kcontrol_get_t *get;
+ snd_kcontrol_put_t *put;
+ union {
+ snd_kcontrol_tlv_rw_t *c;
+ const unsigned int *p;
+ } tlv;
+ unsigned long private_value;
+};
+
+struct snd_kcontrol_volatile {
+ struct snd_ctl_file *owner; /* locked */
+ unsigned int access; /* access rights */
+};
+
+struct snd_kcontrol {
+ struct list_head list; /* list of controls */
+ struct snd_ctl_elem_id id;
+ unsigned int count; /* count of same elements */
+ snd_kcontrol_info_t *info;
+ snd_kcontrol_get_t *get;
+ snd_kcontrol_put_t *put;
+ union {
+ snd_kcontrol_tlv_rw_t *c;
+ const unsigned int *p;
+ } tlv;
+ unsigned long private_value;
+ void *private_data;
+ void (*private_free)(struct snd_kcontrol *kcontrol);
+ struct snd_kcontrol_volatile vd[0]; /* volatile data */
+};
+
+#define snd_kcontrol(n) list_entry(n, struct snd_kcontrol, list)
+
+struct snd_kctl_event {
+ struct list_head list; /* list of events */
+ struct snd_ctl_elem_id id;
+ unsigned int mask;
+};
+
+#define snd_kctl_event(n) list_entry(n, struct snd_kctl_event, list)
+
+struct pid;
+
+enum {
+ SND_CTL_SUBDEV_PCM,
+ SND_CTL_SUBDEV_RAWMIDI,
+ SND_CTL_SUBDEV_ITEMS,
+};
+
+struct snd_ctl_file {
+ struct list_head list; /* list of all control files */
+ struct snd_card *card;
+ struct pid *pid;
+ int preferred_subdevice[SND_CTL_SUBDEV_ITEMS];
+ wait_queue_head_t change_sleep;
+ spinlock_t read_lock;
+ struct fasync_struct *fasync;
+ int subscribed; /* read interface is activated */
+ struct list_head events; /* waiting events for read */
+};
+
+#define snd_ctl_file(n) list_entry(n, struct snd_ctl_file, list)
+
+typedef int (*snd_kctl_ioctl_func_t) (struct snd_card * card,
+ struct snd_ctl_file * control,
+ unsigned int cmd, unsigned long arg);
+
+void snd_ctl_notify(struct snd_card * card, unsigned int mask, struct snd_ctl_elem_id * id);
+
+struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, void * private_data);
+void snd_ctl_free_one(struct snd_kcontrol * kcontrol);
+int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol);
+int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol);
+int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, bool add_on_replace);
+int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id);
+int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id);
+int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id,
+ int active);
+struct snd_kcontrol *snd_ctl_find_numid(struct snd_card * card, unsigned int numid);
+struct snd_kcontrol *snd_ctl_find_id(struct snd_card * card, struct snd_ctl_elem_id *id);
+
+int snd_ctl_create(struct snd_card *card);
+
+int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn);
+int snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn);
+#ifdef CONFIG_COMPAT
+int snd_ctl_register_ioctl_compat(snd_kctl_ioctl_func_t fcn);
+int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn);
+#else
+#define snd_ctl_register_ioctl_compat(fcn)
+#define snd_ctl_unregister_ioctl_compat(fcn)
+#endif
+
+int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type);
+
+static inline unsigned int snd_ctl_get_ioffnum(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
+{
+ return id->numid - kctl->id.numid;
+}
+
+static inline unsigned int snd_ctl_get_ioffidx(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
+{
+ return id->index - kctl->id.index;
+}
+
+static inline unsigned int snd_ctl_get_ioff(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
+{
+ if (id->numid) {
+ return snd_ctl_get_ioffnum(kctl, id);
+ } else {
+ return snd_ctl_get_ioffidx(kctl, id);
+ }
+}
+
+static inline struct snd_ctl_elem_id *snd_ctl_build_ioff(struct snd_ctl_elem_id *dst_id,
+ struct snd_kcontrol *src_kctl,
+ unsigned int offset)
+{
+ *dst_id = src_kctl->id;
+ dst_id->index += offset;
+ dst_id->numid += offset;
+ return dst_id;
+}
+
+/*
+ * Frequently used control callbacks/helpers
+ */
+int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels,
+ unsigned int items, const char *const names[]);
+
+/*
+ * virtual master control
+ */
+struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
+ const unsigned int *tlv);
+int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
+ unsigned int flags);
+/* optional flags for slave */
+#define SND_CTL_SLAVE_NEED_UPDATE (1 << 0)
+
+/**
+ * snd_ctl_add_slave - Add a virtual slave control
+ * @master: vmaster element
+ * @slave: slave element to add
+ *
+ * Add a virtual slave control to the given master element created via
+ * snd_ctl_create_virtual_master() beforehand.
+ *
+ * All slaves must be the same type (returning the same information
+ * via info callback). The function doesn't check it, so it's your
+ * responsibility.
+ *
+ * Also, some additional limitations:
+ * at most two channels,
+ * logarithmic volume control (dB level) thus no linear volume,
+ * master can only attenuate the volume without gain
+ *
+ * Return: Zero if successful or a negative error code.
+ */
+static inline int
+snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
+{
+ return _snd_ctl_add_slave(master, slave, 0);
+}
+
+/**
+ * snd_ctl_add_slave_uncached - Add a virtual slave control
+ * @master: vmaster element
+ * @slave: slave element to add
+ *
+ * Add a virtual slave control to the given master.
+ * Unlike snd_ctl_add_slave(), the element added via this function
+ * is supposed to have volatile values, and get callback is called
+ * at each time queried from the master.
+ *
+ * When the control peeks the hardware values directly and the value
+ * can be changed by other means than the put callback of the element,
+ * this function should be used to keep the value always up-to-date.
+ *
+ * Return: Zero if successful or a negative error code.
+ */
+static inline int
+snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
+ struct snd_kcontrol *slave)
+{
+ return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
+}
+
+int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
+ void (*hook)(void *private_data, int),
+ void *private_data);
+void snd_ctl_sync_vmaster(struct snd_kcontrol *kctl, bool hook_only);
+#define snd_ctl_sync_vmaster_hook(kctl) snd_ctl_sync_vmaster(kctl, true)
+
+/*
+ * Helper functions for jack-detection controls
+ */
+struct snd_kcontrol *
+snd_kctl_jack_new(const char *name, int idx, void *private_data);
+void snd_kctl_jack_report(struct snd_card *card,
+ struct snd_kcontrol *kctl, bool status);
+
+#endif /* __SOUND_CONTROL_H */
diff --git a/kernel/include/sound/core.h b/kernel/include/sound/core.h
new file mode 100644
index 000000000..b12931f51
--- /dev/null
+++ b/kernel/include/sound/core.h
@@ -0,0 +1,463 @@
+#ifndef __SOUND_CORE_H
+#define __SOUND_CORE_H
+
+/*
+ * Main header file for the ALSA driver
+ * Copyright (c) 1994-2001 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/device.h>
+#include <linux/sched.h> /* wake_up() */
+#include <linux/mutex.h> /* struct mutex */
+#include <linux/rwsem.h> /* struct rw_semaphore */
+#include <linux/pm.h> /* pm_message_t */
+#include <linux/stringify.h>
+#include <linux/printk.h>
+
+/* number of supported soundcards */
+#ifdef CONFIG_SND_DYNAMIC_MINORS
+#define SNDRV_CARDS CONFIG_SND_MAX_CARDS
+#else
+#define SNDRV_CARDS 8 /* don't change - minor numbers */
+#endif
+
+#define CONFIG_SND_MAJOR 116 /* standard configuration */
+
+/* forward declarations */
+struct pci_dev;
+struct module;
+struct completion;
+
+/* device allocation stuff */
+
+/* type of the object used in snd_device_*()
+ * this also defines the calling order
+ */
+enum snd_device_type {
+ SNDRV_DEV_LOWLEVEL,
+ SNDRV_DEV_CONTROL,
+ SNDRV_DEV_INFO,
+ SNDRV_DEV_BUS,
+ SNDRV_DEV_CODEC,
+ SNDRV_DEV_PCM,
+ SNDRV_DEV_COMPRESS,
+ SNDRV_DEV_RAWMIDI,
+ SNDRV_DEV_TIMER,
+ SNDRV_DEV_SEQUENCER,
+ SNDRV_DEV_HWDEP,
+ SNDRV_DEV_JACK,
+};
+
+enum snd_device_state {
+ SNDRV_DEV_BUILD,
+ SNDRV_DEV_REGISTERED,
+ SNDRV_DEV_DISCONNECTED,
+};
+
+struct snd_device;
+
+struct snd_device_ops {
+ int (*dev_free)(struct snd_device *dev);
+ int (*dev_register)(struct snd_device *dev);
+ int (*dev_disconnect)(struct snd_device *dev);
+};
+
+struct snd_device {
+ struct list_head list; /* list of registered devices */
+ struct snd_card *card; /* card which holds this device */
+ enum snd_device_state state; /* state of the device */
+ enum snd_device_type type; /* device type */
+ void *device_data; /* device structure */
+ struct snd_device_ops *ops; /* operations */
+};
+
+#define snd_device(n) list_entry(n, struct snd_device, list)
+
+/* main structure for soundcard */
+
+struct snd_card {
+ int number; /* number of soundcard (index to
+ snd_cards) */
+
+ char id[16]; /* id string of this card */
+ char driver[16]; /* driver name */
+ char shortname[32]; /* short name of this soundcard */
+ char longname[80]; /* name of this soundcard */
+ char mixername[80]; /* mixer name */
+ char components[128]; /* card components delimited with
+ space */
+ struct module *module; /* top-level module */
+
+ void *private_data; /* private data for soundcard */
+ void (*private_free) (struct snd_card *card); /* callback for freeing of
+ private data */
+ struct list_head devices; /* devices */
+
+ struct device ctl_dev; /* control device */
+ unsigned int last_numid; /* last used numeric ID */
+ struct rw_semaphore controls_rwsem; /* controls list lock */
+ rwlock_t ctl_files_rwlock; /* ctl_files list lock */
+ int controls_count; /* count of all controls */
+ int user_ctl_count; /* count of all user controls */
+ struct list_head controls; /* all controls for this card */
+ struct list_head ctl_files; /* active control files */
+ struct mutex user_ctl_lock; /* protects user controls against
+ concurrent access */
+
+ struct snd_info_entry *proc_root; /* root for soundcard specific files */
+ struct snd_info_entry *proc_id; /* the card id */
+ struct proc_dir_entry *proc_root_link; /* number link to real id */
+
+ struct list_head files_list; /* all files associated to this card */
+ struct snd_shutdown_f_ops *s_f_ops; /* file operations in the shutdown
+ state */
+ spinlock_t files_lock; /* lock the files for this card */
+ int shutdown; /* this card is going down */
+ struct completion *release_completion;
+ struct device *dev; /* device assigned to this card */
+ struct device card_dev; /* cardX object for sysfs */
+ const struct attribute_group *dev_groups[4]; /* assigned sysfs attr */
+ bool registered; /* card_dev is registered? */
+
+#ifdef CONFIG_PM
+ unsigned int power_state; /* power state */
+ struct mutex power_lock; /* power lock */
+ wait_queue_head_t power_sleep;
+#endif
+
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+ struct snd_mixer_oss *mixer_oss;
+ int mixer_oss_change_count;
+#endif
+};
+
+#define dev_to_snd_card(p) container_of(p, struct snd_card, card_dev)
+
+#ifdef CONFIG_PM
+static inline void snd_power_lock(struct snd_card *card)
+{
+ mutex_lock(&card->power_lock);
+}
+
+static inline void snd_power_unlock(struct snd_card *card)
+{
+ mutex_unlock(&card->power_lock);
+}
+
+static inline unsigned int snd_power_get_state(struct snd_card *card)
+{
+ return card->power_state;
+}
+
+static inline void snd_power_change_state(struct snd_card *card, unsigned int state)
+{
+ card->power_state = state;
+ wake_up(&card->power_sleep);
+}
+
+/* init.c */
+int snd_power_wait(struct snd_card *card, unsigned int power_state);
+
+#else /* ! CONFIG_PM */
+
+#define snd_power_lock(card) do { (void)(card); } while (0)
+#define snd_power_unlock(card) do { (void)(card); } while (0)
+static inline int snd_power_wait(struct snd_card *card, unsigned int state) { return 0; }
+#define snd_power_get_state(card) ({ (void)(card); SNDRV_CTL_POWER_D0; })
+#define snd_power_change_state(card, state) do { (void)(card); } while (0)
+
+#endif /* CONFIG_PM */
+
+struct snd_minor {
+ int type; /* SNDRV_DEVICE_TYPE_XXX */
+ int card; /* card number */
+ int device; /* device number */
+ const struct file_operations *f_ops; /* file operations */
+ void *private_data; /* private data for f_ops->open */
+ struct device *dev; /* device for sysfs */
+ struct snd_card *card_ptr; /* assigned card instance */
+};
+
+/* return a device pointer linked to each sound device as a parent */
+static inline struct device *snd_card_get_device_link(struct snd_card *card)
+{
+ return card ? &card->card_dev : NULL;
+}
+
+/* sound.c */
+
+extern int snd_major;
+extern int snd_ecards_limit;
+extern struct class *sound_class;
+
+void snd_request_card(int card);
+
+void snd_device_initialize(struct device *dev, struct snd_card *card);
+
+int snd_register_device(int type, struct snd_card *card, int dev,
+ const struct file_operations *f_ops,
+ void *private_data, struct device *device);
+int snd_unregister_device(struct device *dev);
+void *snd_lookup_minor_data(unsigned int minor, int type);
+
+#ifdef CONFIG_SND_OSSEMUL
+int snd_register_oss_device(int type, struct snd_card *card, int dev,
+ const struct file_operations *f_ops, void *private_data);
+int snd_unregister_oss_device(int type, struct snd_card *card, int dev);
+void *snd_lookup_oss_minor_data(unsigned int minor, int type);
+#endif
+
+int snd_minor_info_init(void);
+int snd_minor_info_done(void);
+
+/* sound_oss.c */
+
+#ifdef CONFIG_SND_OSSEMUL
+int snd_minor_info_oss_init(void);
+int snd_minor_info_oss_done(void);
+#else
+static inline int snd_minor_info_oss_init(void) { return 0; }
+static inline int snd_minor_info_oss_done(void) { return 0; }
+#endif
+
+/* memory.c */
+
+int copy_to_user_fromio(void __user *dst, const volatile void __iomem *src, size_t count);
+int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size_t count);
+
+/* init.c */
+
+extern struct snd_card *snd_cards[SNDRV_CARDS];
+int snd_card_locked(int card);
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+#define SND_MIXER_OSS_NOTIFY_REGISTER 0
+#define SND_MIXER_OSS_NOTIFY_DISCONNECT 1
+#define SND_MIXER_OSS_NOTIFY_FREE 2
+extern int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int cmd);
+#endif
+
+int snd_card_new(struct device *parent, int idx, const char *xid,
+ struct module *module, int extra_size,
+ struct snd_card **card_ret);
+
+int snd_card_disconnect(struct snd_card *card);
+int snd_card_free(struct snd_card *card);
+int snd_card_free_when_closed(struct snd_card *card);
+void snd_card_set_id(struct snd_card *card, const char *id);
+int snd_card_register(struct snd_card *card);
+int snd_card_info_init(void);
+int snd_card_info_done(void);
+int snd_card_add_dev_attr(struct snd_card *card,
+ const struct attribute_group *group);
+int snd_component_add(struct snd_card *card, const char *component);
+int snd_card_file_add(struct snd_card *card, struct file *file);
+int snd_card_file_remove(struct snd_card *card, struct file *file);
+#define snd_card_unref(card) put_device(&(card)->card_dev)
+
+#define snd_card_set_dev(card, devptr) ((card)->dev = (devptr))
+
+/* device.c */
+
+int snd_device_new(struct snd_card *card, enum snd_device_type type,
+ void *device_data, struct snd_device_ops *ops);
+int snd_device_register(struct snd_card *card, void *device_data);
+int snd_device_register_all(struct snd_card *card);
+void snd_device_disconnect(struct snd_card *card, void *device_data);
+void snd_device_disconnect_all(struct snd_card *card);
+void snd_device_free(struct snd_card *card, void *device_data);
+void snd_device_free_all(struct snd_card *card);
+
+/* isadma.c */
+
+#ifdef CONFIG_ISA_DMA_API
+#define DMA_MODE_NO_ENABLE 0x0100
+
+void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode);
+void snd_dma_disable(unsigned long dma);
+unsigned int snd_dma_pointer(unsigned long dma, unsigned int size);
+#endif
+
+/* misc.c */
+struct resource;
+void release_and_free_resource(struct resource *res);
+
+/* --- */
+
+/* sound printk debug levels */
+enum {
+ SND_PR_ALWAYS,
+ SND_PR_DEBUG,
+ SND_PR_VERBOSE,
+};
+
+#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
+__printf(4, 5)
+void __snd_printk(unsigned int level, const char *file, int line,
+ const char *format, ...);
+#else
+#define __snd_printk(level, file, line, format, args...) \
+ printk(format, ##args)
+#endif
+
+/**
+ * snd_printk - printk wrapper
+ * @fmt: format string
+ *
+ * Works like printk() but prints the file and the line of the caller
+ * when configured with CONFIG_SND_VERBOSE_PRINTK.
+ */
+#define snd_printk(fmt, args...) \
+ __snd_printk(0, __FILE__, __LINE__, fmt, ##args)
+
+#ifdef CONFIG_SND_DEBUG
+/**
+ * snd_printd - debug printk
+ * @fmt: format string
+ *
+ * Works like snd_printk() for debugging purposes.
+ * Ignored when CONFIG_SND_DEBUG is not set.
+ */
+#define snd_printd(fmt, args...) \
+ __snd_printk(1, __FILE__, __LINE__, fmt, ##args)
+#define _snd_printd(level, fmt, args...) \
+ __snd_printk(level, __FILE__, __LINE__, fmt, ##args)
+
+/**
+ * snd_BUG - give a BUG warning message and stack trace
+ *
+ * Calls WARN() if CONFIG_SND_DEBUG is set.
+ * Ignored when CONFIG_SND_DEBUG is not set.
+ */
+#define snd_BUG() WARN(1, "BUG?\n")
+
+/**
+ * Suppress high rates of output when CONFIG_SND_DEBUG is enabled.
+ */
+#define snd_printd_ratelimit() printk_ratelimit()
+
+/**
+ * snd_BUG_ON - debugging check macro
+ * @cond: condition to evaluate
+ *
+ * Has the same behavior as WARN_ON when CONFIG_SND_DEBUG is set,
+ * otherwise just evaluates the conditional and returns the value.
+ */
+#define snd_BUG_ON(cond) WARN_ON((cond))
+
+#else /* !CONFIG_SND_DEBUG */
+
+__printf(1, 2)
+static inline void snd_printd(const char *format, ...) {}
+__printf(2, 3)
+static inline void _snd_printd(int level, const char *format, ...) {}
+
+#define snd_BUG() do { } while (0)
+
+#define snd_BUG_ON(condition) ({ \
+ int __ret_warn_on = !!(condition); \
+ unlikely(__ret_warn_on); \
+})
+
+static inline bool snd_printd_ratelimit(void) { return false; }
+
+#endif /* CONFIG_SND_DEBUG */
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+/**
+ * snd_printdd - debug printk
+ * @format: format string
+ *
+ * Works like snd_printk() for debugging purposes.
+ * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
+ */
+#define snd_printdd(format, args...) \
+ __snd_printk(2, __FILE__, __LINE__, format, ##args)
+#else
+__printf(1, 2)
+static inline void snd_printdd(const char *format, ...) {}
+#endif
+
+
+#define SNDRV_OSS_VERSION ((3<<16)|(8<<8)|(1<<4)|(0)) /* 3.8.1a */
+
+/* for easier backward-porting */
+#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE)
+#define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev))
+#define gameport_set_port_data(gp,r) ((gp)->port_data = (r))
+#define gameport_get_port_data(gp) (gp)->port_data
+#endif
+
+/* PCI quirk list helper */
+struct snd_pci_quirk {
+ unsigned short subvendor; /* PCI subvendor ID */
+ unsigned short subdevice; /* PCI subdevice ID */
+ unsigned short subdevice_mask; /* bitmask to match */
+ int value; /* value */
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ const char *name; /* name of the device (optional) */
+#endif
+};
+
+#define _SND_PCI_QUIRK_ID_MASK(vend, mask, dev) \
+ .subvendor = (vend), .subdevice = (dev), .subdevice_mask = (mask)
+#define _SND_PCI_QUIRK_ID(vend, dev) \
+ _SND_PCI_QUIRK_ID_MASK(vend, 0xffff, dev)
+#define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)}
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+#define SND_PCI_QUIRK(vend,dev,xname,val) \
+ {_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)}
+#define SND_PCI_QUIRK_VENDOR(vend, xname, val) \
+ {_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val), .name = (xname)}
+#define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val) \
+ {_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), \
+ .value = (val), .name = (xname)}
+#define snd_pci_quirk_name(q) ((q)->name)
+#else
+#define SND_PCI_QUIRK(vend,dev,xname,val) \
+ {_SND_PCI_QUIRK_ID(vend, dev), .value = (val)}
+#define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val) \
+ {_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), .value = (val)}
+#define SND_PCI_QUIRK_VENDOR(vend, xname, val) \
+ {_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)}
+#define snd_pci_quirk_name(q) ""
+#endif
+
+#ifdef CONFIG_PCI
+const struct snd_pci_quirk *
+snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list);
+
+const struct snd_pci_quirk *
+snd_pci_quirk_lookup_id(u16 vendor, u16 device,
+ const struct snd_pci_quirk *list);
+#else
+static inline const struct snd_pci_quirk *
+snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list)
+{
+ return NULL;
+}
+
+static inline const struct snd_pci_quirk *
+snd_pci_quirk_lookup_id(u16 vendor, u16 device,
+ const struct snd_pci_quirk *list)
+{
+ return NULL;
+}
+#endif
+
+#endif /* __SOUND_CORE_H */
diff --git a/kernel/include/sound/cs4231-regs.h b/kernel/include/sound/cs4231-regs.h
new file mode 100644
index 000000000..66d28c2cb
--- /dev/null
+++ b/kernel/include/sound/cs4231-regs.h
@@ -0,0 +1,187 @@
+#ifndef __SOUND_CS4231_REGS_H
+#define __SOUND_CS4231_REGS_H
+
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Definitions for CS4231 & InterWave chips & compatible chips registers
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* IO ports */
+
+#define CS4231P(x) (c_d_c_CS4231##x)
+
+#define c_d_c_CS4231REGSEL 0
+#define c_d_c_CS4231REG 1
+#define c_d_c_CS4231STATUS 2
+#define c_d_c_CS4231PIO 3
+
+/* codec registers */
+
+#define CS4231_LEFT_INPUT 0x00 /* left input control */
+#define CS4231_RIGHT_INPUT 0x01 /* right input control */
+#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */
+#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */
+#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */
+#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */
+#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */
+#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */
+#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */
+#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */
+#define CS4231_PIN_CTRL 0x0a /* pin control */
+#define CS4231_TEST_INIT 0x0b /* test and initialization */
+#define CS4231_MISC_INFO 0x0c /* miscellaneous information */
+#define CS4231_LOOPBACK 0x0d /* loopback control */
+#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */
+#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */
+#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */
+#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */
+#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */
+#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */
+#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */
+#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */
+#define CS4231_TIMER_LOW 0x14 /* timer low byte */
+#define CS4231_TIMER_HIGH 0x15 /* timer high byte */
+#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */
+#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */
+#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */
+#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */
+#define CS4236_EXT_REG 0x17 /* extended register access */
+#define CS4231_IRQ_STATUS 0x18 /* irq status register */
+#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */
+#define CS4231_VERSION 0x19 /* CS4231(A) - version values */
+#define CS4231_MONO_CTRL 0x1a /* mono input/output control */
+#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */
+#define AD1845_PWR_DOWN 0x1b /* power down control */
+#define CS4235_LEFT_MASTER 0x1b /* left master output control */
+#define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */
+#define AD1845_CLOCK 0x1d /* crystal clock select and total power down */
+#define CS4235_RIGHT_MASTER 0x1d /* right master output control */
+#define CS4231_REC_UPR_CNT 0x1e /* record upper count */
+#define CS4231_REC_LWR_CNT 0x1f /* record lower count */
+
+/* definitions for codec register select port - CODECP( REGSEL ) */
+
+#define CS4231_INIT 0x80 /* CODEC is initializing */
+#define CS4231_MCE 0x40 /* mode change enable */
+#define CS4231_TRD 0x20 /* transfer request disable */
+
+/* definitions for codec status register - CODECP( STATUS ) */
+
+#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */
+
+/* definitions for codec irq status */
+
+#define CS4231_PLAYBACK_IRQ 0x10
+#define CS4231_RECORD_IRQ 0x20
+#define CS4231_TIMER_IRQ 0x40
+#define CS4231_ALL_IRQS 0x70
+#define CS4231_REC_UNDERRUN 0x08
+#define CS4231_REC_OVERRUN 0x04
+#define CS4231_PLY_OVERRUN 0x02
+#define CS4231_PLY_UNDERRUN 0x01
+
+/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */
+
+#define CS4231_ENABLE_MIC_GAIN 0x20
+
+#define CS4231_MIXS_LINE 0x00
+#define CS4231_MIXS_AUX1 0x40
+#define CS4231_MIXS_MIC 0x80
+#define CS4231_MIXS_ALL 0xc0
+
+/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */
+
+#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */
+#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */
+#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */
+#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */
+#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */
+#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */
+#define CS4231_STEREO 0x10 /* stereo mode */
+/* bits 3-1 define frequency divisor */
+#define CS4231_XTAL1 0x00 /* 24.576 crystal */
+#define CS4231_XTAL2 0x01 /* 16.9344 crystal */
+
+/* definitions for interface control register - CS4231_IFACE_CTRL */
+
+#define CS4231_RECORD_PIO 0x80 /* record PIO enable */
+#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */
+#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */
+#define CS4231_AUTOCALIB 0x08 /* auto calibrate */
+#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */
+#define CS4231_RECORD_ENABLE 0x02 /* record enable */
+#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */
+
+/* definitions for pin control register - CS4231_PIN_CTRL */
+
+#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */
+#define CS4231_XCTL1 0x40 /* external control #1 */
+#define CS4231_XCTL0 0x80 /* external control #0 */
+
+/* definitions for test and init register - CS4231_TEST_INIT */
+
+#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */
+#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */
+
+/* definitions for misc control register - CS4231_MISC_INFO */
+
+#define CS4231_MODE2 0x40 /* MODE 2 */
+#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */
+#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */
+
+/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */
+
+#define CS4231_DACZ 0x01 /* zero DAC when underrun */
+#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */
+#define CS4231_OLB 0x80 /* output level bit */
+
+/* definitions for Extended Registers - CS4236+ */
+
+#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f))
+#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8)
+
+#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */
+#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */
+#define CS4236_LEFT_MIC 0x28 /* left MIC volume */
+#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */
+#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */
+#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */
+#define CS4236_LEFT_FM 0x68 /* left FM volume */
+#define CS4236_RIGHT_FM 0x78 /* right FM volume */
+#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */
+#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */
+#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */
+#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */
+#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */
+#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */
+#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */
+#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */
+#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */
+#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */
+#define CS4236_VERSION 0x9c /* chip version and ID */
+
+/* definitions for extended registers - OPTI93X */
+#define OPTi931_AUX_LEFT_INPUT 0x10
+#define OPTi931_AUX_RIGHT_INPUT 0x11
+#define OPTi93X_MIC_LEFT_INPUT 0x14
+#define OPTi93X_MIC_RIGHT_INPUT 0x15
+#define OPTi93X_OUT_LEFT 0x16
+#define OPTi93X_OUT_RIGHT 0x17
+
+#endif /* __SOUND_CS4231_REGS_H */
diff --git a/kernel/include/sound/cs4271.h b/kernel/include/sound/cs4271.h
new file mode 100644
index 000000000..70f45355a
--- /dev/null
+++ b/kernel/include/sound/cs4271.h
@@ -0,0 +1,40 @@
+/*
+ * Definitions for CS4271 ASoC codec driver
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef __CS4271_H
+#define __CS4271_H
+
+struct cs4271_platform_data {
+ int gpio_nreset; /* GPIO driving Reset pin, if any */
+ bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */
+
+ /*
+ * The CS4271 requires its LRCLK and MCLK to be stable before its RESET
+ * line is de-asserted. That also means that clocks cannot be changed
+ * without putting the chip back into hardware reset, which also requires
+ * a complete re-initialization of all registers.
+ *
+ * One (undocumented) workaround is to assert and de-assert the PDN bit
+ * in the MODE2 register. This workaround can be enabled with the
+ * following flag.
+ *
+ * Note that this is not needed in case the clocks are stable
+ * throughout the entire runtime of the codec.
+ */
+ bool enable_soft_reset;
+};
+
+#endif /* __CS4271_H */
diff --git a/kernel/include/sound/cs42l52.h b/kernel/include/sound/cs42l52.h
new file mode 100644
index 000000000..bbabf84bd
--- /dev/null
+++ b/kernel/include/sound/cs42l52.h
@@ -0,0 +1,32 @@
+/*
+ * linux/sound/cs42l52.h -- Platform data for CS42L52
+ *
+ * Copyright (c) 2012 Cirrus Logic Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __CS42L52_H
+#define __CS42L52_H
+
+struct cs42l52_platform_data {
+
+ /* MICBIAS Level. Check datasheet Pg48 */
+ unsigned int micbias_lvl;
+
+ /* MICA mode selection Differential or Single-ended */
+ bool mica_diff_cfg;
+
+ /* MICB mode selection Differential or Single-ended */
+ bool micb_diff_cfg;
+
+ /* Charge Pump Freq. Check datasheet Pg73 */
+ unsigned int chgfreq;
+
+ /* Reset GPIO */
+ unsigned int reset_gpio;
+};
+
+#endif /* __CS42L52_H */
diff --git a/kernel/include/sound/cs42l56.h b/kernel/include/sound/cs42l56.h
new file mode 100644
index 000000000..2467c8ff1
--- /dev/null
+++ b/kernel/include/sound/cs42l56.h
@@ -0,0 +1,48 @@
+/*
+ * linux/sound/cs42l56.h -- Platform data for CS42L56
+ *
+ * Copyright (c) 2014 Cirrus Logic Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __CS42L56_H
+#define __CS42L56_H
+
+struct cs42l56_platform_data {
+
+ /* GPIO for Reset */
+ unsigned int gpio_nreset;
+
+ /* MICBIAS Level. Check datasheet Pg48 */
+ unsigned int micbias_lvl;
+
+ /* Analog Input 1A Reference 0=Single 1=Pseudo-Differential */
+ unsigned int ain1a_ref_cfg;
+
+ /* Analog Input 2A Reference 0=Single 1=Pseudo-Differential */
+ unsigned int ain2a_ref_cfg;
+
+ /* Analog Input 1B Reference 0=Single 1=Pseudo-Differential */
+ unsigned int ain1b_ref_cfg;
+
+ /* Analog Input 2B Reference 0=Single 1=Pseudo-Differential */
+ unsigned int ain2b_ref_cfg;
+
+ /* Charge Pump Freq. Check datasheet Pg62 */
+ unsigned int chgfreq;
+
+ /* HighPass Filter Right Channel Corner Frequency */
+ unsigned int hpfb_freq;
+
+ /* HighPass Filter Left Channel Corner Frequency */
+ unsigned int hpfa_freq;
+
+ /* Adaptive Power Control for LO/HP */
+ unsigned int adaptive_pwr;
+
+};
+
+#endif /* __CS42L56_H */
diff --git a/kernel/include/sound/cs42l73.h b/kernel/include/sound/cs42l73.h
new file mode 100644
index 000000000..f354be4cd
--- /dev/null
+++ b/kernel/include/sound/cs42l73.h
@@ -0,0 +1,22 @@
+/*
+ * linux/sound/cs42l73.h -- Platform data for CS42L73
+ *
+ * Copyright (c) 2012 Cirrus Logic Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __CS42L73_H
+#define __CS42L73_H
+
+struct cs42l73_platform_data {
+ /* RST GPIO */
+ unsigned int reset_gpio;
+ unsigned int chgfreq;
+ int jack_detection;
+ unsigned int mclk_freq;
+};
+
+#endif /* __CS42L73_H */
diff --git a/kernel/include/sound/cs8403.h b/kernel/include/sound/cs8403.h
new file mode 100644
index 000000000..3a8c174a4
--- /dev/null
+++ b/kernel/include/sound/cs8403.h
@@ -0,0 +1,257 @@
+#ifndef __SOUND_CS8403_H
+#define __SOUND_CS8403_H
+
+/*
+ * Routines for Cirrus Logic CS8403/CS8404A IEC958 (S/PDIF) Transmitter
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifdef SND_CS8403
+
+#ifndef SND_CS8403_DECL
+#define SND_CS8403_DECL static
+#endif
+#ifndef SND_CS8403_DECODE
+#define SND_CS8403_DECODE snd_cs8403_decode_spdif_bits
+#endif
+#ifndef SND_CS8403_ENCODE
+#define SND_CS8403_ENCODE snd_cs8403_encode_spdif_bits
+#endif
+
+
+SND_CS8403_DECL void SND_CS8403_DECODE(struct snd_aes_iec958 *diga, unsigned char bits)
+{
+ if (bits & 0x01) { /* consumer */
+ if (!(bits & 0x02))
+ diga->status[0] |= IEC958_AES0_NONAUDIO;
+ if (!(bits & 0x08))
+ diga->status[0] |= IEC958_AES0_CON_NOT_COPYRIGHT;
+ switch (bits & 0x10) {
+ case 0x10: diga->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE; break;
+ case 0x00: diga->status[0] |= IEC958_AES0_CON_EMPHASIS_5015; break;
+ }
+ if (!(bits & 0x80))
+ diga->status[1] |= IEC958_AES1_CON_ORIGINAL;
+ switch (bits & 0x60) {
+ case 0x00: diga->status[1] |= IEC958_AES1_CON_MAGNETIC_ID; break;
+ case 0x20: diga->status[1] |= IEC958_AES1_CON_DIGDIGCONV_ID; break;
+ case 0x40: diga->status[1] |= IEC958_AES1_CON_LASEROPT_ID; break;
+ case 0x60: diga->status[1] |= IEC958_AES1_CON_GENERAL; break;
+ }
+ switch (bits & 0x06) {
+ case 0x00: diga->status[3] |= IEC958_AES3_CON_FS_44100; break;
+ case 0x02: diga->status[3] |= IEC958_AES3_CON_FS_48000; break;
+ case 0x04: diga->status[3] |= IEC958_AES3_CON_FS_32000; break;
+ }
+ } else {
+ diga->status[0] = IEC958_AES0_PROFESSIONAL;
+ switch (bits & 0x18) {
+ case 0x00: diga->status[0] |= IEC958_AES0_PRO_FS_32000; break;
+ case 0x10: diga->status[0] |= IEC958_AES0_PRO_FS_44100; break;
+ case 0x08: diga->status[0] |= IEC958_AES0_PRO_FS_48000; break;
+ case 0x18: diga->status[0] |= IEC958_AES0_PRO_FS_NOTID; break;
+ }
+ switch (bits & 0x60) {
+ case 0x20: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_NONE; break;
+ case 0x40: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_5015; break;
+ case 0x00: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_CCITT; break;
+ case 0x60: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_NOTID; break;
+ }
+ if (bits & 0x80)
+ diga->status[1] |= IEC958_AES1_PRO_MODE_STEREOPHONIC;
+ }
+}
+
+SND_CS8403_DECL unsigned char SND_CS8403_ENCODE(struct snd_aes_iec958 *diga)
+{
+ unsigned char bits;
+
+ if (!(diga->status[0] & IEC958_AES0_PROFESSIONAL)) {
+ bits = 0x01; /* consumer mode */
+ if (diga->status[0] & IEC958_AES0_NONAUDIO)
+ bits &= ~0x02;
+ else
+ bits |= 0x02;
+ if (diga->status[0] & IEC958_AES0_CON_NOT_COPYRIGHT)
+ bits &= ~0x08;
+ else
+ bits |= 0x08;
+ switch (diga->status[0] & IEC958_AES0_CON_EMPHASIS) {
+ default:
+ case IEC958_AES0_CON_EMPHASIS_NONE: bits |= 0x10; break;
+ case IEC958_AES0_CON_EMPHASIS_5015: bits |= 0x00; break;
+ }
+ if (diga->status[1] & IEC958_AES1_CON_ORIGINAL)
+ bits &= ~0x80;
+ else
+ bits |= 0x80;
+ if ((diga->status[1] & IEC958_AES1_CON_CATEGORY) == IEC958_AES1_CON_GENERAL)
+ bits |= 0x60;
+ else {
+ switch(diga->status[1] & IEC958_AES1_CON_MAGNETIC_MASK) {
+ case IEC958_AES1_CON_MAGNETIC_ID:
+ bits |= 0x00; break;
+ case IEC958_AES1_CON_DIGDIGCONV_ID:
+ bits |= 0x20; break;
+ default:
+ case IEC958_AES1_CON_LASEROPT_ID:
+ bits |= 0x40; break;
+ }
+ }
+ switch (diga->status[3] & IEC958_AES3_CON_FS) {
+ default:
+ case IEC958_AES3_CON_FS_44100: bits |= 0x00; break;
+ case IEC958_AES3_CON_FS_48000: bits |= 0x02; break;
+ case IEC958_AES3_CON_FS_32000: bits |= 0x04; break;
+ }
+ } else {
+ bits = 0x00; /* professional mode */
+ if (diga->status[0] & IEC958_AES0_NONAUDIO)
+ bits &= ~0x02;
+ else
+ bits |= 0x02;
+ /* CHECKME: I'm not sure about the bit order in val here */
+ switch (diga->status[0] & IEC958_AES0_PRO_FS) {
+ case IEC958_AES0_PRO_FS_32000: bits |= 0x00; break;
+ case IEC958_AES0_PRO_FS_44100: bits |= 0x10; break; /* 44.1kHz */
+ case IEC958_AES0_PRO_FS_48000: bits |= 0x08; break; /* 48kHz */
+ default:
+ case IEC958_AES0_PRO_FS_NOTID: bits |= 0x18; break;
+ }
+ switch (diga->status[0] & IEC958_AES0_PRO_EMPHASIS) {
+ case IEC958_AES0_PRO_EMPHASIS_NONE: bits |= 0x20; break;
+ case IEC958_AES0_PRO_EMPHASIS_5015: bits |= 0x40; break;
+ case IEC958_AES0_PRO_EMPHASIS_CCITT: bits |= 0x00; break;
+ default:
+ case IEC958_AES0_PRO_EMPHASIS_NOTID: bits |= 0x60; break;
+ }
+ switch (diga->status[1] & IEC958_AES1_PRO_MODE) {
+ case IEC958_AES1_PRO_MODE_TWO:
+ case IEC958_AES1_PRO_MODE_STEREOPHONIC: bits |= 0x00; break;
+ default: bits |= 0x80; break;
+ }
+ }
+ return bits;
+}
+
+#endif /* SND_CS8403 */
+
+#ifdef SND_CS8404
+
+#ifndef SND_CS8404_DECL
+#define SND_CS8404_DECL static
+#endif
+#ifndef SND_CS8404_DECODE
+#define SND_CS8404_DECODE snd_cs8404_decode_spdif_bits
+#endif
+#ifndef SND_CS8404_ENCODE
+#define SND_CS8404_ENCODE snd_cs8404_encode_spdif_bits
+#endif
+
+
+SND_CS8404_DECL void SND_CS8404_DECODE(struct snd_aes_iec958 *diga, unsigned char bits)
+{
+ if (bits & 0x10) { /* consumer */
+ if (!(bits & 0x20))
+ diga->status[0] |= IEC958_AES0_CON_NOT_COPYRIGHT;
+ if (!(bits & 0x40))
+ diga->status[0] |= IEC958_AES0_CON_EMPHASIS_5015;
+ if (!(bits & 0x80))
+ diga->status[1] |= IEC958_AES1_CON_ORIGINAL;
+ switch (bits & 0x03) {
+ case 0x00: diga->status[1] |= IEC958_AES1_CON_DAT; break;
+ case 0x03: diga->status[1] |= IEC958_AES1_CON_GENERAL; break;
+ }
+ switch (bits & 0x06) {
+ case 0x02: diga->status[3] |= IEC958_AES3_CON_FS_32000; break;
+ case 0x04: diga->status[3] |= IEC958_AES3_CON_FS_48000; break;
+ case 0x06: diga->status[3] |= IEC958_AES3_CON_FS_44100; break;
+ }
+ } else {
+ diga->status[0] = IEC958_AES0_PROFESSIONAL;
+ if (!(bits & 0x04))
+ diga->status[0] |= IEC958_AES0_NONAUDIO;
+ switch (bits & 0x60) {
+ case 0x00: diga->status[0] |= IEC958_AES0_PRO_FS_32000; break;
+ case 0x40: diga->status[0] |= IEC958_AES0_PRO_FS_44100; break;
+ case 0x20: diga->status[0] |= IEC958_AES0_PRO_FS_48000; break;
+ case 0x60: diga->status[0] |= IEC958_AES0_PRO_FS_NOTID; break;
+ }
+ switch (bits & 0x03) {
+ case 0x02: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_NONE; break;
+ case 0x01: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_5015; break;
+ case 0x00: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_CCITT; break;
+ case 0x03: diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_NOTID; break;
+ }
+ if (!(bits & 0x80))
+ diga->status[1] |= IEC958_AES1_PRO_MODE_STEREOPHONIC;
+ }
+}
+
+SND_CS8404_DECL unsigned char SND_CS8404_ENCODE(struct snd_aes_iec958 *diga)
+{
+ unsigned char bits;
+
+ if (!(diga->status[0] & IEC958_AES0_PROFESSIONAL)) {
+ bits = 0x10; /* consumer mode */
+ if (!(diga->status[0] & IEC958_AES0_CON_NOT_COPYRIGHT))
+ bits |= 0x20;
+ if ((diga->status[0] & IEC958_AES0_CON_EMPHASIS) == IEC958_AES0_CON_EMPHASIS_NONE)
+ bits |= 0x40;
+ if (!(diga->status[1] & IEC958_AES1_CON_ORIGINAL))
+ bits |= 0x80;
+ if ((diga->status[1] & IEC958_AES1_CON_CATEGORY) == IEC958_AES1_CON_GENERAL)
+ bits |= 0x03;
+ switch (diga->status[3] & IEC958_AES3_CON_FS) {
+ default:
+ case IEC958_AES3_CON_FS_44100: bits |= 0x06; break;
+ case IEC958_AES3_CON_FS_48000: bits |= 0x04; break;
+ case IEC958_AES3_CON_FS_32000: bits |= 0x02; break;
+ }
+ } else {
+ bits = 0x00; /* professional mode */
+ if (!(diga->status[0] & IEC958_AES0_NONAUDIO))
+ bits |= 0x04;
+ switch (diga->status[0] & IEC958_AES0_PRO_FS) {
+ case IEC958_AES0_PRO_FS_32000: bits |= 0x00; break;
+ case IEC958_AES0_PRO_FS_44100: bits |= 0x40; break; /* 44.1kHz */
+ case IEC958_AES0_PRO_FS_48000: bits |= 0x20; break; /* 48kHz */
+ default:
+ case IEC958_AES0_PRO_FS_NOTID: bits |= 0x00; break;
+ }
+ switch (diga->status[0] & IEC958_AES0_PRO_EMPHASIS) {
+ case IEC958_AES0_PRO_EMPHASIS_NONE: bits |= 0x02; break;
+ case IEC958_AES0_PRO_EMPHASIS_5015: bits |= 0x01; break;
+ case IEC958_AES0_PRO_EMPHASIS_CCITT: bits |= 0x00; break;
+ default:
+ case IEC958_AES0_PRO_EMPHASIS_NOTID: bits |= 0x03; break;
+ }
+ switch (diga->status[1] & IEC958_AES1_PRO_MODE) {
+ case IEC958_AES1_PRO_MODE_TWO:
+ case IEC958_AES1_PRO_MODE_STEREOPHONIC: bits |= 0x00; break;
+ default: bits |= 0x80; break;
+ }
+ }
+ return bits;
+}
+
+#endif /* SND_CS8404 */
+
+#endif /* __SOUND_CS8403_H */
diff --git a/kernel/include/sound/cs8427.h b/kernel/include/sound/cs8427.h
new file mode 100644
index 000000000..0b6a18766
--- /dev/null
+++ b/kernel/include/sound/cs8427.h
@@ -0,0 +1,202 @@
+#ifndef __SOUND_CS8427_H
+#define __SOUND_CS8427_H
+
+/*
+ * Routines for Cirrus Logic CS8427
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/i2c.h>
+
+#define CS8427_BASE_ADDR 0x10 /* base I2C address */
+
+#define CS8427_REG_AUTOINC 0x80 /* flag - autoincrement */
+#define CS8427_REG_CONTROL1 0x01
+#define CS8427_REG_CONTROL2 0x02
+#define CS8427_REG_DATAFLOW 0x03
+#define CS8427_REG_CLOCKSOURCE 0x04
+#define CS8427_REG_SERIALINPUT 0x05
+#define CS8427_REG_SERIALOUTPUT 0x06
+#define CS8427_REG_INT1STATUS 0x07
+#define CS8427_REG_INT2STATUS 0x08
+#define CS8427_REG_INT1MASK 0x09
+#define CS8427_REG_INT1MODEMSB 0x0a
+#define CS8427_REG_INT1MODELSB 0x0b
+#define CS8427_REG_INT2MASK 0x0c
+#define CS8427_REG_INT2MODEMSB 0x0d
+#define CS8427_REG_INT2MODELSB 0x0e
+#define CS8427_REG_RECVCSDATA 0x0f
+#define CS8427_REG_RECVERRORS 0x10
+#define CS8427_REG_RECVERRMASK 0x11
+#define CS8427_REG_CSDATABUF 0x12
+#define CS8427_REG_UDATABUF 0x13
+#define CS8427_REG_QSUBCODE 0x14 /* 0x14-0x1d (10 bytes) */
+#define CS8427_REG_OMCKRMCKRATIO 0x1e
+#define CS8427_REG_CORU_DATABUF 0x20 /* 24 byte buffer area */
+#define CS8427_REG_ID_AND_VER 0x7f
+
+/* CS8427_REG_CONTROL1 bits */
+#define CS8427_SWCLK (1<<7) /* 0 = RMCK default, 1 = OMCK output on RMCK pin */
+#define CS8427_VSET (1<<6) /* 0 = valid PCM data, 1 = invalid PCM data */
+#define CS8427_MUTESAO (1<<5) /* mute control for the serial audio output port, 0 = disabled, 1 = enabled */
+#define CS8427_MUTEAES (1<<4) /* mute control for the AES transmitter output, 0 = disabled, 1 = enabled */
+#define CS8427_INTMASK (3<<1) /* interrupt output pin setup mask */
+#define CS8427_INTACTHIGH (0<<1) /* active high */
+#define CS8427_INTACTLOW (1<<1) /* active low */
+#define CS8427_INTOPENDRAIN (2<<1) /* open drain, active low */
+#define CS8427_TCBLDIR (1<<0) /* 0 = TCBL is an input, 1 = TCBL is an output */
+
+/* CS8427_REQ_CONTROL2 bits */
+#define CS8427_HOLDMASK (3<<5) /* action when a receiver error occurs */
+#define CS8427_HOLDLASTSAMPLE (0<<5) /* hold the last valid sample */
+#define CS8427_HOLDZERO (1<<5) /* replace the current audio sample with zero (mute) */
+#define CS8427_HOLDNOCHANGE (2<<5) /* do not change the received audio sample */
+#define CS8427_RMCKF (1<<4) /* 0 = 256*Fsi, 1 = 128*Fsi */
+#define CS8427_MMR (1<<3) /* AES3 receiver operation, 0 = stereo, 1 = mono */
+#define CS8427_MMT (1<<2) /* AES3 transmitter operation, 0 = stereo, 1 = mono */
+#define CS8427_MMTCS (1<<1) /* 0 = use A + B CS data, 1 = use MMTLR CS data */
+#define CS8427_MMTLR (1<<0) /* 0 = use A CS data, 1 = use B CS data */
+
+/* CS8427_REG_DATAFLOW */
+#define CS8427_TXOFF (1<<6) /* AES3 transmitter Output, 0 = normal operation, 1 = off (0V) */
+#define CS8427_AESBP (1<<5) /* AES3 hardware bypass mode, 0 = normal, 1 = bypass (RX->TX) */
+#define CS8427_TXDMASK (3<<3) /* AES3 Transmitter Data Source Mask */
+#define CS8427_TXDSERIAL (1<<3) /* TXD - serial audio input port */
+#define CS8427_TXAES3DRECEIVER (2<<3) /* TXD - AES3 receiver */
+#define CS8427_SPDMASK (3<<1) /* Serial Audio Output Port Data Source Mask */
+#define CS8427_SPDSERIAL (1<<1) /* SPD - serial audio input port */
+#define CS8427_SPDAES3RECEIVER (2<<1) /* SPD - AES3 receiver */
+
+/* CS8427_REG_CLOCKSOURCE */
+#define CS8427_RUN (1<<6) /* 0 = clock off, 1 = clock on */
+#define CS8427_CLKMASK (3<<4) /* OMCK frequency mask */
+#define CS8427_CLK256 (0<<4) /* 256*Fso */
+#define CS8427_CLK384 (1<<4) /* 384*Fso */
+#define CS8427_CLK512 (2<<4) /* 512*Fso */
+#define CS8427_OUTC (1<<3) /* Output Time Base, 0 = OMCK, 1 = recovered input clock */
+#define CS8427_INC (1<<2) /* Input Time Base Clock Source, 0 = recoverd input clock, 1 = OMCK input pin */
+#define CS8427_RXDMASK (3<<0) /* Recovered Input Clock Source Mask */
+#define CS8427_RXDILRCK (0<<0) /* 256*Fsi from ILRCK pin */
+#define CS8427_RXDAES3INPUT (1<<0) /* 256*Fsi from AES3 input */
+#define CS8427_EXTCLOCKRESET (2<<0) /* bypass PLL, 256*Fsi clock, synchronous reset */
+#define CS8427_EXTCLOCK (3<<0) /* bypass PLL, 256*Fsi clock */
+
+/* CS8427_REG_SERIALINPUT */
+#define CS8427_SIMS (1<<7) /* 0 = slave, 1 = master mode */
+#define CS8427_SISF (1<<6) /* ISCLK freq, 0 = 64*Fsi, 1 = 128*Fsi */
+#define CS8427_SIRESMASK (3<<4) /* Resolution of the input data for right justified formats */
+#define CS8427_SIRES24 (0<<4) /* SIRES 24-bit */
+#define CS8427_SIRES20 (1<<4) /* SIRES 20-bit */
+#define CS8427_SIRES16 (2<<4) /* SIRES 16-bit */
+#define CS8427_SIJUST (1<<3) /* Justification of SDIN data relative to ILRCK, 0 = left-justified, 1 = right-justified */
+#define CS8427_SIDEL (1<<2) /* Delay of SDIN data relative to ILRCK for left-justified data formats, 0 = first ISCLK period, 1 = second ISCLK period */
+#define CS8427_SISPOL (1<<1) /* ICLK clock polarity, 0 = rising edge of ISCLK, 1 = falling edge of ISCLK */
+#define CS8427_SILRPOL (1<<0) /* ILRCK clock polarity, 0 = SDIN data left channel when ILRCK is high, 1 = SDIN right when ILRCK is high */
+
+/* CS8427_REG_SERIALOUTPUT */
+#define CS8427_SOMS (1<<7) /* 0 = slave, 1 = master mode */
+#define CS8427_SOSF (1<<6) /* OSCLK freq, 0 = 64*Fso, 1 = 128*Fso */
+#define CS8427_SORESMASK (3<<4) /* Resolution of the output data on SDOUT and AES3 output */
+#define CS8427_SORES24 (0<<4) /* SIRES 24-bit */
+#define CS8427_SORES20 (1<<4) /* SIRES 20-bit */
+#define CS8427_SORES16 (2<<4) /* SIRES 16-bit */
+#define CS8427_SORESDIRECT (2<<4) /* SIRES direct copy from AES3 receiver */
+#define CS8427_SOJUST (1<<3) /* Justification of SDOUT data relative to OLRCK, 0 = left-justified, 1 = right-justified */
+#define CS8427_SODEL (1<<2) /* Delay of SDOUT data relative to OLRCK for left-justified data formats, 0 = first OSCLK period, 1 = second OSCLK period */
+#define CS8427_SOSPOL (1<<1) /* OSCLK clock polarity, 0 = rising edge of ISCLK, 1 = falling edge of ISCLK */
+#define CS8427_SOLRPOL (1<<0) /* OLRCK clock polarity, 0 = SDOUT data left channel when OLRCK is high, 1 = SDOUT right when OLRCK is high */
+
+/* CS8427_REG_INT1STATUS */
+#define CS8427_TSLIP (1<<7) /* AES3 transmitter source data slip interrupt */
+#define CS8427_OSLIP (1<<6) /* Serial audio output port data slip interrupt */
+#define CS8427_DETC (1<<2) /* D to E C-buffer transfer interrupt */
+#define CS8427_EFTC (1<<1) /* E to F C-buffer transfer interrupt */
+#define CS8427_RERR (1<<0) /* A receiver error has occurred */
+
+/* CS8427_REG_INT2STATUS */
+#define CS8427_DETU (1<<3) /* D to E U-buffer transfer interrupt */
+#define CS8427_EFTU (1<<2) /* E to F U-buffer transfer interrupt */
+#define CS8427_QCH (1<<1) /* A new block of Q-subcode data is available for reading */
+
+/* CS8427_REG_INT1MODEMSB && CS8427_REG_INT1MODELSB */
+/* bits are defined in CS8427_REG_INT1STATUS */
+/* CS8427_REG_INT2MODEMSB && CS8427_REG_INT2MODELSB */
+/* bits are defined in CS8427_REG_INT2STATUS */
+#define CS8427_INTMODERISINGMSB 0
+#define CS8427_INTMODERESINGLSB 0
+#define CS8427_INTMODEFALLINGMSB 0
+#define CS8427_INTMODEFALLINGLSB 1
+#define CS8427_INTMODELEVELMSB 1
+#define CS8427_INTMODELEVELLSB 0
+
+/* CS8427_REG_RECVCSDATA */
+#define CS8427_AUXMASK (15<<4) /* auxiliary data field width */
+#define CS8427_AUXSHIFT 4
+#define CS8427_PRO (1<<3) /* Channel status block format indicator */
+#define CS8427_AUDIO (1<<2) /* Audio indicator (0 = audio, 1 = nonaudio */
+#define CS8427_COPY (1<<1) /* 0 = copyright asserted, 1 = copyright not asserted */
+#define CS8427_ORIG (1<<0) /* SCMS generation indicator, 0 = 1st generation or highter, 1 = original */
+
+/* CS8427_REG_RECVERRORS */
+/* CS8427_REG_RECVERRMASK for CS8427_RERR */
+#define CS8427_QCRC (1<<6) /* Q-subcode data CRC error indicator */
+#define CS8427_CCRC (1<<5) /* Chancnel Status Block Cyclick Redundancy Check Bit */
+#define CS8427_UNLOCK (1<<4) /* PLL lock status bit */
+#define CS8427_V (1<<3) /* 0 = valid data */
+#define CS8427_CONF (1<<2) /* Confidence bit */
+#define CS8427_BIP (1<<1) /* Bi-phase error bit */
+#define CS8427_PAR (1<<0) /* Parity error */
+
+/* CS8427_REG_CSDATABUF */
+#define CS8427_BSEL (1<<5) /* 0 = CS data, 1 = U data */
+#define CS8427_CBMR (1<<4) /* 0 = overwrite first 5 bytes for CS D to E buffer, 1 = prevent */
+#define CS8427_DETCI (1<<3) /* D to E CS data buffer transfer inhibit bit, 0 = allow, 1 = inhibit */
+#define CS8427_EFTCI (1<<2) /* E to F CS data buffer transfer inhibit bit, 0 = allow, 1 = inhibit */
+#define CS8427_CAM (1<<1) /* CS data buffer control port access mode bit, 0 = one byte, 1 = two byte */
+#define CS8427_CHS (1<<0) /* Channel select bit, 0 = Channel A, 1 = Channel B */
+
+/* CS8427_REG_UDATABUF */
+#define CS8427_UD (1<<4) /* User data pin (U) direction, 0 = input, 1 = output */
+#define CS8427_UBMMASK (3<<2) /* Operating mode of the AES3 U bit manager */
+#define CS8427_UBMZEROS (0<<2) /* transmit all zeros mode */
+#define CS8427_UBMBLOCK (1<<2) /* block mode */
+#define CS8427_DETUI (1<<1) /* D to E U-data buffer transfer inhibit bit, 0 = allow, 1 = inhibit */
+#define CS8427_EFTUI (1<<1) /* E to F U-data buffer transfer inhibit bit, 0 = allow, 1 = inhibit */
+
+/* CS8427_REG_ID_AND_VER */
+#define CS8427_IDMASK (15<<4)
+#define CS8427_IDSHIFT 4
+#define CS8427_VERMASK (15<<0)
+#define CS8427_VERSHIFT 0
+#define CS8427_VER8427A 0x71
+
+struct snd_pcm_substream;
+
+int snd_cs8427_init(struct snd_i2c_bus *bus, struct snd_i2c_device *device);
+int snd_cs8427_create(struct snd_i2c_bus *bus, unsigned char addr,
+ unsigned int reset_timeout, struct snd_i2c_device **r_cs8427);
+int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg,
+ unsigned char val);
+int snd_cs8427_iec958_build(struct snd_i2c_device *cs8427,
+ struct snd_pcm_substream *playback_substream,
+ struct snd_pcm_substream *capture_substream);
+int snd_cs8427_iec958_active(struct snd_i2c_device *cs8427, int active);
+int snd_cs8427_iec958_pcm(struct snd_i2c_device *cs8427, unsigned int rate);
+
+#endif /* __SOUND_CS8427_H */
diff --git a/kernel/include/sound/da7213.h b/kernel/include/sound/da7213.h
new file mode 100644
index 000000000..673f5c39c
--- /dev/null
+++ b/kernel/include/sound/da7213.h
@@ -0,0 +1,52 @@
+/*
+ * da7213.h - DA7213 ASoC Codec Driver Platform Data
+ *
+ * Copyright (c) 2013 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _DA7213_PDATA_H
+#define _DA7213_PDATA_H
+
+enum da7213_micbias_voltage {
+ DA7213_MICBIAS_1_6V = 0,
+ DA7213_MICBIAS_2_2V = 1,
+ DA7213_MICBIAS_2_5V = 2,
+ DA7213_MICBIAS_3_0V = 3,
+};
+
+enum da7213_dmic_data_sel {
+ DA7213_DMIC_DATA_LRISE_RFALL = 0,
+ DA7213_DMIC_DATA_LFALL_RRISE = 1,
+};
+
+enum da7213_dmic_samplephase {
+ DA7213_DMIC_SAMPLE_ON_CLKEDGE = 0,
+ DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE = 1,
+};
+
+enum da7213_dmic_clk_rate {
+ DA7213_DMIC_CLK_3_0MHZ = 0,
+ DA7213_DMIC_CLK_1_5MHZ = 1,
+};
+
+struct da7213_platform_data {
+ /* Mic Bias voltage */
+ enum da7213_micbias_voltage micbias1_lvl;
+ enum da7213_micbias_voltage micbias2_lvl;
+
+ /* DMIC config */
+ enum da7213_dmic_data_sel dmic_data_sel;
+ enum da7213_dmic_samplephase dmic_samplephase;
+ enum da7213_dmic_clk_rate dmic_clk_rate;
+
+ /* MCLK squaring config */
+ bool mclk_squaring;
+};
+
+#endif /* _DA7213_PDATA_H */
diff --git a/kernel/include/sound/da9055.h b/kernel/include/sound/da9055.h
new file mode 100644
index 000000000..cf1241b64
--- /dev/null
+++ b/kernel/include/sound/da9055.h
@@ -0,0 +1,33 @@
+/*
+ * DA9055 ALSA Soc codec driver
+ *
+ * Copyright (c) 2012 Dialog Semiconductor
+ *
+ * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C
+ * Written by David Chen <david.chen@diasemi.com> and
+ * Ashish Chavan <ashish.chavan@kpitcummins.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __SOUND_DA9055_H__
+#define __SOUND_DA9055_H__
+
+enum da9055_micbias_voltage {
+ DA9055_MICBIAS_1_6V = 0,
+ DA9055_MICBIAS_1_8V = 1,
+ DA9055_MICBIAS_2_1V = 2,
+ DA9055_MICBIAS_2_2V = 3,
+};
+
+struct da9055_platform_data {
+ /* Selects which of the two MicBias pins acts as the bias source */
+ bool micbias_source;
+ /* Selects the micbias voltage */
+ enum da9055_micbias_voltage micbias;
+};
+
+#endif
diff --git a/kernel/include/sound/designware_i2s.h b/kernel/include/sound/designware_i2s.h
new file mode 100644
index 000000000..3a8fca940
--- /dev/null
+++ b/kernel/include/sound/designware_i2s.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux@gmail.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifndef __SOUND_DESIGNWARE_I2S_H
+#define __SOUND_DESIGNWARE_I2S_H
+
+#include <linux/dmaengine.h>
+#include <linux/types.h>
+
+/*
+ * struct i2s_clk_config_data - represent i2s clk configuration data
+ * @chan_nr: number of channel
+ * @data_width: number of bits per sample (8/16/24/32 bit)
+ * @sample_rate: sampling frequency (8Khz, 16Khz, 32Khz, 44Khz, 48Khz)
+ */
+struct i2s_clk_config_data {
+ int chan_nr;
+ u32 data_width;
+ u32 sample_rate;
+};
+
+struct i2s_platform_data {
+ #define DWC_I2S_PLAY (1 << 0)
+ #define DWC_I2S_RECORD (1 << 1)
+ unsigned int cap;
+ int channel;
+ u32 snd_fmts;
+ u32 snd_rates;
+
+ void *play_dma_data;
+ void *capture_dma_data;
+ bool (*filter)(struct dma_chan *chan, void *slave);
+ int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
+};
+
+struct i2s_dma_data {
+ void *data;
+ dma_addr_t addr;
+ u32 max_burst;
+ enum dma_slave_buswidth addr_width;
+ bool (*filter)(struct dma_chan *chan, void *slave);
+};
+
+/* I2S DMA registers */
+#define I2S_RXDMA 0x01C0
+#define I2S_TXDMA 0x01C8
+
+#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */
+#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */
+#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */
+#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */
+
+#endif /* __SOUND_DESIGNWARE_I2S_H */
diff --git a/kernel/include/sound/dmaengine_pcm.h b/kernel/include/sound/dmaengine_pcm.h
new file mode 100644
index 000000000..eb73a3a39
--- /dev/null
+++ b/kernel/include/sound/dmaengine_pcm.h
@@ -0,0 +1,157 @@
+/*
+ * Copyright (C) 2012, Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+#ifndef __SOUND_DMAENGINE_PCM_H__
+#define __SOUND_DMAENGINE_PCM_H__
+
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <linux/dmaengine.h>
+
+/**
+ * snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM
+ * substream
+ * @substream: PCM substream
+ */
+static inline enum dma_transfer_direction
+snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return DMA_MEM_TO_DEV;
+ else
+ return DMA_DEV_TO_MEM;
+}
+
+int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
+ const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config);
+int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
+snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream);
+snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream);
+
+int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
+ struct dma_chan *chan);
+int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream);
+
+int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
+ dma_filter_fn filter_fn, void *filter_data);
+int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream);
+
+struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn,
+ void *filter_data);
+struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream);
+
+/**
+ * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data
+ * @addr: Address of the DAI data source or destination register.
+ * @addr_width: Width of the DAI data source or destination register.
+ * @maxburst: Maximum number of words(note: words, as in units of the
+ * src_addr_width member, not bytes) that can be send to or received from the
+ * DAI in one burst.
+ * @slave_id: Slave requester id for the DMA channel.
+ * @filter_data: Custom DMA channel filter data, this will usually be used when
+ * requesting the DMA channel.
+ * @chan_name: Custom channel name to use when requesting DMA channel.
+ * @fifo_size: FIFO size of the DAI controller in bytes
+ */
+struct snd_dmaengine_dai_dma_data {
+ dma_addr_t addr;
+ enum dma_slave_buswidth addr_width;
+ u32 maxburst;
+ unsigned int slave_id;
+ void *filter_data;
+ const char *chan_name;
+ unsigned int fifo_size;
+};
+
+void snd_dmaengine_pcm_set_config_from_dai_data(
+ const struct snd_pcm_substream *substream,
+ const struct snd_dmaengine_dai_dma_data *dma_data,
+ struct dma_slave_config *config);
+
+
+/*
+ * Try to request the DMA channel using compat_request_channel or
+ * compat_filter_fn if it couldn't be requested through devicetree.
+ */
+#define SND_DMAENGINE_PCM_FLAG_COMPAT BIT(0)
+/*
+ * Don't try to request the DMA channels through devicetree. This flag only
+ * makes sense if SND_DMAENGINE_PCM_FLAG_COMPAT is set as well.
+ */
+#define SND_DMAENGINE_PCM_FLAG_NO_DT BIT(1)
+/*
+ * The platforms dmaengine driver does not support reporting the amount of
+ * bytes that are still left to transfer.
+ */
+#define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2)
+/*
+ * The PCM is half duplex and the DMA channel is shared between capture and
+ * playback.
+ */
+#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3)
+/*
+ * The PCM streams have custom channel names specified.
+ */
+#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4)
+
+/**
+ * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM
+ * @prepare_slave_config: Callback used to fill in the DMA slave_config for a
+ * PCM substream. Will be called from the PCM drivers hwparams callback.
+ * @compat_request_channel: Callback to request a DMA channel for platforms
+ * which do not use devicetree.
+ * @compat_filter_fn: Will be used as the filter function when requesting a
+ * channel for platforms which do not use devicetree. The filter parameter
+ * will be the DAI's DMA data.
+ * @dma_dev: If set, request DMA channel on this device rather than the DAI
+ * device.
+ * @chan_names: If set, these custom DMA channel names will be requested at
+ * registration time.
+ * @pcm_hardware: snd_pcm_hardware struct to be used for the PCM.
+ * @prealloc_buffer_size: Size of the preallocated audio buffer.
+ *
+ * Note: If both compat_request_channel and compat_filter_fn are set
+ * compat_request_channel will be used to request the channel and
+ * compat_filter_fn will be ignored. Otherwise the channel will be requested
+ * using dma_request_channel with compat_filter_fn as the filter function.
+ */
+struct snd_dmaengine_pcm_config {
+ int (*prepare_slave_config)(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct dma_slave_config *slave_config);
+ struct dma_chan *(*compat_request_channel)(
+ struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_substream *substream);
+ dma_filter_fn compat_filter_fn;
+ struct device *dma_dev;
+ const char *chan_names[SNDRV_PCM_STREAM_LAST + 1];
+
+ const struct snd_pcm_hardware *pcm_hardware;
+ unsigned int prealloc_buffer_size;
+};
+
+int snd_dmaengine_pcm_register(struct device *dev,
+ const struct snd_dmaengine_pcm_config *config,
+ unsigned int flags);
+void snd_dmaengine_pcm_unregister(struct device *dev);
+
+int devm_snd_dmaengine_pcm_register(struct device *dev,
+ const struct snd_dmaengine_pcm_config *config,
+ unsigned int flags);
+
+int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct dma_slave_config *slave_config);
+
+#endif
diff --git a/kernel/include/sound/emu10k1.h b/kernel/include/sound/emu10k1.h
new file mode 100644
index 000000000..5bd134651
--- /dev/null
+++ b/kernel/include/sound/emu10k1.h
@@ -0,0 +1,1906 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Creative Labs, Inc.
+ * Definitions for EMU10K1 (SB Live!) chips
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SOUND_EMU10K1_H
+#define __SOUND_EMU10K1_H
+
+
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/hwdep.h>
+#include <sound/ac97_codec.h>
+#include <sound/util_mem.h>
+#include <sound/pcm-indirect.h>
+#include <sound/timer.h>
+#include <linux/interrupt.h>
+#include <linux/mutex.h>
+#include <linux/firmware.h>
+#include <linux/io.h>
+
+#include <uapi/sound/emu10k1.h>
+
+/* ------------------- DEFINES -------------------- */
+
+#define EMUPAGESIZE 4096
+#define MAXREQVOICES 8
+#define MAXPAGES0 4096 /* 32 bit mode */
+#define MAXPAGES1 8192 /* 31 bit mode */
+#define RESERVED 0
+#define NUM_MIDI 16
+#define NUM_G 64 /* use all channels */
+#define NUM_FXSENDS 4
+#define NUM_EFX_PLAYBACK 16
+
+/* FIXME? - according to the OSS driver the EMU10K1 needs a 29 bit DMA mask */
+#define EMU10K1_DMA_MASK 0x7fffffffUL /* 31bit */
+#define AUDIGY_DMA_MASK 0xffffffffUL /* 32bit mode */
+
+#define TMEMSIZE 256*1024
+#define TMEMSIZEREG 4
+
+#define IP_TO_CP(ip) ((ip == 0) ? 0 : (((0x00001000uL | (ip & 0x00000FFFL)) << (((ip >> 12) & 0x000FL) + 4)) & 0xFFFF0000uL))
+
+// Audigy specify registers are prefixed with 'A_'
+
+/************************************************************************************************/
+/* PCI function 0 registers, address = <val> + PCIBASE0 */
+/************************************************************************************************/
+
+#define PTR 0x00 /* Indexed register set pointer register */
+ /* NOTE: The CHANNELNUM and ADDRESS words can */
+ /* be modified independently of each other. */
+#define PTR_CHANNELNUM_MASK 0x0000003f /* For each per-channel register, indicates the */
+ /* channel number of the register to be */
+ /* accessed. For non per-channel registers the */
+ /* value should be set to zero. */
+#define PTR_ADDRESS_MASK 0x07ff0000 /* Register index */
+#define A_PTR_ADDRESS_MASK 0x0fff0000
+
+#define DATA 0x04 /* Indexed register set data register */
+
+#define IPR 0x08 /* Global interrupt pending register */
+ /* Clear pending interrupts by writing a 1 to */
+ /* the relevant bits and zero to the other bits */
+#define IPR_P16V 0x80000000 /* Bit set when the CA0151 P16V chip wishes
+ to interrupt */
+#define IPR_GPIOMSG 0x20000000 /* GPIO message interrupt (RE'd, still not sure
+ which INTE bits enable it) */
+
+/* The next two interrupts are for the midi port on the Audigy Drive (A_MPU1) */
+#define IPR_A_MIDITRANSBUFEMPTY2 0x10000000 /* MIDI UART transmit buffer empty */
+#define IPR_A_MIDIRECVBUFEMPTY2 0x08000000 /* MIDI UART receive buffer empty */
+
+#define IPR_SPDIFBUFFULL 0x04000000 /* SPDIF capture related, 10k2 only? (RE) */
+#define IPR_SPDIFBUFHALFFULL 0x02000000 /* SPDIF capture related? (RE) */
+
+#define IPR_SAMPLERATETRACKER 0x01000000 /* Sample rate tracker lock status change */
+#define IPR_FXDSP 0x00800000 /* Enable FX DSP interrupts */
+#define IPR_FORCEINT 0x00400000 /* Force Sound Blaster interrupt */
+#define IPR_PCIERROR 0x00200000 /* PCI bus error */
+#define IPR_VOLINCR 0x00100000 /* Volume increment button pressed */
+#define IPR_VOLDECR 0x00080000 /* Volume decrement button pressed */
+#define IPR_MUTE 0x00040000 /* Mute button pressed */
+#define IPR_MICBUFFULL 0x00020000 /* Microphone buffer full */
+#define IPR_MICBUFHALFFULL 0x00010000 /* Microphone buffer half full */
+#define IPR_ADCBUFFULL 0x00008000 /* ADC buffer full */
+#define IPR_ADCBUFHALFFULL 0x00004000 /* ADC buffer half full */
+#define IPR_EFXBUFFULL 0x00002000 /* Effects buffer full */
+#define IPR_EFXBUFHALFFULL 0x00001000 /* Effects buffer half full */
+#define IPR_GPSPDIFSTATUSCHANGE 0x00000800 /* GPSPDIF channel status change */
+#define IPR_CDROMSTATUSCHANGE 0x00000400 /* CD-ROM channel status change */
+#define IPR_INTERVALTIMER 0x00000200 /* Interval timer terminal count */
+#define IPR_MIDITRANSBUFEMPTY 0x00000100 /* MIDI UART transmit buffer empty */
+#define IPR_MIDIRECVBUFEMPTY 0x00000080 /* MIDI UART receive buffer empty */
+#define IPR_CHANNELLOOP 0x00000040 /* Channel (half) loop interrupt(s) pending */
+#define IPR_CHANNELNUMBERMASK 0x0000003f /* When IPR_CHANNELLOOP is set, indicates the */
+ /* highest set channel in CLIPL, CLIPH, HLIPL, */
+ /* or HLIPH. When IP is written with CL set, */
+ /* the bit in H/CLIPL or H/CLIPH corresponding */
+ /* to the CIN value written will be cleared. */
+
+#define INTE 0x0c /* Interrupt enable register */
+#define INTE_VIRTUALSB_MASK 0xc0000000 /* Virtual Soundblaster I/O port capture */
+#define INTE_VIRTUALSB_220 0x00000000 /* Capture at I/O base address 0x220-0x22f */
+#define INTE_VIRTUALSB_240 0x40000000 /* Capture at I/O base address 0x240 */
+#define INTE_VIRTUALSB_260 0x80000000 /* Capture at I/O base address 0x260 */
+#define INTE_VIRTUALSB_280 0xc0000000 /* Capture at I/O base address 0x280 */
+#define INTE_VIRTUALMPU_MASK 0x30000000 /* Virtual MPU I/O port capture */
+#define INTE_VIRTUALMPU_300 0x00000000 /* Capture at I/O base address 0x300-0x301 */
+#define INTE_VIRTUALMPU_310 0x10000000 /* Capture at I/O base address 0x310 */
+#define INTE_VIRTUALMPU_320 0x20000000 /* Capture at I/O base address 0x320 */
+#define INTE_VIRTUALMPU_330 0x30000000 /* Capture at I/O base address 0x330 */
+#define INTE_MASTERDMAENABLE 0x08000000 /* Master DMA emulation at 0x000-0x00f */
+#define INTE_SLAVEDMAENABLE 0x04000000 /* Slave DMA emulation at 0x0c0-0x0df */
+#define INTE_MASTERPICENABLE 0x02000000 /* Master PIC emulation at 0x020-0x021 */
+#define INTE_SLAVEPICENABLE 0x01000000 /* Slave PIC emulation at 0x0a0-0x0a1 */
+#define INTE_VSBENABLE 0x00800000 /* Enable virtual Soundblaster */
+#define INTE_ADLIBENABLE 0x00400000 /* Enable AdLib emulation at 0x388-0x38b */
+#define INTE_MPUENABLE 0x00200000 /* Enable virtual MPU */
+#define INTE_FORCEINT 0x00100000 /* Continuously assert INTAN */
+
+#define INTE_MRHANDENABLE 0x00080000 /* Enable the "Mr. Hand" logic */
+ /* NOTE: There is no reason to use this under */
+ /* Linux, and it will cause odd hardware */
+ /* behavior and possibly random segfaults and */
+ /* lockups if enabled. */
+
+/* The next two interrupts are for the midi port on the Audigy Drive (A_MPU1) */
+#define INTE_A_MIDITXENABLE2 0x00020000 /* Enable MIDI transmit-buffer-empty interrupts */
+#define INTE_A_MIDIRXENABLE2 0x00010000 /* Enable MIDI receive-buffer-empty interrupts */
+
+
+#define INTE_SAMPLERATETRACKER 0x00002000 /* Enable sample rate tracker interrupts */
+ /* NOTE: This bit must always be enabled */
+#define INTE_FXDSPENABLE 0x00001000 /* Enable FX DSP interrupts */
+#define INTE_PCIERRORENABLE 0x00000800 /* Enable PCI bus error interrupts */
+#define INTE_VOLINCRENABLE 0x00000400 /* Enable volume increment button interrupts */
+#define INTE_VOLDECRENABLE 0x00000200 /* Enable volume decrement button interrupts */
+#define INTE_MUTEENABLE 0x00000100 /* Enable mute button interrupts */
+#define INTE_MICBUFENABLE 0x00000080 /* Enable microphone buffer interrupts */
+#define INTE_ADCBUFENABLE 0x00000040 /* Enable ADC buffer interrupts */
+#define INTE_EFXBUFENABLE 0x00000020 /* Enable Effects buffer interrupts */
+#define INTE_GPSPDIFENABLE 0x00000010 /* Enable GPSPDIF status interrupts */
+#define INTE_CDSPDIFENABLE 0x00000008 /* Enable CDSPDIF status interrupts */
+#define INTE_INTERVALTIMERENB 0x00000004 /* Enable interval timer interrupts */
+#define INTE_MIDITXENABLE 0x00000002 /* Enable MIDI transmit-buffer-empty interrupts */
+#define INTE_MIDIRXENABLE 0x00000001 /* Enable MIDI receive-buffer-empty interrupts */
+
+#define WC 0x10 /* Wall Clock register */
+#define WC_SAMPLECOUNTER_MASK 0x03FFFFC0 /* Sample periods elapsed since reset */
+#define WC_SAMPLECOUNTER 0x14060010
+#define WC_CURRENTCHANNEL 0x0000003F /* Channel [0..63] currently being serviced */
+ /* NOTE: Each channel takes 1/64th of a sample */
+ /* period to be serviced. */
+
+#define HCFG 0x14 /* Hardware config register */
+ /* NOTE: There is no reason to use the legacy */
+ /* SoundBlaster emulation stuff described below */
+ /* under Linux, and all kinds of weird hardware */
+ /* behavior can result if you try. Don't. */
+#define HCFG_LEGACYFUNC_MASK 0xe0000000 /* Legacy function number */
+#define HCFG_LEGACYFUNC_MPU 0x00000000 /* Legacy MPU */
+#define HCFG_LEGACYFUNC_SB 0x40000000 /* Legacy SB */
+#define HCFG_LEGACYFUNC_AD 0x60000000 /* Legacy AD */
+#define HCFG_LEGACYFUNC_MPIC 0x80000000 /* Legacy MPIC */
+#define HCFG_LEGACYFUNC_MDMA 0xa0000000 /* Legacy MDMA */
+#define HCFG_LEGACYFUNC_SPCI 0xc0000000 /* Legacy SPCI */
+#define HCFG_LEGACYFUNC_SDMA 0xe0000000 /* Legacy SDMA */
+#define HCFG_IOCAPTUREADDR 0x1f000000 /* The 4 LSBs of the captured I/O address. */
+#define HCFG_LEGACYWRITE 0x00800000 /* 1 = write, 0 = read */
+#define HCFG_LEGACYWORD 0x00400000 /* 1 = word, 0 = byte */
+#define HCFG_LEGACYINT 0x00200000 /* 1 = legacy event captured. Write 1 to clear. */
+ /* NOTE: The rest of the bits in this register */
+ /* _are_ relevant under Linux. */
+#define HCFG_PUSH_BUTTON_ENABLE 0x00100000 /* Enables Volume Inc/Dec and Mute functions */
+#define HCFG_BAUD_RATE 0x00080000 /* 0 = 48kHz, 1 = 44.1kHz */
+#define HCFG_EXPANDED_MEM 0x00040000 /* 1 = any 16M of 4G addr, 0 = 32M of 2G addr */
+#define HCFG_CODECFORMAT_MASK 0x00030000 /* CODEC format */
+
+/* Specific to Alice2, CA0102 */
+#define HCFG_CODECFORMAT_AC97_1 0x00000000 /* AC97 CODEC format -- Ver 1.03 */
+#define HCFG_CODECFORMAT_AC97_2 0x00010000 /* AC97 CODEC format -- Ver 2.1 */
+#define HCFG_AUTOMUTE_ASYNC 0x00008000 /* When set, the async sample rate convertors */
+ /* will automatically mute their output when */
+ /* they are not rate-locked to the external */
+ /* async audio source */
+#define HCFG_AUTOMUTE_SPDIF 0x00004000 /* When set, the async sample rate convertors */
+ /* will automatically mute their output when */
+ /* the SPDIF V-bit indicates invalid audio */
+#define HCFG_EMU32_SLAVE 0x00002000 /* 0 = Master, 1 = Slave. Slave for EMU1010 */
+#define HCFG_SLOW_RAMP 0x00001000 /* Increases Send Smoothing time constant */
+/* 0x00000800 not used on Alice2 */
+#define HCFG_PHASE_TRACK_MASK 0x00000700 /* When set, forces corresponding input to */
+ /* phase track the previous input. */
+ /* I2S0 can phase track the last S/PDIF input */
+#define HCFG_I2S_ASRC_ENABLE 0x00000070 /* When set, enables asynchronous sample rate */
+ /* conversion for the corresponding */
+ /* I2S format input */
+/* Rest of HCFG 0x0000000f same as below. LOCKSOUNDCACHE etc. */
+
+
+
+/* Older chips */
+#define HCFG_CODECFORMAT_AC97 0x00000000 /* AC97 CODEC format -- Primary Output */
+#define HCFG_CODECFORMAT_I2S 0x00010000 /* I2S CODEC format -- Secondary (Rear) Output */
+#define HCFG_GPINPUT0 0x00004000 /* External pin112 */
+#define HCFG_GPINPUT1 0x00002000 /* External pin110 */
+#define HCFG_GPOUTPUT_MASK 0x00001c00 /* External pins which may be controlled */
+#define HCFG_GPOUT0 0x00001000 /* External pin? (spdif enable on 5.1) */
+#define HCFG_GPOUT1 0x00000800 /* External pin? (IR) */
+#define HCFG_GPOUT2 0x00000400 /* External pin? (IR) */
+#define HCFG_JOYENABLE 0x00000200 /* Internal joystick enable */
+#define HCFG_PHASETRACKENABLE 0x00000100 /* Phase tracking enable */
+ /* 1 = Force all 3 async digital inputs to use */
+ /* the same async sample rate tracker (ZVIDEO) */
+#define HCFG_AC3ENABLE_MASK 0x000000e0 /* AC3 async input control - Not implemented */
+#define HCFG_AC3ENABLE_ZVIDEO 0x00000080 /* Channels 0 and 1 replace ZVIDEO */
+#define HCFG_AC3ENABLE_CDSPDIF 0x00000040 /* Channels 0 and 1 replace CDSPDIF */
+#define HCFG_AC3ENABLE_GPSPDIF 0x00000020 /* Channels 0 and 1 replace GPSPDIF */
+#define HCFG_AUTOMUTE 0x00000010 /* When set, the async sample rate convertors */
+ /* will automatically mute their output when */
+ /* they are not rate-locked to the external */
+ /* async audio source */
+#define HCFG_LOCKSOUNDCACHE 0x00000008 /* 1 = Cancel bustmaster accesses to soundcache */
+ /* NOTE: This should generally never be used. */
+#define HCFG_LOCKTANKCACHE_MASK 0x00000004 /* 1 = Cancel bustmaster accesses to tankcache */
+ /* NOTE: This should generally never be used. */
+#define HCFG_LOCKTANKCACHE 0x01020014
+#define HCFG_MUTEBUTTONENABLE 0x00000002 /* 1 = Master mute button sets AUDIOENABLE = 0. */
+ /* NOTE: This is a 'cheap' way to implement a */
+ /* master mute function on the mute button, and */
+ /* in general should not be used unless a more */
+ /* sophisticated master mute function has not */
+ /* been written. */
+#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */
+ /* Should be set to 1 when the EMU10K1 is */
+ /* completely initialized. */
+
+//For Audigy, MPU port move to 0x70-0x74 ptr register
+
+#define MUDATA 0x18 /* MPU401 data register (8 bits) */
+
+#define MUCMD 0x19 /* MPU401 command register (8 bits) */
+#define MUCMD_RESET 0xff /* RESET command */
+#define MUCMD_ENTERUARTMODE 0x3f /* Enter_UART_mode command */
+ /* NOTE: All other commands are ignored */
+
+#define MUSTAT MUCMD /* MPU401 status register (8 bits) */
+#define MUSTAT_IRDYN 0x80 /* 0 = MIDI data or command ACK */
+#define MUSTAT_ORDYN 0x40 /* 0 = MUDATA can accept a command or data */
+
+#define A_IOCFG 0x18 /* GPIO on Audigy card (16bits) */
+#define A_GPINPUT_MASK 0xff00
+#define A_GPOUTPUT_MASK 0x00ff
+
+// Audigy output/GPIO stuff taken from the kX drivers
+#define A_IOCFG_GPOUT0 0x0044 /* analog/digital */
+#define A_IOCFG_DISABLE_ANALOG 0x0040 /* = 'enable' for Audigy2 (chiprev=4) */
+#define A_IOCFG_ENABLE_DIGITAL 0x0004
+#define A_IOCFG_ENABLE_DIGITAL_AUDIGY4 0x0080
+#define A_IOCFG_UNKNOWN_20 0x0020
+#define A_IOCFG_DISABLE_AC97_FRONT 0x0080 /* turn off ac97 front -> front (10k2.1) */
+#define A_IOCFG_GPOUT1 0x0002 /* IR? drive's internal bypass (?) */
+#define A_IOCFG_GPOUT2 0x0001 /* IR */
+#define A_IOCFG_MULTIPURPOSE_JACK 0x2000 /* center+lfe+rear_center (a2/a2ex) */
+ /* + digital for generic 10k2 */
+#define A_IOCFG_DIGITAL_JACK 0x1000 /* digital for a2 platinum */
+#define A_IOCFG_FRONT_JACK 0x4000
+#define A_IOCFG_REAR_JACK 0x8000
+#define A_IOCFG_PHONES_JACK 0x0100 /* LiveDrive */
+
+/* outputs:
+ * for audigy2 platinum: 0xa00
+ * for a2 platinum ex: 0x1c00
+ * for a1 platinum: 0x0
+ */
+
+#define TIMER 0x1a /* Timer terminal count register */
+ /* NOTE: After the rate is changed, a maximum */
+ /* of 1024 sample periods should be allowed */
+ /* before the new rate is guaranteed accurate. */
+#define TIMER_RATE_MASK 0x000003ff /* Timer interrupt rate in sample periods */
+ /* 0 == 1024 periods, [1..4] are not useful */
+#define TIMER_RATE 0x0a00001a
+
+#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */
+
+#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
+#define AC97ADDRESS_READY 0x80 /* Read-only bit, reflects CODEC READY signal */
+#define AC97ADDRESS_ADDRESS 0x7f /* Address of indexed AC97 register */
+
+/* Available on the Audigy 2 and Audigy 4 only. This is the P16V chip. */
+#define PTR2 0x20 /* Indexed register set pointer register */
+#define DATA2 0x24 /* Indexed register set data register */
+#define IPR2 0x28 /* P16V interrupt pending register */
+#define IPR2_PLAYBACK_CH_0_LOOP 0x00001000 /* Playback Channel 0 loop */
+#define IPR2_PLAYBACK_CH_0_HALF_LOOP 0x00000100 /* Playback Channel 0 half loop */
+#define IPR2_CAPTURE_CH_0_LOOP 0x00100000 /* Capture Channel 0 loop */
+#define IPR2_CAPTURE_CH_0_HALF_LOOP 0x00010000 /* Capture Channel 0 half loop */
+ /* 0x00000100 Playback. Only in once per period.
+ * 0x00110000 Capture. Int on half buffer.
+ */
+#define INTE2 0x2c /* P16V Interrupt enable register. */
+#define INTE2_PLAYBACK_CH_0_LOOP 0x00001000 /* Playback Channel 0 loop */
+#define INTE2_PLAYBACK_CH_0_HALF_LOOP 0x00000100 /* Playback Channel 0 half loop */
+#define INTE2_PLAYBACK_CH_1_LOOP 0x00002000 /* Playback Channel 1 loop */
+#define INTE2_PLAYBACK_CH_1_HALF_LOOP 0x00000200 /* Playback Channel 1 half loop */
+#define INTE2_PLAYBACK_CH_2_LOOP 0x00004000 /* Playback Channel 2 loop */
+#define INTE2_PLAYBACK_CH_2_HALF_LOOP 0x00000400 /* Playback Channel 2 half loop */
+#define INTE2_PLAYBACK_CH_3_LOOP 0x00008000 /* Playback Channel 3 loop */
+#define INTE2_PLAYBACK_CH_3_HALF_LOOP 0x00000800 /* Playback Channel 3 half loop */
+#define INTE2_CAPTURE_CH_0_LOOP 0x00100000 /* Capture Channel 0 loop */
+#define INTE2_CAPTURE_CH_0_HALF_LOOP 0x00010000 /* Caputre Channel 0 half loop */
+#define HCFG2 0x34 /* Defaults: 0, win2000 sets it to 00004201 */
+ /* 0x00000000 2-channel output. */
+ /* 0x00000200 8-channel output. */
+ /* 0x00000004 pauses stream/irq fail. */
+ /* Rest of bits no nothing to sound output */
+ /* bit 0: Enable P16V audio.
+ * bit 1: Lock P16V record memory cache.
+ * bit 2: Lock P16V playback memory cache.
+ * bit 3: Dummy record insert zero samples.
+ * bit 8: Record 8-channel in phase.
+ * bit 9: Playback 8-channel in phase.
+ * bit 11-12: Playback mixer attenuation: 0=0dB, 1=-6dB, 2=-12dB, 3=Mute.
+ * bit 13: Playback mixer enable.
+ * bit 14: Route SRC48 mixer output to fx engine.
+ * bit 15: Enable IEEE 1394 chip.
+ */
+#define IPR3 0x38 /* Cdif interrupt pending register */
+#define INTE3 0x3c /* Cdif interrupt enable register. */
+/************************************************************************************************/
+/* PCI function 1 registers, address = <val> + PCIBASE1 */
+/************************************************************************************************/
+
+#define JOYSTICK1 0x00 /* Analog joystick port register */
+#define JOYSTICK2 0x01 /* Analog joystick port register */
+#define JOYSTICK3 0x02 /* Analog joystick port register */
+#define JOYSTICK4 0x03 /* Analog joystick port register */
+#define JOYSTICK5 0x04 /* Analog joystick port register */
+#define JOYSTICK6 0x05 /* Analog joystick port register */
+#define JOYSTICK7 0x06 /* Analog joystick port register */
+#define JOYSTICK8 0x07 /* Analog joystick port register */
+
+/* When writing, any write causes JOYSTICK_COMPARATOR output enable to be pulsed on write. */
+/* When reading, use these bitfields: */
+#define JOYSTICK_BUTTONS 0x0f /* Joystick button data */
+#define JOYSTICK_COMPARATOR 0xf0 /* Joystick comparator data */
+
+
+/********************************************************************************************************/
+/* Emu10k1 pointer-offset register set, accessed through the PTR and DATA registers */
+/********************************************************************************************************/
+
+#define CPF 0x00 /* Current pitch and fraction register */
+#define CPF_CURRENTPITCH_MASK 0xffff0000 /* Current pitch (linear, 0x4000 == unity pitch shift) */
+#define CPF_CURRENTPITCH 0x10100000
+#define CPF_STEREO_MASK 0x00008000 /* 1 = Even channel interleave, odd channel locked */
+#define CPF_STOP_MASK 0x00004000 /* 1 = Current pitch forced to 0 */
+#define CPF_FRACADDRESS_MASK 0x00003fff /* Linear fractional address of the current channel */
+
+#define PTRX 0x01 /* Pitch target and send A/B amounts register */
+#define PTRX_PITCHTARGET_MASK 0xffff0000 /* Pitch target of specified channel */
+#define PTRX_PITCHTARGET 0x10100001
+#define PTRX_FXSENDAMOUNT_A_MASK 0x0000ff00 /* Linear level of channel output sent to FX send bus A */
+#define PTRX_FXSENDAMOUNT_A 0x08080001
+#define PTRX_FXSENDAMOUNT_B_MASK 0x000000ff /* Linear level of channel output sent to FX send bus B */
+#define PTRX_FXSENDAMOUNT_B 0x08000001
+
+#define CVCF 0x02 /* Current volume and filter cutoff register */
+#define CVCF_CURRENTVOL_MASK 0xffff0000 /* Current linear volume of specified channel */
+#define CVCF_CURRENTVOL 0x10100002
+#define CVCF_CURRENTFILTER_MASK 0x0000ffff /* Current filter cutoff frequency of specified channel */
+#define CVCF_CURRENTFILTER 0x10000002
+
+#define VTFT 0x03 /* Volume target and filter cutoff target register */
+#define VTFT_VOLUMETARGET_MASK 0xffff0000 /* Volume target of specified channel */
+#define VTFT_VOLUMETARGET 0x10100003
+#define VTFT_FILTERTARGET_MASK 0x0000ffff /* Filter cutoff target of specified channel */
+#define VTFT_FILTERTARGET 0x10000003
+
+#define Z1 0x05 /* Filter delay memory 1 register */
+
+#define Z2 0x04 /* Filter delay memory 2 register */
+
+#define PSST 0x06 /* Send C amount and loop start address register */
+#define PSST_FXSENDAMOUNT_C_MASK 0xff000000 /* Linear level of channel output sent to FX send bus C */
+
+#define PSST_FXSENDAMOUNT_C 0x08180006
+
+#define PSST_LOOPSTARTADDR_MASK 0x00ffffff /* Loop start address of the specified channel */
+#define PSST_LOOPSTARTADDR 0x18000006
+
+#define DSL 0x07 /* Send D amount and loop start address register */
+#define DSL_FXSENDAMOUNT_D_MASK 0xff000000 /* Linear level of channel output sent to FX send bus D */
+
+#define DSL_FXSENDAMOUNT_D 0x08180007
+
+#define DSL_LOOPENDADDR_MASK 0x00ffffff /* Loop end address of the specified channel */
+#define DSL_LOOPENDADDR 0x18000007
+
+#define CCCA 0x08 /* Filter Q, interp. ROM, byte size, cur. addr register */
+#define CCCA_RESONANCE 0xf0000000 /* Lowpass filter resonance (Q) height */
+#define CCCA_INTERPROMMASK 0x0e000000 /* Selects passband of interpolation ROM */
+ /* 1 == full band, 7 == lowpass */
+ /* ROM 0 is used when pitch shifting downward or less */
+ /* then 3 semitones upward. Increasingly higher ROM */
+ /* numbers are used, typically in steps of 3 semitones, */
+ /* as upward pitch shifting is performed. */
+#define CCCA_INTERPROM_0 0x00000000 /* Select interpolation ROM 0 */
+#define CCCA_INTERPROM_1 0x02000000 /* Select interpolation ROM 1 */
+#define CCCA_INTERPROM_2 0x04000000 /* Select interpolation ROM 2 */
+#define CCCA_INTERPROM_3 0x06000000 /* Select interpolation ROM 3 */
+#define CCCA_INTERPROM_4 0x08000000 /* Select interpolation ROM 4 */
+#define CCCA_INTERPROM_5 0x0a000000 /* Select interpolation ROM 5 */
+#define CCCA_INTERPROM_6 0x0c000000 /* Select interpolation ROM 6 */
+#define CCCA_INTERPROM_7 0x0e000000 /* Select interpolation ROM 7 */
+#define CCCA_8BITSELECT 0x01000000 /* 1 = Sound memory for this channel uses 8-bit samples */
+#define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */
+#define CCCA_CURRADDR 0x18000008
+
+#define CCR 0x09 /* Cache control register */
+#define CCR_CACHEINVALIDSIZE 0x07190009
+#define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples cache for this channel */
+#define CCR_CACHELOOPFLAG 0x01000000 /* 1 = Cache has a loop service pending */
+#define CCR_INTERLEAVEDSAMPLES 0x00800000 /* 1 = A cache service will fetch interleaved samples */
+#define CCR_WORDSIZEDSAMPLES 0x00400000 /* 1 = A cache service will fetch word sized samples */
+#define CCR_READADDRESS 0x06100009
+#define CCR_READADDRESS_MASK 0x003f0000 /* Location of cache just beyond current cache service */
+#define CCR_LOOPINVALSIZE 0x0000fe00 /* Number of invalid samples in cache prior to loop */
+ /* NOTE: This is valid only if CACHELOOPFLAG is set */
+#define CCR_LOOPFLAG 0x00000100 /* Set for a single sample period when a loop occurs */
+#define CCR_CACHELOOPADDRHI 0x000000ff /* DSL_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */
+
+#define CLP 0x0a /* Cache loop register (valid if CCR_CACHELOOPFLAG = 1) */
+ /* NOTE: This register is normally not used */
+#define CLP_CACHELOOPADDR 0x0000ffff /* Cache loop address (DSL_LOOPSTARTADDR [0..15]) */
+
+#define FXRT 0x0b /* Effects send routing register */
+ /* NOTE: It is illegal to assign the same routing to */
+ /* two effects sends. */
+#define FXRT_CHANNELA 0x000f0000 /* Effects send bus number for channel's effects send A */
+#define FXRT_CHANNELB 0x00f00000 /* Effects send bus number for channel's effects send B */
+#define FXRT_CHANNELC 0x0f000000 /* Effects send bus number for channel's effects send C */
+#define FXRT_CHANNELD 0xf0000000 /* Effects send bus number for channel's effects send D */
+
+#define A_HR 0x0b /* High Resolution. 24bit playback from host to DSP. */
+#define MAPA 0x0c /* Cache map A */
+
+#define MAPB 0x0d /* Cache map B */
+
+#define MAP_PTE_MASK0 0xfffff000 /* The 20 MSBs of the PTE indexed by the PTI */
+#define MAP_PTI_MASK0 0x00000fff /* The 12 bit index to one of the 4096 PTE dwords */
+
+#define MAP_PTE_MASK1 0xffffe000 /* The 19 MSBs of the PTE indexed by the PTI */
+#define MAP_PTI_MASK1 0x00001fff /* The 13 bit index to one of the 8192 PTE dwords */
+
+/* 0x0e, 0x0f: Not used */
+
+#define ENVVOL 0x10 /* Volume envelope register */
+#define ENVVOL_MASK 0x0000ffff /* Current value of volume envelope state variable */
+ /* 0x8000-n == 666*n usec delay */
+
+#define ATKHLDV 0x11 /* Volume envelope hold and attack register */
+#define ATKHLDV_PHASE0 0x00008000 /* 0 = Begin attack phase */
+#define ATKHLDV_HOLDTIME_MASK 0x00007f00 /* Envelope hold time (127-n == n*88.2msec) */
+#define ATKHLDV_ATTACKTIME_MASK 0x0000007f /* Envelope attack time, log encoded */
+ /* 0 = infinite, 1 = 10.9msec, ... 0x7f = 5.5msec */
+
+#define DCYSUSV 0x12 /* Volume envelope sustain and decay register */
+#define DCYSUSV_PHASE1_MASK 0x00008000 /* 0 = Begin attack phase, 1 = begin release phase */
+#define DCYSUSV_SUSTAINLEVEL_MASK 0x00007f00 /* 127 = full, 0 = off, 0.75dB increments */
+#define DCYSUSV_CHANNELENABLE_MASK 0x00000080 /* 1 = Inhibit envelope engine from writing values in */
+ /* this channel and from writing to pitch, filter and */
+ /* volume targets. */
+#define DCYSUSV_DECAYTIME_MASK 0x0000007f /* Volume envelope decay time, log encoded */
+ /* 0 = 43.7msec, 1 = 21.8msec, 0x7f = 22msec */
+
+#define LFOVAL1 0x13 /* Modulation LFO value */
+#define LFOVAL_MASK 0x0000ffff /* Current value of modulation LFO state variable */
+ /* 0x8000-n == 666*n usec delay */
+
+#define ENVVAL 0x14 /* Modulation envelope register */
+#define ENVVAL_MASK 0x0000ffff /* Current value of modulation envelope state variable */
+ /* 0x8000-n == 666*n usec delay */
+
+#define ATKHLDM 0x15 /* Modulation envelope hold and attack register */
+#define ATKHLDM_PHASE0 0x00008000 /* 0 = Begin attack phase */
+#define ATKHLDM_HOLDTIME 0x00007f00 /* Envelope hold time (127-n == n*42msec) */
+#define ATKHLDM_ATTACKTIME 0x0000007f /* Envelope attack time, log encoded */
+ /* 0 = infinite, 1 = 11msec, ... 0x7f = 5.5msec */
+
+#define DCYSUSM 0x16 /* Modulation envelope decay and sustain register */
+#define DCYSUSM_PHASE1_MASK 0x00008000 /* 0 = Begin attack phase, 1 = begin release phase */
+#define DCYSUSM_SUSTAINLEVEL_MASK 0x00007f00 /* 127 = full, 0 = off, 0.75dB increments */
+#define DCYSUSM_DECAYTIME_MASK 0x0000007f /* Envelope decay time, log encoded */
+ /* 0 = 43.7msec, 1 = 21.8msec, 0x7f = 22msec */
+
+#define LFOVAL2 0x17 /* Vibrato LFO register */
+#define LFOVAL2_MASK 0x0000ffff /* Current value of vibrato LFO state variable */
+ /* 0x8000-n == 666*n usec delay */
+
+#define IP 0x18 /* Initial pitch register */
+#define IP_MASK 0x0000ffff /* Exponential initial pitch shift */
+ /* 4 bits of octave, 12 bits of fractional octave */
+#define IP_UNITY 0x0000e000 /* Unity pitch shift */
+
+#define IFATN 0x19 /* Initial filter cutoff and attenuation register */
+#define IFATN_FILTERCUTOFF_MASK 0x0000ff00 /* Initial filter cutoff frequency in exponential units */
+ /* 6 most significant bits are semitones */
+ /* 2 least significant bits are fractions */
+#define IFATN_FILTERCUTOFF 0x08080019
+#define IFATN_ATTENUATION_MASK 0x000000ff /* Initial attenuation in 0.375dB steps */
+#define IFATN_ATTENUATION 0x08000019
+
+
+#define PEFE 0x1a /* Pitch envelope and filter envelope amount register */
+#define PEFE_PITCHAMOUNT_MASK 0x0000ff00 /* Pitch envlope amount */
+ /* Signed 2's complement, +/- one octave peak extremes */
+#define PEFE_PITCHAMOUNT 0x0808001a
+#define PEFE_FILTERAMOUNT_MASK 0x000000ff /* Filter envlope amount */
+ /* Signed 2's complement, +/- six octaves peak extremes */
+#define PEFE_FILTERAMOUNT 0x0800001a
+#define FMMOD 0x1b /* Vibrato/filter modulation from LFO register */
+#define FMMOD_MODVIBRATO 0x0000ff00 /* Vibrato LFO modulation depth */
+ /* Signed 2's complement, +/- one octave extremes */
+#define FMMOD_MOFILTER 0x000000ff /* Filter LFO modulation depth */
+ /* Signed 2's complement, +/- three octave extremes */
+
+
+#define TREMFRQ 0x1c /* Tremolo amount and modulation LFO frequency register */
+#define TREMFRQ_DEPTH 0x0000ff00 /* Tremolo depth */
+ /* Signed 2's complement, with +/- 12dB extremes */
+
+#define TREMFRQ_FREQUENCY 0x000000ff /* Tremolo LFO frequency */
+ /* ??Hz steps, maximum of ?? Hz. */
+#define FM2FRQ2 0x1d /* Vibrato amount and vibrato LFO frequency register */
+#define FM2FRQ2_DEPTH 0x0000ff00 /* Vibrato LFO vibrato depth */
+ /* Signed 2's complement, +/- one octave extremes */
+#define FM2FRQ2_FREQUENCY 0x000000ff /* Vibrato LFO frequency */
+ /* 0.039Hz steps, maximum of 9.85 Hz. */
+
+#define TEMPENV 0x1e /* Tempory envelope register */
+#define TEMPENV_MASK 0x0000ffff /* 16-bit value */
+ /* NOTE: All channels contain internal variables; do */
+ /* not write to these locations. */
+
+/* 0x1f: not used */
+
+#define CD0 0x20 /* Cache data 0 register */
+#define CD1 0x21 /* Cache data 1 register */
+#define CD2 0x22 /* Cache data 2 register */
+#define CD3 0x23 /* Cache data 3 register */
+#define CD4 0x24 /* Cache data 4 register */
+#define CD5 0x25 /* Cache data 5 register */
+#define CD6 0x26 /* Cache data 6 register */
+#define CD7 0x27 /* Cache data 7 register */
+#define CD8 0x28 /* Cache data 8 register */
+#define CD9 0x29 /* Cache data 9 register */
+#define CDA 0x2a /* Cache data A register */
+#define CDB 0x2b /* Cache data B register */
+#define CDC 0x2c /* Cache data C register */
+#define CDD 0x2d /* Cache data D register */
+#define CDE 0x2e /* Cache data E register */
+#define CDF 0x2f /* Cache data F register */
+
+/* 0x30-3f seem to be the same as 0x20-2f */
+
+#define PTB 0x40 /* Page table base register */
+#define PTB_MASK 0xfffff000 /* Physical address of the page table in host memory */
+
+#define TCB 0x41 /* Tank cache base register */
+#define TCB_MASK 0xfffff000 /* Physical address of the bottom of host based TRAM */
+
+#define ADCCR 0x42 /* ADC sample rate/stereo control register */
+#define ADCCR_RCHANENABLE 0x00000010 /* Enables right channel for writing to the host */
+#define ADCCR_LCHANENABLE 0x00000008 /* Enables left channel for writing to the host */
+ /* NOTE: To guarantee phase coherency, both channels */
+ /* must be disabled prior to enabling both channels. */
+#define A_ADCCR_RCHANENABLE 0x00000020
+#define A_ADCCR_LCHANENABLE 0x00000010
+
+#define A_ADCCR_SAMPLERATE_MASK 0x0000000F /* Audigy sample rate convertor output rate */
+#define ADCCR_SAMPLERATE_MASK 0x00000007 /* Sample rate convertor output rate */
+#define ADCCR_SAMPLERATE_48 0x00000000 /* 48kHz sample rate */
+#define ADCCR_SAMPLERATE_44 0x00000001 /* 44.1kHz sample rate */
+#define ADCCR_SAMPLERATE_32 0x00000002 /* 32kHz sample rate */
+#define ADCCR_SAMPLERATE_24 0x00000003 /* 24kHz sample rate */
+#define ADCCR_SAMPLERATE_22 0x00000004 /* 22.05kHz sample rate */
+#define ADCCR_SAMPLERATE_16 0x00000005 /* 16kHz sample rate */
+#define ADCCR_SAMPLERATE_11 0x00000006 /* 11.025kHz sample rate */
+#define ADCCR_SAMPLERATE_8 0x00000007 /* 8kHz sample rate */
+#define A_ADCCR_SAMPLERATE_12 0x00000006 /* 12kHz sample rate */
+#define A_ADCCR_SAMPLERATE_11 0x00000007 /* 11.025kHz sample rate */
+#define A_ADCCR_SAMPLERATE_8 0x00000008 /* 8kHz sample rate */
+
+#define FXWC 0x43 /* FX output write channels register */
+ /* When set, each bit enables the writing of the */
+ /* corresponding FX output channel (internal registers */
+ /* 0x20-0x3f) to host memory. This mode of recording */
+ /* is 16bit, 48KHz only. All 32 channels can be enabled */
+ /* simultaneously. */
+
+#define FXWC_DEFAULTROUTE_C (1<<0) /* left emu out? */
+#define FXWC_DEFAULTROUTE_B (1<<1) /* right emu out? */
+#define FXWC_DEFAULTROUTE_A (1<<12)
+#define FXWC_DEFAULTROUTE_D (1<<13)
+#define FXWC_ADCLEFT (1<<18)
+#define FXWC_CDROMSPDIFLEFT (1<<18)
+#define FXWC_ADCRIGHT (1<<19)
+#define FXWC_CDROMSPDIFRIGHT (1<<19)
+#define FXWC_MIC (1<<20)
+#define FXWC_ZOOMLEFT (1<<20)
+#define FXWC_ZOOMRIGHT (1<<21)
+#define FXWC_SPDIFLEFT (1<<22) /* 0x00400000 */
+#define FXWC_SPDIFRIGHT (1<<23) /* 0x00800000 */
+
+#define A_TBLSZ 0x43 /* Effects Tank Internal Table Size. Only low byte or register used */
+
+#define TCBS 0x44 /* Tank cache buffer size register */
+#define TCBS_MASK 0x00000007 /* Tank cache buffer size field */
+#define TCBS_BUFFSIZE_16K 0x00000000
+#define TCBS_BUFFSIZE_32K 0x00000001
+#define TCBS_BUFFSIZE_64K 0x00000002
+#define TCBS_BUFFSIZE_128K 0x00000003
+#define TCBS_BUFFSIZE_256K 0x00000004
+#define TCBS_BUFFSIZE_512K 0x00000005
+#define TCBS_BUFFSIZE_1024K 0x00000006
+#define TCBS_BUFFSIZE_2048K 0x00000007
+
+#define MICBA 0x45 /* AC97 microphone buffer address register */
+#define MICBA_MASK 0xfffff000 /* 20 bit base address */
+
+#define ADCBA 0x46 /* ADC buffer address register */
+#define ADCBA_MASK 0xfffff000 /* 20 bit base address */
+
+#define FXBA 0x47 /* FX Buffer Address */
+#define FXBA_MASK 0xfffff000 /* 20 bit base address */
+
+#define A_HWM 0x48 /* High PCI Water Mark - word access, defaults to 3f */
+
+#define MICBS 0x49 /* Microphone buffer size register */
+
+#define ADCBS 0x4a /* ADC buffer size register */
+
+#define FXBS 0x4b /* FX buffer size register */
+
+/* register: 0x4c..4f: ffff-ffff current amounts, per-channel */
+
+/* The following mask values define the size of the ADC, MIX and FX buffers in bytes */
+#define ADCBS_BUFSIZE_NONE 0x00000000
+#define ADCBS_BUFSIZE_384 0x00000001
+#define ADCBS_BUFSIZE_448 0x00000002
+#define ADCBS_BUFSIZE_512 0x00000003
+#define ADCBS_BUFSIZE_640 0x00000004
+#define ADCBS_BUFSIZE_768 0x00000005
+#define ADCBS_BUFSIZE_896 0x00000006
+#define ADCBS_BUFSIZE_1024 0x00000007
+#define ADCBS_BUFSIZE_1280 0x00000008
+#define ADCBS_BUFSIZE_1536 0x00000009
+#define ADCBS_BUFSIZE_1792 0x0000000a
+#define ADCBS_BUFSIZE_2048 0x0000000b
+#define ADCBS_BUFSIZE_2560 0x0000000c
+#define ADCBS_BUFSIZE_3072 0x0000000d
+#define ADCBS_BUFSIZE_3584 0x0000000e
+#define ADCBS_BUFSIZE_4096 0x0000000f
+#define ADCBS_BUFSIZE_5120 0x00000010
+#define ADCBS_BUFSIZE_6144 0x00000011
+#define ADCBS_BUFSIZE_7168 0x00000012
+#define ADCBS_BUFSIZE_8192 0x00000013
+#define ADCBS_BUFSIZE_10240 0x00000014
+#define ADCBS_BUFSIZE_12288 0x00000015
+#define ADCBS_BUFSIZE_14366 0x00000016
+#define ADCBS_BUFSIZE_16384 0x00000017
+#define ADCBS_BUFSIZE_20480 0x00000018
+#define ADCBS_BUFSIZE_24576 0x00000019
+#define ADCBS_BUFSIZE_28672 0x0000001a
+#define ADCBS_BUFSIZE_32768 0x0000001b
+#define ADCBS_BUFSIZE_40960 0x0000001c
+#define ADCBS_BUFSIZE_49152 0x0000001d
+#define ADCBS_BUFSIZE_57344 0x0000001e
+#define ADCBS_BUFSIZE_65536 0x0000001f
+
+/* Current Send B, A Amounts */
+#define A_CSBA 0x4c
+
+/* Current Send D, C Amounts */
+#define A_CSDC 0x4d
+
+/* Current Send F, E Amounts */
+#define A_CSFE 0x4e
+
+/* Current Send H, G Amounts */
+#define A_CSHG 0x4f
+
+
+#define CDCS 0x50 /* CD-ROM digital channel status register */
+
+#define GPSCS 0x51 /* General Purpose SPDIF channel status register*/
+
+#define DBG 0x52 /* DO NOT PROGRAM THIS REGISTER!!! MAY DESTROY CHIP */
+
+/* S/PDIF Input C Channel Status */
+#define A_SPSC 0x52
+
+#define REG53 0x53 /* DO NOT PROGRAM THIS REGISTER!!! MAY DESTROY CHIP */
+
+#define A_DBG 0x53
+#define A_DBG_SINGLE_STEP 0x00020000 /* Set to zero to start dsp */
+#define A_DBG_ZC 0x40000000 /* zero tram counter */
+#define A_DBG_STEP_ADDR 0x000003ff
+#define A_DBG_SATURATION_OCCURED 0x20000000
+#define A_DBG_SATURATION_ADDR 0x0ffc0000
+
+// NOTE: 0x54,55,56: 64-bit
+#define SPCS0 0x54 /* SPDIF output Channel Status 0 register */
+
+#define SPCS1 0x55 /* SPDIF output Channel Status 1 register */
+
+#define SPCS2 0x56 /* SPDIF output Channel Status 2 register */
+
+#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */
+#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */
+#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */
+#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */
+#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */
+#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */
+#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */
+#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */
+#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */
+#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */
+#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */
+#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */
+#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */
+#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */
+#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */
+#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */
+#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */
+#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */
+#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */
+#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */
+#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */
+#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */
+#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */
+
+/* 0x57: Not used */
+
+/* The 32-bit CLIx and SOLx registers all have one bit per channel control/status */
+#define CLIEL 0x58 /* Channel loop interrupt enable low register */
+
+#define CLIEH 0x59 /* Channel loop interrupt enable high register */
+
+#define CLIPL 0x5a /* Channel loop interrupt pending low register */
+
+#define CLIPH 0x5b /* Channel loop interrupt pending high register */
+
+#define SOLEL 0x5c /* Stop on loop enable low register */
+
+#define SOLEH 0x5d /* Stop on loop enable high register */
+
+#define SPBYPASS 0x5e /* SPDIF BYPASS mode register */
+#define SPBYPASS_SPDIF0_MASK 0x00000003 /* SPDIF 0 bypass mode */
+#define SPBYPASS_SPDIF1_MASK 0x0000000c /* SPDIF 1 bypass mode */
+/* bypass mode: 0 - DSP; 1 - SPDIF A, 2 - SPDIF B, 3 - SPDIF C */
+#define SPBYPASS_FORMAT 0x00000f00 /* If 1, SPDIF XX uses 24 bit, if 0 - 20 bit */
+
+#define AC97SLOT 0x5f /* additional AC97 slots enable bits */
+#define AC97SLOT_REAR_RIGHT 0x01 /* Rear left */
+#define AC97SLOT_REAR_LEFT 0x02 /* Rear right */
+#define AC97SLOT_CNTR 0x10 /* Center enable */
+#define AC97SLOT_LFE 0x20 /* LFE enable */
+
+/* PCB Revision */
+#define A_PCB 0x5f
+
+// NOTE: 0x60,61,62: 64-bit
+#define CDSRCS 0x60 /* CD-ROM Sample Rate Converter status register */
+
+#define GPSRCS 0x61 /* General Purpose SPDIF sample rate cvt status */
+
+#define ZVSRCS 0x62 /* ZVideo sample rate converter status */
+ /* NOTE: This one has no SPDIFLOCKED field */
+ /* Assumes sample lock */
+
+/* These three bitfields apply to CDSRCS, GPSRCS, and (except as noted) ZVSRCS. */
+#define SRCS_SPDIFVALID 0x04000000 /* SPDIF stream valid */
+#define SRCS_SPDIFLOCKED 0x02000000 /* SPDIF stream locked */
+#define SRCS_RATELOCKED 0x01000000 /* Sample rate locked */
+#define SRCS_ESTSAMPLERATE 0x0007ffff /* Do not modify this field. */
+
+/* Note that these values can vary +/- by a small amount */
+#define SRCS_SPDIFRATE_44 0x0003acd9
+#define SRCS_SPDIFRATE_48 0x00040000
+#define SRCS_SPDIFRATE_96 0x00080000
+
+#define MICIDX 0x63 /* Microphone recording buffer index register */
+#define MICIDX_MASK 0x0000ffff /* 16-bit value */
+#define MICIDX_IDX 0x10000063
+
+#define ADCIDX 0x64 /* ADC recording buffer index register */
+#define ADCIDX_MASK 0x0000ffff /* 16 bit index field */
+#define ADCIDX_IDX 0x10000064
+
+#define A_ADCIDX 0x63
+#define A_ADCIDX_IDX 0x10000063
+
+#define A_MICIDX 0x64
+#define A_MICIDX_IDX 0x10000064
+
+#define FXIDX 0x65 /* FX recording buffer index register */
+#define FXIDX_MASK 0x0000ffff /* 16-bit value */
+#define FXIDX_IDX 0x10000065
+
+/* The 32-bit HLIx and HLIPx registers all have one bit per channel control/status */
+#define HLIEL 0x66 /* Channel half loop interrupt enable low register */
+
+#define HLIEH 0x67 /* Channel half loop interrupt enable high register */
+
+#define HLIPL 0x68 /* Channel half loop interrupt pending low register */
+
+#define HLIPH 0x69 /* Channel half loop interrupt pending high register */
+
+/* S/PDIF Host Record Index (bypasses SRC) */
+#define A_SPRI 0x6a
+/* S/PDIF Host Record Address */
+#define A_SPRA 0x6b
+/* S/PDIF Host Record Control */
+#define A_SPRC 0x6c
+/* Delayed Interrupt Counter & Enable */
+#define A_DICE 0x6d
+/* Tank Table Base */
+#define A_TTB 0x6e
+/* Tank Delay Offset */
+#define A_TDOF 0x6f
+
+/* This is the MPU port on the card (via the game port) */
+#define A_MUDATA1 0x70
+#define A_MUCMD1 0x71
+#define A_MUSTAT1 A_MUCMD1
+
+/* This is the MPU port on the Audigy Drive */
+#define A_MUDATA2 0x72
+#define A_MUCMD2 0x73
+#define A_MUSTAT2 A_MUCMD2
+
+/* The next two are the Audigy equivalent of FXWC */
+/* the Audigy can record any output (16bit, 48kHz, up to 64 channel simultaneously) */
+/* Each bit selects a channel for recording */
+#define A_FXWC1 0x74 /* Selects 0x7f-0x60 for FX recording */
+#define A_FXWC2 0x75 /* Selects 0x9f-0x80 for FX recording */
+
+/* Extended Hardware Control */
+#define A_SPDIF_SAMPLERATE 0x76 /* Set the sample rate of SPDIF output */
+#define A_SAMPLE_RATE 0x76 /* Various sample rate settings. */
+#define A_SAMPLE_RATE_NOT_USED 0x0ffc111e /* Bits that are not used and cannot be set. */
+#define A_SAMPLE_RATE_UNKNOWN 0xf0030001 /* Bits that can be set, but have unknown use. */
+#define A_SPDIF_RATE_MASK 0x000000e0 /* Any other values for rates, just use 48000 */
+#define A_SPDIF_48000 0x00000000
+#define A_SPDIF_192000 0x00000020
+#define A_SPDIF_96000 0x00000040
+#define A_SPDIF_44100 0x00000080
+
+#define A_I2S_CAPTURE_RATE_MASK 0x00000e00 /* This sets the capture PCM rate, but it is */
+#define A_I2S_CAPTURE_48000 0x00000000 /* unclear if this sets the ADC rate as well. */
+#define A_I2S_CAPTURE_192000 0x00000200
+#define A_I2S_CAPTURE_96000 0x00000400
+#define A_I2S_CAPTURE_44100 0x00000800
+
+#define A_PCM_RATE_MASK 0x0000e000 /* This sets the playback PCM rate on the P16V */
+#define A_PCM_48000 0x00000000
+#define A_PCM_192000 0x00002000
+#define A_PCM_96000 0x00004000
+#define A_PCM_44100 0x00008000
+
+/* I2S0 Sample Rate Tracker Status */
+#define A_SRT3 0x77
+
+/* I2S1 Sample Rate Tracker Status */
+#define A_SRT4 0x78
+
+/* I2S2 Sample Rate Tracker Status */
+#define A_SRT5 0x79
+/* - default to 0x01080000 on my audigy 2 ZS --rlrevell */
+
+/* Tank Table DMA Address */
+#define A_TTDA 0x7a
+/* Tank Table DMA Data */
+#define A_TTDD 0x7b
+
+#define A_FXRT2 0x7c
+#define A_FXRT_CHANNELE 0x0000003f /* Effects send bus number for channel's effects send E */
+#define A_FXRT_CHANNELF 0x00003f00 /* Effects send bus number for channel's effects send F */
+#define A_FXRT_CHANNELG 0x003f0000 /* Effects send bus number for channel's effects send G */
+#define A_FXRT_CHANNELH 0x3f000000 /* Effects send bus number for channel's effects send H */
+
+#define A_SENDAMOUNTS 0x7d
+#define A_FXSENDAMOUNT_E_MASK 0xFF000000
+#define A_FXSENDAMOUNT_F_MASK 0x00FF0000
+#define A_FXSENDAMOUNT_G_MASK 0x0000FF00
+#define A_FXSENDAMOUNT_H_MASK 0x000000FF
+/* 0x7c, 0x7e "high bit is used for filtering" */
+
+/* The send amounts for this one are the same as used with the emu10k1 */
+#define A_FXRT1 0x7e
+#define A_FXRT_CHANNELA 0x0000003f
+#define A_FXRT_CHANNELB 0x00003f00
+#define A_FXRT_CHANNELC 0x003f0000
+#define A_FXRT_CHANNELD 0x3f000000
+
+/* 0x7f: Not used */
+/* Each FX general purpose register is 32 bits in length, all bits are used */
+#define FXGPREGBASE 0x100 /* FX general purpose registers base */
+#define A_FXGPREGBASE 0x400 /* Audigy GPRs, 0x400 to 0x5ff */
+
+#define A_TANKMEMCTLREGBASE 0x100 /* Tank memory control registers base - only for Audigy */
+#define A_TANKMEMCTLREG_MASK 0x1f /* only 5 bits used - only for Audigy */
+
+/* Tank audio data is logarithmically compressed down to 16 bits before writing to TRAM and is */
+/* decompressed back to 20 bits on a read. There are a total of 160 locations, the last 32 */
+/* locations are for external TRAM. */
+#define TANKMEMDATAREGBASE 0x200 /* Tank memory data registers base */
+#define TANKMEMDATAREG_MASK 0x000fffff /* 20 bit tank audio data field */
+
+/* Combined address field and memory opcode or flag field. 160 locations, last 32 are external */
+#define TANKMEMADDRREGBASE 0x300 /* Tank memory address registers base */
+#define TANKMEMADDRREG_ADDR_MASK 0x000fffff /* 20 bit tank address field */
+#define TANKMEMADDRREG_CLEAR 0x00800000 /* Clear tank memory */
+#define TANKMEMADDRREG_ALIGN 0x00400000 /* Align read or write relative to tank access */
+#define TANKMEMADDRREG_WRITE 0x00200000 /* Write to tank memory */
+#define TANKMEMADDRREG_READ 0x00100000 /* Read from tank memory */
+
+#define MICROCODEBASE 0x400 /* Microcode data base address */
+
+/* Each DSP microcode instruction is mapped into 2 doublewords */
+/* NOTE: When writing, always write the LO doubleword first. Reads can be in either order. */
+#define LOWORD_OPX_MASK 0x000ffc00 /* Instruction operand X */
+#define LOWORD_OPY_MASK 0x000003ff /* Instruction operand Y */
+#define HIWORD_OPCODE_MASK 0x00f00000 /* Instruction opcode */
+#define HIWORD_RESULT_MASK 0x000ffc00 /* Instruction result */
+#define HIWORD_OPA_MASK 0x000003ff /* Instruction operand A */
+
+
+/* Audigy Soundcard have a different instruction format */
+#define A_MICROCODEBASE 0x600
+#define A_LOWORD_OPY_MASK 0x000007ff
+#define A_LOWORD_OPX_MASK 0x007ff000
+#define A_HIWORD_OPCODE_MASK 0x0f000000
+#define A_HIWORD_RESULT_MASK 0x007ff000
+#define A_HIWORD_OPA_MASK 0x000007ff
+
+/************************************************************************************************/
+/* EMU1010m HANA FPGA registers */
+/************************************************************************************************/
+#define EMU_HANA_DESTHI 0x00 /* 0000xxx 3 bits Link Destination */
+#define EMU_HANA_DESTLO 0x01 /* 00xxxxx 5 bits */
+#define EMU_HANA_SRCHI 0x02 /* 0000xxx 3 bits Link Source */
+#define EMU_HANA_SRCLO 0x03 /* 00xxxxx 5 bits */
+#define EMU_HANA_DOCK_PWR 0x04 /* 000000x 1 bits Audio Dock power */
+#define EMU_HANA_DOCK_PWR_ON 0x01 /* Audio Dock power on */
+#define EMU_HANA_WCLOCK 0x05 /* 0000xxx 3 bits Word Clock source select */
+ /* Must be written after power on to reset DLL */
+ /* One is unable to detect the Audio dock without this */
+#define EMU_HANA_WCLOCK_SRC_MASK 0x07
+#define EMU_HANA_WCLOCK_INT_48K 0x00
+#define EMU_HANA_WCLOCK_INT_44_1K 0x01
+#define EMU_HANA_WCLOCK_HANA_SPDIF_IN 0x02
+#define EMU_HANA_WCLOCK_HANA_ADAT_IN 0x03
+#define EMU_HANA_WCLOCK_SYNC_BNCN 0x04
+#define EMU_HANA_WCLOCK_2ND_HANA 0x05
+#define EMU_HANA_WCLOCK_SRC_RESERVED 0x06
+#define EMU_HANA_WCLOCK_OFF 0x07 /* For testing, forces fallback to DEFCLOCK */
+#define EMU_HANA_WCLOCK_MULT_MASK 0x18
+#define EMU_HANA_WCLOCK_1X 0x00
+#define EMU_HANA_WCLOCK_2X 0x08
+#define EMU_HANA_WCLOCK_4X 0x10
+#define EMU_HANA_WCLOCK_MULT_RESERVED 0x18
+
+#define EMU_HANA_DEFCLOCK 0x06 /* 000000x 1 bits Default Word Clock */
+#define EMU_HANA_DEFCLOCK_48K 0x00
+#define EMU_HANA_DEFCLOCK_44_1K 0x01
+
+#define EMU_HANA_UNMUTE 0x07 /* 000000x 1 bits Mute all audio outputs */
+#define EMU_MUTE 0x00
+#define EMU_UNMUTE 0x01
+
+#define EMU_HANA_FPGA_CONFIG 0x08 /* 00000xx 2 bits Config control of FPGAs */
+#define EMU_HANA_FPGA_CONFIG_AUDIODOCK 0x01 /* Set in order to program FPGA on Audio Dock */
+#define EMU_HANA_FPGA_CONFIG_HANA 0x02 /* Set in order to program FPGA on Hana */
+
+#define EMU_HANA_IRQ_ENABLE 0x09 /* 000xxxx 4 bits IRQ Enable */
+#define EMU_HANA_IRQ_WCLK_CHANGED 0x01
+#define EMU_HANA_IRQ_ADAT 0x02
+#define EMU_HANA_IRQ_DOCK 0x04
+#define EMU_HANA_IRQ_DOCK_LOST 0x08
+
+#define EMU_HANA_SPDIF_MODE 0x0a /* 00xxxxx 5 bits SPDIF MODE */
+#define EMU_HANA_SPDIF_MODE_TX_COMSUMER 0x00
+#define EMU_HANA_SPDIF_MODE_TX_PRO 0x01
+#define EMU_HANA_SPDIF_MODE_TX_NOCOPY 0x02
+#define EMU_HANA_SPDIF_MODE_RX_COMSUMER 0x00
+#define EMU_HANA_SPDIF_MODE_RX_PRO 0x04
+#define EMU_HANA_SPDIF_MODE_RX_NOCOPY 0x08
+#define EMU_HANA_SPDIF_MODE_RX_INVALID 0x10
+
+#define EMU_HANA_OPTICAL_TYPE 0x0b /* 00000xx 2 bits ADAT or SPDIF in/out */
+#define EMU_HANA_OPTICAL_IN_SPDIF 0x00
+#define EMU_HANA_OPTICAL_IN_ADAT 0x01
+#define EMU_HANA_OPTICAL_OUT_SPDIF 0x00
+#define EMU_HANA_OPTICAL_OUT_ADAT 0x02
+
+#define EMU_HANA_MIDI_IN 0x0c /* 000000x 1 bit Control MIDI */
+#define EMU_HANA_MIDI_IN_FROM_HAMOA 0x00 /* HAMOA MIDI in to Alice 2 MIDI B */
+#define EMU_HANA_MIDI_IN_FROM_DOCK 0x01 /* Audio Dock MIDI in to Alice 2 MIDI B */
+
+#define EMU_HANA_DOCK_LEDS_1 0x0d /* 000xxxx 4 bit Audio Dock LEDs */
+#define EMU_HANA_DOCK_LEDS_1_MIDI1 0x01 /* MIDI 1 LED on */
+#define EMU_HANA_DOCK_LEDS_1_MIDI2 0x02 /* MIDI 2 LED on */
+#define EMU_HANA_DOCK_LEDS_1_SMPTE_IN 0x04 /* SMPTE IN LED on */
+#define EMU_HANA_DOCK_LEDS_1_SMPTE_OUT 0x08 /* SMPTE OUT LED on */
+
+#define EMU_HANA_DOCK_LEDS_2 0x0e /* 0xxxxxx 6 bit Audio Dock LEDs */
+#define EMU_HANA_DOCK_LEDS_2_44K 0x01 /* 44.1 kHz LED on */
+#define EMU_HANA_DOCK_LEDS_2_48K 0x02 /* 48 kHz LED on */
+#define EMU_HANA_DOCK_LEDS_2_96K 0x04 /* 96 kHz LED on */
+#define EMU_HANA_DOCK_LEDS_2_192K 0x08 /* 192 kHz LED on */
+#define EMU_HANA_DOCK_LEDS_2_LOCK 0x10 /* LOCK LED on */
+#define EMU_HANA_DOCK_LEDS_2_EXT 0x20 /* EXT LED on */
+
+#define EMU_HANA_DOCK_LEDS_3 0x0f /* 0xxxxxx 6 bit Audio Dock LEDs */
+#define EMU_HANA_DOCK_LEDS_3_CLIP_A 0x01 /* Mic A Clip LED on */
+#define EMU_HANA_DOCK_LEDS_3_CLIP_B 0x02 /* Mic B Clip LED on */
+#define EMU_HANA_DOCK_LEDS_3_SIGNAL_A 0x04 /* Signal A Clip LED on */
+#define EMU_HANA_DOCK_LEDS_3_SIGNAL_B 0x08 /* Signal B Clip LED on */
+#define EMU_HANA_DOCK_LEDS_3_MANUAL_CLIP 0x10 /* Manual Clip detection */
+#define EMU_HANA_DOCK_LEDS_3_MANUAL_SIGNAL 0x20 /* Manual Signal detection */
+
+#define EMU_HANA_ADC_PADS 0x10 /* 0000xxx 3 bit Audio Dock ADC 14dB pads */
+#define EMU_HANA_DOCK_ADC_PAD1 0x01 /* 14dB Attenuation on Audio Dock ADC 1 */
+#define EMU_HANA_DOCK_ADC_PAD2 0x02 /* 14dB Attenuation on Audio Dock ADC 2 */
+#define EMU_HANA_DOCK_ADC_PAD3 0x04 /* 14dB Attenuation on Audio Dock ADC 3 */
+#define EMU_HANA_0202_ADC_PAD1 0x08 /* 14dB Attenuation on 0202 ADC 1 */
+
+#define EMU_HANA_DOCK_MISC 0x11 /* 0xxxxxx 6 bit Audio Dock misc bits */
+#define EMU_HANA_DOCK_DAC1_MUTE 0x01 /* DAC 1 Mute */
+#define EMU_HANA_DOCK_DAC2_MUTE 0x02 /* DAC 2 Mute */
+#define EMU_HANA_DOCK_DAC3_MUTE 0x04 /* DAC 3 Mute */
+#define EMU_HANA_DOCK_DAC4_MUTE 0x08 /* DAC 4 Mute */
+#define EMU_HANA_DOCK_PHONES_192_DAC1 0x00 /* DAC 1 Headphones source at 192kHz */
+#define EMU_HANA_DOCK_PHONES_192_DAC2 0x10 /* DAC 2 Headphones source at 192kHz */
+#define EMU_HANA_DOCK_PHONES_192_DAC3 0x20 /* DAC 3 Headphones source at 192kHz */
+#define EMU_HANA_DOCK_PHONES_192_DAC4 0x30 /* DAC 4 Headphones source at 192kHz */
+
+#define EMU_HANA_MIDI_OUT 0x12 /* 00xxxxx 5 bit Source for each MIDI out port */
+#define EMU_HANA_MIDI_OUT_0202 0x01 /* 0202 MIDI from Alice 2. 0 = A, 1 = B */
+#define EMU_HANA_MIDI_OUT_DOCK1 0x02 /* Audio Dock MIDI1 front, from Alice 2. 0 = A, 1 = B */
+#define EMU_HANA_MIDI_OUT_DOCK2 0x04 /* Audio Dock MIDI2 rear, from Alice 2. 0 = A, 1 = B */
+#define EMU_HANA_MIDI_OUT_SYNC2 0x08 /* Sync card. Not the actual MIDI out jack. 0 = A, 1 = B */
+#define EMU_HANA_MIDI_OUT_LOOP 0x10 /* 0 = bits (3:0) normal. 1 = MIDI loopback enabled. */
+
+#define EMU_HANA_DAC_PADS 0x13 /* 00xxxxx 5 bit DAC 14dB attenuation pads */
+#define EMU_HANA_DOCK_DAC_PAD1 0x01 /* 14dB Attenuation on AudioDock DAC 1. Left and Right */
+#define EMU_HANA_DOCK_DAC_PAD2 0x02 /* 14dB Attenuation on AudioDock DAC 2. Left and Right */
+#define EMU_HANA_DOCK_DAC_PAD3 0x04 /* 14dB Attenuation on AudioDock DAC 3. Left and Right */
+#define EMU_HANA_DOCK_DAC_PAD4 0x08 /* 14dB Attenuation on AudioDock DAC 4. Left and Right */
+#define EMU_HANA_0202_DAC_PAD1 0x10 /* 14dB Attenuation on 0202 DAC 1. Left and Right */
+
+/* 0x14 - 0x1f Unused R/W registers */
+#define EMU_HANA_IRQ_STATUS 0x20 /* 000xxxx 4 bits IRQ Status */
+#if 0 /* Already defined for reg 0x09 IRQ_ENABLE */
+#define EMU_HANA_IRQ_WCLK_CHANGED 0x01
+#define EMU_HANA_IRQ_ADAT 0x02
+#define EMU_HANA_IRQ_DOCK 0x04
+#define EMU_HANA_IRQ_DOCK_LOST 0x08
+#endif
+
+#define EMU_HANA_OPTION_CARDS 0x21 /* 000xxxx 4 bits Presence of option cards */
+#define EMU_HANA_OPTION_HAMOA 0x01 /* HAMOA card present */
+#define EMU_HANA_OPTION_SYNC 0x02 /* Sync card present */
+#define EMU_HANA_OPTION_DOCK_ONLINE 0x04 /* Audio Dock online and FPGA configured */
+#define EMU_HANA_OPTION_DOCK_OFFLINE 0x08 /* Audio Dock online and FPGA not configured */
+
+#define EMU_HANA_ID 0x22 /* 1010101 7 bits ID byte & 0x7f = 0x55 */
+
+#define EMU_HANA_MAJOR_REV 0x23 /* 0000xxx 3 bit Hana FPGA Major rev */
+#define EMU_HANA_MINOR_REV 0x24 /* 0000xxx 3 bit Hana FPGA Minor rev */
+
+#define EMU_DOCK_MAJOR_REV 0x25 /* 0000xxx 3 bit Audio Dock FPGA Major rev */
+#define EMU_DOCK_MINOR_REV 0x26 /* 0000xxx 3 bit Audio Dock FPGA Minor rev */
+
+#define EMU_DOCK_BOARD_ID 0x27 /* 00000xx 2 bits Audio Dock ID pins */
+#define EMU_DOCK_BOARD_ID0 0x00 /* ID bit 0 */
+#define EMU_DOCK_BOARD_ID1 0x03 /* ID bit 1 */
+
+#define EMU_HANA_WC_SPDIF_HI 0x28 /* 0xxxxxx 6 bit SPDIF IN Word clock, upper 6 bits */
+#define EMU_HANA_WC_SPDIF_LO 0x29 /* 0xxxxxx 6 bit SPDIF IN Word clock, lower 6 bits */
+
+#define EMU_HANA_WC_ADAT_HI 0x2a /* 0xxxxxx 6 bit ADAT IN Word clock, upper 6 bits */
+#define EMU_HANA_WC_ADAT_LO 0x2b /* 0xxxxxx 6 bit ADAT IN Word clock, lower 6 bits */
+
+#define EMU_HANA_WC_BNC_LO 0x2c /* 0xxxxxx 6 bit BNC IN Word clock, lower 6 bits */
+#define EMU_HANA_WC_BNC_HI 0x2d /* 0xxxxxx 6 bit BNC IN Word clock, upper 6 bits */
+
+#define EMU_HANA2_WC_SPDIF_HI 0x2e /* 0xxxxxx 6 bit HANA2 SPDIF IN Word clock, upper 6 bits */
+#define EMU_HANA2_WC_SPDIF_LO 0x2f /* 0xxxxxx 6 bit HANA2 SPDIF IN Word clock, lower 6 bits */
+/* 0x30 - 0x3f Unused Read only registers */
+
+/************************************************************************************************/
+/* EMU1010m HANA Destinations */
+/************************************************************************************************/
+/* Hana, original 1010,1212,1820 using Alice2
+ * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2
+ * 0x01, 0x10-0x1f: 32 Elink channels to Audio Dock
+ * 0x01, 0x00: Dock DAC 1 Left
+ * 0x01, 0x04: Dock DAC 1 Right
+ * 0x01, 0x08: Dock DAC 2 Left
+ * 0x01, 0x0c: Dock DAC 2 Right
+ * 0x01, 0x10: Dock DAC 3 Left
+ * 0x01, 0x12: PHONES Left
+ * 0x01, 0x14: Dock DAC 3 Right
+ * 0x01, 0x16: PHONES Right
+ * 0x01, 0x18: Dock DAC 4 Left
+ * 0x01, 0x1a: S/PDIF Left
+ * 0x01, 0x1c: Dock DAC 4 Right
+ * 0x01, 0x1e: S/PDIF Right
+ * 0x02, 0x00: Hana S/PDIF Left
+ * 0x02, 0x01: Hana S/PDIF Right
+ * 0x03, 0x00: Hanoa DAC Left
+ * 0x03, 0x01: Hanoa DAC Right
+ * 0x04, 0x00-0x07: Hana ADAT
+ * 0x05, 0x00: I2S0 Left to Alice2
+ * 0x05, 0x01: I2S0 Right to Alice2
+ * 0x06, 0x00: I2S0 Left to Alice2
+ * 0x06, 0x01: I2S0 Right to Alice2
+ * 0x07, 0x00: I2S0 Left to Alice2
+ * 0x07, 0x01: I2S0 Right to Alice2
+ *
+ * Hana2 never released, but used Tina
+ * Not needed.
+ *
+ * Hana3, rev2 1010,1212,1616 using Tina
+ * Destinations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00, 0x00-0x0f: 16 EMU32A channels to Tina
+ * 0x01, 0x10-0x1f: 32 EDI channels to Micro Dock
+ * 0x01, 0x00: Dock DAC 1 Left
+ * 0x01, 0x04: Dock DAC 1 Right
+ * 0x01, 0x08: Dock DAC 2 Left
+ * 0x01, 0x0c: Dock DAC 2 Right
+ * 0x01, 0x10: Dock DAC 3 Left
+ * 0x01, 0x12: Dock S/PDIF Left
+ * 0x01, 0x14: Dock DAC 3 Right
+ * 0x01, 0x16: Dock S/PDIF Right
+ * 0x01, 0x18-0x1f: Dock ADAT 0-7
+ * 0x02, 0x00: Hana3 S/PDIF Left
+ * 0x02, 0x01: Hana3 S/PDIF Right
+ * 0x03, 0x00: Hanoa DAC Left
+ * 0x03, 0x01: Hanoa DAC Right
+ * 0x04, 0x00-0x07: Hana3 ADAT 0-7
+ * 0x05, 0x00-0x0f: 16 EMU32B channels to Tina
+ * 0x06-0x07: Not used
+ *
+ * HanaLite, rev1 0404 using Alice2
+ * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2
+ * 0x01: Not used
+ * 0x02, 0x00: S/PDIF Left
+ * 0x02, 0x01: S/PDIF Right
+ * 0x03, 0x00: DAC Left
+ * 0x03, 0x01: DAC Right
+ * 0x04-0x07: Not used
+ *
+ * HanaLiteLite, rev2 0404 using Alice2
+ * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2
+ * 0x01: Not used
+ * 0x02, 0x00: S/PDIF Left
+ * 0x02, 0x01: S/PDIF Right
+ * 0x03, 0x00: DAC Left
+ * 0x03, 0x01: DAC Right
+ * 0x04-0x07: Not used
+ *
+ * Mana, Cardbus 1616 using Tina2
+ * Destinations for SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00, 0x00-0x0f: 16 EMU32A channels to Tina2
+ * 0x01, 0x10-0x1f: 32 EDI channels to Micro Dock
+ * 0x01, 0x00: Dock DAC 1 Left
+ * 0x01, 0x04: Dock DAC 1 Right
+ * 0x01, 0x08: Dock DAC 2 Left
+ * 0x01, 0x0c: Dock DAC 2 Right
+ * 0x01, 0x10: Dock DAC 3 Left
+ * 0x01, 0x12: Dock S/PDIF Left
+ * 0x01, 0x14: Dock DAC 3 Right
+ * 0x01, 0x16: Dock S/PDIF Right
+ * 0x01, 0x18-0x1f: Dock ADAT 0-7
+ * 0x02: Not used
+ * 0x03, 0x00: Mana DAC Left
+ * 0x03, 0x01: Mana DAC Right
+ * 0x04, 0x00-0x0f: 16 EMU32B channels to Tina2
+ * 0x05-0x07: Not used
+ *
+ *
+ */
+/* 32-bit destinations of signal in the Hana FPGA. Destinations are either
+ * physical outputs of Hana, or outputs going to Alice2 (audigy) for capture
+ * - 16 x EMU_DST_ALICE2_EMU32_X.
+ */
+/* EMU32 = 32-bit serial channel between Alice2 (audigy) and Hana (FPGA) */
+/* EMU_DST_ALICE2_EMU32_X - data channels from Hana to Alice2 used for capture.
+ * Which data is fed into a EMU_DST_ALICE2_EMU32_X channel in Hana depends on
+ * setup of mixer control for each destination - see emumixer.c -
+ * snd_emu1010_output_enum_ctls[], snd_emu1010_input_enum_ctls[]
+ */
+#define EMU_DST_ALICE2_EMU32_0 0x000f /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_1 0x0000 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_2 0x0001 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_3 0x0002 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_4 0x0003 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_5 0x0004 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_6 0x0005 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_7 0x0006 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_8 0x0007 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_9 0x0008 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_A 0x0009 /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_B 0x000a /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_C 0x000b /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_D 0x000c /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_E 0x000d /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_ALICE2_EMU32_F 0x000e /* 16 EMU32 channels to Alice2 +0 to +0xf */
+#define EMU_DST_DOCK_DAC1_LEFT1 0x0100 /* Audio Dock DAC1 Left, 1st or 48kHz only */
+#define EMU_DST_DOCK_DAC1_LEFT2 0x0101 /* Audio Dock DAC1 Left, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC1_LEFT3 0x0102 /* Audio Dock DAC1 Left, 3rd or 192kHz */
+#define EMU_DST_DOCK_DAC1_LEFT4 0x0103 /* Audio Dock DAC1 Left, 4th or 192kHz */
+#define EMU_DST_DOCK_DAC1_RIGHT1 0x0104 /* Audio Dock DAC1 Right, 1st or 48kHz only */
+#define EMU_DST_DOCK_DAC1_RIGHT2 0x0105 /* Audio Dock DAC1 Right, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC1_RIGHT3 0x0106 /* Audio Dock DAC1 Right, 3rd or 192kHz */
+#define EMU_DST_DOCK_DAC1_RIGHT4 0x0107 /* Audio Dock DAC1 Right, 4th or 192kHz */
+#define EMU_DST_DOCK_DAC2_LEFT1 0x0108 /* Audio Dock DAC2 Left, 1st or 48kHz only */
+#define EMU_DST_DOCK_DAC2_LEFT2 0x0109 /* Audio Dock DAC2 Left, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC2_LEFT3 0x010a /* Audio Dock DAC2 Left, 3rd or 192kHz */
+#define EMU_DST_DOCK_DAC2_LEFT4 0x010b /* Audio Dock DAC2 Left, 4th or 192kHz */
+#define EMU_DST_DOCK_DAC2_RIGHT1 0x010c /* Audio Dock DAC2 Right, 1st or 48kHz only */
+#define EMU_DST_DOCK_DAC2_RIGHT2 0x010d /* Audio Dock DAC2 Right, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC2_RIGHT3 0x010e /* Audio Dock DAC2 Right, 3rd or 192kHz */
+#define EMU_DST_DOCK_DAC2_RIGHT4 0x010f /* Audio Dock DAC2 Right, 4th or 192kHz */
+#define EMU_DST_DOCK_DAC3_LEFT1 0x0110 /* Audio Dock DAC1 Left, 1st or 48kHz only */
+#define EMU_DST_DOCK_DAC3_LEFT2 0x0111 /* Audio Dock DAC1 Left, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC3_LEFT3 0x0112 /* Audio Dock DAC1 Left, 3rd or 192kHz */
+#define EMU_DST_DOCK_DAC3_LEFT4 0x0113 /* Audio Dock DAC1 Left, 4th or 192kHz */
+#define EMU_DST_DOCK_PHONES_LEFT1 0x0112 /* Audio Dock PHONES Left, 1st or 48kHz only */
+#define EMU_DST_DOCK_PHONES_LEFT2 0x0113 /* Audio Dock PHONES Left, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC3_RIGHT1 0x0114 /* Audio Dock DAC1 Right, 1st or 48kHz only */
+#define EMU_DST_DOCK_DAC3_RIGHT2 0x0115 /* Audio Dock DAC1 Right, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC3_RIGHT3 0x0116 /* Audio Dock DAC1 Right, 3rd or 192kHz */
+#define EMU_DST_DOCK_DAC3_RIGHT4 0x0117 /* Audio Dock DAC1 Right, 4th or 192kHz */
+#define EMU_DST_DOCK_PHONES_RIGHT1 0x0116 /* Audio Dock PHONES Right, 1st or 48kHz only */
+#define EMU_DST_DOCK_PHONES_RIGHT2 0x0117 /* Audio Dock PHONES Right, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC4_LEFT1 0x0118 /* Audio Dock DAC2 Left, 1st or 48kHz only */
+#define EMU_DST_DOCK_DAC4_LEFT2 0x0119 /* Audio Dock DAC2 Left, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC4_LEFT3 0x011a /* Audio Dock DAC2 Left, 3rd or 192kHz */
+#define EMU_DST_DOCK_DAC4_LEFT4 0x011b /* Audio Dock DAC2 Left, 4th or 192kHz */
+#define EMU_DST_DOCK_SPDIF_LEFT1 0x011a /* Audio Dock SPDIF Left, 1st or 48kHz only */
+#define EMU_DST_DOCK_SPDIF_LEFT2 0x011b /* Audio Dock SPDIF Left, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC4_RIGHT1 0x011c /* Audio Dock DAC2 Right, 1st or 48kHz only */
+#define EMU_DST_DOCK_DAC4_RIGHT2 0x011d /* Audio Dock DAC2 Right, 2nd or 96kHz */
+#define EMU_DST_DOCK_DAC4_RIGHT3 0x011e /* Audio Dock DAC2 Right, 3rd or 192kHz */
+#define EMU_DST_DOCK_DAC4_RIGHT4 0x011f /* Audio Dock DAC2 Right, 4th or 192kHz */
+#define EMU_DST_DOCK_SPDIF_RIGHT1 0x011e /* Audio Dock SPDIF Right, 1st or 48kHz only */
+#define EMU_DST_DOCK_SPDIF_RIGHT2 0x011f /* Audio Dock SPDIF Right, 2nd or 96kHz */
+#define EMU_DST_HANA_SPDIF_LEFT1 0x0200 /* Hana SPDIF Left, 1st or 48kHz only */
+#define EMU_DST_HANA_SPDIF_LEFT2 0x0202 /* Hana SPDIF Left, 2nd or 96kHz */
+#define EMU_DST_HANA_SPDIF_RIGHT1 0x0201 /* Hana SPDIF Right, 1st or 48kHz only */
+#define EMU_DST_HANA_SPDIF_RIGHT2 0x0203 /* Hana SPDIF Right, 2nd or 96kHz */
+#define EMU_DST_HAMOA_DAC_LEFT1 0x0300 /* Hamoa DAC Left, 1st or 48kHz only */
+#define EMU_DST_HAMOA_DAC_LEFT2 0x0302 /* Hamoa DAC Left, 2nd or 96kHz */
+#define EMU_DST_HAMOA_DAC_LEFT3 0x0304 /* Hamoa DAC Left, 3rd or 192kHz */
+#define EMU_DST_HAMOA_DAC_LEFT4 0x0306 /* Hamoa DAC Left, 4th or 192kHz */
+#define EMU_DST_HAMOA_DAC_RIGHT1 0x0301 /* Hamoa DAC Right, 1st or 48kHz only */
+#define EMU_DST_HAMOA_DAC_RIGHT2 0x0303 /* Hamoa DAC Right, 2nd or 96kHz */
+#define EMU_DST_HAMOA_DAC_RIGHT3 0x0305 /* Hamoa DAC Right, 3rd or 192kHz */
+#define EMU_DST_HAMOA_DAC_RIGHT4 0x0307 /* Hamoa DAC Right, 4th or 192kHz */
+#define EMU_DST_HANA_ADAT 0x0400 /* Hana ADAT 8 channel out +0 to +7 */
+#define EMU_DST_ALICE_I2S0_LEFT 0x0500 /* Alice2 I2S0 Left */
+#define EMU_DST_ALICE_I2S0_RIGHT 0x0501 /* Alice2 I2S0 Right */
+#define EMU_DST_ALICE_I2S1_LEFT 0x0600 /* Alice2 I2S1 Left */
+#define EMU_DST_ALICE_I2S1_RIGHT 0x0601 /* Alice2 I2S1 Right */
+#define EMU_DST_ALICE_I2S2_LEFT 0x0700 /* Alice2 I2S2 Left */
+#define EMU_DST_ALICE_I2S2_RIGHT 0x0701 /* Alice2 I2S2 Right */
+
+/* Additional destinations for 1616(M)/Microdock */
+/* Microdock S/PDIF OUT Left, 1st or 48kHz only */
+#define EMU_DST_MDOCK_SPDIF_LEFT1 0x0112
+/* Microdock S/PDIF OUT Left, 2nd or 96kHz */
+#define EMU_DST_MDOCK_SPDIF_LEFT2 0x0113
+/* Microdock S/PDIF OUT Right, 1st or 48kHz only */
+#define EMU_DST_MDOCK_SPDIF_RIGHT1 0x0116
+/* Microdock S/PDIF OUT Right, 2nd or 96kHz */
+#define EMU_DST_MDOCK_SPDIF_RIGHT2 0x0117
+/* Microdock S/PDIF ADAT 8 channel out +8 to +f */
+#define EMU_DST_MDOCK_ADAT 0x0118
+
+/* Headphone jack on 1010 cardbus? 44.1/48kHz only? */
+#define EMU_DST_MANA_DAC_LEFT 0x0300
+/* Headphone jack on 1010 cardbus? 44.1/48kHz only? */
+#define EMU_DST_MANA_DAC_RIGHT 0x0301
+
+/************************************************************************************************/
+/* EMU1010m HANA Sources */
+/************************************************************************************************/
+/* Hana, original 1010,1212,1820 using Alice2
+ * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00,0x00-0x1f: Silence
+ * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock
+ * 0x01, 0x00: Dock Mic A
+ * 0x01, 0x04: Dock Mic B
+ * 0x01, 0x08: Dock ADC 1 Left
+ * 0x01, 0x0c: Dock ADC 1 Right
+ * 0x01, 0x10: Dock ADC 2 Left
+ * 0x01, 0x14: Dock ADC 2 Right
+ * 0x01, 0x18: Dock ADC 3 Left
+ * 0x01, 0x1c: Dock ADC 3 Right
+ * 0x02, 0x00: Hana ADC Left
+ * 0x02, 0x01: Hana ADC Right
+ * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output
+ * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output
+ * 0x04, 0x00-0x07: Hana ADAT
+ * 0x05, 0x00: Hana S/PDIF Left
+ * 0x05, 0x01: Hana S/PDIF Right
+ * 0x06-0x07: Not used
+ *
+ * Hana2 never released, but used Tina
+ * Not needed.
+ *
+ * Hana3, rev2 1010,1212,1616 using Tina
+ * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00,0x00-0x1f: Silence
+ * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock
+ * 0x01, 0x00: Dock Mic A
+ * 0x01, 0x04: Dock Mic B
+ * 0x01, 0x08: Dock ADC 1 Left
+ * 0x01, 0x0c: Dock ADC 1 Right
+ * 0x01, 0x10: Dock ADC 2 Left
+ * 0x01, 0x12: Dock S/PDIF Left
+ * 0x01, 0x14: Dock ADC 2 Right
+ * 0x01, 0x16: Dock S/PDIF Right
+ * 0x01, 0x18-0x1f: Dock ADAT 0-7
+ * 0x01, 0x18: Dock ADC 3 Left
+ * 0x01, 0x1c: Dock ADC 3 Right
+ * 0x02, 0x00: Hanoa ADC Left
+ * 0x02, 0x01: Hanoa ADC Right
+ * 0x03, 0x00-0x0f: 16 inputs from Tina Emu32A output
+ * 0x03, 0x10-0x1f: 16 inputs from Tina Emu32B output
+ * 0x04, 0x00-0x07: Hana3 ADAT
+ * 0x05, 0x00: Hana3 S/PDIF Left
+ * 0x05, 0x01: Hana3 S/PDIF Right
+ * 0x06-0x07: Not used
+ *
+ * HanaLite, rev1 0404 using Alice2
+ * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00,0x00-0x1f: Silence
+ * 0x01: Not used
+ * 0x02, 0x00: ADC Left
+ * 0x02, 0x01: ADC Right
+ * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output
+ * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output
+ * 0x04: Not used
+ * 0x05, 0x00: S/PDIF Left
+ * 0x05, 0x01: S/PDIF Right
+ * 0x06-0x07: Not used
+ *
+ * HanaLiteLite, rev2 0404 using Alice2
+ * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00,0x00-0x1f: Silence
+ * 0x01: Not used
+ * 0x02, 0x00: ADC Left
+ * 0x02, 0x01: ADC Right
+ * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output
+ * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output
+ * 0x04: Not used
+ * 0x05, 0x00: S/PDIF Left
+ * 0x05, 0x01: S/PDIF Right
+ * 0x06-0x07: Not used
+ *
+ * Mana, Cardbus 1616 using Tina2
+ * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz
+ * 0x00,0x00-0x1f: Silence
+ * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock
+ * 0x01, 0x00: Dock Mic A
+ * 0x01, 0x04: Dock Mic B
+ * 0x01, 0x08: Dock ADC 1 Left
+ * 0x01, 0x0c: Dock ADC 1 Right
+ * 0x01, 0x10: Dock ADC 2 Left
+ * 0x01, 0x12: Dock S/PDIF Left
+ * 0x01, 0x14: Dock ADC 2 Right
+ * 0x01, 0x16: Dock S/PDIF Right
+ * 0x01, 0x18-0x1f: Dock ADAT 0-7
+ * 0x01, 0x18: Dock ADC 3 Left
+ * 0x01, 0x1c: Dock ADC 3 Right
+ * 0x02: Not used
+ * 0x03, 0x00-0x0f: 16 inputs from Tina Emu32A output
+ * 0x03, 0x10-0x1f: 16 inputs from Tina Emu32B output
+ * 0x04-0x07: Not used
+ *
+ */
+
+/* 32-bit sources of signal in the Hana FPGA. The sources are routed to
+ * destinations using mixer control for each destination - see emumixer.c
+ * Sources are either physical inputs of FPGA,
+ * or outputs from Alice (audigy) - 16 x EMU_SRC_ALICE_EMU32A +
+ * 16 x EMU_SRC_ALICE_EMU32B
+ */
+#define EMU_SRC_SILENCE 0x0000 /* Silence */
+#define EMU_SRC_DOCK_MIC_A1 0x0100 /* Audio Dock Mic A, 1st or 48kHz only */
+#define EMU_SRC_DOCK_MIC_A2 0x0101 /* Audio Dock Mic A, 2nd or 96kHz */
+#define EMU_SRC_DOCK_MIC_A3 0x0102 /* Audio Dock Mic A, 3rd or 192kHz */
+#define EMU_SRC_DOCK_MIC_A4 0x0103 /* Audio Dock Mic A, 4th or 192kHz */
+#define EMU_SRC_DOCK_MIC_B1 0x0104 /* Audio Dock Mic B, 1st or 48kHz only */
+#define EMU_SRC_DOCK_MIC_B2 0x0105 /* Audio Dock Mic B, 2nd or 96kHz */
+#define EMU_SRC_DOCK_MIC_B3 0x0106 /* Audio Dock Mic B, 3rd or 192kHz */
+#define EMU_SRC_DOCK_MIC_B4 0x0107 /* Audio Dock Mic B, 4th or 192kHz */
+#define EMU_SRC_DOCK_ADC1_LEFT1 0x0108 /* Audio Dock ADC1 Left, 1st or 48kHz only */
+#define EMU_SRC_DOCK_ADC1_LEFT2 0x0109 /* Audio Dock ADC1 Left, 2nd or 96kHz */
+#define EMU_SRC_DOCK_ADC1_LEFT3 0x010a /* Audio Dock ADC1 Left, 3rd or 192kHz */
+#define EMU_SRC_DOCK_ADC1_LEFT4 0x010b /* Audio Dock ADC1 Left, 4th or 192kHz */
+#define EMU_SRC_DOCK_ADC1_RIGHT1 0x010c /* Audio Dock ADC1 Right, 1st or 48kHz only */
+#define EMU_SRC_DOCK_ADC1_RIGHT2 0x010d /* Audio Dock ADC1 Right, 2nd or 96kHz */
+#define EMU_SRC_DOCK_ADC1_RIGHT3 0x010e /* Audio Dock ADC1 Right, 3rd or 192kHz */
+#define EMU_SRC_DOCK_ADC1_RIGHT4 0x010f /* Audio Dock ADC1 Right, 4th or 192kHz */
+#define EMU_SRC_DOCK_ADC2_LEFT1 0x0110 /* Audio Dock ADC2 Left, 1st or 48kHz only */
+#define EMU_SRC_DOCK_ADC2_LEFT2 0x0111 /* Audio Dock ADC2 Left, 2nd or 96kHz */
+#define EMU_SRC_DOCK_ADC2_LEFT3 0x0112 /* Audio Dock ADC2 Left, 3rd or 192kHz */
+#define EMU_SRC_DOCK_ADC2_LEFT4 0x0113 /* Audio Dock ADC2 Left, 4th or 192kHz */
+#define EMU_SRC_DOCK_ADC2_RIGHT1 0x0114 /* Audio Dock ADC2 Right, 1st or 48kHz only */
+#define EMU_SRC_DOCK_ADC2_RIGHT2 0x0115 /* Audio Dock ADC2 Right, 2nd or 96kHz */
+#define EMU_SRC_DOCK_ADC2_RIGHT3 0x0116 /* Audio Dock ADC2 Right, 3rd or 192kHz */
+#define EMU_SRC_DOCK_ADC2_RIGHT4 0x0117 /* Audio Dock ADC2 Right, 4th or 192kHz */
+#define EMU_SRC_DOCK_ADC3_LEFT1 0x0118 /* Audio Dock ADC3 Left, 1st or 48kHz only */
+#define EMU_SRC_DOCK_ADC3_LEFT2 0x0119 /* Audio Dock ADC3 Left, 2nd or 96kHz */
+#define EMU_SRC_DOCK_ADC3_LEFT3 0x011a /* Audio Dock ADC3 Left, 3rd or 192kHz */
+#define EMU_SRC_DOCK_ADC3_LEFT4 0x011b /* Audio Dock ADC3 Left, 4th or 192kHz */
+#define EMU_SRC_DOCK_ADC3_RIGHT1 0x011c /* Audio Dock ADC3 Right, 1st or 48kHz only */
+#define EMU_SRC_DOCK_ADC3_RIGHT2 0x011d /* Audio Dock ADC3 Right, 2nd or 96kHz */
+#define EMU_SRC_DOCK_ADC3_RIGHT3 0x011e /* Audio Dock ADC3 Right, 3rd or 192kHz */
+#define EMU_SRC_DOCK_ADC3_RIGHT4 0x011f /* Audio Dock ADC3 Right, 4th or 192kHz */
+#define EMU_SRC_HAMOA_ADC_LEFT1 0x0200 /* Hamoa ADC Left, 1st or 48kHz only */
+#define EMU_SRC_HAMOA_ADC_LEFT2 0x0202 /* Hamoa ADC Left, 2nd or 96kHz */
+#define EMU_SRC_HAMOA_ADC_LEFT3 0x0204 /* Hamoa ADC Left, 3rd or 192kHz */
+#define EMU_SRC_HAMOA_ADC_LEFT4 0x0206 /* Hamoa ADC Left, 4th or 192kHz */
+#define EMU_SRC_HAMOA_ADC_RIGHT1 0x0201 /* Hamoa ADC Right, 1st or 48kHz only */
+#define EMU_SRC_HAMOA_ADC_RIGHT2 0x0203 /* Hamoa ADC Right, 2nd or 96kHz */
+#define EMU_SRC_HAMOA_ADC_RIGHT3 0x0205 /* Hamoa ADC Right, 3rd or 192kHz */
+#define EMU_SRC_HAMOA_ADC_RIGHT4 0x0207 /* Hamoa ADC Right, 4th or 192kHz */
+#define EMU_SRC_ALICE_EMU32A 0x0300 /* Alice2 EMU32a 16 outputs. +0 to +0xf */
+#define EMU_SRC_ALICE_EMU32B 0x0310 /* Alice2 EMU32b 16 outputs. +0 to +0xf */
+#define EMU_SRC_HANA_ADAT 0x0400 /* Hana ADAT 8 channel in +0 to +7 */
+#define EMU_SRC_HANA_SPDIF_LEFT1 0x0500 /* Hana SPDIF Left, 1st or 48kHz only */
+#define EMU_SRC_HANA_SPDIF_LEFT2 0x0502 /* Hana SPDIF Left, 2nd or 96kHz */
+#define EMU_SRC_HANA_SPDIF_RIGHT1 0x0501 /* Hana SPDIF Right, 1st or 48kHz only */
+#define EMU_SRC_HANA_SPDIF_RIGHT2 0x0503 /* Hana SPDIF Right, 2nd or 96kHz */
+
+/* Additional inputs for 1616(M)/Microdock */
+/* Microdock S/PDIF Left, 1st or 48kHz only */
+#define EMU_SRC_MDOCK_SPDIF_LEFT1 0x0112
+/* Microdock S/PDIF Left, 2nd or 96kHz */
+#define EMU_SRC_MDOCK_SPDIF_LEFT2 0x0113
+/* Microdock S/PDIF Right, 1st or 48kHz only */
+#define EMU_SRC_MDOCK_SPDIF_RIGHT1 0x0116
+/* Microdock S/PDIF Right, 2nd or 96kHz */
+#define EMU_SRC_MDOCK_SPDIF_RIGHT2 0x0117
+/* Microdock ADAT 8 channel in +8 to +f */
+#define EMU_SRC_MDOCK_ADAT 0x0118
+
+/* 0x600 and 0x700 no used */
+
+/* ------------------- STRUCTURES -------------------- */
+
+enum {
+ EMU10K1_EFX,
+ EMU10K1_PCM,
+ EMU10K1_SYNTH,
+ EMU10K1_MIDI
+};
+
+struct snd_emu10k1;
+
+struct snd_emu10k1_voice {
+ struct snd_emu10k1 *emu;
+ int number;
+ unsigned int use: 1,
+ pcm: 1,
+ efx: 1,
+ synth: 1,
+ midi: 1;
+ void (*interrupt)(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *pvoice);
+
+ struct snd_emu10k1_pcm *epcm;
+};
+
+enum {
+ PLAYBACK_EMUVOICE,
+ PLAYBACK_EFX,
+ CAPTURE_AC97ADC,
+ CAPTURE_AC97MIC,
+ CAPTURE_EFX
+};
+
+struct snd_emu10k1_pcm {
+ struct snd_emu10k1 *emu;
+ int type;
+ struct snd_pcm_substream *substream;
+ struct snd_emu10k1_voice *voices[NUM_EFX_PLAYBACK];
+ struct snd_emu10k1_voice *extra;
+ unsigned short running;
+ unsigned short first_ptr;
+ struct snd_util_memblk *memblk;
+ unsigned int start_addr;
+ unsigned int ccca_start_addr;
+ unsigned int capture_ipr; /* interrupt acknowledge mask */
+ unsigned int capture_inte; /* interrupt enable mask */
+ unsigned int capture_ba_reg; /* buffer address register */
+ unsigned int capture_bs_reg; /* buffer size register */
+ unsigned int capture_idx_reg; /* buffer index register */
+ unsigned int capture_cr_val; /* control value */
+ unsigned int capture_cr_val2; /* control value2 (for audigy) */
+ unsigned int capture_bs_val; /* buffer size value */
+ unsigned int capture_bufsize; /* buffer size in bytes */
+};
+
+struct snd_emu10k1_pcm_mixer {
+ /* mono, left, right x 8 sends (4 on emu10k1) */
+ unsigned char send_routing[3][8];
+ unsigned char send_volume[3][8];
+ unsigned short attn[3];
+ struct snd_emu10k1_pcm *epcm;
+};
+
+#define snd_emu10k1_compose_send_routing(route) \
+((route[0] | (route[1] << 4) | (route[2] << 8) | (route[3] << 12)) << 16)
+
+#define snd_emu10k1_compose_audigy_fxrt1(route) \
+((unsigned int)route[0] | ((unsigned int)route[1] << 8) | ((unsigned int)route[2] << 16) | ((unsigned int)route[3] << 24))
+
+#define snd_emu10k1_compose_audigy_fxrt2(route) \
+((unsigned int)route[4] | ((unsigned int)route[5] << 8) | ((unsigned int)route[6] << 16) | ((unsigned int)route[7] << 24))
+
+struct snd_emu10k1_memblk {
+ struct snd_util_memblk mem;
+ /* private part */
+ int first_page, last_page, pages, mapped_page;
+ unsigned int map_locked;
+ struct list_head mapped_link;
+ struct list_head mapped_order_link;
+};
+
+#define snd_emu10k1_memblk_offset(blk) (((blk)->mapped_page << PAGE_SHIFT) | ((blk)->mem.offset & (PAGE_SIZE - 1)))
+
+#define EMU10K1_MAX_TRAM_BLOCKS_PER_CODE 16
+
+struct snd_emu10k1_fx8010_ctl {
+ struct list_head list; /* list link container */
+ unsigned int vcount;
+ unsigned int count; /* count of GPR (1..16) */
+ unsigned short gpr[32]; /* GPR number(s) */
+ unsigned int value[32];
+ unsigned int min; /* minimum range */
+ unsigned int max; /* maximum range */
+ unsigned int translation; /* translation type (EMU10K1_GPR_TRANSLATION*) */
+ struct snd_kcontrol *kcontrol;
+};
+
+typedef void (snd_fx8010_irq_handler_t)(struct snd_emu10k1 *emu, void *private_data);
+
+struct snd_emu10k1_fx8010_irq {
+ struct snd_emu10k1_fx8010_irq *next;
+ snd_fx8010_irq_handler_t *handler;
+ unsigned short gpr_running;
+ void *private_data;
+};
+
+struct snd_emu10k1_fx8010_pcm {
+ unsigned int valid: 1,
+ opened: 1,
+ active: 1;
+ unsigned int channels; /* 16-bit channels count */
+ unsigned int tram_start; /* initial ring buffer position in TRAM (in samples) */
+ unsigned int buffer_size; /* count of buffered samples */
+ unsigned short gpr_size; /* GPR containing size of ring buffer in samples (host) */
+ unsigned short gpr_ptr; /* GPR containing current pointer in the ring buffer (host = reset, FX8010) */
+ unsigned short gpr_count; /* GPR containing count of samples between two interrupts (host) */
+ unsigned short gpr_tmpcount; /* GPR containing current count of samples to interrupt (host = set, FX8010) */
+ unsigned short gpr_trigger; /* GPR containing trigger (activate) information (host) */
+ unsigned short gpr_running; /* GPR containing info if PCM is running (FX8010) */
+ unsigned char etram[32]; /* external TRAM address & data */
+ struct snd_pcm_indirect pcm_rec;
+ unsigned int tram_pos;
+ unsigned int tram_shift;
+ struct snd_emu10k1_fx8010_irq *irq;
+};
+
+struct snd_emu10k1_fx8010 {
+ unsigned short fxbus_mask; /* used FX buses (bitmask) */
+ unsigned short extin_mask; /* used external inputs (bitmask) */
+ unsigned short extout_mask; /* used external outputs (bitmask) */
+ unsigned short pad1;
+ unsigned int itram_size; /* internal TRAM size in samples */
+ struct snd_dma_buffer etram_pages; /* external TRAM pages and size */
+ unsigned int dbg; /* FX debugger register */
+ unsigned char name[128];
+ int gpr_size; /* size of allocated GPR controls */
+ int gpr_count; /* count of used kcontrols */
+ struct list_head gpr_ctl; /* GPR controls */
+ struct mutex lock;
+ struct snd_emu10k1_fx8010_pcm pcm[8];
+ spinlock_t irq_lock;
+ struct snd_emu10k1_fx8010_irq *irq_handlers;
+};
+
+struct snd_emu10k1_midi {
+ struct snd_emu10k1 *emu;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *substream_input;
+ struct snd_rawmidi_substream *substream_output;
+ unsigned int midi_mode;
+ spinlock_t input_lock;
+ spinlock_t output_lock;
+ spinlock_t open_lock;
+ int tx_enable, rx_enable;
+ int port;
+ int ipr_tx, ipr_rx;
+ void (*interrupt)(struct snd_emu10k1 *emu, unsigned int status);
+};
+
+enum {
+ EMU_MODEL_SB,
+ EMU_MODEL_EMU1010,
+ EMU_MODEL_EMU1010B,
+ EMU_MODEL_EMU1616,
+ EMU_MODEL_EMU0404,
+};
+
+struct snd_emu_chip_details {
+ u32 vendor;
+ u32 device;
+ u32 subsystem;
+ unsigned char revision;
+ unsigned char emu10k1_chip; /* Original SB Live. Not SB Live 24bit. */
+ unsigned char emu10k2_chip; /* Audigy 1 or Audigy 2. */
+ unsigned char ca0102_chip; /* Audigy 1 or Audigy 2. Not SB Audigy 2 Value. */
+ unsigned char ca0108_chip; /* Audigy 2 Value */
+ unsigned char ca_cardbus_chip; /* Audigy 2 ZS Notebook */
+ unsigned char ca0151_chip; /* P16V */
+ unsigned char spk71; /* Has 7.1 speakers */
+ unsigned char sblive51; /* SBLive! 5.1 - extout 0x11 -> center, 0x12 -> lfe */
+ unsigned char spdif_bug; /* Has Spdif phasing bug */
+ unsigned char ac97_chip; /* Has an AC97 chip: 1 = mandatory, 2 = optional */
+ unsigned char ecard; /* APS EEPROM */
+ unsigned char emu_model; /* EMU model type */
+ unsigned char spi_dac; /* SPI interface for DAC */
+ unsigned char i2c_adc; /* I2C interface for ADC */
+ unsigned char adc_1361t; /* Use Philips 1361T ADC */
+ unsigned char invert_shared_spdif; /* analog/digital switch inverted */
+ const char *driver;
+ const char *name;
+ const char *id; /* for backward compatibility - can be NULL if not needed */
+};
+
+struct snd_emu1010 {
+ unsigned int output_source[64];
+ unsigned int input_source[64];
+ unsigned int adc_pads; /* bit mask */
+ unsigned int dac_pads; /* bit mask */
+ unsigned int internal_clock; /* 44100 or 48000 */
+ unsigned int optical_in; /* 0:SPDIF, 1:ADAT */
+ unsigned int optical_out; /* 0:SPDIF, 1:ADAT */
+ struct task_struct *firmware_thread;
+};
+
+struct snd_emu10k1 {
+ int irq;
+
+ unsigned long port; /* I/O port number */
+ unsigned int tos_link: 1, /* tos link detected */
+ rear_ac97: 1, /* rear channels are on AC'97 */
+ enable_ir: 1;
+ unsigned int support_tlv :1;
+ /* Contains profile of card capabilities */
+ const struct snd_emu_chip_details *card_capabilities;
+ unsigned int audigy; /* is Audigy? */
+ unsigned int revision; /* chip revision */
+ unsigned int serial; /* serial number */
+ unsigned short model; /* subsystem id */
+ unsigned int card_type; /* EMU10K1_CARD_* */
+ unsigned int ecard_ctrl; /* ecard control bits */
+ unsigned int address_mode; /* address mode */
+ unsigned long dma_mask; /* PCI DMA mask */
+ unsigned int delay_pcm_irq; /* in samples */
+ int max_cache_pages; /* max memory size / PAGE_SIZE */
+ struct snd_dma_buffer silent_page; /* silent page */
+ struct snd_dma_buffer ptb_pages; /* page table pages */
+ struct snd_dma_device p16v_dma_dev;
+ struct snd_dma_buffer p16v_buffer;
+
+ struct snd_util_memhdr *memhdr; /* page allocation list */
+ struct snd_emu10k1_memblk *reserved_page; /* reserved page */
+
+ struct list_head mapped_link_head;
+ struct list_head mapped_order_link_head;
+ void **page_ptr_table;
+ unsigned long *page_addr_table;
+ spinlock_t memblk_lock;
+
+ unsigned int spdif_bits[3]; /* s/pdif out setup */
+ unsigned int i2c_capture_source;
+ u8 i2c_capture_volume[4][2];
+
+ struct snd_emu10k1_fx8010 fx8010; /* FX8010 info */
+ int gpr_base;
+
+ struct snd_ac97 *ac97;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm *pcm_mic;
+ struct snd_pcm *pcm_efx;
+ struct snd_pcm *pcm_multi;
+ struct snd_pcm *pcm_p16v;
+
+ spinlock_t synth_lock;
+ void *synth;
+ int (*get_synth_voice)(struct snd_emu10k1 *emu);
+
+ spinlock_t reg_lock;
+ spinlock_t emu_lock;
+ spinlock_t voice_lock;
+ spinlock_t spi_lock; /* serialises access to spi port */
+ spinlock_t i2c_lock; /* serialises access to i2c port */
+
+ struct snd_emu10k1_voice voices[NUM_G];
+ struct snd_emu10k1_voice p16v_voices[4];
+ struct snd_emu10k1_voice p16v_capture_voice;
+ int p16v_device_offset;
+ u32 p16v_capture_source;
+ u32 p16v_capture_channel;
+ struct snd_emu1010 emu1010;
+ struct snd_emu10k1_pcm_mixer pcm_mixer[32];
+ struct snd_emu10k1_pcm_mixer efx_pcm_mixer[NUM_EFX_PLAYBACK];
+ struct snd_kcontrol *ctl_send_routing;
+ struct snd_kcontrol *ctl_send_volume;
+ struct snd_kcontrol *ctl_attn;
+ struct snd_kcontrol *ctl_efx_send_routing;
+ struct snd_kcontrol *ctl_efx_send_volume;
+ struct snd_kcontrol *ctl_efx_attn;
+
+ void (*hwvol_interrupt)(struct snd_emu10k1 *emu, unsigned int status);
+ void (*capture_interrupt)(struct snd_emu10k1 *emu, unsigned int status);
+ void (*capture_mic_interrupt)(struct snd_emu10k1 *emu, unsigned int status);
+ void (*capture_efx_interrupt)(struct snd_emu10k1 *emu, unsigned int status);
+ void (*spdif_interrupt)(struct snd_emu10k1 *emu, unsigned int status);
+ void (*dsp_interrupt)(struct snd_emu10k1 *emu);
+
+ struct snd_pcm_substream *pcm_capture_substream;
+ struct snd_pcm_substream *pcm_capture_mic_substream;
+ struct snd_pcm_substream *pcm_capture_efx_substream;
+ struct snd_pcm_substream *pcm_playback_efx_substream;
+
+ struct snd_timer *timer;
+
+ struct snd_emu10k1_midi midi;
+ struct snd_emu10k1_midi midi2; /* for audigy */
+
+ unsigned int efx_voices_mask[2];
+ unsigned int next_free_voice;
+
+ const struct firmware *firmware;
+ const struct firmware *dock_fw;
+
+#ifdef CONFIG_PM_SLEEP
+ unsigned int *saved_ptr;
+ unsigned int *saved_gpr;
+ unsigned int *tram_val_saved;
+ unsigned int *tram_addr_saved;
+ unsigned int *saved_icode;
+ unsigned int *p16v_saved;
+ unsigned int saved_a_iocfg, saved_hcfg;
+ bool suspend;
+#endif
+
+};
+
+int snd_emu10k1_create(struct snd_card *card,
+ struct pci_dev *pci,
+ unsigned short extin_mask,
+ unsigned short extout_mask,
+ long max_cache_bytes,
+ int enable_ir,
+ uint subsystem,
+ struct snd_emu10k1 ** remu);
+
+int snd_emu10k1_pcm(struct snd_emu10k1 *emu, int device);
+int snd_emu10k1_pcm_mic(struct snd_emu10k1 *emu, int device);
+int snd_emu10k1_pcm_efx(struct snd_emu10k1 *emu, int device);
+int snd_p16v_pcm(struct snd_emu10k1 *emu, int device);
+int snd_p16v_free(struct snd_emu10k1 * emu);
+int snd_p16v_mixer(struct snd_emu10k1 * emu);
+int snd_emu10k1_pcm_multi(struct snd_emu10k1 *emu, int device);
+int snd_emu10k1_fx8010_pcm(struct snd_emu10k1 *emu, int device);
+int snd_emu10k1_mixer(struct snd_emu10k1 * emu, int pcm_device, int multi_device);
+int snd_emu10k1_timer(struct snd_emu10k1 * emu, int device);
+int snd_emu10k1_fx8010_new(struct snd_emu10k1 *emu, int device);
+
+irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id);
+
+void snd_emu10k1_voice_init(struct snd_emu10k1 * emu, int voice);
+int snd_emu10k1_init_efx(struct snd_emu10k1 *emu);
+void snd_emu10k1_free_efx(struct snd_emu10k1 *emu);
+int snd_emu10k1_fx8010_tram_setup(struct snd_emu10k1 *emu, u32 size);
+int snd_emu10k1_done(struct snd_emu10k1 * emu);
+
+/* I/O functions */
+unsigned int snd_emu10k1_ptr_read(struct snd_emu10k1 * emu, unsigned int reg, unsigned int chn);
+void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data);
+unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg, unsigned int chn);
+void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data);
+int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data);
+int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, u32 reg, u32 value);
+int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value);
+int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value);
+int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src);
+unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc);
+void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb);
+void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb);
+void snd_emu10k1_voice_intr_enable(struct snd_emu10k1 *emu, unsigned int voicenum);
+void snd_emu10k1_voice_intr_disable(struct snd_emu10k1 *emu, unsigned int voicenum);
+void snd_emu10k1_voice_intr_ack(struct snd_emu10k1 *emu, unsigned int voicenum);
+void snd_emu10k1_voice_half_loop_intr_enable(struct snd_emu10k1 *emu, unsigned int voicenum);
+void snd_emu10k1_voice_half_loop_intr_disable(struct snd_emu10k1 *emu, unsigned int voicenum);
+void snd_emu10k1_voice_half_loop_intr_ack(struct snd_emu10k1 *emu, unsigned int voicenum);
+void snd_emu10k1_voice_set_loop_stop(struct snd_emu10k1 *emu, unsigned int voicenum);
+void snd_emu10k1_voice_clear_loop_stop(struct snd_emu10k1 *emu, unsigned int voicenum);
+void snd_emu10k1_wait(struct snd_emu10k1 *emu, unsigned int wait);
+static inline unsigned int snd_emu10k1_wc(struct snd_emu10k1 *emu) { return (inl(emu->port + WC) >> 6) & 0xfffff; }
+unsigned short snd_emu10k1_ac97_read(struct snd_ac97 *ac97, unsigned short reg);
+void snd_emu10k1_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short data);
+unsigned int snd_emu10k1_rate_to_pitch(unsigned int rate);
+
+#ifdef CONFIG_PM_SLEEP
+void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu);
+void snd_emu10k1_resume_init(struct snd_emu10k1 *emu);
+void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu);
+int snd_emu10k1_efx_alloc_pm_buffer(struct snd_emu10k1 *emu);
+void snd_emu10k1_efx_free_pm_buffer(struct snd_emu10k1 *emu);
+void snd_emu10k1_efx_suspend(struct snd_emu10k1 *emu);
+void snd_emu10k1_efx_resume(struct snd_emu10k1 *emu);
+int snd_p16v_alloc_pm_buffer(struct snd_emu10k1 *emu);
+void snd_p16v_free_pm_buffer(struct snd_emu10k1 *emu);
+void snd_p16v_suspend(struct snd_emu10k1 *emu);
+void snd_p16v_resume(struct snd_emu10k1 *emu);
+#endif
+
+/* memory allocation */
+struct snd_util_memblk *snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *substream);
+int snd_emu10k1_free_pages(struct snd_emu10k1 *emu, struct snd_util_memblk *blk);
+struct snd_util_memblk *snd_emu10k1_synth_alloc(struct snd_emu10k1 *emu, unsigned int size);
+int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *blk);
+int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size);
+int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size);
+int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk);
+
+/* voice allocation */
+int snd_emu10k1_voice_alloc(struct snd_emu10k1 *emu, int type, int pair, struct snd_emu10k1_voice **rvoice);
+int snd_emu10k1_voice_free(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *pvoice);
+
+/* MIDI uart */
+int snd_emu10k1_midi(struct snd_emu10k1 * emu);
+int snd_emu10k1_audigy_midi(struct snd_emu10k1 * emu);
+
+/* proc interface */
+int snd_emu10k1_proc_init(struct snd_emu10k1 * emu);
+
+/* fx8010 irq handler */
+int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
+ snd_fx8010_irq_handler_t *handler,
+ unsigned char gpr_running,
+ void *private_data,
+ struct snd_emu10k1_fx8010_irq **r_irq);
+int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_irq *irq);
+
+#endif /* __SOUND_EMU10K1_H */
diff --git a/kernel/include/sound/emu10k1_synth.h b/kernel/include/sound/emu10k1_synth.h
new file mode 100644
index 000000000..9f211e957
--- /dev/null
+++ b/kernel/include/sound/emu10k1_synth.h
@@ -0,0 +1,39 @@
+#ifndef __EMU10K1_SYNTH_H
+#define __EMU10K1_SYNTH_H
+/*
+ * Defines for the Emu10k1 WaveTable synth
+ *
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/emu10k1.h>
+#include <sound/emux_synth.h>
+
+/* sequencer device id */
+#define SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH "emu10k1-synth"
+
+/* argument for snd_seq_device_new */
+struct snd_emu10k1_synth_arg {
+ struct snd_emu10k1 *hwptr; /* chip */
+ int index; /* sequencer client index */
+ int seq_ports; /* number of sequencer ports to be created */
+ int max_voices; /* maximum number of voices for wavetable */
+};
+
+#define EMU10K1_MAX_MEMSIZE (32 * 1024 * 1024) /* 32MB */
+
+#endif
diff --git a/kernel/include/sound/emu8000.h b/kernel/include/sound/emu8000.h
new file mode 100644
index 000000000..c321302a9
--- /dev/null
+++ b/kernel/include/sound/emu8000.h
@@ -0,0 +1,121 @@
+#ifndef __SOUND_EMU8000_H
+#define __SOUND_EMU8000_H
+/*
+ * Defines for the emu8000 (AWE32/64)
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/emux_synth.h>
+#include <sound/seq_kernel.h>
+
+/*
+ * Hardware parameters.
+ */
+#define EMU8000_MAX_DRAM (28 * 1024 * 1024) /* Max on-board mem is 28Mb ???*/
+#define EMU8000_DRAM_OFFSET 0x200000 /* Beginning of on board ram */
+#define EMU8000_CHANNELS 32 /* Number of hardware channels */
+#define EMU8000_DRAM_VOICES 30 /* number of normal voices */
+
+/* Flags to set a dma channel to read or write */
+#define EMU8000_RAM_READ 0
+#define EMU8000_RAM_WRITE 1
+#define EMU8000_RAM_CLOSE 2
+#define EMU8000_RAM_MODE_MASK 0x03
+#define EMU8000_RAM_RIGHT 0x10 /* use 'right' DMA channel */
+
+enum {
+ EMU8000_CONTROL_BASS = 0,
+ EMU8000_CONTROL_TREBLE,
+ EMU8000_CONTROL_CHORUS_MODE,
+ EMU8000_CONTROL_REVERB_MODE,
+ EMU8000_CONTROL_FM_CHORUS_DEPTH,
+ EMU8000_CONTROL_FM_REVERB_DEPTH,
+ EMU8000_NUM_CONTROLS,
+};
+
+/*
+ * Structure to hold all state information for the emu8000 driver.
+ *
+ * Note 1: The chip supports 32 channels in hardware this is max_channels
+ * some of the channels may be used for other things so max_channels is
+ * the number in use for wave voices.
+ */
+struct snd_emu8000 {
+
+ struct snd_emux *emu;
+
+ int index; /* sequencer client index */
+ int seq_ports; /* number of sequencer ports */
+ int fm_chorus_depth; /* FM OPL3 chorus depth */
+ int fm_reverb_depth; /* FM OPL3 reverb depth */
+
+ int mem_size; /* memory size */
+ unsigned long port1; /* Port usually base+0 */
+ unsigned long port2; /* Port usually at base+0x400 */
+ unsigned long port3; /* Port usually at base+0x800 */
+ struct resource *res_port1;
+ struct resource *res_port2;
+ struct resource *res_port3;
+ unsigned short last_reg;/* Last register command */
+ spinlock_t reg_lock;
+
+ int dram_checked;
+
+ struct snd_card *card; /* The card that this belongs to */
+
+ int chorus_mode;
+ int reverb_mode;
+ int bass_level;
+ int treble_level;
+
+ struct snd_util_memhdr *memhdr;
+
+ spinlock_t control_lock;
+ struct snd_kcontrol *controls[EMU8000_NUM_CONTROLS];
+
+ struct snd_pcm *pcm; /* pcm on emu8000 wavetable */
+
+};
+
+/* sequencer device id */
+#define SNDRV_SEQ_DEV_ID_EMU8000 "emu8000-synth"
+
+
+/* exported functions */
+int snd_emu8000_new(struct snd_card *card, int device, long port, int seq_ports,
+ struct snd_seq_device **ret);
+void snd_emu8000_poke(struct snd_emu8000 *emu, unsigned int port, unsigned int reg,
+ unsigned int val);
+unsigned short snd_emu8000_peek(struct snd_emu8000 *emu, unsigned int port,
+ unsigned int reg);
+void snd_emu8000_poke_dw(struct snd_emu8000 *emu, unsigned int port, unsigned int reg,
+ unsigned int val);
+unsigned int snd_emu8000_peek_dw(struct snd_emu8000 *emu, unsigned int port,
+ unsigned int reg);
+void snd_emu8000_dma_chan(struct snd_emu8000 *emu, int ch, int mode);
+
+void snd_emu8000_init_fm(struct snd_emu8000 *emu);
+
+void snd_emu8000_update_chorus_mode(struct snd_emu8000 *emu);
+void snd_emu8000_update_reverb_mode(struct snd_emu8000 *emu);
+void snd_emu8000_update_equalizer(struct snd_emu8000 *emu);
+int snd_emu8000_load_chorus_fx(struct snd_emu8000 *emu, int mode, const void __user *buf, long len);
+int snd_emu8000_load_reverb_fx(struct snd_emu8000 *emu, int mode, const void __user *buf, long len);
+
+#endif /* __SOUND_EMU8000_H */
diff --git a/kernel/include/sound/emu8000_reg.h b/kernel/include/sound/emu8000_reg.h
new file mode 100644
index 000000000..4b9827ac4
--- /dev/null
+++ b/kernel/include/sound/emu8000_reg.h
@@ -0,0 +1,207 @@
+#ifndef __SOUND_EMU8000_REG_H
+#define __SOUND_EMU8000_REG_H
+/*
+ * Register operations for the EMU8000
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ *
+ * Based on awe_wave.c by Takashi Iwai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * Data port addresses relative to the EMU base.
+ */
+#define EMU8000_DATA0(e) ((e)->port1)
+#define EMU8000_DATA1(e) ((e)->port2)
+#define EMU8000_DATA2(e) ((e)->port2+2)
+#define EMU8000_DATA3(e) ((e)->port3)
+#define EMU8000_PTR(e) ((e)->port3+2)
+
+/*
+ * Make a command from a register and channel.
+ */
+#define EMU8000_CMD(reg, chan) ((reg)<<5 | (chan))
+
+/*
+ * Commands to read and write the EMU8000 registers.
+ * These macros should be used for all register accesses.
+ */
+#define EMU8000_CPF_READ(emu, chan) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(0, (chan)))
+#define EMU8000_PTRX_READ(emu, chan) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(1, (chan)))
+#define EMU8000_CVCF_READ(emu, chan) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(2, (chan)))
+#define EMU8000_VTFT_READ(emu, chan) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(3, (chan)))
+#define EMU8000_PSST_READ(emu, chan) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(6, (chan)))
+#define EMU8000_CSL_READ(emu, chan) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(7, (chan)))
+#define EMU8000_CCCA_READ(emu, chan) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(0, (chan)))
+#define EMU8000_HWCF4_READ(emu) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 9))
+#define EMU8000_HWCF5_READ(emu) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 10))
+#define EMU8000_HWCF6_READ(emu) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 13))
+#define EMU8000_SMALR_READ(emu) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 20))
+#define EMU8000_SMARR_READ(emu) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 21))
+#define EMU8000_SMALW_READ(emu) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 22))
+#define EMU8000_SMARW_READ(emu) \
+ snd_emu8000_peek_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 23))
+#define EMU8000_SMLD_READ(emu) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 26))
+#define EMU8000_SMRD_READ(emu) \
+ snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(1, 26))
+#define EMU8000_WC_READ(emu) \
+ snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(1, 27))
+#define EMU8000_HWCF1_READ(emu) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 29))
+#define EMU8000_HWCF2_READ(emu) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 30))
+#define EMU8000_HWCF3_READ(emu) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 31))
+#define EMU8000_INIT1_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(2, (chan)))
+#define EMU8000_INIT2_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(2, (chan)))
+#define EMU8000_INIT3_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(3, (chan)))
+#define EMU8000_INIT4_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(3, (chan)))
+#define EMU8000_ENVVOL_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(4, (chan)))
+#define EMU8000_DCYSUSV_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(5, (chan)))
+#define EMU8000_ENVVAL_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(6, (chan)))
+#define EMU8000_DCYSUS_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA1(emu), EMU8000_CMD(7, (chan)))
+#define EMU8000_ATKHLDV_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(4, (chan)))
+#define EMU8000_LFO1VAL_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(5, (chan)))
+#define EMU8000_ATKHLD_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(6, (chan)))
+#define EMU8000_LFO2VAL_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA2(emu), EMU8000_CMD(7, (chan)))
+#define EMU8000_IP_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(0, (chan)))
+#define EMU8000_IFATN_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(1, (chan)))
+#define EMU8000_PEFE_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(2, (chan)))
+#define EMU8000_FMMOD_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(3, (chan)))
+#define EMU8000_TREMFRQ_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(4, (chan)))
+#define EMU8000_FM2FRQ2_READ(emu, chan) \
+ snd_emu8000_peek((emu), EMU8000_DATA3(emu), EMU8000_CMD(5, (chan)))
+
+
+#define EMU8000_CPF_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(0, (chan)), (val))
+#define EMU8000_PTRX_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(1, (chan)), (val))
+#define EMU8000_CVCF_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(2, (chan)), (val))
+#define EMU8000_VTFT_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(3, (chan)), (val))
+#define EMU8000_PSST_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(6, (chan)), (val))
+#define EMU8000_CSL_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(7, (chan)), (val))
+#define EMU8000_CCCA_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(0, (chan)), (val))
+#define EMU8000_HWCF4_WRITE(emu, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 9), (val))
+#define EMU8000_HWCF5_WRITE(emu, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 10), (val))
+#define EMU8000_HWCF6_WRITE(emu, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 13), (val))
+/* this register is not documented */
+#define EMU8000_HWCF7_WRITE(emu, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 14), (val))
+#define EMU8000_SMALR_WRITE(emu, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 20), (val))
+#define EMU8000_SMARR_WRITE(emu, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 21), (val))
+#define EMU8000_SMALW_WRITE(emu, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 22), (val))
+#define EMU8000_SMARW_WRITE(emu, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 23), (val))
+#define EMU8000_SMLD_WRITE(emu, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 26), (val))
+#define EMU8000_SMRD_WRITE(emu, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(1, 26), (val))
+#define EMU8000_WC_WRITE(emu, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(1, 27), (val))
+#define EMU8000_HWCF1_WRITE(emu, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 29), (val))
+#define EMU8000_HWCF2_WRITE(emu, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 30), (val))
+#define EMU8000_HWCF3_WRITE(emu, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(1, 31), (val))
+#define EMU8000_INIT1_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(2, (chan)), (val))
+#define EMU8000_INIT2_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(2, (chan)), (val))
+#define EMU8000_INIT3_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(3, (chan)), (val))
+#define EMU8000_INIT4_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(3, (chan)), (val))
+#define EMU8000_ENVVOL_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(4, (chan)), (val))
+#define EMU8000_DCYSUSV_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(5, (chan)), (val))
+#define EMU8000_ENVVAL_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(6, (chan)), (val))
+#define EMU8000_DCYSUS_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA1(emu), EMU8000_CMD(7, (chan)), (val))
+#define EMU8000_ATKHLDV_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(4, (chan)), (val))
+#define EMU8000_LFO1VAL_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(5, (chan)), (val))
+#define EMU8000_ATKHLD_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(6, (chan)), (val))
+#define EMU8000_LFO2VAL_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA2(emu), EMU8000_CMD(7, (chan)), (val))
+#define EMU8000_IP_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(0, (chan)), (val))
+#define EMU8000_IFATN_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(1, (chan)), (val))
+#define EMU8000_PEFE_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(2, (chan)), (val))
+#define EMU8000_FMMOD_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(3, (chan)), (val))
+#define EMU8000_TREMFRQ_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(4, (chan)), (val))
+#define EMU8000_FM2FRQ2_WRITE(emu, chan, val) \
+ snd_emu8000_poke((emu), EMU8000_DATA3(emu), EMU8000_CMD(5, (chan)), (val))
+
+#define EMU8000_0080_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(4, (chan)), (val))
+#define EMU8000_00A0_WRITE(emu, chan, val) \
+ snd_emu8000_poke_dw((emu), EMU8000_DATA0(emu), EMU8000_CMD(5, (chan)), (val))
+
+#endif /* __SOUND_EMU8000_REG_H */
diff --git a/kernel/include/sound/emux_legacy.h b/kernel/include/sound/emux_legacy.h
new file mode 100644
index 000000000..baf43fc24
--- /dev/null
+++ b/kernel/include/sound/emux_legacy.h
@@ -0,0 +1,146 @@
+#ifndef __SOUND_EMUX_LEGACY_H
+#define __SOUND_EMUX_LEGACY_H
+
+/*
+ * Copyright (c) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Definitions of OSS compatible headers for Emu8000 device informations
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/seq_oss_legacy.h>
+
+/*
+ * awe hardware controls
+ */
+
+#define _EMUX_OSS_DEBUG_MODE 0x00
+#define _EMUX_OSS_REVERB_MODE 0x01
+#define _EMUX_OSS_CHORUS_MODE 0x02
+#define _EMUX_OSS_REMOVE_LAST_SAMPLES 0x03
+#define _EMUX_OSS_INITIALIZE_CHIP 0x04
+#define _EMUX_OSS_SEND_EFFECT 0x05
+#define _EMUX_OSS_TERMINATE_CHANNEL 0x06
+#define _EMUX_OSS_TERMINATE_ALL 0x07
+#define _EMUX_OSS_INITIAL_VOLUME 0x08
+#define _EMUX_OSS_INITIAL_ATTEN _EMUX_OSS_INITIAL_VOLUME
+#define _EMUX_OSS_RESET_CHANNEL 0x09
+#define _EMUX_OSS_CHANNEL_MODE 0x0a
+#define _EMUX_OSS_DRUM_CHANNELS 0x0b
+#define _EMUX_OSS_MISC_MODE 0x0c
+#define _EMUX_OSS_RELEASE_ALL 0x0d
+#define _EMUX_OSS_NOTEOFF_ALL 0x0e
+#define _EMUX_OSS_CHN_PRESSURE 0x0f
+#define _EMUX_OSS_EQUALIZER 0x11
+
+#define _EMUX_OSS_MODE_FLAG 0x80
+#define _EMUX_OSS_COOKED_FLAG 0x40 /* not supported */
+#define _EMUX_OSS_MODE_VALUE_MASK 0x3F
+
+
+/*
+ * mode type definitions
+ */
+enum {
+/* 0*/ EMUX_MD_EXCLUSIVE_OFF, /* obsolete */
+/* 1*/ EMUX_MD_EXCLUSIVE_ON, /* obsolete */
+/* 2*/ EMUX_MD_VERSION, /* read only */
+/* 3*/ EMUX_MD_EXCLUSIVE_SOUND, /* 0/1: exclusive note on (default=1) */
+/* 4*/ EMUX_MD_REALTIME_PAN, /* 0/1: do realtime pan change (default=1) */
+/* 5*/ EMUX_MD_GUS_BANK, /* bank number for GUS patches (default=0) */
+/* 6*/ EMUX_MD_KEEP_EFFECT, /* 0/1: keep effect values, (default=0) */
+/* 7*/ EMUX_MD_ZERO_ATTEN, /* attenuation of max volume (default=32) */
+/* 8*/ EMUX_MD_CHN_PRIOR, /* 0/1: set MIDI channel priority mode (default=1) */
+/* 9*/ EMUX_MD_MOD_SENSE, /* integer: modwheel sensitivity (def=18) */
+/*10*/ EMUX_MD_DEF_PRESET, /* integer: default preset number (def=0) */
+/*11*/ EMUX_MD_DEF_BANK, /* integer: default bank number (def=0) */
+/*12*/ EMUX_MD_DEF_DRUM, /* integer: default drumset number (def=0) */
+/*13*/ EMUX_MD_TOGGLE_DRUM_BANK, /* 0/1: toggle drum flag with bank# (def=0) */
+/*14*/ EMUX_MD_NEW_VOLUME_CALC, /* 0/1: volume calculation mode (def=1) */
+/*15*/ EMUX_MD_CHORUS_MODE, /* integer: chorus mode (def=2) */
+/*16*/ EMUX_MD_REVERB_MODE, /* integer: chorus mode (def=4) */
+/*17*/ EMUX_MD_BASS_LEVEL, /* integer: bass level (def=5) */
+/*18*/ EMUX_MD_TREBLE_LEVEL, /* integer: treble level (def=9) */
+/*19*/ EMUX_MD_DEBUG_MODE, /* integer: debug level (def=0) */
+/*20*/ EMUX_MD_PAN_EXCHANGE, /* 0/1: exchange panning direction (def=0) */
+ EMUX_MD_END,
+};
+
+
+/*
+ * effect parameters
+ */
+enum {
+
+/* modulation envelope parameters */
+/* 0*/ EMUX_FX_ENV1_DELAY, /* WORD: ENVVAL */
+/* 1*/ EMUX_FX_ENV1_ATTACK, /* BYTE: up ATKHLD */
+/* 2*/ EMUX_FX_ENV1_HOLD, /* BYTE: lw ATKHLD */
+/* 3*/ EMUX_FX_ENV1_DECAY, /* BYTE: lw DCYSUS */
+/* 4*/ EMUX_FX_ENV1_RELEASE, /* BYTE: lw DCYSUS */
+/* 5*/ EMUX_FX_ENV1_SUSTAIN, /* BYTE: up DCYSUS */
+/* 6*/ EMUX_FX_ENV1_PITCH, /* BYTE: up PEFE */
+/* 7*/ EMUX_FX_ENV1_CUTOFF, /* BYTE: lw PEFE */
+
+/* volume envelope parameters */
+/* 8*/ EMUX_FX_ENV2_DELAY, /* WORD: ENVVOL */
+/* 9*/ EMUX_FX_ENV2_ATTACK, /* BYTE: up ATKHLDV */
+/*10*/ EMUX_FX_ENV2_HOLD, /* BYTE: lw ATKHLDV */
+/*11*/ EMUX_FX_ENV2_DECAY, /* BYTE: lw DCYSUSV */
+/*12*/ EMUX_FX_ENV2_RELEASE, /* BYTE: lw DCYSUSV */
+/*13*/ EMUX_FX_ENV2_SUSTAIN, /* BYTE: up DCYSUSV */
+
+/* LFO1 (tremolo & vibrato) parameters */
+/*14*/ EMUX_FX_LFO1_DELAY, /* WORD: LFO1VAL */
+/*15*/ EMUX_FX_LFO1_FREQ, /* BYTE: lo TREMFRQ */
+/*16*/ EMUX_FX_LFO1_VOLUME, /* BYTE: up TREMFRQ */
+/*17*/ EMUX_FX_LFO1_PITCH, /* BYTE: up FMMOD */
+/*18*/ EMUX_FX_LFO1_CUTOFF, /* BYTE: lo FMMOD */
+
+/* LFO2 (vibrato) parameters */
+/*19*/ EMUX_FX_LFO2_DELAY, /* WORD: LFO2VAL */
+/*20*/ EMUX_FX_LFO2_FREQ, /* BYTE: lo FM2FRQ2 */
+/*21*/ EMUX_FX_LFO2_PITCH, /* BYTE: up FM2FRQ2 */
+
+/* Other overall effect parameters */
+/*22*/ EMUX_FX_INIT_PITCH, /* SHORT: pitch offset */
+/*23*/ EMUX_FX_CHORUS, /* BYTE: chorus effects send (0-255) */
+/*24*/ EMUX_FX_REVERB, /* BYTE: reverb effects send (0-255) */
+/*25*/ EMUX_FX_CUTOFF, /* BYTE: up IFATN */
+/*26*/ EMUX_FX_FILTERQ, /* BYTE: up CCCA */
+
+/* Sample / loop offset changes */
+/*27*/ EMUX_FX_SAMPLE_START, /* SHORT: offset */
+/*28*/ EMUX_FX_LOOP_START, /* SHORT: offset */
+/*29*/ EMUX_FX_LOOP_END, /* SHORT: offset */
+/*30*/ EMUX_FX_COARSE_SAMPLE_START, /* SHORT: upper word offset */
+/*31*/ EMUX_FX_COARSE_LOOP_START, /* SHORT: upper word offset */
+/*32*/ EMUX_FX_COARSE_LOOP_END, /* SHORT: upper word offset */
+/*33*/ EMUX_FX_ATTEN, /* BYTE: lo IFATN */
+
+ EMUX_FX_END,
+};
+/* number of effects */
+#define EMUX_NUM_EFFECTS EMUX_FX_END
+
+/* effect flag values */
+#define EMUX_FX_FLAG_OFF 0
+#define EMUX_FX_FLAG_SET 1
+#define EMUX_FX_FLAG_ADD 2
+
+
+#endif /* __SOUND_EMUX_LEGACY_H */
diff --git a/kernel/include/sound/emux_synth.h b/kernel/include/sound/emux_synth.h
new file mode 100644
index 000000000..fb81f3722
--- /dev/null
+++ b/kernel/include/sound/emux_synth.h
@@ -0,0 +1,244 @@
+#ifndef __SOUND_EMUX_SYNTH_H
+#define __SOUND_EMUX_SYNTH_H
+
+/*
+ * Defines for the Emu-series WaveTable chip
+ *
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/seq_kernel.h>
+#include <sound/seq_device.h>
+#include <sound/soundfont.h>
+#include <sound/seq_midi_emul.h>
+#ifdef CONFIG_SND_SEQUENCER_OSS
+#include <sound/seq_oss.h>
+#endif
+#include <sound/emux_legacy.h>
+#include <sound/seq_virmidi.h>
+
+/*
+ * compile flags
+ */
+#define SNDRV_EMUX_USE_RAW_EFFECT
+
+struct snd_emux;
+struct snd_emux_port;
+struct snd_emux_voice;
+struct snd_emux_effect_table;
+
+/*
+ * operators
+ */
+struct snd_emux_operators {
+ struct module *owner;
+ struct snd_emux_voice *(*get_voice)(struct snd_emux *emu,
+ struct snd_emux_port *port);
+ int (*prepare)(struct snd_emux_voice *vp);
+ void (*trigger)(struct snd_emux_voice *vp);
+ void (*release)(struct snd_emux_voice *vp);
+ void (*update)(struct snd_emux_voice *vp, int update);
+ void (*terminate)(struct snd_emux_voice *vp);
+ void (*free_voice)(struct snd_emux_voice *vp);
+ void (*reset)(struct snd_emux *emu, int ch);
+ /* the first parameters are struct snd_emux */
+ int (*sample_new)(struct snd_emux *emu, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr,
+ const void __user *data, long count);
+ int (*sample_free)(struct snd_emux *emu, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr);
+ void (*sample_reset)(struct snd_emux *emu);
+ int (*load_fx)(struct snd_emux *emu, int type, int arg,
+ const void __user *data, long count);
+ void (*sysex)(struct snd_emux *emu, char *buf, int len, int parsed,
+ struct snd_midi_channel_set *chset);
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ int (*oss_ioctl)(struct snd_emux *emu, int cmd, int p1, int p2);
+#endif
+};
+
+
+/*
+ * constant values
+ */
+#define SNDRV_EMUX_MAX_PORTS 32 /* max # of sequencer ports */
+#define SNDRV_EMUX_MAX_VOICES 64 /* max # of voices */
+#define SNDRV_EMUX_MAX_MULTI_VOICES 16 /* max # of playable voices
+ * simultineously
+ */
+
+/*
+ * flags
+ */
+#define SNDRV_EMUX_ACCEPT_ROM (1<<0)
+
+/*
+ * emuX wavetable
+ */
+struct snd_emux {
+
+ struct snd_card *card; /* assigned card */
+
+ /* following should be initialized before registration */
+ int max_voices; /* Number of voices */
+ int mem_size; /* memory size (in byte) */
+ int num_ports; /* number of ports to be created */
+ int pitch_shift; /* pitch shift value (for Emu10k1) */
+ struct snd_emux_operators ops; /* operators */
+ void *hw; /* hardware */
+ unsigned long flags; /* other conditions */
+ int midi_ports; /* number of virtual midi devices */
+ int midi_devidx; /* device offset of virtual midi */
+ unsigned int linear_panning: 1; /* panning is linear (sbawe = 1, emu10k1 = 0) */
+ int hwdep_idx; /* hwdep device index */
+ struct snd_hwdep *hwdep; /* hwdep device */
+
+ /* private */
+ int num_voices; /* current number of voices */
+ struct snd_sf_list *sflist; /* root of SoundFont list */
+ struct snd_emux_voice *voices; /* Voices (EMU 'channel') */
+ int use_time; /* allocation counter */
+ spinlock_t voice_lock; /* Lock for voice access */
+ struct mutex register_mutex;
+ int client; /* For the sequencer client */
+ int ports[SNDRV_EMUX_MAX_PORTS]; /* The ports for this device */
+ struct snd_emux_port *portptrs[SNDRV_EMUX_MAX_PORTS];
+ int used; /* use counter */
+ char *name; /* name of the device (internal) */
+ struct snd_rawmidi **vmidi;
+ struct timer_list tlist; /* for pending note-offs */
+ int timer_active;
+
+ struct snd_util_memhdr *memhdr; /* memory chunk information */
+
+#ifdef CONFIG_PROC_FS
+ struct snd_info_entry *proc;
+#endif
+
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ struct snd_seq_device *oss_synth;
+#endif
+};
+
+
+/*
+ * sequencer port information
+ */
+struct snd_emux_port {
+
+ struct snd_midi_channel_set chset;
+ struct snd_emux *emu;
+
+ char port_mode; /* operation mode */
+ int volume_atten; /* emuX raw attenuation */
+ unsigned long drum_flags; /* drum bitmaps */
+ int ctrls[EMUX_MD_END]; /* control parameters */
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ struct snd_emux_effect_table *effect;
+#endif
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ struct snd_seq_oss_arg *oss_arg;
+#endif
+};
+
+/* port_mode */
+#define SNDRV_EMUX_PORT_MODE_MIDI 0 /* normal MIDI port */
+#define SNDRV_EMUX_PORT_MODE_OSS_SYNTH 1 /* OSS synth port */
+#define SNDRV_EMUX_PORT_MODE_OSS_MIDI 2 /* OSS multi channel synth port */
+
+/*
+ * A structure to keep track of each hardware voice
+ */
+struct snd_emux_voice {
+ int ch; /* Hardware channel number */
+
+ int state; /* status */
+#define SNDRV_EMUX_ST_OFF 0x00 /* Not playing, and inactive */
+#define SNDRV_EMUX_ST_ON 0x01 /* Note on */
+#define SNDRV_EMUX_ST_RELEASED (0x02|SNDRV_EMUX_ST_ON) /* Note released */
+#define SNDRV_EMUX_ST_SUSTAINED (0x04|SNDRV_EMUX_ST_ON) /* Note sustained */
+#define SNDRV_EMUX_ST_STANDBY (0x08|SNDRV_EMUX_ST_ON) /* Waiting to be triggered */
+#define SNDRV_EMUX_ST_PENDING (0x10|SNDRV_EMUX_ST_ON) /* Note will be released */
+#define SNDRV_EMUX_ST_LOCKED 0x100 /* Not accessible */
+
+ unsigned int time; /* An allocation time */
+ unsigned char note; /* Note currently assigned to this voice */
+ unsigned char key;
+ unsigned char velocity; /* Velocity of current note */
+
+ struct snd_sf_zone *zone; /* Zone assigned to this note */
+ void *block; /* sample block pointer (optional) */
+ struct snd_midi_channel *chan; /* Midi channel for this note */
+ struct snd_emux_port *port; /* associated port */
+ struct snd_emux *emu; /* assigned root info */
+ void *hw; /* hardware pointer (emu8000 or emu10k1) */
+ unsigned long ontime; /* jiffies at note triggered */
+
+ /* Emu8k/Emu10k1 registers */
+ struct soundfont_voice_info reg;
+
+ /* additional registers */
+ int avol; /* volume attenuation */
+ int acutoff; /* cutoff target */
+ int apitch; /* pitch offset */
+ int apan; /* pan/aux pair */
+ int aaux;
+ int ptarget; /* pitch target */
+ int vtarget; /* volume target */
+ int ftarget; /* filter target */
+
+};
+
+/*
+ * update flags (can be combined)
+ */
+#define SNDRV_EMUX_UPDATE_VOLUME (1<<0)
+#define SNDRV_EMUX_UPDATE_PITCH (1<<1)
+#define SNDRV_EMUX_UPDATE_PAN (1<<2)
+#define SNDRV_EMUX_UPDATE_FMMOD (1<<3)
+#define SNDRV_EMUX_UPDATE_TREMFREQ (1<<4)
+#define SNDRV_EMUX_UPDATE_FM2FRQ2 (1<<5)
+#define SNDRV_EMUX_UPDATE_Q (1<<6)
+
+
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+/*
+ * effect table
+ */
+struct snd_emux_effect_table {
+ /* Emu8000 specific effects */
+ short val[EMUX_NUM_EFFECTS];
+ unsigned char flag[EMUX_NUM_EFFECTS];
+};
+#endif /* SNDRV_EMUX_USE_RAW_EFFECT */
+
+
+/*
+ * prototypes - interface to Emu10k1 and Emu8k routines
+ */
+int snd_emux_new(struct snd_emux **remu);
+int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, char *name);
+int snd_emux_free(struct snd_emux *emu);
+
+/*
+ * exported functions
+ */
+void snd_emux_terminate_all(struct snd_emux *emu);
+void snd_emux_lock_voice(struct snd_emux *emu, int voice);
+void snd_emux_unlock_voice(struct snd_emux *emu, int voice);
+
+#endif /* __SOUND_EMUX_SYNTH_H */
diff --git a/kernel/include/sound/es1688.h b/kernel/include/sound/es1688.h
new file mode 100644
index 000000000..b34f23a5b
--- /dev/null
+++ b/kernel/include/sound/es1688.h
@@ -0,0 +1,122 @@
+#ifndef __SOUND_ES1688_H
+#define __SOUND_ES1688_H
+
+/*
+ * Header file for ES488/ES1688
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <linux/interrupt.h>
+
+#define ES1688_HW_AUTO 0x0000
+#define ES1688_HW_688 0x0001
+#define ES1688_HW_1688 0x0002
+#define ES1688_HW_UNDEF 0x0003
+
+struct snd_es1688 {
+ unsigned long port; /* port of ESS chip */
+ struct resource *res_port;
+ unsigned long mpu_port; /* MPU-401 port of ESS chip */
+ int irq; /* IRQ number of ESS chip */
+ int mpu_irq; /* MPU IRQ */
+ int dma8; /* 8-bit DMA */
+ unsigned short version; /* version of ESS chip */
+ unsigned short hardware; /* see to ES1688_HW_XXXX */
+
+ unsigned short trigger_value;
+ unsigned char pad;
+ unsigned int dma_size;
+
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ spinlock_t reg_lock;
+ spinlock_t mixer_lock;
+};
+
+/* I/O ports */
+
+#define ES1688P(codec, x) ((codec)->port + e_s_s_ESS1688##x)
+
+#define e_s_s_ESS1688RESET 0x6
+#define e_s_s_ESS1688READ 0xa
+#define e_s_s_ESS1688WRITE 0xc
+#define e_s_s_ESS1688COMMAND 0xc
+#define e_s_s_ESS1688STATUS 0xc
+#define e_s_s_ESS1688DATA_AVAIL 0xe
+#define e_s_s_ESS1688DATA_AVAIL_16 0xf
+#define e_s_s_ESS1688MIXER_ADDR 0x4
+#define e_s_s_ESS1688MIXER_DATA 0x5
+#define e_s_s_ESS1688OPL3_LEFT 0x0
+#define e_s_s_ESS1688OPL3_RIGHT 0x2
+#define e_s_s_ESS1688OPL3_BOTH 0x8
+#define e_s_s_ESS1688ENABLE0 0x0
+#define e_s_s_ESS1688ENABLE1 0x9
+#define e_s_s_ESS1688ENABLE2 0xb
+#define e_s_s_ESS1688INIT1 0x7
+
+#define ES1688_DSP_CMD_DMAOFF 0xd0
+#define ES1688_DSP_CMD_SPKON 0xd1
+#define ES1688_DSP_CMD_SPKOFF 0xd3
+#define ES1688_DSP_CMD_DMAON 0xd4
+
+#define ES1688_PCM_DEV 0x14
+#define ES1688_MIC_DEV 0x1a
+#define ES1688_REC_DEV 0x1c
+#define ES1688_MASTER_DEV 0x32
+#define ES1688_FM_DEV 0x36
+#define ES1688_CD_DEV 0x38
+#define ES1688_AUX_DEV 0x3a
+#define ES1688_SPEAKER_DEV 0x3c
+#define ES1688_LINE_DEV 0x3e
+#define ES1688_RECLEV_DEV 0xb4
+
+#define ES1688_MIXS_MASK 0x17
+#define ES1688_MIXS_MIC 0x00
+#define ES1688_MIXS_MIC_MASTER 0x01
+#define ES1688_MIXS_CD 0x02
+#define ES1688_MIXS_AOUT 0x03
+#define ES1688_MIXS_MIC1 0x04
+#define ES1688_MIXS_REC_MIX 0x05
+#define ES1688_MIXS_LINE 0x06
+#define ES1688_MIXS_MASTER 0x07
+#define ES1688_MIXS_MUTE 0x10
+
+/*
+
+ */
+
+void snd_es1688_mixer_write(struct snd_es1688 *chip, unsigned char reg, unsigned char data);
+
+int snd_es1688_create(struct snd_card *card,
+ struct snd_es1688 *chip,
+ unsigned long port,
+ unsigned long mpu_port,
+ int irq,
+ int mpu_irq,
+ int dma8,
+ unsigned short hardware);
+int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, int device);
+int snd_es1688_mixer(struct snd_card *card, struct snd_es1688 *chip);
+int snd_es1688_reset(struct snd_es1688 *chip);
+
+#endif /* __SOUND_ES1688_H */
diff --git a/kernel/include/sound/gus.h b/kernel/include/sound/gus.h
new file mode 100644
index 000000000..07c116fe7
--- /dev/null
+++ b/kernel/include/sound/gus.h
@@ -0,0 +1,631 @@
+#ifndef __SOUND_GUS_H
+#define __SOUND_GUS_H
+
+/*
+ * Global structures used for GUS part of ALSA driver
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/timer.h>
+#include <sound/seq_midi_emul.h>
+#include <sound/seq_device.h>
+#include <linux/io.h>
+
+/* IO ports */
+
+#define GUSP(gus, x) ((gus)->gf1.port + SNDRV_g_u_s_##x)
+
+#define SNDRV_g_u_s_MIDICTRL (0x320-0x220)
+#define SNDRV_g_u_s_MIDISTAT (0x320-0x220)
+#define SNDRV_g_u_s_MIDIDATA (0x321-0x220)
+
+#define SNDRV_g_u_s_GF1PAGE (0x322-0x220)
+#define SNDRV_g_u_s_GF1REGSEL (0x323-0x220)
+#define SNDRV_g_u_s_GF1DATALOW (0x324-0x220)
+#define SNDRV_g_u_s_GF1DATAHIGH (0x325-0x220)
+#define SNDRV_g_u_s_IRQSTAT (0x226-0x220)
+#define SNDRV_g_u_s_TIMERCNTRL (0x228-0x220)
+#define SNDRV_g_u_s_TIMERDATA (0x229-0x220)
+#define SNDRV_g_u_s_DRAM (0x327-0x220)
+#define SNDRV_g_u_s_MIXCNTRLREG (0x220-0x220)
+#define SNDRV_g_u_s_IRQDMACNTRLREG (0x22b-0x220)
+#define SNDRV_g_u_s_REGCNTRLS (0x22f-0x220)
+#define SNDRV_g_u_s_BOARDVERSION (0x726-0x220)
+#define SNDRV_g_u_s_MIXCNTRLPORT (0x726-0x220)
+#define SNDRV_g_u_s_IVER (0x325-0x220)
+#define SNDRV_g_u_s_MIXDATAPORT (0x326-0x220)
+#define SNDRV_g_u_s_MAXCNTRLPORT (0x326-0x220)
+
+/* GF1 registers */
+
+/* global registers */
+#define SNDRV_GF1_GB_ACTIVE_VOICES 0x0e
+#define SNDRV_GF1_GB_VOICES_IRQ 0x0f
+#define SNDRV_GF1_GB_GLOBAL_MODE 0x19
+#define SNDRV_GF1_GW_LFO_BASE 0x1a
+#define SNDRV_GF1_GB_VOICES_IRQ_READ 0x1f
+#define SNDRV_GF1_GB_DRAM_DMA_CONTROL 0x41
+#define SNDRV_GF1_GW_DRAM_DMA_LOW 0x42
+#define SNDRV_GF1_GW_DRAM_IO_LOW 0x43
+#define SNDRV_GF1_GB_DRAM_IO_HIGH 0x44
+#define SNDRV_GF1_GB_SOUND_BLASTER_CONTROL 0x45
+#define SNDRV_GF1_GB_ADLIB_TIMER_1 0x46
+#define SNDRV_GF1_GB_ADLIB_TIMER_2 0x47
+#define SNDRV_GF1_GB_RECORD_RATE 0x48
+#define SNDRV_GF1_GB_REC_DMA_CONTROL 0x49
+#define SNDRV_GF1_GB_JOYSTICK_DAC_LEVEL 0x4b
+#define SNDRV_GF1_GB_RESET 0x4c
+#define SNDRV_GF1_GB_DRAM_DMA_HIGH 0x50
+#define SNDRV_GF1_GW_DRAM_IO16 0x51
+#define SNDRV_GF1_GW_MEMORY_CONFIG 0x52
+#define SNDRV_GF1_GB_MEMORY_CONTROL 0x53
+#define SNDRV_GF1_GW_FIFO_RECORD_BASE_ADDR 0x54
+#define SNDRV_GF1_GW_FIFO_PLAY_BASE_ADDR 0x55
+#define SNDRV_GF1_GW_FIFO_SIZE 0x56
+#define SNDRV_GF1_GW_INTERLEAVE 0x57
+#define SNDRV_GF1_GB_COMPATIBILITY 0x59
+#define SNDRV_GF1_GB_DECODE_CONTROL 0x5a
+#define SNDRV_GF1_GB_VERSION_NUMBER 0x5b
+#define SNDRV_GF1_GB_MPU401_CONTROL_A 0x5c
+#define SNDRV_GF1_GB_MPU401_CONTROL_B 0x5d
+#define SNDRV_GF1_GB_EMULATION_IRQ 0x60
+/* voice specific registers */
+#define SNDRV_GF1_VB_ADDRESS_CONTROL 0x00
+#define SNDRV_GF1_VW_FREQUENCY 0x01
+#define SNDRV_GF1_VW_START_HIGH 0x02
+#define SNDRV_GF1_VW_START_LOW 0x03
+#define SNDRV_GF1_VA_START SNDRV_GF1_VW_START_HIGH
+#define SNDRV_GF1_VW_END_HIGH 0x04
+#define SNDRV_GF1_VW_END_LOW 0x05
+#define SNDRV_GF1_VA_END SNDRV_GF1_VW_END_HIGH
+#define SNDRV_GF1_VB_VOLUME_RATE 0x06
+#define SNDRV_GF1_VB_VOLUME_START 0x07
+#define SNDRV_GF1_VB_VOLUME_END 0x08
+#define SNDRV_GF1_VW_VOLUME 0x09
+#define SNDRV_GF1_VW_CURRENT_HIGH 0x0a
+#define SNDRV_GF1_VW_CURRENT_LOW 0x0b
+#define SNDRV_GF1_VA_CURRENT SNDRV_GF1_VW_CURRENT_HIGH
+#define SNDRV_GF1_VB_PAN 0x0c
+#define SNDRV_GF1_VW_OFFSET_RIGHT 0x0c
+#define SNDRV_GF1_VB_VOLUME_CONTROL 0x0d
+#define SNDRV_GF1_VB_UPPER_ADDRESS 0x10
+#define SNDRV_GF1_VW_EFFECT_HIGH 0x11
+#define SNDRV_GF1_VW_EFFECT_LOW 0x12
+#define SNDRV_GF1_VA_EFFECT SNDRV_GF1_VW_EFFECT_HIGH
+#define SNDRV_GF1_VW_OFFSET_LEFT 0x13
+#define SNDRV_GF1_VB_ACCUMULATOR 0x14
+#define SNDRV_GF1_VB_MODE 0x15
+#define SNDRV_GF1_VW_EFFECT_VOLUME 0x16
+#define SNDRV_GF1_VB_FREQUENCY_LFO 0x17
+#define SNDRV_GF1_VB_VOLUME_LFO 0x18
+#define SNDRV_GF1_VW_OFFSET_RIGHT_FINAL 0x1b
+#define SNDRV_GF1_VW_OFFSET_LEFT_FINAL 0x1c
+#define SNDRV_GF1_VW_EFFECT_VOLUME_FINAL 0x1d
+
+/* ICS registers */
+
+#define SNDRV_ICS_MIC_DEV 0
+#define SNDRV_ICS_LINE_DEV 1
+#define SNDRV_ICS_CD_DEV 2
+#define SNDRV_ICS_GF1_DEV 3
+#define SNDRV_ICS_NONE_DEV 4
+#define SNDRV_ICS_MASTER_DEV 5
+
+/* LFO */
+
+#define SNDRV_LFO_TREMOLO 0
+#define SNDRV_LFO_VIBRATO 1
+
+/* misc */
+
+#define SNDRV_GF1_DMA_UNSIGNED 0x80
+#define SNDRV_GF1_DMA_16BIT 0x40
+#define SNDRV_GF1_DMA_IRQ 0x20
+#define SNDRV_GF1_DMA_WIDTH16 0x04
+#define SNDRV_GF1_DMA_READ 0x02 /* read from GUS's DRAM */
+#define SNDRV_GF1_DMA_ENABLE 0x01
+
+/* ramp ranges */
+
+#define SNDRV_GF1_ATTEN(x) (snd_gf1_atten_table[x])
+#define SNDRV_GF1_MIN_VOLUME 1800
+#define SNDRV_GF1_MAX_VOLUME 4095
+#define SNDRV_GF1_MIN_OFFSET (SNDRV_GF1_MIN_VOLUME>>4)
+#define SNDRV_GF1_MAX_OFFSET 255
+#define SNDRV_GF1_MAX_TDEPTH 90
+
+/* defines for memory manager */
+
+#define SNDRV_GF1_MEM_BLOCK_16BIT 0x0001
+
+#define SNDRV_GF1_MEM_OWNER_DRIVER 0x0001
+#define SNDRV_GF1_MEM_OWNER_WAVE_SIMPLE 0x0002
+#define SNDRV_GF1_MEM_OWNER_WAVE_GF1 0x0003
+#define SNDRV_GF1_MEM_OWNER_WAVE_IWFFFF 0x0004
+
+/* constants for interrupt handlers */
+
+#define SNDRV_GF1_HANDLER_MIDI_OUT 0x00010000
+#define SNDRV_GF1_HANDLER_MIDI_IN 0x00020000
+#define SNDRV_GF1_HANDLER_TIMER1 0x00040000
+#define SNDRV_GF1_HANDLER_TIMER2 0x00080000
+#define SNDRV_GF1_HANDLER_VOICE 0x00100000
+#define SNDRV_GF1_HANDLER_DMA_WRITE 0x00200000
+#define SNDRV_GF1_HANDLER_DMA_READ 0x00400000
+#define SNDRV_GF1_HANDLER_ALL (0xffff0000&~SNDRV_GF1_HANDLER_VOICE)
+
+/* constants for DMA flags */
+
+#define SNDRV_GF1_DMA_TRIGGER 1
+
+/* --- */
+
+struct snd_gus_card;
+
+/* GF1 specific structure */
+
+struct snd_gf1_bank_info {
+ unsigned int address;
+ unsigned int size;
+};
+
+struct snd_gf1_mem_block {
+ unsigned short flags; /* flags - SNDRV_GF1_MEM_BLOCK_XXXX */
+ unsigned short owner; /* owner - SNDRV_GF1_MEM_OWNER_XXXX */
+ unsigned int share; /* share count */
+ unsigned int share_id[4]; /* share ID */
+ unsigned int ptr;
+ unsigned int size;
+ char *name;
+ struct snd_gf1_mem_block *next;
+ struct snd_gf1_mem_block *prev;
+};
+
+struct snd_gf1_mem {
+ struct snd_gf1_bank_info banks_8[4];
+ struct snd_gf1_bank_info banks_16[4];
+ struct snd_gf1_mem_block *first;
+ struct snd_gf1_mem_block *last;
+ struct mutex memory_mutex;
+};
+
+struct snd_gf1_dma_block {
+ void *buffer; /* buffer in computer's RAM */
+ unsigned long buf_addr; /* buffer address */
+ unsigned int addr; /* address in onboard memory */
+ unsigned int count; /* count in bytes */
+ unsigned int cmd; /* DMA command (format) */
+ void (*ack)(struct snd_gus_card * gus, void *private_data);
+ void *private_data;
+ struct snd_gf1_dma_block *next;
+};
+
+struct snd_gus_port {
+ struct snd_midi_channel_set * chset;
+ struct snd_gus_card * gus;
+ int mode; /* operation mode */
+ int client; /* sequencer client number */
+ int port; /* sequencer port number */
+ unsigned int midi_has_voices: 1;
+};
+
+struct snd_gus_voice;
+
+#define SNDRV_GF1_VOICE_TYPE_PCM 0
+#define SNDRV_GF1_VOICE_TYPE_SYNTH 1
+#define SNDRV_GF1_VOICE_TYPE_MIDI 2
+
+#define SNDRV_GF1_VFLG_RUNNING (1<<0)
+#define SNDRV_GF1_VFLG_EFFECT_TIMER1 (1<<1)
+#define SNDRV_GF1_VFLG_PAN (1<<2)
+
+enum snd_gus_volume_state {
+ VENV_BEFORE,
+ VENV_ATTACK,
+ VENV_SUSTAIN,
+ VENV_RELEASE,
+ VENV_DONE,
+ VENV_VOLUME
+};
+
+struct snd_gus_voice {
+ int number;
+ unsigned int use: 1,
+ pcm: 1,
+ synth:1,
+ midi: 1;
+ unsigned int flags;
+ unsigned char client;
+ unsigned char port;
+ unsigned char index;
+ unsigned char pad;
+
+#ifdef CONFIG_SND_DEBUG
+ unsigned int interrupt_stat_wave;
+ unsigned int interrupt_stat_volume;
+#endif
+ void (*handler_wave) (struct snd_gus_card * gus, struct snd_gus_voice * voice);
+ void (*handler_volume) (struct snd_gus_card * gus, struct snd_gus_voice * voice);
+ void (*handler_effect) (struct snd_gus_card * gus, struct snd_gus_voice * voice);
+ void (*volume_change) (struct snd_gus_card * gus);
+
+ struct snd_gus_sample_ops *sample_ops;
+
+ /* running status / registers */
+
+ unsigned short fc_register;
+ unsigned short fc_lfo;
+ unsigned short gf1_volume;
+ unsigned char control;
+ unsigned char mode;
+ unsigned char gf1_pan;
+ unsigned char effect_accumulator;
+ unsigned char volume_control;
+ unsigned char venv_value_next;
+ enum snd_gus_volume_state venv_state;
+ enum snd_gus_volume_state venv_state_prev;
+ unsigned short vlo;
+ unsigned short vro;
+ unsigned short gf1_effect_volume;
+
+ /* --- */
+
+ void *private_data;
+ void (*private_free)(struct snd_gus_voice *voice);
+};
+
+struct snd_gf1 {
+
+ unsigned int enh_mode:1, /* enhanced mode (GFA1) */
+ hw_lfo:1, /* use hardware LFO */
+ sw_lfo:1, /* use software LFO */
+ effect:1; /* use effect voices */
+
+ unsigned long port; /* port of GF1 chip */
+ struct resource *res_port1;
+ struct resource *res_port2;
+ int irq; /* IRQ number */
+ int dma1; /* DMA1 number */
+ int dma2; /* DMA2 number */
+ unsigned int memory; /* GUS's DRAM size in bytes */
+ unsigned int rom_memory; /* GUS's ROM size in bytes */
+ unsigned int rom_present; /* bitmask */
+ unsigned int rom_banks; /* GUS's ROM banks */
+
+ struct snd_gf1_mem mem_alloc;
+
+ /* registers */
+ unsigned short reg_page;
+ unsigned short reg_regsel;
+ unsigned short reg_data8;
+ unsigned short reg_data16;
+ unsigned short reg_irqstat;
+ unsigned short reg_dram;
+ unsigned short reg_timerctrl;
+ unsigned short reg_timerdata;
+ unsigned char ics_regs[6][2];
+ /* --------- */
+
+ unsigned char active_voices; /* active voices */
+ unsigned char active_voice; /* selected voice (GF1PAGE register) */
+
+ struct snd_gus_voice voices[32]; /* GF1 voices */
+
+ unsigned int default_voice_address;
+
+ unsigned short playback_freq; /* GF1 playback (mixing) frequency */
+ unsigned short mode; /* see to SNDRV_GF1_MODE_XXXX */
+ unsigned char volume_ramp;
+ unsigned char smooth_pan;
+ unsigned char full_range_pan;
+ unsigned char pad0;
+
+ unsigned char *lfos;
+
+ /* interrupt handlers */
+
+ void (*interrupt_handler_midi_out) (struct snd_gus_card * gus);
+ void (*interrupt_handler_midi_in) (struct snd_gus_card * gus);
+ void (*interrupt_handler_timer1) (struct snd_gus_card * gus);
+ void (*interrupt_handler_timer2) (struct snd_gus_card * gus);
+ void (*interrupt_handler_dma_write) (struct snd_gus_card * gus);
+ void (*interrupt_handler_dma_read) (struct snd_gus_card * gus);
+
+#ifdef CONFIG_SND_DEBUG
+ unsigned int interrupt_stat_midi_out;
+ unsigned int interrupt_stat_midi_in;
+ unsigned int interrupt_stat_timer1;
+ unsigned int interrupt_stat_timer2;
+ unsigned int interrupt_stat_dma_write;
+ unsigned int interrupt_stat_dma_read;
+ unsigned int interrupt_stat_voice_lost;
+#endif
+
+ /* synthesizer */
+
+ int seq_client;
+ struct snd_gus_port seq_ports[4];
+
+ /* timer */
+
+ unsigned short timer_enabled;
+ struct snd_timer *timer1;
+ struct snd_timer *timer2;
+
+ /* midi */
+
+ unsigned short uart_cmd;
+ unsigned int uart_framing;
+ unsigned int uart_overrun;
+
+ /* dma operations */
+
+ unsigned int dma_flags;
+ unsigned int dma_shared;
+ struct snd_gf1_dma_block *dma_data_pcm;
+ struct snd_gf1_dma_block *dma_data_pcm_last;
+ struct snd_gf1_dma_block *dma_data_synth;
+ struct snd_gf1_dma_block *dma_data_synth_last;
+ void (*dma_ack)(struct snd_gus_card * gus, void *private_data);
+ void *dma_private_data;
+
+ /* pcm */
+ int pcm_channels;
+ int pcm_alloc_voices;
+ unsigned short pcm_volume_level_left;
+ unsigned short pcm_volume_level_right;
+ unsigned short pcm_volume_level_left1;
+ unsigned short pcm_volume_level_right1;
+
+ unsigned char pcm_rcntrl_reg;
+ unsigned char pad_end;
+};
+
+/* main structure for GUS card */
+
+struct snd_gus_card {
+ struct snd_card *card;
+
+ unsigned int
+ initialized: 1, /* resources were initialized */
+ equal_irq:1, /* GF1 and CODEC shares IRQ (GUS MAX only) */
+ equal_dma:1, /* if dma channels are equal (not valid for daughter board) */
+ ics_flag:1, /* have we ICS mixer chip */
+ ics_flipped:1, /* ICS mixer have flipped some channels? */
+ codec_flag:1, /* have we CODEC chip? */
+ max_flag:1, /* have we GUS MAX card? */
+ max_ctrl_flag:1, /* have we original GUS MAX card? */
+ daughter_flag:1, /* have we daughter board? */
+ interwave:1, /* hey - we have InterWave card */
+ ess_flag:1, /* ESS chip found... GUS Extreme */
+ ace_flag:1, /* GUS ACE detected */
+ uart_enable:1; /* enable MIDI UART */
+ unsigned short revision; /* revision of chip */
+ unsigned short max_cntrl_val; /* GUS MAX control value */
+ unsigned short mix_cntrl_reg; /* mixer control register */
+ unsigned short joystick_dac; /* joystick DAC level */
+ int timer_dev; /* timer device */
+
+ struct snd_gf1 gf1; /* gf1 specific variables */
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *pcm_cap_substream;
+ unsigned int c_dma_size;
+ unsigned int c_period_size;
+ unsigned int c_pos;
+
+ struct snd_rawmidi *midi_uart;
+ struct snd_rawmidi_substream *midi_substream_output;
+ struct snd_rawmidi_substream *midi_substream_input;
+
+ spinlock_t reg_lock;
+ spinlock_t voice_alloc;
+ spinlock_t active_voice_lock;
+ spinlock_t event_lock;
+ spinlock_t dma_lock;
+ spinlock_t pcm_volume_level_lock;
+ spinlock_t uart_cmd_lock;
+ struct mutex dma_mutex;
+ struct mutex register_mutex;
+};
+
+/* I/O functions for GF1/InterWave chip - gus_io.c */
+
+static inline void snd_gf1_select_voice(struct snd_gus_card * gus, int voice)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->active_voice_lock, flags);
+ if (voice != gus->gf1.active_voice) {
+ gus->gf1.active_voice = voice;
+ outb(voice, GUSP(gus, GF1PAGE));
+ }
+ spin_unlock_irqrestore(&gus->active_voice_lock, flags);
+}
+
+static inline void snd_gf1_uart_cmd(struct snd_gus_card * gus, unsigned char b)
+{
+ outb(gus->gf1.uart_cmd = b, GUSP(gus, MIDICTRL));
+}
+
+static inline unsigned char snd_gf1_uart_stat(struct snd_gus_card * gus)
+{
+ return inb(GUSP(gus, MIDISTAT));
+}
+
+static inline void snd_gf1_uart_put(struct snd_gus_card * gus, unsigned char b)
+{
+ outb(b, GUSP(gus, MIDIDATA));
+}
+
+static inline unsigned char snd_gf1_uart_get(struct snd_gus_card * gus)
+{
+ return inb(GUSP(gus, MIDIDATA));
+}
+
+extern void snd_gf1_delay(struct snd_gus_card * gus);
+
+extern void snd_gf1_ctrl_stop(struct snd_gus_card * gus, unsigned char reg);
+
+extern void snd_gf1_write8(struct snd_gus_card * gus, unsigned char reg, unsigned char data);
+extern unsigned char snd_gf1_look8(struct snd_gus_card * gus, unsigned char reg);
+static inline unsigned char snd_gf1_read8(struct snd_gus_card * gus, unsigned char reg)
+{
+ return snd_gf1_look8(gus, reg | 0x80);
+}
+extern void snd_gf1_write16(struct snd_gus_card * gus, unsigned char reg, unsigned int data);
+extern unsigned short snd_gf1_look16(struct snd_gus_card * gus, unsigned char reg);
+static inline unsigned short snd_gf1_read16(struct snd_gus_card * gus, unsigned char reg)
+{
+ return snd_gf1_look16(gus, reg | 0x80);
+}
+extern void snd_gf1_adlib_write(struct snd_gus_card * gus, unsigned char reg, unsigned char data);
+extern void snd_gf1_dram_addr(struct snd_gus_card * gus, unsigned int addr);
+extern void snd_gf1_poke(struct snd_gus_card * gus, unsigned int addr, unsigned char data);
+extern unsigned char snd_gf1_peek(struct snd_gus_card * gus, unsigned int addr);
+extern void snd_gf1_write_addr(struct snd_gus_card * gus, unsigned char reg, unsigned int addr, short w_16bit);
+extern unsigned int snd_gf1_read_addr(struct snd_gus_card * gus, unsigned char reg, short w_16bit);
+extern void snd_gf1_i_ctrl_stop(struct snd_gus_card * gus, unsigned char reg);
+extern void snd_gf1_i_write8(struct snd_gus_card * gus, unsigned char reg, unsigned char data);
+extern unsigned char snd_gf1_i_look8(struct snd_gus_card * gus, unsigned char reg);
+extern void snd_gf1_i_write16(struct snd_gus_card * gus, unsigned char reg, unsigned int data);
+static inline unsigned char snd_gf1_i_read8(struct snd_gus_card * gus, unsigned char reg)
+{
+ return snd_gf1_i_look8(gus, reg | 0x80);
+}
+extern unsigned short snd_gf1_i_look16(struct snd_gus_card * gus, unsigned char reg);
+static inline unsigned short snd_gf1_i_read16(struct snd_gus_card * gus, unsigned char reg)
+{
+ return snd_gf1_i_look16(gus, reg | 0x80);
+}
+
+extern void snd_gf1_select_active_voices(struct snd_gus_card * gus);
+
+/* gus_lfo.c */
+
+struct _SND_IW_LFO_PROGRAM {
+ unsigned short freq_and_control;
+ unsigned char depth_final;
+ unsigned char depth_inc;
+ unsigned short twave;
+ unsigned short depth;
+};
+
+#if 0
+extern irqreturn_t snd_gf1_lfo_effect_interrupt(struct snd_gus_card * gus, snd_gf1_voice_t * voice);
+#endif
+extern void snd_gf1_lfo_init(struct snd_gus_card * gus);
+extern void snd_gf1_lfo_done(struct snd_gus_card * gus);
+extern void snd_gf1_lfo_program(struct snd_gus_card * gus, int voice, int lfo_type, struct _SND_IW_LFO_PROGRAM *program);
+extern void snd_gf1_lfo_enable(struct snd_gus_card * gus, int voice, int lfo_type);
+extern void snd_gf1_lfo_disable(struct snd_gus_card * gus, int voice, int lfo_type);
+extern void snd_gf1_lfo_change_freq(struct snd_gus_card * gus, int voice, int lfo_type, int freq);
+extern void snd_gf1_lfo_change_depth(struct snd_gus_card * gus, int voice, int lfo_type, int depth);
+extern void snd_gf1_lfo_setup(struct snd_gus_card * gus, int voice, int lfo_type, int freq, int current_depth, int depth, int sweep, int shape);
+extern void snd_gf1_lfo_shutdown(struct snd_gus_card * gus, int voice, int lfo_type);
+#if 0
+extern void snd_gf1_lfo_command(struct snd_gus_card * gus, int voice, unsigned char *command);
+#endif
+
+/* gus_mem.c */
+
+void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup);
+int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * block);
+struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owner,
+ char *name, int size, int w_16,
+ int align, unsigned int *share_id);
+int snd_gf1_mem_free(struct snd_gf1_mem * alloc, unsigned int address);
+int snd_gf1_mem_free_owner(struct snd_gf1_mem * alloc, int owner);
+int snd_gf1_mem_init(struct snd_gus_card * gus);
+int snd_gf1_mem_done(struct snd_gus_card * gus);
+
+/* gus_mem_proc.c */
+
+int snd_gf1_mem_proc_init(struct snd_gus_card * gus);
+
+/* gus_dma.c */
+
+int snd_gf1_dma_init(struct snd_gus_card * gus);
+int snd_gf1_dma_done(struct snd_gus_card * gus);
+int snd_gf1_dma_transfer_block(struct snd_gus_card * gus,
+ struct snd_gf1_dma_block * block,
+ int atomic,
+ int synth);
+
+/* gus_volume.c */
+
+unsigned short snd_gf1_lvol_to_gvol_raw(unsigned int vol);
+unsigned short snd_gf1_translate_freq(struct snd_gus_card * gus, unsigned int freq2);
+
+/* gus_reset.c */
+
+void snd_gf1_set_default_handlers(struct snd_gus_card * gus, unsigned int what);
+void snd_gf1_smart_stop_voice(struct snd_gus_card * gus, unsigned short voice);
+void snd_gf1_stop_voice(struct snd_gus_card * gus, unsigned short voice);
+void snd_gf1_stop_voices(struct snd_gus_card * gus, unsigned short v_min, unsigned short v_max);
+struct snd_gus_voice *snd_gf1_alloc_voice(struct snd_gus_card * gus, int type, int client, int port);
+void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice);
+int snd_gf1_start(struct snd_gus_card * gus);
+int snd_gf1_stop(struct snd_gus_card * gus);
+
+/* gus_mixer.c */
+
+int snd_gf1_new_mixer(struct snd_gus_card * gus);
+
+/* gus_pcm.c */
+
+int snd_gf1_pcm_new(struct snd_gus_card *gus, int pcm_dev, int control_index);
+
+#ifdef CONFIG_SND_DEBUG
+extern void snd_gf1_print_voice_registers(struct snd_gus_card * gus);
+#endif
+
+/* gus.c */
+
+int snd_gus_use_inc(struct snd_gus_card * gus);
+void snd_gus_use_dec(struct snd_gus_card * gus);
+int snd_gus_create(struct snd_card *card,
+ unsigned long port,
+ int irq, int dma1, int dma2,
+ int timer_dev,
+ int voices,
+ int pcm_channels,
+ int effect,
+ struct snd_gus_card ** rgus);
+int snd_gus_initialize(struct snd_gus_card * gus);
+
+/* gus_irq.c */
+
+irqreturn_t snd_gus_interrupt(int irq, void *dev_id);
+#ifdef CONFIG_SND_DEBUG
+void snd_gus_irq_profile_init(struct snd_gus_card *gus);
+#endif
+
+/* gus_uart.c */
+
+int snd_gf1_rawmidi_new(struct snd_gus_card *gus, int device);
+
+/* gus_dram.c */
+int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr,
+ unsigned int addr, unsigned int size);
+int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr,
+ unsigned int addr, unsigned int size, int rom);
+
+#endif /* __SOUND_GUS_H */
diff --git a/kernel/include/sound/hda_hwdep.h b/kernel/include/sound/hda_hwdep.h
new file mode 100644
index 000000000..1c0034e87
--- /dev/null
+++ b/kernel/include/sound/hda_hwdep.h
@@ -0,0 +1,44 @@
+/*
+ * HWDEP Interface for HD-audio codec
+ *
+ * Copyright (c) 2007 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_HDA_HWDEP_H
+#define __SOUND_HDA_HWDEP_H
+
+#define HDA_HWDEP_VERSION ((1 << 16) | (0 << 8) | (0 << 0)) /* 1.0.0 */
+
+/* verb */
+#define HDA_REG_NID_SHIFT 24
+#define HDA_REG_VERB_SHIFT 8
+#define HDA_REG_VAL_SHIFT 0
+#define HDA_VERB(nid,verb,param) ((nid)<<24 | (verb)<<8 | (param))
+
+struct hda_verb_ioctl {
+ u32 verb; /* HDA_VERB() */
+ u32 res; /* response */
+};
+
+/*
+ * ioctls
+ */
+#define HDA_IOCTL_PVERSION _IOR('H', 0x10, int)
+#define HDA_IOCTL_VERB_WRITE _IOWR('H', 0x11, struct hda_verb_ioctl)
+#define HDA_IOCTL_GET_WCAP _IOWR('H', 0x12, struct hda_verb_ioctl)
+
+#endif
diff --git a/kernel/include/sound/hda_regmap.h b/kernel/include/sound/hda_regmap.h
new file mode 100644
index 000000000..df7059084
--- /dev/null
+++ b/kernel/include/sound/hda_regmap.h
@@ -0,0 +1,219 @@
+/*
+ * HD-audio regmap helpers
+ */
+
+#ifndef __SOUND_HDA_REGMAP_H
+#define __SOUND_HDA_REGMAP_H
+
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/hdaudio.h>
+
+#define AC_AMP_FAKE_MUTE 0x10 /* fake mute bit set to amp verbs */
+
+int snd_hdac_regmap_init(struct hdac_device *codec);
+void snd_hdac_regmap_exit(struct hdac_device *codec);
+int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec,
+ unsigned int verb);
+int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg,
+ unsigned int *val);
+int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg,
+ unsigned int val);
+int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg,
+ unsigned int mask, unsigned int val);
+
+/**
+ * snd_hdac_regmap_encode_verb - encode the verb to a pseudo register
+ * @nid: widget NID
+ * @verb: codec verb
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_verb(nid, verb) \
+ (((verb) << 8) | 0x80000 | ((unsigned int)(nid) << 20))
+
+/**
+ * snd_hdac_regmap_encode_amp - encode the AMP verb to a pseudo register
+ * @nid: widget NID
+ * @ch: channel (left = 0, right = 1)
+ * @dir: direction (#HDA_INPUT, #HDA_OUTPUT)
+ * @idx: input index value
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_amp(nid, ch, dir, idx) \
+ (snd_hdac_regmap_encode_verb(nid, AC_VERB_GET_AMP_GAIN_MUTE) | \
+ ((ch) ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT) | \
+ ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \
+ (idx))
+
+/**
+ * snd_hdac_regmap_encode_amp_stereo - encode a pseudo register for stereo AMPs
+ * @nid: widget NID
+ * @dir: direction (#HDA_INPUT, #HDA_OUTPUT)
+ * @idx: input index value
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_amp_stereo(nid, dir, idx) \
+ (snd_hdac_regmap_encode_verb(nid, AC_VERB_GET_AMP_GAIN_MUTE) | \
+ AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT | /* both bits set! */ \
+ ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \
+ (idx))
+
+/**
+ * snd_hdac_regmap_write - Write a verb with caching
+ * @nid: codec NID
+ * @reg: verb to write
+ * @val: value to write
+ *
+ * For writing an amp value, use snd_hda_regmap_amp_update().
+ */
+static inline int
+snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid,
+ unsigned int verb, unsigned int val)
+{
+ unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+ return snd_hdac_regmap_write_raw(codec, cmd, val);
+}
+
+/**
+ * snd_hda_regmap_update - Update a verb value with caching
+ * @nid: codec NID
+ * @verb: verb to update
+ * @mask: bit mask to update
+ * @val: value to update
+ *
+ * For updating an amp value, use snd_hda_regmap_amp_update().
+ */
+static inline int
+snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid,
+ unsigned int verb, unsigned int mask,
+ unsigned int val)
+{
+ unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+ return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hda_regmap_read - Read a verb with caching
+ * @nid: codec NID
+ * @verb: verb to read
+ * @val: pointer to store the value
+ *
+ * For reading an amp value, use snd_hda_regmap_get_amp().
+ */
+static inline int
+snd_hdac_regmap_read(struct hdac_device *codec, hda_nid_t nid,
+ unsigned int verb, unsigned int *val)
+{
+ unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+ return snd_hdac_regmap_read_raw(codec, cmd, val);
+}
+
+/**
+ * snd_hdac_regmap_get_amp - Read AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @index: the index value (only for input direction)
+ * @val: the pointer to store the value
+ *
+ * Read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit.
+ * Returns the value or a negative error.
+ */
+static inline int
+snd_hdac_regmap_get_amp(struct hdac_device *codec, hda_nid_t nid,
+ int ch, int dir, int idx)
+{
+ unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx);
+ int err, val;
+
+ err = snd_hdac_regmap_read_raw(codec, cmd, &val);
+ return err < 0 ? err : val;
+}
+
+/**
+ * snd_hdac_regmap_update_amp - update the AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Update the AMP value with a bit mask.
+ * Returns 0 if the value is unchanged, 1 if changed, or a negative error.
+ */
+static inline int
+snd_hdac_regmap_update_amp(struct hdac_device *codec, hda_nid_t nid,
+ int ch, int dir, int idx, int mask, int val)
+{
+ unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx);
+
+ return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hdac_regmap_get_amp_stereo - Read stereo AMP values
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @index: the index value (only for input direction)
+ * @val: the pointer to store the value
+ *
+ * Read stereo AMP values. The lower byte is left, the upper byte is right.
+ * Returns the value or a negative error.
+ */
+static inline int
+snd_hdac_regmap_get_amp_stereo(struct hdac_device *codec, hda_nid_t nid,
+ int dir, int idx)
+{
+ unsigned int cmd = snd_hdac_regmap_encode_amp_stereo(nid, dir, idx);
+ int err, val;
+
+ err = snd_hdac_regmap_read_raw(codec, cmd, &val);
+ return err < 0 ? err : val;
+}
+
+/**
+ * snd_hdac_regmap_update_amp_stereo - update the stereo AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Update the stereo AMP value with a bit mask.
+ * The lower byte is left, the upper byte is right.
+ * Returns 0 if the value is unchanged, 1 if changed, or a negative error.
+ */
+static inline int
+snd_hdac_regmap_update_amp_stereo(struct hdac_device *codec, hda_nid_t nid,
+ int dir, int idx, int mask, int val)
+{
+ unsigned int cmd = snd_hdac_regmap_encode_amp_stereo(nid, dir, idx);
+
+ return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hdac_regmap_sync_node - sync the widget node attributes
+ * @codec: HD-audio codec
+ * @nid: NID to sync
+ */
+static inline void
+snd_hdac_regmap_sync_node(struct hdac_device *codec, hda_nid_t nid)
+{
+ regcache_mark_dirty(codec->regmap);
+ regcache_sync_region(codec->regmap, nid << 20, ((nid + 1) << 20) - 1);
+}
+
+#endif /* __SOUND_HDA_REGMAP_H */
diff --git a/kernel/include/sound/hda_verbs.h b/kernel/include/sound/hda_verbs.h
new file mode 100644
index 000000000..d0509db6d
--- /dev/null
+++ b/kernel/include/sound/hda_verbs.h
@@ -0,0 +1,554 @@
+/*
+ * HD-audio codec verbs
+ */
+
+#ifndef __SOUND_HDA_VERBS_H
+#define __SOUND_HDA_VERBS_H
+
+/*
+ * nodes
+ */
+#define AC_NODE_ROOT 0x00
+
+/*
+ * function group types
+ */
+enum {
+ AC_GRP_AUDIO_FUNCTION = 0x01,
+ AC_GRP_MODEM_FUNCTION = 0x02,
+};
+
+/*
+ * widget types
+ */
+enum {
+ AC_WID_AUD_OUT, /* Audio Out */
+ AC_WID_AUD_IN, /* Audio In */
+ AC_WID_AUD_MIX, /* Audio Mixer */
+ AC_WID_AUD_SEL, /* Audio Selector */
+ AC_WID_PIN, /* Pin Complex */
+ AC_WID_POWER, /* Power */
+ AC_WID_VOL_KNB, /* Volume Knob */
+ AC_WID_BEEP, /* Beep Generator */
+ AC_WID_VENDOR = 0x0f /* Vendor specific */
+};
+
+/*
+ * GET verbs
+ */
+#define AC_VERB_GET_STREAM_FORMAT 0x0a00
+#define AC_VERB_GET_AMP_GAIN_MUTE 0x0b00
+#define AC_VERB_GET_PROC_COEF 0x0c00
+#define AC_VERB_GET_COEF_INDEX 0x0d00
+#define AC_VERB_PARAMETERS 0x0f00
+#define AC_VERB_GET_CONNECT_SEL 0x0f01
+#define AC_VERB_GET_CONNECT_LIST 0x0f02
+#define AC_VERB_GET_PROC_STATE 0x0f03
+#define AC_VERB_GET_SDI_SELECT 0x0f04
+#define AC_VERB_GET_POWER_STATE 0x0f05
+#define AC_VERB_GET_CONV 0x0f06
+#define AC_VERB_GET_PIN_WIDGET_CONTROL 0x0f07
+#define AC_VERB_GET_UNSOLICITED_RESPONSE 0x0f08
+#define AC_VERB_GET_PIN_SENSE 0x0f09
+#define AC_VERB_GET_BEEP_CONTROL 0x0f0a
+#define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c
+#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d
+#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */
+#define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f
+/* f10-f1a: GPIO */
+#define AC_VERB_GET_GPIO_DATA 0x0f15
+#define AC_VERB_GET_GPIO_MASK 0x0f16
+#define AC_VERB_GET_GPIO_DIRECTION 0x0f17
+#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18
+#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19
+#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a
+#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c
+/* f20: AFG/MFG */
+#define AC_VERB_GET_SUBSYSTEM_ID 0x0f20
+#define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d
+#define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e
+#define AC_VERB_GET_HDMI_ELDD 0x0f2f
+#define AC_VERB_GET_HDMI_DIP_INDEX 0x0f30
+#define AC_VERB_GET_HDMI_DIP_DATA 0x0f31
+#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32
+#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33
+#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34
+#define AC_VERB_GET_DEVICE_SEL 0xf35
+#define AC_VERB_GET_DEVICE_LIST 0xf36
+
+/*
+ * SET verbs
+ */
+#define AC_VERB_SET_STREAM_FORMAT 0x200
+#define AC_VERB_SET_AMP_GAIN_MUTE 0x300
+#define AC_VERB_SET_PROC_COEF 0x400
+#define AC_VERB_SET_COEF_INDEX 0x500
+#define AC_VERB_SET_CONNECT_SEL 0x701
+#define AC_VERB_SET_PROC_STATE 0x703
+#define AC_VERB_SET_SDI_SELECT 0x704
+#define AC_VERB_SET_POWER_STATE 0x705
+#define AC_VERB_SET_CHANNEL_STREAMID 0x706
+#define AC_VERB_SET_PIN_WIDGET_CONTROL 0x707
+#define AC_VERB_SET_UNSOLICITED_ENABLE 0x708
+#define AC_VERB_SET_PIN_SENSE 0x709
+#define AC_VERB_SET_BEEP_CONTROL 0x70a
+#define AC_VERB_SET_EAPD_BTLENABLE 0x70c
+#define AC_VERB_SET_DIGI_CONVERT_1 0x70d
+#define AC_VERB_SET_DIGI_CONVERT_2 0x70e
+#define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f
+#define AC_VERB_SET_GPIO_DATA 0x715
+#define AC_VERB_SET_GPIO_MASK 0x716
+#define AC_VERB_SET_GPIO_DIRECTION 0x717
+#define AC_VERB_SET_GPIO_WAKE_MASK 0x718
+#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719
+#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f
+#define AC_VERB_SET_EAPD 0x788
+#define AC_VERB_SET_CODEC_RESET 0x7ff
+#define AC_VERB_SET_CVT_CHAN_COUNT 0x72d
+#define AC_VERB_SET_HDMI_DIP_INDEX 0x730
+#define AC_VERB_SET_HDMI_DIP_DATA 0x731
+#define AC_VERB_SET_HDMI_DIP_XMIT 0x732
+#define AC_VERB_SET_HDMI_CP_CTRL 0x733
+#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734
+#define AC_VERB_SET_DEVICE_SEL 0x735
+
+/*
+ * Parameter IDs
+ */
+#define AC_PAR_VENDOR_ID 0x00
+#define AC_PAR_SUBSYSTEM_ID 0x01
+#define AC_PAR_REV_ID 0x02
+#define AC_PAR_NODE_COUNT 0x04
+#define AC_PAR_FUNCTION_TYPE 0x05
+#define AC_PAR_AUDIO_FG_CAP 0x08
+#define AC_PAR_AUDIO_WIDGET_CAP 0x09
+#define AC_PAR_PCM 0x0a
+#define AC_PAR_STREAM 0x0b
+#define AC_PAR_PIN_CAP 0x0c
+#define AC_PAR_AMP_IN_CAP 0x0d
+#define AC_PAR_CONNLIST_LEN 0x0e
+#define AC_PAR_POWER_STATE 0x0f
+#define AC_PAR_PROC_CAP 0x10
+#define AC_PAR_GPIO_CAP 0x11
+#define AC_PAR_AMP_OUT_CAP 0x12
+#define AC_PAR_VOL_KNB_CAP 0x13
+#define AC_PAR_DEVLIST_LEN 0x15
+#define AC_PAR_HDMI_LPCM_CAP 0x20
+
+/*
+ * AC_VERB_PARAMETERS results (32bit)
+ */
+
+/* Function Group Type */
+#define AC_FGT_TYPE (0xff<<0)
+#define AC_FGT_TYPE_SHIFT 0
+#define AC_FGT_UNSOL_CAP (1<<8)
+
+/* Audio Function Group Capabilities */
+#define AC_AFG_OUT_DELAY (0xf<<0)
+#define AC_AFG_IN_DELAY (0xf<<8)
+#define AC_AFG_BEEP_GEN (1<<16)
+
+/* Audio Widget Capabilities */
+#define AC_WCAP_STEREO (1<<0) /* stereo I/O */
+#define AC_WCAP_IN_AMP (1<<1) /* AMP-in present */
+#define AC_WCAP_OUT_AMP (1<<2) /* AMP-out present */
+#define AC_WCAP_AMP_OVRD (1<<3) /* AMP-parameter override */
+#define AC_WCAP_FORMAT_OVRD (1<<4) /* format override */
+#define AC_WCAP_STRIPE (1<<5) /* stripe */
+#define AC_WCAP_PROC_WID (1<<6) /* Proc Widget */
+#define AC_WCAP_UNSOL_CAP (1<<7) /* Unsol capable */
+#define AC_WCAP_CONN_LIST (1<<8) /* connection list */
+#define AC_WCAP_DIGITAL (1<<9) /* digital I/O */
+#define AC_WCAP_POWER (1<<10) /* power control */
+#define AC_WCAP_LR_SWAP (1<<11) /* L/R swap */
+#define AC_WCAP_CP_CAPS (1<<12) /* content protection */
+#define AC_WCAP_CHAN_CNT_EXT (7<<13) /* channel count ext */
+#define AC_WCAP_DELAY (0xf<<16)
+#define AC_WCAP_DELAY_SHIFT 16
+#define AC_WCAP_TYPE (0xf<<20)
+#define AC_WCAP_TYPE_SHIFT 20
+
+/* supported PCM rates and bits */
+#define AC_SUPPCM_RATES (0xfff << 0)
+#define AC_SUPPCM_BITS_8 (1<<16)
+#define AC_SUPPCM_BITS_16 (1<<17)
+#define AC_SUPPCM_BITS_20 (1<<18)
+#define AC_SUPPCM_BITS_24 (1<<19)
+#define AC_SUPPCM_BITS_32 (1<<20)
+
+/* supported PCM stream format */
+#define AC_SUPFMT_PCM (1<<0)
+#define AC_SUPFMT_FLOAT32 (1<<1)
+#define AC_SUPFMT_AC3 (1<<2)
+
+/* GP I/O count */
+#define AC_GPIO_IO_COUNT (0xff<<0)
+#define AC_GPIO_O_COUNT (0xff<<8)
+#define AC_GPIO_O_COUNT_SHIFT 8
+#define AC_GPIO_I_COUNT (0xff<<16)
+#define AC_GPIO_I_COUNT_SHIFT 16
+#define AC_GPIO_UNSOLICITED (1<<30)
+#define AC_GPIO_WAKE (1<<31)
+
+/* Converter stream, channel */
+#define AC_CONV_CHANNEL (0xf<<0)
+#define AC_CONV_STREAM (0xf<<4)
+#define AC_CONV_STREAM_SHIFT 4
+
+/* Input converter SDI select */
+#define AC_SDI_SELECT (0xf<<0)
+
+/* stream format id */
+#define AC_FMT_CHAN_SHIFT 0
+#define AC_FMT_CHAN_MASK (0x0f << 0)
+#define AC_FMT_BITS_SHIFT 4
+#define AC_FMT_BITS_MASK (7 << 4)
+#define AC_FMT_BITS_8 (0 << 4)
+#define AC_FMT_BITS_16 (1 << 4)
+#define AC_FMT_BITS_20 (2 << 4)
+#define AC_FMT_BITS_24 (3 << 4)
+#define AC_FMT_BITS_32 (4 << 4)
+#define AC_FMT_DIV_SHIFT 8
+#define AC_FMT_DIV_MASK (7 << 8)
+#define AC_FMT_MULT_SHIFT 11
+#define AC_FMT_MULT_MASK (7 << 11)
+#define AC_FMT_BASE_SHIFT 14
+#define AC_FMT_BASE_48K (0 << 14)
+#define AC_FMT_BASE_44K (1 << 14)
+#define AC_FMT_TYPE_SHIFT 15
+#define AC_FMT_TYPE_PCM (0 << 15)
+#define AC_FMT_TYPE_NON_PCM (1 << 15)
+
+/* Unsolicited response control */
+#define AC_UNSOL_TAG (0x3f<<0)
+#define AC_UNSOL_ENABLED (1<<7)
+#define AC_USRSP_EN AC_UNSOL_ENABLED
+
+/* Unsolicited responses */
+#define AC_UNSOL_RES_TAG (0x3f<<26)
+#define AC_UNSOL_RES_TAG_SHIFT 26
+#define AC_UNSOL_RES_SUBTAG (0x1f<<21)
+#define AC_UNSOL_RES_SUBTAG_SHIFT 21
+#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry
+ * (for DP1.2 MST)
+ */
+#define AC_UNSOL_RES_DE_SHIFT 15
+#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */
+#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */
+#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */
+#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */
+#define AC_UNSOL_RES_CP_READY (1<<0) /* content protection */
+
+/* Pin widget capabilies */
+#define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */
+#define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */
+#define AC_PINCAP_PRES_DETECT (1<<2) /* presence detect capable */
+#define AC_PINCAP_HP_DRV (1<<3) /* headphone drive capable */
+#define AC_PINCAP_OUT (1<<4) /* output capable */
+#define AC_PINCAP_IN (1<<5) /* input capable */
+#define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */
+/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification,
+ * but is marked reserved in the Intel HDA specification.
+ */
+#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */
+/* Note: The same bit as LR_SWAP is newly defined as HDMI capability
+ * in HD-audio specification
+ */
+#define AC_PINCAP_HDMI (1<<7) /* HDMI pin */
+#define AC_PINCAP_DP (1<<24) /* DisplayPort pin, can
+ * coexist with AC_PINCAP_HDMI
+ */
+#define AC_PINCAP_VREF (0x37<<8)
+#define AC_PINCAP_VREF_SHIFT 8
+#define AC_PINCAP_EAPD (1<<16) /* EAPD capable */
+#define AC_PINCAP_HBR (1<<27) /* High Bit Rate */
+/* Vref status (used in pin cap) */
+#define AC_PINCAP_VREF_HIZ (1<<0) /* Hi-Z */
+#define AC_PINCAP_VREF_50 (1<<1) /* 50% */
+#define AC_PINCAP_VREF_GRD (1<<2) /* ground */
+#define AC_PINCAP_VREF_80 (1<<4) /* 80% */
+#define AC_PINCAP_VREF_100 (1<<5) /* 100% */
+
+/* Amplifier capabilities */
+#define AC_AMPCAP_OFFSET (0x7f<<0) /* 0dB offset */
+#define AC_AMPCAP_OFFSET_SHIFT 0
+#define AC_AMPCAP_NUM_STEPS (0x7f<<8) /* number of steps */
+#define AC_AMPCAP_NUM_STEPS_SHIFT 8
+#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB
+ * in 0.25dB
+ */
+#define AC_AMPCAP_STEP_SIZE_SHIFT 16
+#define AC_AMPCAP_MUTE (1<<31) /* mute capable */
+#define AC_AMPCAP_MUTE_SHIFT 31
+
+/* driver-specific amp-caps: using bits 24-30 */
+#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */
+
+/* Connection list */
+#define AC_CLIST_LENGTH (0x7f<<0)
+#define AC_CLIST_LONG (1<<7)
+
+/* Supported power status */
+#define AC_PWRST_D0SUP (1<<0)
+#define AC_PWRST_D1SUP (1<<1)
+#define AC_PWRST_D2SUP (1<<2)
+#define AC_PWRST_D3SUP (1<<3)
+#define AC_PWRST_D3COLDSUP (1<<4)
+#define AC_PWRST_S3D3COLDSUP (1<<29)
+#define AC_PWRST_CLKSTOP (1<<30)
+#define AC_PWRST_EPSS (1U<<31)
+
+/* Power state values */
+#define AC_PWRST_SETTING (0xf<<0)
+#define AC_PWRST_ACTUAL (0xf<<4)
+#define AC_PWRST_ACTUAL_SHIFT 4
+#define AC_PWRST_D0 0x00
+#define AC_PWRST_D1 0x01
+#define AC_PWRST_D2 0x02
+#define AC_PWRST_D3 0x03
+#define AC_PWRST_ERROR (1<<8)
+#define AC_PWRST_CLK_STOP_OK (1<<9)
+#define AC_PWRST_SETTING_RESET (1<<10)
+
+/* Processing capabilies */
+#define AC_PCAP_BENIGN (1<<0)
+#define AC_PCAP_NUM_COEF (0xff<<8)
+#define AC_PCAP_NUM_COEF_SHIFT 8
+
+/* Volume knobs capabilities */
+#define AC_KNBCAP_NUM_STEPS (0x7f<<0)
+#define AC_KNBCAP_DELTA (1<<7)
+
+/* HDMI LPCM capabilities */
+#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */
+#define AC_LPCMCAP_48K_NO_CHNS (0x0f<<4) /* max channels w/o CP-on */
+#define AC_LPCMCAP_48K_20BIT (1<<8) /* 20b bitrate supported */
+#define AC_LPCMCAP_48K_24BIT (1<<9) /* 24b bitrate supported */
+#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */
+#define AC_LPCMCAP_96K_NO_CHNS (0x0f<<14) /* max channels w/o CP-on */
+#define AC_LPCMCAP_96K_20BIT (1<<18) /* 20b bitrate supported */
+#define AC_LPCMCAP_96K_24BIT (1<<19) /* 24b bitrate supported */
+#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */
+#define AC_LPCMCAP_192K_NO_CHNS (0x0f<<24) /* max channels w/o CP-on */
+#define AC_LPCMCAP_192K_20BIT (1<<28) /* 20b bitrate supported */
+#define AC_LPCMCAP_192K_24BIT (1<<29) /* 24b bitrate supported */
+#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */
+#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */
+
+/* Display pin's device list length */
+#define AC_DEV_LIST_LEN_MASK 0x3f
+#define AC_MAX_DEV_LIST_LEN 64
+
+/*
+ * Control Parameters
+ */
+
+/* Amp gain/mute */
+#define AC_AMP_MUTE (1<<7)
+#define AC_AMP_GAIN (0x7f)
+#define AC_AMP_GET_INDEX (0xf<<0)
+
+#define AC_AMP_GET_LEFT (1<<13)
+#define AC_AMP_GET_RIGHT (0<<13)
+#define AC_AMP_GET_OUTPUT (1<<15)
+#define AC_AMP_GET_INPUT (0<<15)
+
+#define AC_AMP_SET_INDEX (0xf<<8)
+#define AC_AMP_SET_INDEX_SHIFT 8
+#define AC_AMP_SET_RIGHT (1<<12)
+#define AC_AMP_SET_LEFT (1<<13)
+#define AC_AMP_SET_INPUT (1<<14)
+#define AC_AMP_SET_OUTPUT (1<<15)
+
+/* DIGITAL1 bits */
+#define AC_DIG1_ENABLE (1<<0)
+#define AC_DIG1_V (1<<1)
+#define AC_DIG1_VCFG (1<<2)
+#define AC_DIG1_EMPHASIS (1<<3)
+#define AC_DIG1_COPYRIGHT (1<<4)
+#define AC_DIG1_NONAUDIO (1<<5)
+#define AC_DIG1_PROFESSIONAL (1<<6)
+#define AC_DIG1_LEVEL (1<<7)
+
+/* DIGITAL2 bits */
+#define AC_DIG2_CC (0x7f<<0)
+
+/* DIGITAL3 bits */
+#define AC_DIG3_ICT (0xf<<0)
+#define AC_DIG3_KAE (1<<7)
+
+/* Pin widget control - 8bit */
+#define AC_PINCTL_EPT (0x3<<0)
+#define AC_PINCTL_EPT_NATIVE 0
+#define AC_PINCTL_EPT_HBR 3
+#define AC_PINCTL_VREFEN (0x7<<0)
+#define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */
+#define AC_PINCTL_VREF_50 1 /* 50% */
+#define AC_PINCTL_VREF_GRD 2 /* ground */
+#define AC_PINCTL_VREF_80 4 /* 80% */
+#define AC_PINCTL_VREF_100 5 /* 100% */
+#define AC_PINCTL_IN_EN (1<<5)
+#define AC_PINCTL_OUT_EN (1<<6)
+#define AC_PINCTL_HP_EN (1<<7)
+
+/* Pin sense - 32bit */
+#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff)
+#define AC_PINSENSE_PRESENCE (1<<31)
+#define AC_PINSENSE_ELDV (1<<30) /* ELD valid (HDMI) */
+
+/* EAPD/BTL enable - 32bit */
+#define AC_EAPDBTL_BALANCED (1<<0)
+#define AC_EAPDBTL_EAPD (1<<1)
+#define AC_EAPDBTL_LR_SWAP (1<<2)
+
+/* HDMI ELD data */
+#define AC_ELDD_ELD_VALID (1<<31)
+#define AC_ELDD_ELD_DATA 0xff
+
+/* HDMI DIP size */
+#define AC_DIPSIZE_ELD_BUF (1<<3) /* ELD buf size of packet size */
+#define AC_DIPSIZE_PACK_IDX (0x07<<0) /* packet index */
+
+/* HDMI DIP index */
+#define AC_DIPIDX_PACK_IDX (0x07<<5) /* packet idnex */
+#define AC_DIPIDX_BYTE_IDX (0x1f<<0) /* byte index */
+
+/* HDMI DIP xmit (transmit) control */
+#define AC_DIPXMIT_MASK (0x3<<6)
+#define AC_DIPXMIT_DISABLE (0x0<<6) /* disable xmit */
+#define AC_DIPXMIT_ONCE (0x2<<6) /* xmit once then disable */
+#define AC_DIPXMIT_BEST (0x3<<6) /* best effort */
+
+/* HDMI content protection (CP) control */
+#define AC_CPCTRL_CES (1<<9) /* current encryption state */
+#define AC_CPCTRL_READY (1<<8) /* ready bit */
+#define AC_CPCTRL_SUBTAG (0x1f<<3) /* subtag for unsol-resp */
+#define AC_CPCTRL_STATE (3<<0) /* current CP request state */
+
+/* Converter channel <-> HDMI slot mapping */
+#define AC_CVTMAP_HDMI_SLOT (0xf<<0) /* HDMI slot number */
+#define AC_CVTMAP_CHAN (0xf<<4) /* converter channel number */
+
+/* configuration default - 32bit */
+#define AC_DEFCFG_SEQUENCE (0xf<<0)
+#define AC_DEFCFG_DEF_ASSOC (0xf<<4)
+#define AC_DEFCFG_ASSOC_SHIFT 4
+#define AC_DEFCFG_MISC (0xf<<8)
+#define AC_DEFCFG_MISC_SHIFT 8
+#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0)
+#define AC_DEFCFG_COLOR (0xf<<12)
+#define AC_DEFCFG_COLOR_SHIFT 12
+#define AC_DEFCFG_CONN_TYPE (0xf<<16)
+#define AC_DEFCFG_CONN_TYPE_SHIFT 16
+#define AC_DEFCFG_DEVICE (0xf<<20)
+#define AC_DEFCFG_DEVICE_SHIFT 20
+#define AC_DEFCFG_LOCATION (0x3f<<24)
+#define AC_DEFCFG_LOCATION_SHIFT 24
+#define AC_DEFCFG_PORT_CONN (0x3<<30)
+#define AC_DEFCFG_PORT_CONN_SHIFT 30
+
+/* Display pin's device list entry */
+#define AC_DE_PD (1<<0)
+#define AC_DE_ELDV (1<<1)
+#define AC_DE_IA (1<<2)
+
+/* device device types (0x0-0xf) */
+enum {
+ AC_JACK_LINE_OUT,
+ AC_JACK_SPEAKER,
+ AC_JACK_HP_OUT,
+ AC_JACK_CD,
+ AC_JACK_SPDIF_OUT,
+ AC_JACK_DIG_OTHER_OUT,
+ AC_JACK_MODEM_LINE_SIDE,
+ AC_JACK_MODEM_HAND_SIDE,
+ AC_JACK_LINE_IN,
+ AC_JACK_AUX,
+ AC_JACK_MIC_IN,
+ AC_JACK_TELEPHONY,
+ AC_JACK_SPDIF_IN,
+ AC_JACK_DIG_OTHER_IN,
+ AC_JACK_OTHER = 0xf,
+};
+
+/* jack connection types (0x0-0xf) */
+enum {
+ AC_JACK_CONN_UNKNOWN,
+ AC_JACK_CONN_1_8,
+ AC_JACK_CONN_1_4,
+ AC_JACK_CONN_ATAPI,
+ AC_JACK_CONN_RCA,
+ AC_JACK_CONN_OPTICAL,
+ AC_JACK_CONN_OTHER_DIGITAL,
+ AC_JACK_CONN_OTHER_ANALOG,
+ AC_JACK_CONN_DIN,
+ AC_JACK_CONN_XLR,
+ AC_JACK_CONN_RJ11,
+ AC_JACK_CONN_COMB,
+ AC_JACK_CONN_OTHER = 0xf,
+};
+
+/* jack colors (0x0-0xf) */
+enum {
+ AC_JACK_COLOR_UNKNOWN,
+ AC_JACK_COLOR_BLACK,
+ AC_JACK_COLOR_GREY,
+ AC_JACK_COLOR_BLUE,
+ AC_JACK_COLOR_GREEN,
+ AC_JACK_COLOR_RED,
+ AC_JACK_COLOR_ORANGE,
+ AC_JACK_COLOR_YELLOW,
+ AC_JACK_COLOR_PURPLE,
+ AC_JACK_COLOR_PINK,
+ AC_JACK_COLOR_WHITE = 0xe,
+ AC_JACK_COLOR_OTHER,
+};
+
+/* Jack location (0x0-0x3f) */
+/* common case */
+enum {
+ AC_JACK_LOC_NONE,
+ AC_JACK_LOC_REAR,
+ AC_JACK_LOC_FRONT,
+ AC_JACK_LOC_LEFT,
+ AC_JACK_LOC_RIGHT,
+ AC_JACK_LOC_TOP,
+ AC_JACK_LOC_BOTTOM,
+};
+/* bits 4-5 */
+enum {
+ AC_JACK_LOC_EXTERNAL = 0x00,
+ AC_JACK_LOC_INTERNAL = 0x10,
+ AC_JACK_LOC_SEPARATE = 0x20,
+ AC_JACK_LOC_OTHER = 0x30,
+};
+enum {
+ /* external on primary chasis */
+ AC_JACK_LOC_REAR_PANEL = 0x07,
+ AC_JACK_LOC_DRIVE_BAY,
+ /* internal */
+ AC_JACK_LOC_RISER = 0x17,
+ AC_JACK_LOC_HDMI,
+ AC_JACK_LOC_ATAPI,
+ /* others */
+ AC_JACK_LOC_MOBILE_IN = 0x37,
+ AC_JACK_LOC_MOBILE_OUT,
+};
+
+/* Port connectivity (0-3) */
+enum {
+ AC_JACK_PORT_COMPLEX,
+ AC_JACK_PORT_NONE,
+ AC_JACK_PORT_FIXED,
+ AC_JACK_PORT_BOTH,
+};
+
+/* max. codec address */
+#define HDA_MAX_CODEC_ADDRESS 0x0f
+
+#endif /* __SOUND_HDA_VERBS_H */
diff --git a/kernel/include/sound/hdaudio.h b/kernel/include/sound/hdaudio.h
new file mode 100644
index 000000000..2a8aa9dfb
--- /dev/null
+++ b/kernel/include/sound/hdaudio.h
@@ -0,0 +1,247 @@
+/*
+ * HD-audio core stuff
+ */
+
+#ifndef __SOUND_HDAUDIO_H
+#define __SOUND_HDAUDIO_H
+
+#include <linux/device.h>
+#include <sound/hda_verbs.h>
+
+/* codec node id */
+typedef u16 hda_nid_t;
+
+struct hdac_bus;
+struct hdac_device;
+struct hdac_driver;
+struct hdac_widget_tree;
+
+/*
+ * exported bus type
+ */
+extern struct bus_type snd_hda_bus_type;
+
+/*
+ * generic arrays
+ */
+struct snd_array {
+ unsigned int used;
+ unsigned int alloced;
+ unsigned int elem_size;
+ unsigned int alloc_align;
+ void *list;
+};
+
+/*
+ * HD-audio codec base device
+ */
+struct hdac_device {
+ struct device dev;
+ int type;
+ struct hdac_bus *bus;
+ unsigned int addr; /* codec address */
+ struct list_head list; /* list point for bus codec_list */
+
+ hda_nid_t afg; /* AFG node id */
+ hda_nid_t mfg; /* MFG node id */
+
+ /* ids */
+ unsigned int vendor_id;
+ unsigned int subsystem_id;
+ unsigned int revision_id;
+ unsigned int afg_function_id;
+ unsigned int mfg_function_id;
+ unsigned int afg_unsol:1;
+ unsigned int mfg_unsol:1;
+
+ unsigned int power_caps; /* FG power caps */
+
+ const char *vendor_name; /* codec vendor name */
+ const char *chip_name; /* codec chip name */
+
+ /* verb exec op override */
+ int (*exec_verb)(struct hdac_device *dev, unsigned int cmd,
+ unsigned int flags, unsigned int *res);
+
+ /* widgets */
+ unsigned int num_nodes;
+ hda_nid_t start_nid, end_nid;
+
+ /* misc flags */
+ atomic_t in_pm; /* suspend/resume being performed */
+
+ /* sysfs */
+ struct hdac_widget_tree *widgets;
+
+ /* regmap */
+ struct regmap *regmap;
+ struct snd_array vendor_verbs;
+ bool lazy_cache:1; /* don't wake up for writes */
+ bool caps_overwriting:1; /* caps overwrite being in process */
+ bool cache_coef:1; /* cache COEF read/write too */
+};
+
+/* device/driver type used for matching */
+enum {
+ HDA_DEV_CORE,
+ HDA_DEV_LEGACY,
+};
+
+/* direction */
+enum {
+ HDA_INPUT, HDA_OUTPUT
+};
+
+#define dev_to_hdac_dev(_dev) container_of(_dev, struct hdac_device, dev)
+
+int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus,
+ const char *name, unsigned int addr);
+void snd_hdac_device_exit(struct hdac_device *dev);
+int snd_hdac_device_register(struct hdac_device *codec);
+void snd_hdac_device_unregister(struct hdac_device *codec);
+
+int snd_hdac_refresh_widgets(struct hdac_device *codec);
+
+unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
+ unsigned int verb, unsigned int parm);
+int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd,
+ unsigned int flags, unsigned int *res);
+int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid,
+ unsigned int verb, unsigned int parm, unsigned int *res);
+int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm,
+ unsigned int *res);
+int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid,
+ int parm);
+int snd_hdac_override_parm(struct hdac_device *codec, hda_nid_t nid,
+ unsigned int parm, unsigned int val);
+int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid,
+ hda_nid_t *conn_list, int max_conns);
+int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid,
+ hda_nid_t *start_id);
+
+/**
+ * snd_hdac_read_parm - read a codec parameter
+ * @codec: the codec object
+ * @nid: NID to read a parameter
+ * @parm: parameter to read
+ *
+ * Returns -1 for error. If you need to distinguish the error more
+ * strictly, use _snd_hdac_read_parm() directly.
+ */
+static inline int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid,
+ int parm)
+{
+ unsigned int val;
+
+ return _snd_hdac_read_parm(codec, nid, parm, &val) < 0 ? -1 : val;
+}
+
+#ifdef CONFIG_PM
+void snd_hdac_power_up(struct hdac_device *codec);
+void snd_hdac_power_down(struct hdac_device *codec);
+void snd_hdac_power_up_pm(struct hdac_device *codec);
+void snd_hdac_power_down_pm(struct hdac_device *codec);
+#else
+static inline void snd_hdac_power_up(struct hdac_device *codec) {}
+static inline void snd_hdac_power_down(struct hdac_device *codec) {}
+static inline void snd_hdac_power_up_pm(struct hdac_device *codec) {}
+static inline void snd_hdac_power_down_pm(struct hdac_device *codec) {}
+#endif
+
+/*
+ * HD-audio codec base driver
+ */
+struct hdac_driver {
+ struct device_driver driver;
+ int type;
+ int (*match)(struct hdac_device *dev, struct hdac_driver *drv);
+ void (*unsol_event)(struct hdac_device *dev, unsigned int event);
+};
+
+#define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver)
+
+/*
+ * HD-audio bus base driver
+ */
+struct hdac_bus_ops {
+ /* send a single command */
+ int (*command)(struct hdac_bus *bus, unsigned int cmd);
+ /* get a response from the last command */
+ int (*get_response)(struct hdac_bus *bus, unsigned int addr,
+ unsigned int *res);
+};
+
+#define HDA_UNSOL_QUEUE_SIZE 64
+
+struct hdac_bus {
+ struct device *dev;
+ const struct hdac_bus_ops *ops;
+
+ /* codec linked list */
+ struct list_head codec_list;
+ unsigned int num_codecs;
+
+ /* link caddr -> codec */
+ struct hdac_device *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1];
+
+ /* unsolicited event queue */
+ u32 unsol_queue[HDA_UNSOL_QUEUE_SIZE * 2]; /* ring buffer */
+ unsigned int unsol_rp, unsol_wp;
+ struct work_struct unsol_work;
+
+ /* bit flags of powered codecs */
+ unsigned long codec_powered;
+
+ /* flags */
+ bool sync_write:1; /* sync after verb write */
+
+ /* locks */
+ struct mutex cmd_mutex;
+};
+
+int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
+ const struct hdac_bus_ops *ops);
+void snd_hdac_bus_exit(struct hdac_bus *bus);
+int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr,
+ unsigned int cmd, unsigned int *res);
+int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr,
+ unsigned int cmd, unsigned int *res);
+void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex);
+
+int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec);
+void snd_hdac_bus_remove_device(struct hdac_bus *bus,
+ struct hdac_device *codec);
+
+static inline void snd_hdac_codec_link_up(struct hdac_device *codec)
+{
+ set_bit(codec->addr, &codec->bus->codec_powered);
+}
+
+static inline void snd_hdac_codec_link_down(struct hdac_device *codec)
+{
+ clear_bit(codec->addr, &codec->bus->codec_powered);
+}
+
+/*
+ * generic array helpers
+ */
+void *snd_array_new(struct snd_array *array);
+void snd_array_free(struct snd_array *array);
+static inline void snd_array_init(struct snd_array *array, unsigned int size,
+ unsigned int align)
+{
+ array->elem_size = size;
+ array->alloc_align = align;
+}
+
+static inline void *snd_array_elem(struct snd_array *array, unsigned int idx)
+{
+ return array->list + idx * array->elem_size;
+}
+
+static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
+{
+ return (unsigned long)(ptr - array->list) / array->elem_size;
+}
+
+#endif /* __SOUND_HDAUDIO_H */
diff --git a/kernel/include/sound/hwdep.h b/kernel/include/sound/hwdep.h
new file mode 100644
index 000000000..ab9fcb2f9
--- /dev/null
+++ b/kernel/include/sound/hwdep.h
@@ -0,0 +1,82 @@
+#ifndef __SOUND_HWDEP_H
+#define __SOUND_HWDEP_H
+
+/*
+ * Hardware dependent layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/asound.h>
+#include <linux/poll.h>
+
+struct snd_hwdep;
+
+/* hwdep file ops; all ops can be NULL */
+struct snd_hwdep_ops {
+ long long (*llseek)(struct snd_hwdep *hw, struct file *file,
+ long long offset, int orig);
+ long (*read)(struct snd_hwdep *hw, char __user *buf,
+ long count, loff_t *offset);
+ long (*write)(struct snd_hwdep *hw, const char __user *buf,
+ long count, loff_t *offset);
+ int (*open)(struct snd_hwdep *hw, struct file * file);
+ int (*release)(struct snd_hwdep *hw, struct file * file);
+ unsigned int (*poll)(struct snd_hwdep *hw, struct file *file,
+ poll_table *wait);
+ int (*ioctl)(struct snd_hwdep *hw, struct file *file,
+ unsigned int cmd, unsigned long arg);
+ int (*ioctl_compat)(struct snd_hwdep *hw, struct file *file,
+ unsigned int cmd, unsigned long arg);
+ int (*mmap)(struct snd_hwdep *hw, struct file *file,
+ struct vm_area_struct *vma);
+ int (*dsp_status)(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_status *status);
+ int (*dsp_load)(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_image *image);
+};
+
+struct snd_hwdep {
+ struct snd_card *card;
+ struct list_head list;
+ int device;
+ char id[32];
+ char name[80];
+ int iface;
+
+#ifdef CONFIG_SND_OSSEMUL
+ int oss_type;
+ int ossreg;
+#endif
+
+ struct snd_hwdep_ops ops;
+ wait_queue_head_t open_wait;
+ void *private_data;
+ void (*private_free) (struct snd_hwdep *hwdep);
+ struct device dev;
+
+ struct mutex open_mutex;
+ int used; /* reference counter */
+ unsigned int dsp_loaded; /* bit fields of loaded dsp indices */
+ unsigned int exclusive:1; /* exclusive access mode */
+};
+
+extern int snd_hwdep_new(struct snd_card *card, char *id, int device,
+ struct snd_hwdep **rhwdep);
+
+#endif /* __SOUND_HWDEP_H */
diff --git a/kernel/include/sound/i2c.h b/kernel/include/sound/i2c.h
new file mode 100644
index 000000000..d125ff8c8
--- /dev/null
+++ b/kernel/include/sound/i2c.h
@@ -0,0 +1,104 @@
+#ifndef __SOUND_I2C_H
+#define __SOUND_I2C_H
+
+/*
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ */
+
+#define SND_I2C_DEVICE_ADDRTEN (1<<0) /* 10-bit I2C address */
+
+struct snd_i2c_device {
+ struct list_head list;
+ struct snd_i2c_bus *bus; /* I2C bus */
+ char name[32]; /* some useful device name */
+ unsigned short flags; /* device flags */
+ unsigned short addr; /* device address (might be 10-bit) */
+ unsigned long private_value;
+ void *private_data;
+ void (*private_free)(struct snd_i2c_device *device);
+};
+
+#define snd_i2c_device(n) list_entry(n, struct snd_i2c_device, list)
+
+struct snd_i2c_bit_ops {
+ void (*start)(struct snd_i2c_bus *bus); /* transfer start */
+ void (*stop)(struct snd_i2c_bus *bus); /* transfer stop */
+ void (*direction)(struct snd_i2c_bus *bus, int clock, int data); /* set line direction (0 = write, 1 = read) */
+ void (*setlines)(struct snd_i2c_bus *bus, int clock, int data);
+ int (*getclock)(struct snd_i2c_bus *bus);
+ int (*getdata)(struct snd_i2c_bus *bus, int ack);
+};
+
+struct snd_i2c_ops {
+ int (*sendbytes)(struct snd_i2c_device *device, unsigned char *bytes, int count);
+ int (*readbytes)(struct snd_i2c_device *device, unsigned char *bytes, int count);
+ int (*probeaddr)(struct snd_i2c_bus *bus, unsigned short addr);
+};
+
+struct snd_i2c_bus {
+ struct snd_card *card; /* card which I2C belongs to */
+ char name[32]; /* some useful label */
+
+ struct mutex lock_mutex;
+
+ struct snd_i2c_bus *master; /* master bus when SCK/SCL is shared */
+ struct list_head buses; /* master: slave buses sharing SCK/SCL, slave: link list */
+
+ struct list_head devices; /* attached devices to this bus */
+
+ union {
+ struct snd_i2c_bit_ops *bit;
+ void *ops;
+ } hw_ops; /* lowlevel operations */
+ struct snd_i2c_ops *ops; /* midlevel operations */
+
+ unsigned long private_value;
+ void *private_data;
+ void (*private_free)(struct snd_i2c_bus *bus);
+};
+
+#define snd_i2c_slave_bus(n) list_entry(n, struct snd_i2c_bus, buses)
+
+int snd_i2c_bus_create(struct snd_card *card, const char *name,
+ struct snd_i2c_bus *master, struct snd_i2c_bus **ri2c);
+int snd_i2c_device_create(struct snd_i2c_bus *bus, const char *name,
+ unsigned char addr, struct snd_i2c_device **rdevice);
+int snd_i2c_device_free(struct snd_i2c_device *device);
+
+static inline void snd_i2c_lock(struct snd_i2c_bus *bus)
+{
+ if (bus->master)
+ mutex_lock(&bus->master->lock_mutex);
+ else
+ mutex_lock(&bus->lock_mutex);
+}
+
+static inline void snd_i2c_unlock(struct snd_i2c_bus *bus)
+{
+ if (bus->master)
+ mutex_unlock(&bus->master->lock_mutex);
+ else
+ mutex_unlock(&bus->lock_mutex);
+}
+
+int snd_i2c_sendbytes(struct snd_i2c_device *device, unsigned char *bytes, int count);
+int snd_i2c_readbytes(struct snd_i2c_device *device, unsigned char *bytes, int count);
+int snd_i2c_probeaddr(struct snd_i2c_bus *bus, unsigned short addr);
+
+#endif /* __SOUND_I2C_H */
diff --git a/kernel/include/sound/info.h b/kernel/include/sound/info.h
new file mode 100644
index 000000000..9ca1a493d
--- /dev/null
+++ b/kernel/include/sound/info.h
@@ -0,0 +1,202 @@
+#ifndef __SOUND_INFO_H
+#define __SOUND_INFO_H
+
+/*
+ * Header file for info interface
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/poll.h>
+
+/* buffer for information */
+struct snd_info_buffer {
+ char *buffer; /* pointer to begin of buffer */
+ unsigned int curr; /* current position in buffer */
+ unsigned int size; /* current size */
+ unsigned int len; /* total length of buffer */
+ int stop; /* stop flag */
+ int error; /* error code */
+};
+
+#define SNDRV_INFO_CONTENT_TEXT 0
+#define SNDRV_INFO_CONTENT_DATA 1
+
+struct snd_info_entry;
+
+struct snd_info_entry_text {
+ void (*read)(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer);
+ void (*write)(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer);
+};
+
+struct snd_info_entry_ops {
+ int (*open)(struct snd_info_entry *entry,
+ unsigned short mode, void **file_private_data);
+ int (*release)(struct snd_info_entry *entry,
+ unsigned short mode, void *file_private_data);
+ ssize_t (*read)(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, char __user *buf,
+ size_t count, loff_t pos);
+ ssize_t (*write)(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, const char __user *buf,
+ size_t count, loff_t pos);
+ loff_t (*llseek)(struct snd_info_entry *entry,
+ void *file_private_data, struct file *file,
+ loff_t offset, int orig);
+ unsigned int (*poll)(struct snd_info_entry *entry,
+ void *file_private_data, struct file *file,
+ poll_table *wait);
+ int (*ioctl)(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, unsigned int cmd, unsigned long arg);
+ int (*mmap)(struct snd_info_entry *entry, void *file_private_data,
+ struct inode *inode, struct file *file,
+ struct vm_area_struct *vma);
+};
+
+struct snd_info_entry {
+ const char *name;
+ umode_t mode;
+ long size;
+ unsigned short content;
+ union {
+ struct snd_info_entry_text text;
+ struct snd_info_entry_ops *ops;
+ } c;
+ struct snd_info_entry *parent;
+ struct snd_card *card;
+ struct module *module;
+ void *private_data;
+ void (*private_free)(struct snd_info_entry *entry);
+ struct proc_dir_entry *p;
+ struct mutex access;
+ struct list_head children;
+ struct list_head list;
+};
+
+#if defined(CONFIG_SND_OSSEMUL) && defined(CONFIG_PROC_FS)
+int snd_info_minor_register(void);
+int snd_info_minor_unregister(void);
+#else
+#define snd_info_minor_register() /* NOP */
+#define snd_info_minor_unregister() /* NOP */
+#endif
+
+
+#ifdef CONFIG_PROC_FS
+
+extern struct snd_info_entry *snd_seq_root;
+#ifdef CONFIG_SND_OSSEMUL
+extern struct snd_info_entry *snd_oss_root;
+void snd_card_info_read_oss(struct snd_info_buffer *buffer);
+#else
+#define snd_oss_root NULL
+static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
+#endif
+
+__printf(2, 3)
+int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...);
+int snd_info_init(void);
+int snd_info_done(void);
+
+int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len);
+const char *snd_info_get_str(char *dest, const char *src, int len);
+struct snd_info_entry *snd_info_create_module_entry(struct module *module,
+ const char *name,
+ struct snd_info_entry *parent);
+struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card,
+ const char *name,
+ struct snd_info_entry *parent);
+void snd_info_free_entry(struct snd_info_entry *entry);
+int snd_info_store_text(struct snd_info_entry *entry);
+int snd_info_restore_text(struct snd_info_entry *entry);
+
+int snd_info_card_create(struct snd_card *card);
+int snd_info_card_register(struct snd_card *card);
+int snd_info_card_free(struct snd_card *card);
+void snd_info_card_disconnect(struct snd_card *card);
+void snd_info_card_id_change(struct snd_card *card);
+int snd_info_register(struct snd_info_entry *entry);
+
+/* for card drivers */
+int snd_card_proc_new(struct snd_card *card, const char *name,
+ struct snd_info_entry **entryp);
+
+static inline void snd_info_set_text_ops(struct snd_info_entry *entry,
+ void *private_data,
+ void (*read)(struct snd_info_entry *, struct snd_info_buffer *))
+{
+ entry->private_data = private_data;
+ entry->c.text.read = read;
+}
+
+int snd_info_check_reserved_words(const char *str);
+
+#else
+
+#define snd_seq_root NULL
+#define snd_oss_root NULL
+
+static inline int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) { return 0; }
+static inline int snd_info_init(void) { return 0; }
+static inline int snd_info_done(void) { return 0; }
+
+static inline int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { return 0; }
+static inline char *snd_info_get_str(char *dest, char *src, int len) { return NULL; }
+static inline struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent) { return NULL; }
+static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent) { return NULL; }
+static inline void snd_info_free_entry(struct snd_info_entry *entry) { ; }
+
+static inline int snd_info_card_create(struct snd_card *card) { return 0; }
+static inline int snd_info_card_register(struct snd_card *card) { return 0; }
+static inline int snd_info_card_free(struct snd_card *card) { return 0; }
+static inline void snd_info_card_disconnect(struct snd_card *card) { }
+static inline void snd_info_card_id_change(struct snd_card *card) { }
+static inline int snd_info_register(struct snd_info_entry *entry) { return 0; }
+
+static inline int snd_card_proc_new(struct snd_card *card, const char *name,
+ struct snd_info_entry **entryp) { return -EINVAL; }
+static inline void snd_info_set_text_ops(struct snd_info_entry *entry __attribute__((unused)),
+ void *private_data,
+ void (*read)(struct snd_info_entry *, struct snd_info_buffer *)) {}
+
+static inline int snd_info_check_reserved_words(const char *str) { return 1; }
+
+#endif
+
+/*
+ * OSS info part
+ */
+
+#if defined(CONFIG_SND_OSSEMUL) && defined(CONFIG_PROC_FS)
+
+#define SNDRV_OSS_INFO_DEV_AUDIO 0
+#define SNDRV_OSS_INFO_DEV_SYNTH 1
+#define SNDRV_OSS_INFO_DEV_MIDI 2
+#define SNDRV_OSS_INFO_DEV_TIMERS 4
+#define SNDRV_OSS_INFO_DEV_MIXERS 5
+
+#define SNDRV_OSS_INFO_DEV_COUNT 6
+
+int snd_oss_info_register(int dev, int num, char *string);
+#define snd_oss_info_unregister(dev, num) snd_oss_info_register(dev, num, NULL)
+
+#endif /* CONFIG_SND_OSSEMUL && CONFIG_PROC_FS */
+
+#endif /* __SOUND_INFO_H */
diff --git a/kernel/include/sound/initval.h b/kernel/include/sound/initval.h
new file mode 100644
index 000000000..ac62c67e6
--- /dev/null
+++ b/kernel/include/sound/initval.h
@@ -0,0 +1,104 @@
+#ifndef __SOUND_INITVAL_H
+#define __SOUND_INITVAL_H
+
+/*
+ * Init values for soundcard modules
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define SNDRV_AUTO_PORT 1
+#define SNDRV_AUTO_IRQ 0xffff
+#define SNDRV_AUTO_DMA 0xffff
+#define SNDRV_AUTO_DMA_SIZE (0x7fffffff)
+
+#define SNDRV_DEFAULT_IDX1 (-1)
+#define SNDRV_DEFAULT_STR1 NULL
+#define SNDRV_DEFAULT_ENABLE1 1
+#define SNDRV_DEFAULT_PORT1 SNDRV_AUTO_PORT
+#define SNDRV_DEFAULT_IRQ1 SNDRV_AUTO_IRQ
+#define SNDRV_DEFAULT_DMA1 SNDRV_AUTO_DMA
+#define SNDRV_DEFAULT_DMA_SIZE1 SNDRV_AUTO_DMA_SIZE
+#define SNDRV_DEFAULT_PTR1 SNDRV_DEFAULT_STR1
+
+#define SNDRV_DEFAULT_IDX { [0 ... (SNDRV_CARDS-1)] = -1 }
+#define SNDRV_DEFAULT_STR { [0 ... (SNDRV_CARDS-1)] = NULL }
+#define SNDRV_DEFAULT_ENABLE { 1, [1 ... (SNDRV_CARDS-1)] = 0 }
+#define SNDRV_DEFAULT_ENABLE_PNP { [0 ... (SNDRV_CARDS-1)] = 1 }
+#ifdef CONFIG_PNP
+#define SNDRV_DEFAULT_ENABLE_ISAPNP SNDRV_DEFAULT_ENABLE_PNP
+#else
+#define SNDRV_DEFAULT_ENABLE_ISAPNP SNDRV_DEFAULT_ENABLE
+#endif
+#define SNDRV_DEFAULT_PORT { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_PORT }
+#define SNDRV_DEFAULT_IRQ { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_IRQ }
+#define SNDRV_DEFAULT_DMA { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_DMA }
+#define SNDRV_DEFAULT_DMA_SIZE { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_DMA_SIZE }
+#define SNDRV_DEFAULT_PTR SNDRV_DEFAULT_STR
+
+#ifdef SNDRV_LEGACY_FIND_FREE_IOPORT
+static long snd_legacy_find_free_ioport(long *port_table, long size)
+{
+ while (*port_table != -1) {
+ if (request_region(*port_table, size, "ALSA test")) {
+ release_region(*port_table, size);
+ return *port_table;
+ }
+ port_table++;
+ }
+ return -1;
+}
+#endif
+
+#ifdef SNDRV_LEGACY_FIND_FREE_IRQ
+#include <linux/interrupt.h>
+
+static irqreturn_t snd_legacy_empty_irq_handler(int irq, void *dev_id)
+{
+ return IRQ_HANDLED;
+}
+
+static int snd_legacy_find_free_irq(int *irq_table)
+{
+ while (*irq_table != -1) {
+ if (!request_irq(*irq_table, snd_legacy_empty_irq_handler,
+ IRQF_PROBE_SHARED, "ALSA Test IRQ",
+ (void *) irq_table)) {
+ free_irq(*irq_table, (void *) irq_table);
+ return *irq_table;
+ }
+ irq_table++;
+ }
+ return -1;
+}
+#endif
+
+#ifdef SNDRV_LEGACY_FIND_FREE_DMA
+static int snd_legacy_find_free_dma(int *dma_table)
+{
+ while (*dma_table != -1) {
+ if (!request_dma(*dma_table, "ALSA Test DMA")) {
+ free_dma(*dma_table);
+ return *dma_table;
+ }
+ dma_table++;
+ }
+ return -1;
+}
+#endif
+
+#endif /* __SOUND_INITVAL_H */
diff --git a/kernel/include/sound/jack.h b/kernel/include/sound/jack.h
new file mode 100644
index 000000000..218235030
--- /dev/null
+++ b/kernel/include/sound/jack.h
@@ -0,0 +1,121 @@
+#ifndef __SOUND_JACK_H
+#define __SOUND_JACK_H
+
+/*
+ * Jack abstraction layer
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+
+struct input_dev;
+
+/**
+ * enum snd_jack_types - Jack types which can be reported
+ * @SND_JACK_HEADPHONE: Headphone
+ * @SND_JACK_MICROPHONE: Microphone
+ * @SND_JACK_HEADSET: Headset
+ * @SND_JACK_LINEOUT: Line out
+ * @SND_JACK_MECHANICAL: Mechanical switch
+ * @SND_JACK_VIDEOOUT: Video out
+ * @SND_JACK_AVOUT: AV (Audio Video) out
+ * @SND_JACK_LINEIN: Line in
+ * @SND_JACK_BTN_0: Button 0
+ * @SND_JACK_BTN_1: Button 1
+ * @SND_JACK_BTN_2: Button 2
+ * @SND_JACK_BTN_3: Button 3
+ * @SND_JACK_BTN_4: Button 4
+ * @SND_JACK_BTN_5: Button 5
+ *
+ * These values are used as a bitmask.
+ *
+ * Note that this must be kept in sync with the lookup table in
+ * sound/core/jack.c.
+ */
+enum snd_jack_types {
+ SND_JACK_HEADPHONE = 0x0001,
+ SND_JACK_MICROPHONE = 0x0002,
+ SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE,
+ SND_JACK_LINEOUT = 0x0004,
+ SND_JACK_MECHANICAL = 0x0008, /* If detected separately */
+ SND_JACK_VIDEOOUT = 0x0010,
+ SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT,
+ SND_JACK_LINEIN = 0x0020,
+
+ /* Kept separate from switches to facilitate implementation */
+ SND_JACK_BTN_0 = 0x4000,
+ SND_JACK_BTN_1 = 0x2000,
+ SND_JACK_BTN_2 = 0x1000,
+ SND_JACK_BTN_3 = 0x0800,
+ SND_JACK_BTN_4 = 0x0400,
+ SND_JACK_BTN_5 = 0x0200,
+};
+
+/* Keep in sync with definitions above */
+#define SND_JACK_SWITCH_TYPES 6
+
+struct snd_jack {
+ struct input_dev *input_dev;
+ int registered;
+ int type;
+ const char *id;
+ char name[100];
+ unsigned int key[6]; /* Keep in sync with definitions above */
+ void *private_data;
+ void (*private_free)(struct snd_jack *);
+};
+
+#ifdef CONFIG_SND_JACK
+
+int snd_jack_new(struct snd_card *card, const char *id, int type,
+ struct snd_jack **jack);
+void snd_jack_set_parent(struct snd_jack *jack, struct device *parent);
+int snd_jack_set_key(struct snd_jack *jack, enum snd_jack_types type,
+ int keytype);
+
+void snd_jack_report(struct snd_jack *jack, int status);
+
+#else
+
+static inline int snd_jack_new(struct snd_card *card, const char *id, int type,
+ struct snd_jack **jack)
+{
+ return 0;
+}
+
+static inline void snd_jack_set_parent(struct snd_jack *jack,
+ struct device *parent)
+{
+}
+
+static inline int snd_jack_set_key(struct snd_jack *jack,
+ enum snd_jack_types type,
+ int keytype)
+{
+ return 0;
+}
+
+static inline void snd_jack_report(struct snd_jack *jack, int status)
+{
+}
+
+#endif
+
+#endif
diff --git a/kernel/include/sound/l3.h b/kernel/include/sound/l3.h
new file mode 100644
index 000000000..423a08f0f
--- /dev/null
+++ b/kernel/include/sound/l3.h
@@ -0,0 +1,18 @@
+#ifndef _L3_H_
+#define _L3_H_ 1
+
+struct l3_pins {
+ void (*setdat)(int);
+ void (*setclk)(int);
+ void (*setmode)(int);
+ int data_hold;
+ int data_setup;
+ int clock_high;
+ int mode_hold;
+ int mode;
+ int mode_setup;
+};
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
+
+#endif
diff --git a/kernel/include/sound/max9768.h b/kernel/include/sound/max9768.h
new file mode 100644
index 000000000..0f78b41d0
--- /dev/null
+++ b/kernel/include/sound/max9768.h
@@ -0,0 +1,24 @@
+/*
+ * Platform data for MAX9768
+ * Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K.
+ * same licence as the driver
+ */
+
+#ifndef __SOUND_MAX9768_PDATA_H__
+#define __SOUND_MAX9768_PDATA_H__
+
+/**
+ * struct max9768_pdata - optional platform specific MAX9768 configuration
+ * @shdn_gpio: GPIO to SHDN pin. If not valid, pin must be hardwired HIGH
+ * @mute_gpio: GPIO to MUTE pin. If not valid, control for mute won't be added
+ * @flags: configuration flags, e.g. set classic PWM mode (check datasheet
+ * regarding "filterless modulation" which is default).
+ */
+struct max9768_pdata {
+ int shdn_gpio;
+ int mute_gpio;
+ unsigned flags;
+#define MAX9768_FLAG_CLASSIC_PWM (1 << 0)
+};
+
+#endif /* __SOUND_MAX9768_PDATA_H__*/
diff --git a/kernel/include/sound/max98088.h b/kernel/include/sound/max98088.h
new file mode 100644
index 000000000..c3ba82391
--- /dev/null
+++ b/kernel/include/sound/max98088.h
@@ -0,0 +1,50 @@
+/*
+ * Platform data for MAX98088
+ *
+ * Copyright 2010 Maxim Integrated Products
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef __SOUND_MAX98088_PDATA_H__
+#define __SOUND_MAX98088_PDATA_H__
+
+/* Equalizer filter response configuration */
+struct max98088_eq_cfg {
+ const char *name;
+ unsigned int rate;
+ u16 band1[5];
+ u16 band2[5];
+ u16 band3[5];
+ u16 band4[5];
+ u16 band5[5];
+};
+
+/* codec platform data */
+struct max98088_pdata {
+
+ /* Equalizers for DAI1 and DAI2 */
+ struct max98088_eq_cfg *eq_cfg;
+ unsigned int eq_cfgcnt;
+
+ /* Receiver output can be configured as power amplifier or LINE out */
+ /* Set receiver_mode to:
+ * 0 = amplifier output, or
+ * 1 = LINE level output
+ */
+ unsigned int receiver_mode:1;
+
+ /* Analog/digital microphone configuration:
+ * 0 = analog microphone input (normal setting)
+ * 1 = digital microphone input
+ */
+ unsigned int digmic_left_mode:1;
+ unsigned int digmic_right_mode:1;
+
+};
+
+#endif
diff --git a/kernel/include/sound/max98090.h b/kernel/include/sound/max98090.h
new file mode 100644
index 000000000..95efb13f8
--- /dev/null
+++ b/kernel/include/sound/max98090.h
@@ -0,0 +1,29 @@
+/*
+ * Platform data for MAX98090
+ *
+ * Copyright 2011-2012 Maxim Integrated Products
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef __SOUND_MAX98090_PDATA_H__
+#define __SOUND_MAX98090_PDATA_H__
+
+/* codec platform data */
+struct max98090_pdata {
+
+ /* Analog/digital microphone configuration:
+ * 0 = analog microphone input (normal setting)
+ * 1 = digital microphone input
+ */
+ unsigned int digmic_left_mode:1;
+ unsigned int digmic_right_mode:1;
+ unsigned int digmic_3_mode:1;
+ unsigned int digmic_4_mode:1;
+};
+
+#endif
diff --git a/kernel/include/sound/max98095.h b/kernel/include/sound/max98095.h
new file mode 100644
index 000000000..e87ae67b0
--- /dev/null
+++ b/kernel/include/sound/max98095.h
@@ -0,0 +1,66 @@
+/*
+ * Platform data for MAX98095
+ *
+ * Copyright 2011 Maxim Integrated Products
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef __SOUND_MAX98095_PDATA_H__
+#define __SOUND_MAX98095_PDATA_H__
+
+/* Equalizer filter response configuration */
+struct max98095_eq_cfg {
+ const char *name;
+ unsigned int rate;
+ u16 band1[5];
+ u16 band2[5];
+ u16 band3[5];
+ u16 band4[5];
+ u16 band5[5];
+};
+
+/* Biquad filter response configuration */
+struct max98095_biquad_cfg {
+ const char *name;
+ unsigned int rate;
+ u16 band1[5];
+ u16 band2[5];
+};
+
+/* codec platform data */
+struct max98095_pdata {
+
+ /* Equalizers for DAI1 and DAI2 */
+ struct max98095_eq_cfg *eq_cfg;
+ unsigned int eq_cfgcnt;
+
+ /* Biquad filter for DAI1 and DAI2 */
+ struct max98095_biquad_cfg *bq_cfg;
+ unsigned int bq_cfgcnt;
+
+ /* Analog/digital microphone configuration:
+ * 0 = analog microphone input (normal setting)
+ * 1 = digital microphone input
+ */
+ unsigned int digmic_left_mode:1;
+ unsigned int digmic_right_mode:1;
+
+ /* Pin5 is the mechanical method of sensing jack insertion
+ * but it is something that might not be supported.
+ * 0 = PIN5 not supported
+ * 1 = PIN5 supported
+ */
+ unsigned int jack_detect_pin5en:1;
+
+ /* Slew amount for jack detection. Calculated as 4 * (delay + 1).
+ * Default delay is 24 to get a time of 100ms.
+ */
+ unsigned int jack_detect_delay;
+};
+
+#endif
diff --git a/kernel/include/sound/memalloc.h b/kernel/include/sound/memalloc.h
new file mode 100644
index 000000000..782d1df34
--- /dev/null
+++ b/kernel/include/sound/memalloc.h
@@ -0,0 +1,157 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ * Generic memory allocators
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifndef __SOUND_MEMALLOC_H
+#define __SOUND_MEMALLOC_H
+
+struct device;
+
+/*
+ * buffer device info
+ */
+struct snd_dma_device {
+ int type; /* SNDRV_DMA_TYPE_XXX */
+ struct device *dev; /* generic device */
+};
+
+#ifndef snd_dma_pci_data
+#define snd_dma_pci_data(pci) (&(pci)->dev)
+#define snd_dma_isa_data() NULL
+#define snd_dma_continuous_data(x) ((struct device *)(__force unsigned long)(x))
+#endif
+
+
+/*
+ * buffer types
+ */
+#define SNDRV_DMA_TYPE_UNKNOWN 0 /* not defined */
+#define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */
+#define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */
+#ifdef CONFIG_SND_DMA_SGBUF
+#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */
+#else
+#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */
+#endif
+#ifdef CONFIG_GENERIC_ALLOCATOR
+#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */
+#else
+#define SNDRV_DMA_TYPE_DEV_IRAM SNDRV_DMA_TYPE_DEV
+#endif
+
+/*
+ * info for buffer allocation
+ */
+struct snd_dma_buffer {
+ struct snd_dma_device dev; /* device type */
+ unsigned char *area; /* virtual pointer */
+ dma_addr_t addr; /* physical address */
+ size_t bytes; /* buffer size in bytes */
+ void *private_data; /* private for allocator; don't touch */
+};
+
+#ifdef CONFIG_SND_DMA_SGBUF
+/*
+ * Scatter-Gather generic device pages
+ */
+void *snd_malloc_sgbuf_pages(struct device *device,
+ size_t size, struct snd_dma_buffer *dmab,
+ size_t *res_size);
+int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab);
+
+struct snd_sg_page {
+ void *buf;
+ dma_addr_t addr;
+};
+
+struct snd_sg_buf {
+ int size; /* allocated byte size */
+ int pages; /* allocated pages */
+ int tblsize; /* allocated table size */
+ struct snd_sg_page *table; /* address table */
+ struct page **page_table; /* page table (for vmap/vunmap) */
+ struct device *dev;
+};
+
+/*
+ * return the pages matching with the given byte size
+ */
+static inline unsigned int snd_sgbuf_aligned_pages(size_t size)
+{
+ return (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
+}
+
+/*
+ * return the physical address at the corresponding offset
+ */
+static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
+ size_t offset)
+{
+ struct snd_sg_buf *sgbuf = dmab->private_data;
+ dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
+ addr &= ~((dma_addr_t)PAGE_SIZE - 1);
+ return addr + offset % PAGE_SIZE;
+}
+
+/*
+ * return the virtual address at the corresponding offset
+ */
+static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
+ size_t offset)
+{
+ struct snd_sg_buf *sgbuf = dmab->private_data;
+ return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
+}
+
+unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab,
+ unsigned int ofs, unsigned int size);
+#else
+/* non-SG versions */
+static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
+ size_t offset)
+{
+ return dmab->addr + offset;
+}
+
+static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
+ size_t offset)
+{
+ return dmab->area + offset;
+}
+
+#define snd_sgbuf_get_chunk_size(dmab, ofs, size) (size)
+
+#endif /* CONFIG_SND_DMA_SGBUF */
+
+/* allocate/release a buffer */
+int snd_dma_alloc_pages(int type, struct device *dev, size_t size,
+ struct snd_dma_buffer *dmab);
+int snd_dma_alloc_pages_fallback(int type, struct device *dev, size_t size,
+ struct snd_dma_buffer *dmab);
+void snd_dma_free_pages(struct snd_dma_buffer *dmab);
+
+/* basic memory allocation functions */
+void *snd_malloc_pages(size_t size, gfp_t gfp_flags);
+void snd_free_pages(void *ptr, size_t size);
+
+#endif /* __SOUND_MEMALLOC_H */
+
diff --git a/kernel/include/sound/minors.h b/kernel/include/sound/minors.h
new file mode 100644
index 000000000..5978f9a8c
--- /dev/null
+++ b/kernel/include/sound/minors.h
@@ -0,0 +1,112 @@
+#ifndef __SOUND_MINORS_H
+#define __SOUND_MINORS_H
+
+/*
+ * MINOR numbers
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define SNDRV_OS_MINORS 256
+
+#define SNDRV_MINOR_DEVICES 32
+#define SNDRV_MINOR_CARD(minor) ((minor) >> 5)
+#define SNDRV_MINOR_DEVICE(minor) ((minor) & 0x001f)
+#define SNDRV_MINOR(card, dev) (((card) << 5) | (dev))
+
+/* these minors can still be used for autoloading devices (/dev/aload*) */
+#define SNDRV_MINOR_CONTROL 0 /* 0 */
+#define SNDRV_MINOR_GLOBAL 1 /* 1 */
+#define SNDRV_MINOR_SEQUENCER 1 /* SNDRV_MINOR_GLOBAL + 0 * 32 */
+#define SNDRV_MINOR_TIMER 33 /* SNDRV_MINOR_GLOBAL + 1 * 32 */
+
+#ifndef CONFIG_SND_DYNAMIC_MINORS
+#define SNDRV_MINOR_COMPRESS 2 /* 2 - 3 */
+#define SNDRV_MINOR_HWDEP 4 /* 4 - 7 */
+#define SNDRV_MINOR_RAWMIDI 8 /* 8 - 15 */
+#define SNDRV_MINOR_PCM_PLAYBACK 16 /* 16 - 23 */
+#define SNDRV_MINOR_PCM_CAPTURE 24 /* 24 - 31 */
+
+/* same as first respective minor number to make minor allocation easier */
+#define SNDRV_DEVICE_TYPE_CONTROL SNDRV_MINOR_CONTROL
+#define SNDRV_DEVICE_TYPE_HWDEP SNDRV_MINOR_HWDEP
+#define SNDRV_DEVICE_TYPE_RAWMIDI SNDRV_MINOR_RAWMIDI
+#define SNDRV_DEVICE_TYPE_PCM_PLAYBACK SNDRV_MINOR_PCM_PLAYBACK
+#define SNDRV_DEVICE_TYPE_PCM_CAPTURE SNDRV_MINOR_PCM_CAPTURE
+#define SNDRV_DEVICE_TYPE_SEQUENCER SNDRV_MINOR_SEQUENCER
+#define SNDRV_DEVICE_TYPE_TIMER SNDRV_MINOR_TIMER
+#define SNDRV_DEVICE_TYPE_COMPRESS SNDRV_MINOR_COMPRESS
+
+#else /* CONFIG_SND_DYNAMIC_MINORS */
+
+enum {
+ SNDRV_DEVICE_TYPE_CONTROL,
+ SNDRV_DEVICE_TYPE_SEQUENCER,
+ SNDRV_DEVICE_TYPE_TIMER,
+ SNDRV_DEVICE_TYPE_HWDEP,
+ SNDRV_DEVICE_TYPE_RAWMIDI,
+ SNDRV_DEVICE_TYPE_PCM_PLAYBACK,
+ SNDRV_DEVICE_TYPE_PCM_CAPTURE,
+ SNDRV_DEVICE_TYPE_COMPRESS,
+};
+
+#endif /* CONFIG_SND_DYNAMIC_MINORS */
+
+#define SNDRV_MINOR_HWDEPS 4
+#define SNDRV_MINOR_RAWMIDIS 8
+#define SNDRV_MINOR_PCMS 8
+
+
+#ifdef CONFIG_SND_OSSEMUL
+
+#define SNDRV_MINOR_OSS_DEVICES 16
+#define SNDRV_MINOR_OSS_CARD(minor) ((minor) >> 4)
+#define SNDRV_MINOR_OSS_DEVICE(minor) ((minor) & 0x000f)
+#define SNDRV_MINOR_OSS(card, dev) (((card) << 4) | (dev))
+
+#define SNDRV_MINOR_OSS_MIXER 0 /* /dev/mixer - OSS 3.XX compatible */
+#define SNDRV_MINOR_OSS_SEQUENCER 1 /* /dev/sequencer - OSS 3.XX compatible */
+#define SNDRV_MINOR_OSS_MIDI 2 /* /dev/midi - native midi interface - OSS 3.XX compatible - UART */
+#define SNDRV_MINOR_OSS_PCM 3 /* alias */
+#define SNDRV_MINOR_OSS_PCM_8 3 /* /dev/dsp - 8bit PCM - OSS 3.XX compatible */
+#define SNDRV_MINOR_OSS_AUDIO 4 /* /dev/audio - SunSparc compatible */
+#define SNDRV_MINOR_OSS_PCM_16 5 /* /dev/dsp16 - 16bit PCM - OSS 3.XX compatible */
+#define SNDRV_MINOR_OSS_SNDSTAT 6 /* /dev/sndstat - for compatibility with OSS */
+#define SNDRV_MINOR_OSS_RESERVED7 7 /* reserved for future use */
+#define SNDRV_MINOR_OSS_MUSIC 8 /* /dev/music - OSS 3.XX compatible */
+#define SNDRV_MINOR_OSS_DMMIDI 9 /* /dev/dmmidi0 - this device can have another minor # with OSS */
+#define SNDRV_MINOR_OSS_DMFM 10 /* /dev/dmfm0 - this device can have another minor # with OSS */
+#define SNDRV_MINOR_OSS_MIXER1 11 /* alternate mixer */
+#define SNDRV_MINOR_OSS_PCM1 12 /* alternate PCM (GF-A-1) */
+#define SNDRV_MINOR_OSS_MIDI1 13 /* alternate midi - SYNTH */
+#define SNDRV_MINOR_OSS_DMMIDI1 14 /* alternate dmmidi - SYNTH */
+#define SNDRV_MINOR_OSS_RESERVED15 15 /* reserved for future use */
+
+#define SNDRV_OSS_DEVICE_TYPE_MIXER 0
+#define SNDRV_OSS_DEVICE_TYPE_SEQUENCER 1
+#define SNDRV_OSS_DEVICE_TYPE_PCM 2
+#define SNDRV_OSS_DEVICE_TYPE_MIDI 3
+#define SNDRV_OSS_DEVICE_TYPE_DMFM 4
+#define SNDRV_OSS_DEVICE_TYPE_SNDSTAT 5
+#define SNDRV_OSS_DEVICE_TYPE_MUSIC 6
+
+#define MODULE_ALIAS_SNDRV_MINOR(type) \
+ MODULE_ALIAS("sound-service-?-" __stringify(type))
+
+#endif
+
+#endif /* __SOUND_MINORS_H */
diff --git a/kernel/include/sound/mixer_oss.h b/kernel/include/sound/mixer_oss.h
new file mode 100644
index 000000000..13cb0b430
--- /dev/null
+++ b/kernel/include/sound/mixer_oss.h
@@ -0,0 +1,81 @@
+#ifndef __SOUND_MIXER_OSS_H
+#define __SOUND_MIXER_OSS_H
+
+/*
+ * OSS MIXER API
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+
+#define SNDRV_OSS_MAX_MIXERS 32
+
+struct snd_mixer_oss_file;
+
+struct snd_mixer_oss_slot {
+ int number;
+ unsigned int stereo: 1;
+ int (*get_volume)(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *chn,
+ int *left, int *right);
+ int (*put_volume)(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *chn,
+ int left, int right);
+ int (*get_recsrc)(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *chn,
+ int *active);
+ int (*put_recsrc)(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *chn,
+ int active);
+ unsigned long private_value;
+ void *private_data;
+ void (*private_free)(struct snd_mixer_oss_slot *slot);
+ int volume[2];
+};
+
+struct snd_mixer_oss {
+ struct snd_card *card;
+ char id[16];
+ char name[32];
+ struct snd_mixer_oss_slot slots[SNDRV_OSS_MAX_MIXERS]; /* OSS mixer slots */
+ unsigned int mask_recsrc; /* exclusive recsrc mask */
+ int (*get_recsrc)(struct snd_mixer_oss_file *fmixer,
+ unsigned int *active_index);
+ int (*put_recsrc)(struct snd_mixer_oss_file *fmixer,
+ unsigned int active_index);
+ void *private_data_recsrc;
+ void (*private_free_recsrc)(struct snd_mixer_oss *mixer);
+ struct mutex reg_mutex;
+ struct snd_info_entry *proc_entry;
+ int oss_dev_alloc;
+ /* --- */
+ int oss_recsrc;
+};
+
+struct snd_mixer_oss_file {
+ struct snd_card *card;
+ struct snd_mixer_oss *mixer;
+};
+
+int snd_mixer_oss_ioctl_card(struct snd_card *card,
+ unsigned int cmd, unsigned long arg);
+
+#endif /* CONFIG_SND_MIXER_OSS */
+
+#endif /* __SOUND_MIXER_OSS_H */
diff --git a/kernel/include/sound/mpu401.h b/kernel/include/sound/mpu401.h
new file mode 100644
index 000000000..e94209692
--- /dev/null
+++ b/kernel/include/sound/mpu401.h
@@ -0,0 +1,138 @@
+#ifndef __SOUND_MPU401_H
+#define __SOUND_MPU401_H
+
+/*
+ * Header file for MPU-401 and compatible cards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/rawmidi.h>
+#include <linux/interrupt.h>
+
+#define MPU401_HW_MPU401 1 /* native MPU401 */
+#define MPU401_HW_SB 2 /* SoundBlaster MPU-401 UART */
+#define MPU401_HW_ES1688 3 /* AudioDrive ES1688 MPU-401 UART */
+#define MPU401_HW_OPL3SA2 4 /* Yamaha OPL3-SA2 */
+#define MPU401_HW_SONICVIBES 5 /* S3 SonicVibes */
+#define MPU401_HW_CS4232 6 /* CS4232 */
+#define MPU401_HW_ES18XX 7 /* AudioDrive ES18XX MPU-401 UART */
+#define MPU401_HW_FM801 8 /* ForteMedia FM801 */
+#define MPU401_HW_TRID4DWAVE 9 /* Trident 4DWave */
+#define MPU401_HW_AZT2320 10 /* Aztech AZT2320 */
+#define MPU401_HW_ALS100 11 /* Avance Logic ALS100 */
+#define MPU401_HW_ICE1712 12 /* Envy24 */
+#define MPU401_HW_VIA686A 13 /* VIA 82C686A */
+#define MPU401_HW_YMFPCI 14 /* YMF DS-XG PCI */
+#define MPU401_HW_CMIPCI 15 /* CMIPCI MPU-401 UART */
+#define MPU401_HW_ALS4000 16 /* Avance Logic ALS4000 */
+#define MPU401_HW_INTEL8X0 17 /* Intel8x0 driver */
+#define MPU401_HW_PC98II 18 /* Roland PC98II */
+#define MPU401_HW_AUREAL 19 /* Aureal Vortex */
+
+#define MPU401_INFO_INPUT (1 << 0) /* input stream */
+#define MPU401_INFO_OUTPUT (1 << 1) /* output stream */
+#define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */
+#define MPU401_INFO_MMIO (1 << 3) /* MMIO access */
+#define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */
+#define MPU401_INFO_IRQ_HOOK (1 << 5) /* mpu401 irq handler is called
+ from driver irq handler */
+#define MPU401_INFO_NO_ACK (1 << 6) /* No ACK cmd needed */
+#define MPU401_INFO_USE_TIMER (1 << 15) /* internal */
+
+#define MPU401_MODE_BIT_INPUT 0
+#define MPU401_MODE_BIT_OUTPUT 1
+#define MPU401_MODE_BIT_INPUT_TRIGGER 2
+#define MPU401_MODE_BIT_OUTPUT_TRIGGER 3
+
+#define MPU401_MODE_INPUT (1<<MPU401_MODE_BIT_INPUT)
+#define MPU401_MODE_OUTPUT (1<<MPU401_MODE_BIT_OUTPUT)
+#define MPU401_MODE_INPUT_TRIGGER (1<<MPU401_MODE_BIT_INPUT_TRIGGER)
+#define MPU401_MODE_OUTPUT_TRIGGER (1<<MPU401_MODE_BIT_OUTPUT_TRIGGER)
+
+#define MPU401_MODE_INPUT_TIMER (1<<0)
+#define MPU401_MODE_OUTPUT_TIMER (1<<1)
+
+struct snd_mpu401 {
+ struct snd_rawmidi *rmidi;
+
+ unsigned short hardware; /* MPU401_HW_XXXX */
+ unsigned int info_flags; /* MPU401_INFO_XXX */
+ unsigned long port; /* base port of MPU-401 chip */
+ unsigned long cport; /* port + 1 (usually) */
+ struct resource *res; /* port resource */
+ int irq; /* IRQ number of MPU-401 chip */
+
+ unsigned long mode; /* MPU401_MODE_XXXX */
+ int timer_invoked;
+
+ int (*open_input) (struct snd_mpu401 * mpu);
+ void (*close_input) (struct snd_mpu401 * mpu);
+ int (*open_output) (struct snd_mpu401 * mpu);
+ void (*close_output) (struct snd_mpu401 * mpu);
+ void *private_data;
+
+ struct snd_rawmidi_substream *substream_input;
+ struct snd_rawmidi_substream *substream_output;
+
+ spinlock_t input_lock;
+ spinlock_t output_lock;
+ spinlock_t timer_lock;
+
+ struct timer_list timer;
+
+ void (*write) (struct snd_mpu401 * mpu, unsigned char data, unsigned long addr);
+ unsigned char (*read) (struct snd_mpu401 *mpu, unsigned long addr);
+};
+
+/* I/O ports */
+
+#define MPU401C(mpu) (mpu)->cport
+#define MPU401D(mpu) (mpu)->port
+
+/*
+ * control register bits
+ */
+/* read MPU401C() */
+#define MPU401_RX_EMPTY 0x80
+#define MPU401_TX_FULL 0x40
+
+/* write MPU401C() */
+#define MPU401_RESET 0xff
+#define MPU401_ENTER_UART 0x3f
+
+/* read MPU401D() */
+#define MPU401_ACK 0xfe
+
+
+/*
+
+ */
+
+irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id);
+irqreturn_t snd_mpu401_uart_interrupt_tx(int irq, void *dev_id);
+
+int snd_mpu401_uart_new(struct snd_card *card,
+ int device,
+ unsigned short hardware,
+ unsigned long port,
+ unsigned int info_flags,
+ int irq,
+ struct snd_rawmidi ** rrawmidi);
+
+#endif /* __SOUND_MPU401_H */
diff --git a/kernel/include/sound/omap-hdmi-audio.h b/kernel/include/sound/omap-hdmi-audio.h
new file mode 100644
index 000000000..afdb41689
--- /dev/null
+++ b/kernel/include/sound/omap-hdmi-audio.h
@@ -0,0 +1,43 @@
+/*
+ * hdmi-audio.c -- OMAP4+ DSS HDMI audio support library
+ *
+ * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ */
+
+#include <video/omapdss.h>
+
+#ifndef __OMAP_HDMI_AUDIO_H__
+#define __OMAP_HDMI_AUDIO_H__
+
+struct omap_hdmi_audio_ops {
+ int (*audio_startup)(struct device *dev,
+ void (*abort_cb)(struct device *dev));
+ int (*audio_shutdown)(struct device *dev);
+ int (*audio_start)(struct device *dev);
+ void (*audio_stop)(struct device *dev);
+ int (*audio_config)(struct device *dev,
+ struct omap_dss_audio *dss_audio);
+};
+
+/* HDMI audio initalization data */
+struct omap_hdmi_audio_pdata {
+ struct device *dev;
+ enum omapdss_version dss_version;
+ phys_addr_t audio_dma_addr;
+
+ const struct omap_hdmi_audio_ops *ops;
+};
+
+#endif /* __OMAP_HDMI_AUDIO_H__ */
diff --git a/kernel/include/sound/omap-pcm.h b/kernel/include/sound/omap-pcm.h
new file mode 100644
index 000000000..c1d2f31d7
--- /dev/null
+++ b/kernel/include/sound/omap-pcm.h
@@ -0,0 +1,30 @@
+/*
+ * omap-pcm.h - OMAP PCM driver
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef __OMAP_PCM_H__
+#define __OMAP_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_OMAP_SOC)
+int omap_pcm_platform_register(struct device *dev);
+#else
+static inline int omap_pcm_platform_register(struct device *dev)
+{
+ return 0;
+}
+#endif /* CONFIG_SND_OMAP_SOC */
+
+#endif /* __OMAP_PCM_H__ */
diff --git a/kernel/include/sound/opl3.h b/kernel/include/sound/opl3.h
new file mode 100644
index 000000000..6ba670707
--- /dev/null
+++ b/kernel/include/sound/opl3.h
@@ -0,0 +1,393 @@
+#ifndef __SOUND_OPL3_H
+#define __SOUND_OPL3_H
+
+/*
+ * Definitions of the OPL-3 registers.
+ *
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Hannu Savolainen 1993-1996
+ *
+ *
+ * The OPL-3 mode is switched on by writing 0x01, to the offset 5
+ * of the right side.
+ *
+ * Another special register at the right side is at offset 4. It contains
+ * a bit mask defining which voices are used as 4 OP voices.
+ *
+ * The percussive mode is implemented in the left side only.
+ *
+ * With the above exceptions the both sides can be operated independently.
+ *
+ * A 4 OP voice can be created by setting the corresponding
+ * bit at offset 4 of the right side.
+ *
+ * For example setting the rightmost bit (0x01) changes the
+ * first voice on the right side to the 4 OP mode. The fourth
+ * voice is made inaccessible.
+ *
+ * If a voice is set to the 2 OP mode, it works like 2 OP modes
+ * of the original YM3812 (AdLib). In addition the voice can
+ * be connected the left, right or both stereo channels. It can
+ * even be left unconnected. This works with 4 OP voices also.
+ *
+ * The stereo connection bits are located in the FEEDBACK_CONNECTION
+ * register of the voice (0xC0-0xC8). In 4 OP voices these bits are
+ * in the second half of the voice.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include <sound/hwdep.h>
+#include <sound/timer.h>
+#include <sound/seq_midi_emul.h>
+#ifdef CONFIG_SND_SEQUENCER_OSS
+#include <sound/seq_oss.h>
+#include <sound/seq_oss_legacy.h>
+#endif
+#include <sound/seq_device.h>
+#include <sound/asound_fm.h>
+
+/*
+ * Register numbers for the global registers
+ */
+
+#define OPL3_REG_TEST 0x01
+#define OPL3_ENABLE_WAVE_SELECT 0x20
+
+#define OPL3_REG_TIMER1 0x02
+#define OPL3_REG_TIMER2 0x03
+#define OPL3_REG_TIMER_CONTROL 0x04 /* Left side */
+#define OPL3_IRQ_RESET 0x80
+#define OPL3_TIMER1_MASK 0x40
+#define OPL3_TIMER2_MASK 0x20
+#define OPL3_TIMER1_START 0x01
+#define OPL3_TIMER2_START 0x02
+
+#define OPL3_REG_CONNECTION_SELECT 0x04 /* Right side */
+#define OPL3_LEFT_4OP_0 0x01
+#define OPL3_LEFT_4OP_1 0x02
+#define OPL3_LEFT_4OP_2 0x04
+#define OPL3_RIGHT_4OP_0 0x08
+#define OPL3_RIGHT_4OP_1 0x10
+#define OPL3_RIGHT_4OP_2 0x20
+
+#define OPL3_REG_MODE 0x05 /* Right side */
+#define OPL3_OPL3_ENABLE 0x01 /* OPL3 mode */
+#define OPL3_OPL4_ENABLE 0x02 /* OPL4 mode */
+
+#define OPL3_REG_KBD_SPLIT 0x08 /* Left side */
+#define OPL3_COMPOSITE_SINE_WAVE_MODE 0x80 /* Don't use with OPL-3? */
+#define OPL3_KEYBOARD_SPLIT 0x40
+
+#define OPL3_REG_PERCUSSION 0xbd /* Left side only */
+#define OPL3_TREMOLO_DEPTH 0x80
+#define OPL3_VIBRATO_DEPTH 0x40
+#define OPL3_PERCUSSION_ENABLE 0x20
+#define OPL3_BASSDRUM_ON 0x10
+#define OPL3_SNAREDRUM_ON 0x08
+#define OPL3_TOMTOM_ON 0x04
+#define OPL3_CYMBAL_ON 0x02
+#define OPL3_HIHAT_ON 0x01
+
+/*
+ * Offsets to the register banks for operators. To get the
+ * register number just add the operator offset to the bank offset
+ *
+ * AM/VIB/EG/KSR/Multiple (0x20 to 0x35)
+ */
+#define OPL3_REG_AM_VIB 0x20
+#define OPL3_TREMOLO_ON 0x80
+#define OPL3_VIBRATO_ON 0x40
+#define OPL3_SUSTAIN_ON 0x20
+#define OPL3_KSR 0x10 /* Key scaling rate */
+#define OPL3_MULTIPLE_MASK 0x0f /* Frequency multiplier */
+
+ /*
+ * KSL/Total level (0x40 to 0x55)
+ */
+#define OPL3_REG_KSL_LEVEL 0x40
+#define OPL3_KSL_MASK 0xc0 /* Envelope scaling bits */
+#define OPL3_TOTAL_LEVEL_MASK 0x3f /* Strength (volume) of OP */
+
+/*
+ * Attack / Decay rate (0x60 to 0x75)
+ */
+#define OPL3_REG_ATTACK_DECAY 0x60
+#define OPL3_ATTACK_MASK 0xf0
+#define OPL3_DECAY_MASK 0x0f
+
+/*
+ * Sustain level / Release rate (0x80 to 0x95)
+ */
+#define OPL3_REG_SUSTAIN_RELEASE 0x80
+#define OPL3_SUSTAIN_MASK 0xf0
+#define OPL3_RELEASE_MASK 0x0f
+
+/*
+ * Wave select (0xE0 to 0xF5)
+ */
+#define OPL3_REG_WAVE_SELECT 0xe0
+#define OPL3_WAVE_SELECT_MASK 0x07
+
+/*
+ * Offsets to the register banks for voices. Just add to the
+ * voice number to get the register number.
+ *
+ * F-Number low bits (0xA0 to 0xA8).
+ */
+#define OPL3_REG_FNUM_LOW 0xa0
+
+/*
+ * F-number high bits / Key on / Block (octave) (0xB0 to 0xB8)
+ */
+#define OPL3_REG_KEYON_BLOCK 0xb0
+#define OPL3_KEYON_BIT 0x20
+#define OPL3_BLOCKNUM_MASK 0x1c
+#define OPL3_FNUM_HIGH_MASK 0x03
+
+/*
+ * Feedback / Connection (0xc0 to 0xc8)
+ *
+ * These registers have two new bits when the OPL-3 mode
+ * is selected. These bits controls connecting the voice
+ * to the stereo channels. For 4 OP voices this bit is
+ * defined in the second half of the voice (add 3 to the
+ * register offset).
+ *
+ * For 4 OP voices the connection bit is used in the
+ * both halves (gives 4 ways to connect the operators).
+ */
+#define OPL3_REG_FEEDBACK_CONNECTION 0xc0
+#define OPL3_FEEDBACK_MASK 0x0e /* Valid just for 1st OP of a voice */
+#define OPL3_CONNECTION_BIT 0x01
+/*
+ * In the 4 OP mode there is four possible configurations how the
+ * operators can be connected together (in 2 OP modes there is just
+ * AM or FM). The 4 OP connection mode is defined by the rightmost
+ * bit of the FEEDBACK_CONNECTION (0xC0-0xC8) on the both halves.
+ *
+ * First half Second half Mode
+ *
+ * +---+
+ * v |
+ * 0 0 >+-1-+--2--3--4-->
+ *
+ *
+ *
+ * +---+
+ * | |
+ * 0 1 >+-1-+--2-+
+ * |->
+ * >--3----4-+
+ *
+ * +---+
+ * | |
+ * 1 0 >+-1-+-----+
+ * |->
+ * >--2--3--4-+
+ *
+ * +---+
+ * | |
+ * 1 1 >+-1-+--+
+ * |
+ * >--2--3-+->
+ * |
+ * >--4----+
+ */
+#define OPL3_STEREO_BITS 0x30 /* OPL-3 only */
+#define OPL3_VOICE_TO_LEFT 0x10
+#define OPL3_VOICE_TO_RIGHT 0x20
+
+/*
+
+ */
+
+#define OPL3_LEFT 0x0000
+#define OPL3_RIGHT 0x0100
+
+#define OPL3_HW_AUTO 0x0000
+#define OPL3_HW_OPL2 0x0200
+#define OPL3_HW_OPL3 0x0300
+#define OPL3_HW_OPL3_SV 0x0301 /* S3 SonicVibes */
+#define OPL3_HW_OPL3_CS 0x0302 /* CS4232/CS4236+ */
+#define OPL3_HW_OPL3_FM801 0x0303 /* FM801 */
+#define OPL3_HW_OPL3_CS4281 0x0304 /* CS4281 */
+#define OPL3_HW_OPL4 0x0400 /* YMF278B/YMF295 */
+#define OPL3_HW_OPL4_ML 0x0401 /* YMF704/YMF721 */
+#define OPL3_HW_MASK 0xff00
+
+#define MAX_OPL2_VOICES 9
+#define MAX_OPL3_VOICES 18
+
+struct snd_opl3;
+
+/*
+ * Instrument record, aka "Patch"
+ */
+
+/* FM operator */
+struct fm_operator {
+ unsigned char am_vib;
+ unsigned char ksl_level;
+ unsigned char attack_decay;
+ unsigned char sustain_release;
+ unsigned char wave_select;
+} __attribute__((packed));
+
+/* Instrument data */
+struct fm_instrument {
+ struct fm_operator op[4];
+ unsigned char feedback_connection[2];
+ unsigned char echo_delay;
+ unsigned char echo_atten;
+ unsigned char chorus_spread;
+ unsigned char trnsps;
+ unsigned char fix_dur;
+ unsigned char modes;
+ unsigned char fix_key;
+};
+
+/* type */
+#define FM_PATCH_OPL2 0x01 /* OPL2 2 operators FM instrument */
+#define FM_PATCH_OPL3 0x02 /* OPL3 4 operators FM instrument */
+
+/* Instrument record */
+struct fm_patch {
+ unsigned char prog;
+ unsigned char bank;
+ unsigned char type;
+ struct fm_instrument inst;
+ char name[24];
+ struct fm_patch *next;
+};
+
+
+/*
+ * A structure to keep track of each hardware voice
+ */
+struct snd_opl3_voice {
+ int state; /* status */
+#define SNDRV_OPL3_ST_OFF 0 /* Not playing */
+#define SNDRV_OPL3_ST_ON_2OP 1 /* 2op voice is allocated */
+#define SNDRV_OPL3_ST_ON_4OP 2 /* 4op voice is allocated */
+#define SNDRV_OPL3_ST_NOT_AVAIL -1 /* voice is not available */
+
+ unsigned int time; /* An allocation time */
+ unsigned char note; /* Note currently assigned to this voice */
+
+ unsigned long note_off; /* note-off time */
+ int note_off_check; /* check note-off time */
+
+ unsigned char keyon_reg; /* KON register shadow */
+
+ struct snd_midi_channel *chan; /* Midi channel for this note */
+};
+
+struct snd_opl3 {
+ unsigned long l_port;
+ unsigned long r_port;
+ struct resource *res_l_port;
+ struct resource *res_r_port;
+ unsigned short hardware;
+ /* hardware access */
+ void (*command) (struct snd_opl3 * opl3, unsigned short cmd, unsigned char val);
+ unsigned short timer_enable;
+ int seq_dev_num; /* sequencer device number */
+ struct snd_timer *timer1;
+ struct snd_timer *timer2;
+ spinlock_t timer_lock;
+
+ void *private_data;
+ void (*private_free)(struct snd_opl3 *);
+
+ struct snd_hwdep *hwdep;
+ spinlock_t reg_lock;
+ struct snd_card *card; /* The card that this belongs to */
+ unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */
+ unsigned char rhythm; /* percussion mode flag */
+ unsigned char max_voices; /* max number of voices */
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+#define SNDRV_OPL3_MODE_SYNTH 0 /* OSS - voices allocated by application */
+#define SNDRV_OPL3_MODE_SEQ 1 /* ALSA - driver handles voice allocation */
+ int synth_mode; /* synth mode */
+ int seq_client;
+
+ struct snd_seq_device *seq_dev; /* sequencer device */
+ struct snd_midi_channel_set * chset;
+
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ struct snd_seq_device *oss_seq_dev; /* OSS sequencer device */
+ struct snd_midi_channel_set * oss_chset;
+#endif
+
+#define OPL3_PATCH_HASH_SIZE 32
+ struct fm_patch *patch_table[OPL3_PATCH_HASH_SIZE];
+
+ struct snd_opl3_voice voices[MAX_OPL3_VOICES]; /* Voices (OPL3 'channel') */
+ int use_time; /* allocation counter */
+
+ unsigned short connection_reg; /* connection reg shadow */
+ unsigned char drum_reg; /* percussion reg shadow */
+
+ spinlock_t voice_lock; /* Lock for voice access */
+
+ struct timer_list tlist; /* timer for note-offs and effects */
+ int sys_timer_status; /* system timer run status */
+ spinlock_t sys_timer_lock; /* Lock for system timer access */
+#endif
+};
+
+/* opl3.c */
+void snd_opl3_interrupt(struct snd_hwdep * hw);
+int snd_opl3_new(struct snd_card *card, unsigned short hardware,
+ struct snd_opl3 **ropl3);
+int snd_opl3_init(struct snd_opl3 *opl3);
+int snd_opl3_create(struct snd_card *card,
+ unsigned long l_port, unsigned long r_port,
+ unsigned short hardware,
+ int integrated,
+ struct snd_opl3 ** opl3);
+int snd_opl3_timer_new(struct snd_opl3 * opl3, int timer1_dev, int timer2_dev);
+int snd_opl3_hwdep_new(struct snd_opl3 * opl3, int device, int seq_device,
+ struct snd_hwdep ** rhwdep);
+
+/* opl3_synth */
+int snd_opl3_open(struct snd_hwdep * hw, struct file *file);
+int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
+ unsigned int cmd, unsigned long arg);
+int snd_opl3_release(struct snd_hwdep * hw, struct file *file);
+
+void snd_opl3_reset(struct snd_opl3 * opl3);
+
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count,
+ loff_t *offset);
+int snd_opl3_load_patch(struct snd_opl3 *opl3,
+ int prog, int bank, int type,
+ const char *name,
+ const unsigned char *ext,
+ const unsigned char *data);
+struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank,
+ int create_patch);
+void snd_opl3_clear_patches(struct snd_opl3 *opl3);
+#else
+#define snd_opl3_write NULL
+static inline void snd_opl3_clear_patches(struct snd_opl3 *opl3) {}
+#endif
+
+#endif /* __SOUND_OPL3_H */
diff --git a/kernel/include/sound/opl4.h b/kernel/include/sound/opl4.h
new file mode 100644
index 000000000..60ae8454b
--- /dev/null
+++ b/kernel/include/sound/opl4.h
@@ -0,0 +1,32 @@
+#ifndef __SOUND_OPL4_H
+#define __SOUND_OPL4_H
+
+/*
+ * Global definitions for the OPL4 driver
+ * Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/opl3.h>
+
+struct snd_opl4;
+
+extern int snd_opl4_create(struct snd_card *card,
+ unsigned long fm_port, unsigned long pcm_port,
+ int seq_device,
+ struct snd_opl3 **opl3, struct snd_opl4 **opl4);
+
+#endif /* __SOUND_OPL4_H */
diff --git a/kernel/include/sound/pcm-indirect.h b/kernel/include/sound/pcm-indirect.h
new file mode 100644
index 000000000..1df7acaaa
--- /dev/null
+++ b/kernel/include/sound/pcm-indirect.h
@@ -0,0 +1,177 @@
+/*
+ * Helper functions for indirect PCM data transfer
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ * Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#ifndef __SOUND_PCM_INDIRECT_H
+#define __SOUND_PCM_INDIRECT_H
+
+#include <sound/pcm.h>
+
+struct snd_pcm_indirect {
+ unsigned int hw_buffer_size; /* Byte size of hardware buffer */
+ unsigned int hw_queue_size; /* Max queue size of hw buffer (0 = buffer size) */
+ unsigned int hw_data; /* Offset to next dst (or src) in hw ring buffer */
+ unsigned int hw_io; /* Ring buffer hw pointer */
+ int hw_ready; /* Bytes ready for play (or captured) in hw ring buffer */
+ unsigned int sw_buffer_size; /* Byte size of software buffer */
+ unsigned int sw_data; /* Offset to next dst (or src) in sw ring buffer */
+ unsigned int sw_io; /* Current software pointer in bytes */
+ int sw_ready; /* Bytes ready to be transferred to/from hw */
+ snd_pcm_uframes_t appl_ptr; /* Last seen appl_ptr */
+};
+
+typedef void (*snd_pcm_indirect_copy_t)(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, size_t bytes);
+
+/*
+ * helper function for playback ack callback
+ */
+static inline void
+snd_pcm_indirect_playback_transfer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec,
+ snd_pcm_indirect_copy_t copy)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr;
+ snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr;
+ int qsize;
+
+ if (diff) {
+ if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2))
+ diff += runtime->boundary;
+ rec->sw_ready += (int)frames_to_bytes(runtime, diff);
+ rec->appl_ptr = appl_ptr;
+ }
+ qsize = rec->hw_queue_size ? rec->hw_queue_size : rec->hw_buffer_size;
+ while (rec->hw_ready < qsize && rec->sw_ready > 0) {
+ unsigned int hw_to_end = rec->hw_buffer_size - rec->hw_data;
+ unsigned int sw_to_end = rec->sw_buffer_size - rec->sw_data;
+ unsigned int bytes = qsize - rec->hw_ready;
+ if (rec->sw_ready < (int)bytes)
+ bytes = rec->sw_ready;
+ if (hw_to_end < bytes)
+ bytes = hw_to_end;
+ if (sw_to_end < bytes)
+ bytes = sw_to_end;
+ if (! bytes)
+ break;
+ copy(substream, rec, bytes);
+ rec->hw_data += bytes;
+ if (rec->hw_data == rec->hw_buffer_size)
+ rec->hw_data = 0;
+ rec->sw_data += bytes;
+ if (rec->sw_data == rec->sw_buffer_size)
+ rec->sw_data = 0;
+ rec->hw_ready += bytes;
+ rec->sw_ready -= bytes;
+ }
+}
+
+/*
+ * helper function for playback pointer callback
+ * ptr = current byte pointer
+ */
+static inline snd_pcm_uframes_t
+snd_pcm_indirect_playback_pointer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, unsigned int ptr)
+{
+ int bytes = ptr - rec->hw_io;
+ if (bytes < 0)
+ bytes += rec->hw_buffer_size;
+ rec->hw_io = ptr;
+ rec->hw_ready -= bytes;
+ rec->sw_io += bytes;
+ if (rec->sw_io >= rec->sw_buffer_size)
+ rec->sw_io -= rec->sw_buffer_size;
+ if (substream->ops->ack)
+ substream->ops->ack(substream);
+ return bytes_to_frames(substream->runtime, rec->sw_io);
+}
+
+
+/*
+ * helper function for capture ack callback
+ */
+static inline void
+snd_pcm_indirect_capture_transfer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec,
+ snd_pcm_indirect_copy_t copy)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr;
+ snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr;
+
+ if (diff) {
+ if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2))
+ diff += runtime->boundary;
+ rec->sw_ready -= frames_to_bytes(runtime, diff);
+ rec->appl_ptr = appl_ptr;
+ }
+ while (rec->hw_ready > 0 &&
+ rec->sw_ready < (int)rec->sw_buffer_size) {
+ size_t hw_to_end = rec->hw_buffer_size - rec->hw_data;
+ size_t sw_to_end = rec->sw_buffer_size - rec->sw_data;
+ size_t bytes = rec->sw_buffer_size - rec->sw_ready;
+ if (rec->hw_ready < (int)bytes)
+ bytes = rec->hw_ready;
+ if (hw_to_end < bytes)
+ bytes = hw_to_end;
+ if (sw_to_end < bytes)
+ bytes = sw_to_end;
+ if (! bytes)
+ break;
+ copy(substream, rec, bytes);
+ rec->hw_data += bytes;
+ if ((int)rec->hw_data == rec->hw_buffer_size)
+ rec->hw_data = 0;
+ rec->sw_data += bytes;
+ if (rec->sw_data == rec->sw_buffer_size)
+ rec->sw_data = 0;
+ rec->hw_ready -= bytes;
+ rec->sw_ready += bytes;
+ }
+}
+
+/*
+ * helper function for capture pointer callback,
+ * ptr = current byte pointer
+ */
+static inline snd_pcm_uframes_t
+snd_pcm_indirect_capture_pointer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, unsigned int ptr)
+{
+ int qsize;
+ int bytes = ptr - rec->hw_io;
+ if (bytes < 0)
+ bytes += rec->hw_buffer_size;
+ rec->hw_io = ptr;
+ rec->hw_ready += bytes;
+ qsize = rec->hw_queue_size ? rec->hw_queue_size : rec->hw_buffer_size;
+ if (rec->hw_ready > qsize)
+ return SNDRV_PCM_POS_XRUN;
+ rec->sw_io += bytes;
+ if (rec->sw_io >= rec->sw_buffer_size)
+ rec->sw_io -= rec->sw_buffer_size;
+ if (substream->ops->ack)
+ substream->ops->ack(substream);
+ return bytes_to_frames(substream->runtime, rec->sw_io);
+}
+
+#endif /* __SOUND_PCM_INDIRECT_H */
diff --git a/kernel/include/sound/pcm.h b/kernel/include/sound/pcm.h
new file mode 100644
index 000000000..0cb7f3f5d
--- /dev/null
+++ b/kernel/include/sound/pcm.h
@@ -0,0 +1,1397 @@
+#ifndef __SOUND_PCM_H
+#define __SOUND_PCM_H
+
+/*
+ * Digital Audio (PCM) abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/asound.h>
+#include <sound/memalloc.h>
+#include <sound/minors.h>
+#include <linux/poll.h>
+#include <linux/mm.h>
+#include <linux/bitops.h>
+#include <linux/pm_qos.h>
+
+#define snd_pcm_substream_chip(substream) ((substream)->private_data)
+#define snd_pcm_chip(pcm) ((pcm)->private_data)
+
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+#include <sound/pcm_oss.h>
+#endif
+
+/*
+ * Hardware (lowlevel) section
+ */
+
+struct snd_pcm_hardware {
+ unsigned int info; /* SNDRV_PCM_INFO_* */
+ u64 formats; /* SNDRV_PCM_FMTBIT_* */
+ unsigned int rates; /* SNDRV_PCM_RATE_* */
+ unsigned int rate_min; /* min rate */
+ unsigned int rate_max; /* max rate */
+ unsigned int channels_min; /* min channels */
+ unsigned int channels_max; /* max channels */
+ size_t buffer_bytes_max; /* max buffer size */
+ size_t period_bytes_min; /* min period size */
+ size_t period_bytes_max; /* max period size */
+ unsigned int periods_min; /* min # of periods */
+ unsigned int periods_max; /* max # of periods */
+ size_t fifo_size; /* fifo size in bytes */
+};
+
+struct snd_pcm_substream;
+
+struct snd_pcm_audio_tstamp_config; /* definitions further down */
+struct snd_pcm_audio_tstamp_report;
+
+struct snd_pcm_ops {
+ int (*open)(struct snd_pcm_substream *substream);
+ int (*close)(struct snd_pcm_substream *substream);
+ int (*ioctl)(struct snd_pcm_substream * substream,
+ unsigned int cmd, void *arg);
+ int (*hw_params)(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params);
+ int (*hw_free)(struct snd_pcm_substream *substream);
+ int (*prepare)(struct snd_pcm_substream *substream);
+ int (*trigger)(struct snd_pcm_substream *substream, int cmd);
+ snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream);
+ int (*get_time_info)(struct snd_pcm_substream *substream,
+ struct timespec *system_ts, struct timespec *audio_ts,
+ struct snd_pcm_audio_tstamp_config *audio_tstamp_config,
+ struct snd_pcm_audio_tstamp_report *audio_tstamp_report);
+ int (*copy)(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos,
+ void __user *buf, snd_pcm_uframes_t count);
+ int (*silence)(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, snd_pcm_uframes_t count);
+ struct page *(*page)(struct snd_pcm_substream *substream,
+ unsigned long offset);
+ int (*mmap)(struct snd_pcm_substream *substream, struct vm_area_struct *vma);
+ int (*ack)(struct snd_pcm_substream *substream);
+};
+
+/*
+ *
+ */
+
+#if defined(CONFIG_SND_DYNAMIC_MINORS)
+#define SNDRV_PCM_DEVICES (SNDRV_OS_MINORS-2)
+#else
+#define SNDRV_PCM_DEVICES 8
+#endif
+
+#define SNDRV_PCM_IOCTL1_RESET 0
+#define SNDRV_PCM_IOCTL1_INFO 1
+#define SNDRV_PCM_IOCTL1_CHANNEL_INFO 2
+#define SNDRV_PCM_IOCTL1_GSTATE 3
+#define SNDRV_PCM_IOCTL1_FIFO_SIZE 4
+
+#define SNDRV_PCM_TRIGGER_STOP 0
+#define SNDRV_PCM_TRIGGER_START 1
+#define SNDRV_PCM_TRIGGER_PAUSE_PUSH 3
+#define SNDRV_PCM_TRIGGER_PAUSE_RELEASE 4
+#define SNDRV_PCM_TRIGGER_SUSPEND 5
+#define SNDRV_PCM_TRIGGER_RESUME 6
+#define SNDRV_PCM_TRIGGER_DRAIN 7
+
+#define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1)
+
+/* If you change this don't forget to change rates[] table in pcm_native.c */
+#define SNDRV_PCM_RATE_5512 (1<<0) /* 5512Hz */
+#define SNDRV_PCM_RATE_8000 (1<<1) /* 8000Hz */
+#define SNDRV_PCM_RATE_11025 (1<<2) /* 11025Hz */
+#define SNDRV_PCM_RATE_16000 (1<<3) /* 16000Hz */
+#define SNDRV_PCM_RATE_22050 (1<<4) /* 22050Hz */
+#define SNDRV_PCM_RATE_32000 (1<<5) /* 32000Hz */
+#define SNDRV_PCM_RATE_44100 (1<<6) /* 44100Hz */
+#define SNDRV_PCM_RATE_48000 (1<<7) /* 48000Hz */
+#define SNDRV_PCM_RATE_64000 (1<<8) /* 64000Hz */
+#define SNDRV_PCM_RATE_88200 (1<<9) /* 88200Hz */
+#define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */
+#define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */
+#define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */
+
+#define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */
+#define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */
+
+#define SNDRV_PCM_RATE_8000_44100 (SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_11025|\
+ SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_22050|\
+ SNDRV_PCM_RATE_32000|SNDRV_PCM_RATE_44100)
+#define SNDRV_PCM_RATE_8000_48000 (SNDRV_PCM_RATE_8000_44100|SNDRV_PCM_RATE_48000)
+#define SNDRV_PCM_RATE_8000_96000 (SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_64000|\
+ SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000)
+#define SNDRV_PCM_RATE_8000_192000 (SNDRV_PCM_RATE_8000_96000|SNDRV_PCM_RATE_176400|\
+ SNDRV_PCM_RATE_192000)
+#define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt)
+#define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8)
+#define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8)
+#define SNDRV_PCM_FMTBIT_S16_LE _SNDRV_PCM_FMTBIT(S16_LE)
+#define SNDRV_PCM_FMTBIT_S16_BE _SNDRV_PCM_FMTBIT(S16_BE)
+#define SNDRV_PCM_FMTBIT_U16_LE _SNDRV_PCM_FMTBIT(U16_LE)
+#define SNDRV_PCM_FMTBIT_U16_BE _SNDRV_PCM_FMTBIT(U16_BE)
+#define SNDRV_PCM_FMTBIT_S24_LE _SNDRV_PCM_FMTBIT(S24_LE)
+#define SNDRV_PCM_FMTBIT_S24_BE _SNDRV_PCM_FMTBIT(S24_BE)
+#define SNDRV_PCM_FMTBIT_U24_LE _SNDRV_PCM_FMTBIT(U24_LE)
+#define SNDRV_PCM_FMTBIT_U24_BE _SNDRV_PCM_FMTBIT(U24_BE)
+#define SNDRV_PCM_FMTBIT_S32_LE _SNDRV_PCM_FMTBIT(S32_LE)
+#define SNDRV_PCM_FMTBIT_S32_BE _SNDRV_PCM_FMTBIT(S32_BE)
+#define SNDRV_PCM_FMTBIT_U32_LE _SNDRV_PCM_FMTBIT(U32_LE)
+#define SNDRV_PCM_FMTBIT_U32_BE _SNDRV_PCM_FMTBIT(U32_BE)
+#define SNDRV_PCM_FMTBIT_FLOAT_LE _SNDRV_PCM_FMTBIT(FLOAT_LE)
+#define SNDRV_PCM_FMTBIT_FLOAT_BE _SNDRV_PCM_FMTBIT(FLOAT_BE)
+#define SNDRV_PCM_FMTBIT_FLOAT64_LE _SNDRV_PCM_FMTBIT(FLOAT64_LE)
+#define SNDRV_PCM_FMTBIT_FLOAT64_BE _SNDRV_PCM_FMTBIT(FLOAT64_BE)
+#define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE _SNDRV_PCM_FMTBIT(IEC958_SUBFRAME_LE)
+#define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE _SNDRV_PCM_FMTBIT(IEC958_SUBFRAME_BE)
+#define SNDRV_PCM_FMTBIT_MU_LAW _SNDRV_PCM_FMTBIT(MU_LAW)
+#define SNDRV_PCM_FMTBIT_A_LAW _SNDRV_PCM_FMTBIT(A_LAW)
+#define SNDRV_PCM_FMTBIT_IMA_ADPCM _SNDRV_PCM_FMTBIT(IMA_ADPCM)
+#define SNDRV_PCM_FMTBIT_MPEG _SNDRV_PCM_FMTBIT(MPEG)
+#define SNDRV_PCM_FMTBIT_GSM _SNDRV_PCM_FMTBIT(GSM)
+#define SNDRV_PCM_FMTBIT_SPECIAL _SNDRV_PCM_FMTBIT(SPECIAL)
+#define SNDRV_PCM_FMTBIT_S24_3LE _SNDRV_PCM_FMTBIT(S24_3LE)
+#define SNDRV_PCM_FMTBIT_U24_3LE _SNDRV_PCM_FMTBIT(U24_3LE)
+#define SNDRV_PCM_FMTBIT_S24_3BE _SNDRV_PCM_FMTBIT(S24_3BE)
+#define SNDRV_PCM_FMTBIT_U24_3BE _SNDRV_PCM_FMTBIT(U24_3BE)
+#define SNDRV_PCM_FMTBIT_S20_3LE _SNDRV_PCM_FMTBIT(S20_3LE)
+#define SNDRV_PCM_FMTBIT_U20_3LE _SNDRV_PCM_FMTBIT(U20_3LE)
+#define SNDRV_PCM_FMTBIT_S20_3BE _SNDRV_PCM_FMTBIT(S20_3BE)
+#define SNDRV_PCM_FMTBIT_U20_3BE _SNDRV_PCM_FMTBIT(U20_3BE)
+#define SNDRV_PCM_FMTBIT_S18_3LE _SNDRV_PCM_FMTBIT(S18_3LE)
+#define SNDRV_PCM_FMTBIT_U18_3LE _SNDRV_PCM_FMTBIT(U18_3LE)
+#define SNDRV_PCM_FMTBIT_S18_3BE _SNDRV_PCM_FMTBIT(S18_3BE)
+#define SNDRV_PCM_FMTBIT_U18_3BE _SNDRV_PCM_FMTBIT(U18_3BE)
+#define SNDRV_PCM_FMTBIT_G723_24 _SNDRV_PCM_FMTBIT(G723_24)
+#define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B)
+#define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40)
+#define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B)
+#define SNDRV_PCM_FMTBIT_DSD_U8 _SNDRV_PCM_FMTBIT(DSD_U8)
+#define SNDRV_PCM_FMTBIT_DSD_U16_LE _SNDRV_PCM_FMTBIT(DSD_U16_LE)
+#define SNDRV_PCM_FMTBIT_DSD_U32_LE _SNDRV_PCM_FMTBIT(DSD_U32_LE)
+#define SNDRV_PCM_FMTBIT_DSD_U16_BE _SNDRV_PCM_FMTBIT(DSD_U16_BE)
+#define SNDRV_PCM_FMTBIT_DSD_U32_BE _SNDRV_PCM_FMTBIT(DSD_U32_BE)
+
+#ifdef SNDRV_LITTLE_ENDIAN
+#define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE
+#define SNDRV_PCM_FMTBIT_U16 SNDRV_PCM_FMTBIT_U16_LE
+#define SNDRV_PCM_FMTBIT_S24 SNDRV_PCM_FMTBIT_S24_LE
+#define SNDRV_PCM_FMTBIT_U24 SNDRV_PCM_FMTBIT_U24_LE
+#define SNDRV_PCM_FMTBIT_S32 SNDRV_PCM_FMTBIT_S32_LE
+#define SNDRV_PCM_FMTBIT_U32 SNDRV_PCM_FMTBIT_U32_LE
+#define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_LE
+#define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_LE
+#define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
+#endif
+#ifdef SNDRV_BIG_ENDIAN
+#define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_BE
+#define SNDRV_PCM_FMTBIT_U16 SNDRV_PCM_FMTBIT_U16_BE
+#define SNDRV_PCM_FMTBIT_S24 SNDRV_PCM_FMTBIT_S24_BE
+#define SNDRV_PCM_FMTBIT_U24 SNDRV_PCM_FMTBIT_U24_BE
+#define SNDRV_PCM_FMTBIT_S32 SNDRV_PCM_FMTBIT_S32_BE
+#define SNDRV_PCM_FMTBIT_U32 SNDRV_PCM_FMTBIT_U32_BE
+#define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_BE
+#define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_BE
+#define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE
+#endif
+
+struct snd_pcm_file {
+ struct snd_pcm_substream *substream;
+ int no_compat_mmap;
+};
+
+struct snd_pcm_hw_rule;
+typedef int (*snd_pcm_hw_rule_func_t)(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule);
+
+struct snd_pcm_hw_rule {
+ unsigned int cond;
+ snd_pcm_hw_rule_func_t func;
+ int var;
+ int deps[4];
+ void *private;
+};
+
+struct snd_pcm_hw_constraints {
+ struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK + 1];
+ struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL -
+ SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1];
+ unsigned int rules_num;
+ unsigned int rules_all;
+ struct snd_pcm_hw_rule *rules;
+};
+
+static inline struct snd_mask *constrs_mask(struct snd_pcm_hw_constraints *constrs,
+ snd_pcm_hw_param_t var)
+{
+ return &constrs->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK];
+}
+
+static inline struct snd_interval *constrs_interval(struct snd_pcm_hw_constraints *constrs,
+ snd_pcm_hw_param_t var)
+{
+ return &constrs->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL];
+}
+
+struct snd_ratnum {
+ unsigned int num;
+ unsigned int den_min, den_max, den_step;
+};
+
+struct snd_ratden {
+ unsigned int num_min, num_max, num_step;
+ unsigned int den;
+};
+
+struct snd_pcm_hw_constraint_ratnums {
+ int nrats;
+ struct snd_ratnum *rats;
+};
+
+struct snd_pcm_hw_constraint_ratdens {
+ int nrats;
+ struct snd_ratden *rats;
+};
+
+struct snd_pcm_hw_constraint_list {
+ unsigned int count;
+ const unsigned int *list;
+ unsigned int mask;
+};
+
+struct snd_pcm_hw_constraint_ranges {
+ unsigned int count;
+ const struct snd_interval *ranges;
+ unsigned int mask;
+};
+
+struct snd_pcm_hwptr_log;
+
+/*
+ * userspace-provided audio timestamp config to kernel,
+ * structure is for internal use only and filled with dedicated unpack routine
+ */
+struct snd_pcm_audio_tstamp_config {
+ /* 5 of max 16 bits used */
+ u32 type_requested:4;
+ u32 report_delay:1; /* add total delay to A/D or D/A */
+};
+
+static inline void snd_pcm_unpack_audio_tstamp_config(__u32 data,
+ struct snd_pcm_audio_tstamp_config *config)
+{
+ config->type_requested = data & 0xF;
+ config->report_delay = (data >> 4) & 1;
+}
+
+/*
+ * kernel-provided audio timestamp report to user-space
+ * structure is for internal use only and read by dedicated pack routine
+ */
+struct snd_pcm_audio_tstamp_report {
+ /* 6 of max 16 bits used for bit-fields */
+
+ /* for backwards compatibility */
+ u32 valid:1;
+
+ /* actual type if hardware could not support requested timestamp */
+ u32 actual_type:4;
+
+ /* accuracy represented in ns units */
+ u32 accuracy_report:1; /* 0 if accuracy unknown, 1 if accuracy field is valid */
+ u32 accuracy; /* up to 4.29s, will be packed in separate field */
+};
+
+static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy,
+ const struct snd_pcm_audio_tstamp_report *report)
+{
+ u32 tmp;
+
+ tmp = report->accuracy_report;
+ tmp <<= 4;
+ tmp |= report->actual_type;
+ tmp <<= 1;
+ tmp |= report->valid;
+
+ *data &= 0xffff; /* zero-clear MSBs */
+ *data |= (tmp << 16);
+ *accuracy = report->accuracy;
+}
+
+
+struct snd_pcm_runtime {
+ /* -- Status -- */
+ struct snd_pcm_substream *trigger_master;
+ struct timespec trigger_tstamp; /* trigger timestamp */
+ bool trigger_tstamp_latched; /* trigger timestamp latched in low-level driver/hardware */
+ int overrange;
+ snd_pcm_uframes_t avail_max;
+ snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */
+ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */
+ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */
+ unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */
+ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */
+ u64 hw_ptr_wrap; /* offset for hw_ptr due to boundary wrap-around */
+
+ /* -- HW params -- */
+ snd_pcm_access_t access; /* access mode */
+ snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */
+ snd_pcm_subformat_t subformat; /* subformat */
+ unsigned int rate; /* rate in Hz */
+ unsigned int channels; /* channels */
+ snd_pcm_uframes_t period_size; /* period size */
+ unsigned int periods; /* periods */
+ snd_pcm_uframes_t buffer_size; /* buffer size */
+ snd_pcm_uframes_t min_align; /* Min alignment for the format */
+ size_t byte_align;
+ unsigned int frame_bits;
+ unsigned int sample_bits;
+ unsigned int info;
+ unsigned int rate_num;
+ unsigned int rate_den;
+ unsigned int no_period_wakeup: 1;
+
+ /* -- SW params -- */
+ int tstamp_mode; /* mmap timestamp is updated */
+ unsigned int period_step;
+ snd_pcm_uframes_t start_threshold;
+ snd_pcm_uframes_t stop_threshold;
+ snd_pcm_uframes_t silence_threshold; /* Silence filling happens when
+ noise is nearest than this */
+ snd_pcm_uframes_t silence_size; /* Silence filling size */
+ snd_pcm_uframes_t boundary; /* pointers wrap point */
+
+ snd_pcm_uframes_t silence_start; /* starting pointer to silence area */
+ snd_pcm_uframes_t silence_filled; /* size filled with silence */
+
+ union snd_pcm_sync_id sync; /* hardware synchronization ID */
+
+ /* -- mmap -- */
+ struct snd_pcm_mmap_status *status;
+ struct snd_pcm_mmap_control *control;
+
+ /* -- locking / scheduling -- */
+ snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */
+ wait_queue_head_t sleep; /* poll sleep */
+ wait_queue_head_t tsleep; /* transfer sleep */
+ struct fasync_struct *fasync;
+
+ /* -- private section -- */
+ void *private_data;
+ void (*private_free)(struct snd_pcm_runtime *runtime);
+
+ /* -- hardware description -- */
+ struct snd_pcm_hardware hw;
+ struct snd_pcm_hw_constraints hw_constraints;
+
+ /* -- interrupt callbacks -- */
+ void (*transfer_ack_begin)(struct snd_pcm_substream *substream);
+ void (*transfer_ack_end)(struct snd_pcm_substream *substream);
+
+ /* -- timer -- */
+ unsigned int timer_resolution; /* timer resolution */
+ int tstamp_type; /* timestamp type */
+
+ /* -- DMA -- */
+ unsigned char *dma_area; /* DMA area */
+ dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */
+ size_t dma_bytes; /* size of DMA area */
+
+ struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */
+
+ /* -- audio timestamp config -- */
+ struct snd_pcm_audio_tstamp_config audio_tstamp_config;
+ struct snd_pcm_audio_tstamp_report audio_tstamp_report;
+ struct timespec driver_tstamp;
+
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ /* -- OSS things -- */
+ struct snd_pcm_oss_runtime oss;
+#endif
+
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+ struct snd_pcm_hwptr_log *hwptr_log;
+#endif
+};
+
+struct snd_pcm_group { /* keep linked substreams */
+ spinlock_t lock;
+ struct mutex mutex;
+ struct list_head substreams;
+ int count;
+};
+
+struct pid;
+
+struct snd_pcm_substream {
+ struct snd_pcm *pcm;
+ struct snd_pcm_str *pstr;
+ void *private_data; /* copied from pcm->private_data */
+ int number;
+ char name[32]; /* substream name */
+ int stream; /* stream (direction) */
+ struct pm_qos_request latency_pm_qos_req; /* pm_qos request */
+ size_t buffer_bytes_max; /* limit ring buffer size */
+ struct snd_dma_buffer dma_buffer;
+ size_t dma_max;
+ /* -- hardware operations -- */
+ const struct snd_pcm_ops *ops;
+ /* -- runtime information -- */
+ struct snd_pcm_runtime *runtime;
+ /* -- timer section -- */
+ struct snd_timer *timer; /* timer */
+ unsigned timer_running: 1; /* time is running */
+ /* -- next substream -- */
+ struct snd_pcm_substream *next;
+ /* -- linked substreams -- */
+ struct list_head link_list; /* linked list member */
+ struct snd_pcm_group self_group; /* fake group for non linked substream (with substream lock inside) */
+ struct snd_pcm_group *group; /* pointer to current group */
+ /* -- assigned files -- */
+ void *file;
+ int ref_count;
+ atomic_t mmap_count;
+ unsigned int f_flags;
+ void (*pcm_release)(struct snd_pcm_substream *);
+ struct pid *pid;
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ /* -- OSS things -- */
+ struct snd_pcm_oss_substream oss;
+#endif
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+ struct snd_info_entry *proc_root;
+ struct snd_info_entry *proc_info_entry;
+ struct snd_info_entry *proc_hw_params_entry;
+ struct snd_info_entry *proc_sw_params_entry;
+ struct snd_info_entry *proc_status_entry;
+ struct snd_info_entry *proc_prealloc_entry;
+ struct snd_info_entry *proc_prealloc_max_entry;
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+ struct snd_info_entry *proc_xrun_injection_entry;
+#endif
+#endif /* CONFIG_SND_VERBOSE_PROCFS */
+ /* misc flags */
+ unsigned int hw_opened: 1;
+};
+
+#define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0)
+
+
+struct snd_pcm_str {
+ int stream; /* stream (direction) */
+ struct snd_pcm *pcm;
+ /* -- substreams -- */
+ unsigned int substream_count;
+ unsigned int substream_opened;
+ struct snd_pcm_substream *substream;
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ /* -- OSS things -- */
+ struct snd_pcm_oss_stream oss;
+#endif
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+ struct snd_info_entry *proc_root;
+ struct snd_info_entry *proc_info_entry;
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+ unsigned int xrun_debug; /* 0 = disabled, 1 = verbose, 2 = stacktrace */
+ struct snd_info_entry *proc_xrun_debug_entry;
+#endif
+#endif
+ struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */
+ struct device dev;
+};
+
+struct snd_pcm {
+ struct snd_card *card;
+ struct list_head list;
+ int device; /* device number */
+ unsigned int info_flags;
+ unsigned short dev_class;
+ unsigned short dev_subclass;
+ char id[64];
+ char name[80];
+ struct snd_pcm_str streams[2];
+ struct mutex open_mutex;
+ wait_queue_head_t open_wait;
+ void *private_data;
+ void (*private_free) (struct snd_pcm *pcm);
+ bool internal; /* pcm is for internal use only */
+ bool nonatomic; /* whole PCM operations are in non-atomic context */
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ struct snd_pcm_oss oss;
+#endif
+};
+
+struct snd_pcm_notify {
+ int (*n_register) (struct snd_pcm * pcm);
+ int (*n_disconnect) (struct snd_pcm * pcm);
+ int (*n_unregister) (struct snd_pcm * pcm);
+ struct list_head list;
+};
+
+/*
+ * Registering
+ */
+
+extern const struct file_operations snd_pcm_f_ops[2];
+
+int snd_pcm_new(struct snd_card *card, const char *id, int device,
+ int playback_count, int capture_count,
+ struct snd_pcm **rpcm);
+int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
+ int playback_count, int capture_count,
+ struct snd_pcm **rpcm);
+int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count);
+
+int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree);
+
+/*
+ * Native I/O
+ */
+
+int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info);
+int snd_pcm_info_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_info __user *info);
+int snd_pcm_status(struct snd_pcm_substream *substream,
+ struct snd_pcm_status *status);
+int snd_pcm_start(struct snd_pcm_substream *substream);
+int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status);
+int snd_pcm_drain_done(struct snd_pcm_substream *substream);
+int snd_pcm_stop_xrun(struct snd_pcm_substream *substream);
+#ifdef CONFIG_PM
+int snd_pcm_suspend(struct snd_pcm_substream *substream);
+int snd_pcm_suspend_all(struct snd_pcm *pcm);
+#endif
+int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg);
+int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, struct file *file,
+ struct snd_pcm_substream **rsubstream);
+void snd_pcm_release_substream(struct snd_pcm_substream *substream);
+int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, struct file *file,
+ struct snd_pcm_substream **rsubstream);
+void snd_pcm_detach_substream(struct snd_pcm_substream *substream);
+int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area);
+
+
+#ifdef CONFIG_SND_DEBUG
+void snd_pcm_debug_name(struct snd_pcm_substream *substream,
+ char *name, size_t len);
+#else
+static inline void
+snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
+{
+ *buf = 0;
+}
+#endif
+
+/*
+ * PCM library
+ */
+
+/**
+ * snd_pcm_stream_linked - Check whether the substream is linked with others
+ * @substream: substream to check
+ *
+ * Returns true if the given substream is being linked with others.
+ */
+static inline int snd_pcm_stream_linked(struct snd_pcm_substream *substream)
+{
+ return substream->group != &substream->self_group;
+}
+
+void snd_pcm_stream_lock(struct snd_pcm_substream *substream);
+void snd_pcm_stream_unlock(struct snd_pcm_substream *substream);
+void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream);
+void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream);
+unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream);
+
+/**
+ * snd_pcm_stream_lock_irqsave - Lock the PCM stream
+ * @substream: PCM substream
+ * @flags: irq flags
+ *
+ * This locks the PCM stream like snd_pcm_stream_lock() but with the local
+ * IRQ (only when nonatomic is false). In nonatomic case, this is identical
+ * as snd_pcm_stream_lock().
+ */
+#define snd_pcm_stream_lock_irqsave(substream, flags) \
+ do { \
+ typecheck(unsigned long, flags); \
+ flags = _snd_pcm_stream_lock_irqsave(substream); \
+ } while (0)
+void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
+ unsigned long flags);
+
+/**
+ * snd_pcm_group_for_each_entry - iterate over the linked substreams
+ * @s: the iterator
+ * @substream: the substream
+ *
+ * Iterate over the all linked substreams to the given @substream.
+ * When @substream isn't linked with any others, this gives returns @substream
+ * itself once.
+ */
+#define snd_pcm_group_for_each_entry(s, substream) \
+ list_for_each_entry(s, &substream->group->substreams, link_list)
+
+/**
+ * snd_pcm_running - Check whether the substream is in a running state
+ * @substream: substream to check
+ *
+ * Returns true if the given substream is in the state RUNNING, or in the
+ * state DRAINING for playback.
+ */
+static inline int snd_pcm_running(struct snd_pcm_substream *substream)
+{
+ return (substream->runtime->status->state == SNDRV_PCM_STATE_RUNNING ||
+ (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING &&
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK));
+}
+
+/**
+ * bytes_to_samples - Unit conversion of the size from bytes to samples
+ * @runtime: PCM runtime instance
+ * @size: size in bytes
+ */
+static inline ssize_t bytes_to_samples(struct snd_pcm_runtime *runtime, ssize_t size)
+{
+ return size * 8 / runtime->sample_bits;
+}
+
+/**
+ * bytes_to_frames - Unit conversion of the size from bytes to frames
+ * @runtime: PCM runtime instance
+ * @size: size in bytes
+ */
+static inline snd_pcm_sframes_t bytes_to_frames(struct snd_pcm_runtime *runtime, ssize_t size)
+{
+ return size * 8 / runtime->frame_bits;
+}
+
+/**
+ * samples_to_bytes - Unit conversion of the size from samples to bytes
+ * @runtime: PCM runtime instance
+ * @size: size in samples
+ */
+static inline ssize_t samples_to_bytes(struct snd_pcm_runtime *runtime, ssize_t size)
+{
+ return size * runtime->sample_bits / 8;
+}
+
+/**
+ * frames_to_bytes - Unit conversion of the size from frames to bytes
+ * @runtime: PCM runtime instance
+ * @size: size in frames
+ */
+static inline ssize_t frames_to_bytes(struct snd_pcm_runtime *runtime, snd_pcm_sframes_t size)
+{
+ return size * runtime->frame_bits / 8;
+}
+
+/**
+ * frame_aligned - Check whether the byte size is aligned to frames
+ * @runtime: PCM runtime instance
+ * @bytes: size in bytes
+ */
+static inline int frame_aligned(struct snd_pcm_runtime *runtime, ssize_t bytes)
+{
+ return bytes % runtime->byte_align == 0;
+}
+
+/**
+ * snd_pcm_lib_buffer_bytes - Get the buffer size of the current PCM in bytes
+ * @substream: PCM substream
+ */
+static inline size_t snd_pcm_lib_buffer_bytes(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return frames_to_bytes(runtime, runtime->buffer_size);
+}
+
+/**
+ * snd_pcm_lib_period_bytes - Get the period size of the current PCM in bytes
+ * @substream: PCM substream
+ */
+static inline size_t snd_pcm_lib_period_bytes(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return frames_to_bytes(runtime, runtime->period_size);
+}
+
+/**
+ * snd_pcm_playback_avail - Get the available (writable) space for playback
+ * @runtime: PCM runtime instance
+ *
+ * Result is between 0 ... (boundary - 1)
+ */
+static inline snd_pcm_uframes_t snd_pcm_playback_avail(struct snd_pcm_runtime *runtime)
+{
+ snd_pcm_sframes_t avail = runtime->status->hw_ptr + runtime->buffer_size - runtime->control->appl_ptr;
+ if (avail < 0)
+ avail += runtime->boundary;
+ else if ((snd_pcm_uframes_t) avail >= runtime->boundary)
+ avail -= runtime->boundary;
+ return avail;
+}
+
+/**
+ * snd_pcm_playback_avail - Get the available (readable) space for capture
+ * @runtime: PCM runtime instance
+ *
+ * Result is between 0 ... (boundary - 1)
+ */
+static inline snd_pcm_uframes_t snd_pcm_capture_avail(struct snd_pcm_runtime *runtime)
+{
+ snd_pcm_sframes_t avail = runtime->status->hw_ptr - runtime->control->appl_ptr;
+ if (avail < 0)
+ avail += runtime->boundary;
+ return avail;
+}
+
+/**
+ * snd_pcm_playback_hw_avail - Get the queued space for playback
+ * @runtime: PCM runtime instance
+ */
+static inline snd_pcm_sframes_t snd_pcm_playback_hw_avail(struct snd_pcm_runtime *runtime)
+{
+ return runtime->buffer_size - snd_pcm_playback_avail(runtime);
+}
+
+/**
+ * snd_pcm_capture_hw_avail - Get the free space for capture
+ * @runtime: PCM runtime instance
+ */
+static inline snd_pcm_sframes_t snd_pcm_capture_hw_avail(struct snd_pcm_runtime *runtime)
+{
+ return runtime->buffer_size - snd_pcm_capture_avail(runtime);
+}
+
+/**
+ * snd_pcm_playback_ready - check whether the playback buffer is available
+ * @substream: the pcm substream instance
+ *
+ * Checks whether enough free space is available on the playback buffer.
+ *
+ * Return: Non-zero if available, or zero if not.
+ */
+static inline int snd_pcm_playback_ready(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return snd_pcm_playback_avail(runtime) >= runtime->control->avail_min;
+}
+
+/**
+ * snd_pcm_capture_ready - check whether the capture buffer is available
+ * @substream: the pcm substream instance
+ *
+ * Checks whether enough capture data is available on the capture buffer.
+ *
+ * Return: Non-zero if available, or zero if not.
+ */
+static inline int snd_pcm_capture_ready(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return snd_pcm_capture_avail(runtime) >= runtime->control->avail_min;
+}
+
+/**
+ * snd_pcm_playback_data - check whether any data exists on the playback buffer
+ * @substream: the pcm substream instance
+ *
+ * Checks whether any data exists on the playback buffer.
+ *
+ * Return: Non-zero if any data exists, or zero if not. If stop_threshold
+ * is bigger or equal to boundary, then this function returns always non-zero.
+ */
+static inline int snd_pcm_playback_data(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (runtime->stop_threshold >= runtime->boundary)
+ return 1;
+ return snd_pcm_playback_avail(runtime) < runtime->buffer_size;
+}
+
+/**
+ * snd_pcm_playback_empty - check whether the playback buffer is empty
+ * @substream: the pcm substream instance
+ *
+ * Checks whether the playback buffer is empty.
+ *
+ * Return: Non-zero if empty, or zero if not.
+ */
+static inline int snd_pcm_playback_empty(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return snd_pcm_playback_avail(runtime) >= runtime->buffer_size;
+}
+
+/**
+ * snd_pcm_capture_empty - check whether the capture buffer is empty
+ * @substream: the pcm substream instance
+ *
+ * Checks whether the capture buffer is empty.
+ *
+ * Return: Non-zero if empty, or zero if not.
+ */
+static inline int snd_pcm_capture_empty(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return snd_pcm_capture_avail(runtime) == 0;
+}
+
+/**
+ * snd_pcm_trigger_done - Mark the master substream
+ * @substream: the pcm substream instance
+ * @master: the linked master substream
+ *
+ * When multiple substreams of the same card are linked and the hardware
+ * supports the single-shot operation, the driver calls this in the loop
+ * in snd_pcm_group_for_each_entry() for marking the substream as "done".
+ * Then most of trigger operations are performed only to the given master
+ * substream.
+ *
+ * The trigger_master mark is cleared at timestamp updates at the end
+ * of trigger operations.
+ */
+static inline void snd_pcm_trigger_done(struct snd_pcm_substream *substream,
+ struct snd_pcm_substream *master)
+{
+ substream->runtime->trigger_master = master;
+}
+
+static inline int hw_is_mask(int var)
+{
+ return var >= SNDRV_PCM_HW_PARAM_FIRST_MASK &&
+ var <= SNDRV_PCM_HW_PARAM_LAST_MASK;
+}
+
+static inline int hw_is_interval(int var)
+{
+ return var >= SNDRV_PCM_HW_PARAM_FIRST_INTERVAL &&
+ var <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL;
+}
+
+static inline struct snd_mask *hw_param_mask(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ return &params->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK];
+}
+
+static inline struct snd_interval *hw_param_interval(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ return &params->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL];
+}
+
+static inline const struct snd_mask *hw_param_mask_c(const struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ return &params->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK];
+}
+
+static inline const struct snd_interval *hw_param_interval_c(const struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ return &params->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL];
+}
+
+/**
+ * params_channels - Get the number of channels from the hw params
+ * @p: hw params
+ */
+static inline unsigned int params_channels(const struct snd_pcm_hw_params *p)
+{
+ return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_CHANNELS)->min;
+}
+
+/**
+ * params_rate - Get the sample rate from the hw params
+ * @p: hw params
+ */
+static inline unsigned int params_rate(const struct snd_pcm_hw_params *p)
+{
+ return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_RATE)->min;
+}
+
+/**
+ * params_period_size - Get the period size (in frames) from the hw params
+ * @p: hw params
+ */
+static inline unsigned int params_period_size(const struct snd_pcm_hw_params *p)
+{
+ return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_PERIOD_SIZE)->min;
+}
+
+/**
+ * params_periods - Get the number of periods from the hw params
+ * @p: hw params
+ */
+static inline unsigned int params_periods(const struct snd_pcm_hw_params *p)
+{
+ return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_PERIODS)->min;
+}
+
+/**
+ * params_buffer_size - Get the buffer size (in frames) from the hw params
+ * @p: hw params
+ */
+static inline unsigned int params_buffer_size(const struct snd_pcm_hw_params *p)
+{
+ return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min;
+}
+
+/**
+ * params_buffer_bytes - Get the buffer size (in bytes) from the hw params
+ * @p: hw params
+ */
+static inline unsigned int params_buffer_bytes(const struct snd_pcm_hw_params *p)
+{
+ return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min;
+}
+
+int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v);
+void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c);
+void snd_interval_div(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c);
+void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interval *b,
+ unsigned int k, struct snd_interval *c);
+void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k,
+ const struct snd_interval *b, struct snd_interval *c);
+int snd_interval_list(struct snd_interval *i, unsigned int count,
+ const unsigned int *list, unsigned int mask);
+int snd_interval_ranges(struct snd_interval *i, unsigned int count,
+ const struct snd_interval *list, unsigned int mask);
+int snd_interval_ratnum(struct snd_interval *i,
+ unsigned int rats_count, struct snd_ratnum *rats,
+ unsigned int *nump, unsigned int *denp);
+
+void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params);
+void _snd_pcm_hw_param_setempty(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var);
+int snd_pcm_hw_params_choose(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params);
+
+int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params);
+
+int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream);
+int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream);
+
+int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
+ u_int32_t mask);
+int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
+ u_int64_t mask);
+int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
+ unsigned int min, unsigned int max);
+int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var);
+int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ const struct snd_pcm_hw_constraint_list *l);
+int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ const struct snd_pcm_hw_constraint_ranges *r);
+int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ struct snd_pcm_hw_constraint_ratnums *r);
+int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ struct snd_pcm_hw_constraint_ratdens *r);
+int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ unsigned int width,
+ unsigned int msbits);
+int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ unsigned long step);
+int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var);
+int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime,
+ unsigned int base_rate);
+int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ int var,
+ snd_pcm_hw_rule_func_t func, void *private,
+ int dep, ...);
+
+int snd_pcm_format_signed(snd_pcm_format_t format);
+int snd_pcm_format_unsigned(snd_pcm_format_t format);
+int snd_pcm_format_linear(snd_pcm_format_t format);
+int snd_pcm_format_little_endian(snd_pcm_format_t format);
+int snd_pcm_format_big_endian(snd_pcm_format_t format);
+#if 0 /* just for DocBook */
+/**
+ * snd_pcm_format_cpu_endian - Check the PCM format is CPU-endian
+ * @format: the format to check
+ *
+ * Return: 1 if the given PCM format is CPU-endian, 0 if
+ * opposite, or a negative error code if endian not specified.
+ */
+int snd_pcm_format_cpu_endian(snd_pcm_format_t format);
+#endif /* DocBook */
+#ifdef SNDRV_LITTLE_ENDIAN
+#define snd_pcm_format_cpu_endian(format) snd_pcm_format_little_endian(format)
+#else
+#define snd_pcm_format_cpu_endian(format) snd_pcm_format_big_endian(format)
+#endif
+int snd_pcm_format_width(snd_pcm_format_t format); /* in bits */
+int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */
+ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples);
+const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format);
+int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int frames);
+
+void snd_pcm_set_ops(struct snd_pcm * pcm, int direction,
+ const struct snd_pcm_ops *ops);
+void snd_pcm_set_sync(struct snd_pcm_substream *substream);
+int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg);
+int snd_pcm_update_state(struct snd_pcm_substream *substream,
+ struct snd_pcm_runtime *runtime);
+int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream);
+void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr);
+void snd_pcm_period_elapsed(struct snd_pcm_substream *substream);
+snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream,
+ const void __user *buf,
+ snd_pcm_uframes_t frames);
+snd_pcm_sframes_t snd_pcm_lib_read(struct snd_pcm_substream *substream,
+ void __user *buf, snd_pcm_uframes_t frames);
+snd_pcm_sframes_t snd_pcm_lib_writev(struct snd_pcm_substream *substream,
+ void __user **bufs, snd_pcm_uframes_t frames);
+snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream,
+ void __user **bufs, snd_pcm_uframes_t frames);
+
+extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates;
+
+int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
+unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
+unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
+unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
+ unsigned int rates_b);
+
+/**
+ * snd_pcm_set_runtime_buffer - Set the PCM runtime buffer
+ * @substream: PCM substream to set
+ * @bufp: the buffer information, NULL to clear
+ *
+ * Copy the buffer information to runtime->dma_buffer when @bufp is non-NULL.
+ * Otherwise it clears the current buffer information.
+ */
+static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream,
+ struct snd_dma_buffer *bufp)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (bufp) {
+ runtime->dma_buffer_p = bufp;
+ runtime->dma_area = bufp->area;
+ runtime->dma_addr = bufp->addr;
+ runtime->dma_bytes = bufp->bytes;
+ } else {
+ runtime->dma_buffer_p = NULL;
+ runtime->dma_area = NULL;
+ runtime->dma_addr = 0;
+ runtime->dma_bytes = 0;
+ }
+}
+
+/*
+ * Timer interface
+ */
+
+void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream);
+void snd_pcm_timer_init(struct snd_pcm_substream *substream);
+void snd_pcm_timer_done(struct snd_pcm_substream *substream);
+
+/**
+ * snd_pcm_gettime - Fill the timespec depending on the timestamp mode
+ * @runtime: PCM runtime instance
+ * @tv: timespec to fill
+ */
+static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime,
+ struct timespec *tv)
+{
+ switch (runtime->tstamp_type) {
+ case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC:
+ ktime_get_ts(tv);
+ break;
+ case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW:
+ getrawmonotonic(tv);
+ break;
+ default:
+ getnstimeofday(tv);
+ break;
+ }
+}
+
+/*
+ * Memory
+ */
+
+int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream);
+int snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm);
+int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream,
+ int type, struct device *data,
+ size_t size, size_t max);
+int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
+ int type, void *data,
+ size_t size, size_t max);
+int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size);
+int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream);
+
+int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream,
+ size_t size, gfp_t gfp_flags);
+int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream);
+struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream,
+ unsigned long offset);
+/**
+ * snd_pcm_lib_alloc_vmalloc_buffer - allocate virtual DMA buffer
+ * @substream: the substream to allocate the buffer to
+ * @size: the requested buffer size, in bytes
+ *
+ * Allocates the PCM substream buffer using vmalloc(), i.e., the memory is
+ * contiguous in kernel virtual space, but not in physical memory. Use this
+ * if the buffer is accessed by kernel code but not by device DMA.
+ *
+ * Return: 1 if the buffer was changed, 0 if not changed, or a negative error
+ * code.
+ */
+static inline int snd_pcm_lib_alloc_vmalloc_buffer
+ (struct snd_pcm_substream *substream, size_t size)
+{
+ return _snd_pcm_lib_alloc_vmalloc_buffer(substream, size,
+ GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO);
+}
+
+/**
+ * snd_pcm_lib_alloc_vmalloc_32_buffer - allocate 32-bit-addressable buffer
+ * @substream: the substream to allocate the buffer to
+ * @size: the requested buffer size, in bytes
+ *
+ * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses
+ * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory.
+ *
+ * Return: 1 if the buffer was changed, 0 if not changed, or a negative error
+ * code.
+ */
+static inline int snd_pcm_lib_alloc_vmalloc_32_buffer
+ (struct snd_pcm_substream *substream, size_t size)
+{
+ return _snd_pcm_lib_alloc_vmalloc_buffer(substream, size,
+ GFP_KERNEL | GFP_DMA32 | __GFP_ZERO);
+}
+
+#define snd_pcm_get_dma_buf(substream) ((substream)->runtime->dma_buffer_p)
+
+#ifdef CONFIG_SND_DMA_SGBUF
+/*
+ * SG-buffer handling
+ */
+#define snd_pcm_substream_sgbuf(substream) \
+ snd_pcm_get_dma_buf(substream)->private_data
+
+struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream,
+ unsigned long offset);
+#else /* !SND_DMA_SGBUF */
+/*
+ * fake using a continuous buffer
+ */
+#define snd_pcm_sgbuf_ops_page NULL
+#endif /* SND_DMA_SGBUF */
+
+/**
+ * snd_pcm_sgbuf_get_addr - Get the DMA address at the corresponding offset
+ * @substream: PCM substream
+ * @ofs: byte offset
+ */
+static inline dma_addr_t
+snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
+{
+ return snd_sgbuf_get_addr(snd_pcm_get_dma_buf(substream), ofs);
+}
+
+/**
+ * snd_pcm_sgbuf_get_ptr - Get the virtual address at the corresponding offset
+ * @substream: PCM substream
+ * @ofs: byte offset
+ */
+static inline void *
+snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs)
+{
+ return snd_sgbuf_get_ptr(snd_pcm_get_dma_buf(substream), ofs);
+}
+
+/**
+ * snd_pcm_sgbuf_chunk_size - Compute the max size that fits within the contig.
+ * page from the given size
+ * @substream: PCM substream
+ * @ofs: byte offset
+ * @size: byte size to examine
+ */
+static inline unsigned int
+snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
+ unsigned int ofs, unsigned int size)
+{
+ return snd_sgbuf_get_chunk_size(snd_pcm_get_dma_buf(substream), ofs, size);
+}
+
+/**
+ * snd_pcm_mmap_data_open - increase the mmap counter
+ * @area: VMA
+ *
+ * PCM mmap callback should handle this counter properly
+ */
+static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area)
+{
+ struct snd_pcm_substream *substream = (struct snd_pcm_substream *)area->vm_private_data;
+ atomic_inc(&substream->mmap_count);
+}
+
+/**
+ * snd_pcm_mmap_data_close - decrease the mmap counter
+ * @area: VMA
+ *
+ * PCM mmap callback should handle this counter properly
+ */
+static inline void snd_pcm_mmap_data_close(struct vm_area_struct *area)
+{
+ struct snd_pcm_substream *substream = (struct snd_pcm_substream *)area->vm_private_data;
+ atomic_dec(&substream->mmap_count);
+}
+
+int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *area);
+/* mmap for io-memory area */
+#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA)
+#define SNDRV_PCM_INFO_MMAP_IOMEM SNDRV_PCM_INFO_MMAP
+int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area);
+#else
+#define SNDRV_PCM_INFO_MMAP_IOMEM 0
+#define snd_pcm_lib_mmap_iomem NULL
+#endif
+
+#define snd_pcm_lib_mmap_vmalloc NULL
+
+/**
+ * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer
+ * @dma: DMA number
+ * @max: pointer to store the max size
+ */
+static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
+{
+ *max = dma < 4 ? 64 * 1024 : 128 * 1024;
+}
+
+/*
+ * Misc
+ */
+
+#define SNDRV_PCM_DEFAULT_CON_SPDIF (IEC958_AES0_CON_EMPHASIS_NONE|\
+ (IEC958_AES1_CON_ORIGINAL<<8)|\
+ (IEC958_AES1_CON_PCM_CODER<<8)|\
+ (IEC958_AES3_CON_FS_48000<<24))
+
+#define PCM_RUNTIME_CHECK(sub) snd_BUG_ON(!(sub) || !(sub)->runtime)
+
+const char *snd_pcm_format_name(snd_pcm_format_t format);
+
+/**
+ * snd_pcm_stream_str - Get a string naming the direction of a stream
+ * @substream: the pcm substream instance
+ *
+ * Return: A string naming the direction of the stream.
+ */
+static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return "Playback";
+ else
+ return "Capture";
+}
+
+/*
+ * PCM channel-mapping control API
+ */
+/* array element of channel maps */
+struct snd_pcm_chmap_elem {
+ unsigned char channels;
+ unsigned char map[15];
+};
+
+/* channel map information; retrieved via snd_kcontrol_chip() */
+struct snd_pcm_chmap {
+ struct snd_pcm *pcm; /* assigned PCM instance */
+ int stream; /* PLAYBACK or CAPTURE */
+ struct snd_kcontrol *kctl;
+ const struct snd_pcm_chmap_elem *chmap;
+ unsigned int max_channels;
+ unsigned int channel_mask; /* optional: active channels bitmask */
+ void *private_data; /* optional: private data pointer */
+};
+
+/**
+ * snd_pcm_chmap_substream - get the PCM substream assigned to the given chmap info
+ * @info: chmap information
+ * @idx: the substream number index
+ */
+static inline struct snd_pcm_substream *
+snd_pcm_chmap_substream(struct snd_pcm_chmap *info, unsigned int idx)
+{
+ struct snd_pcm_substream *s;
+ for (s = info->pcm->streams[info->stream].substream; s; s = s->next)
+ if (s->number == idx)
+ return s;
+ return NULL;
+}
+
+/* ALSA-standard channel maps (RL/RR prior to C/LFE) */
+extern const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[];
+/* Other world's standard channel maps (C/LFE prior to RL/RR) */
+extern const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[];
+
+/* bit masks to be passed to snd_pcm_chmap.channel_mask field */
+#define SND_PCM_CHMAP_MASK_24 ((1U << 2) | (1U << 4))
+#define SND_PCM_CHMAP_MASK_246 (SND_PCM_CHMAP_MASK_24 | (1U << 6))
+#define SND_PCM_CHMAP_MASK_2468 (SND_PCM_CHMAP_MASK_246 | (1U << 8))
+
+int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
+ const struct snd_pcm_chmap_elem *chmap,
+ int max_channels,
+ unsigned long private_value,
+ struct snd_pcm_chmap **info_ret);
+
+/**
+ * pcm_format_to_bits - Strong-typed conversion of pcm_format to bitwise
+ * @pcm_format: PCM format
+ */
+static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format)
+{
+ return 1ULL << (__force int) pcm_format;
+}
+
+/* printk helpers */
+#define pcm_err(pcm, fmt, args...) \
+ dev_err((pcm)->card->dev, fmt, ##args)
+#define pcm_warn(pcm, fmt, args...) \
+ dev_warn((pcm)->card->dev, fmt, ##args)
+#define pcm_dbg(pcm, fmt, args...) \
+ dev_dbg((pcm)->card->dev, fmt, ##args)
+
+#endif /* __SOUND_PCM_H */
diff --git a/kernel/include/sound/pcm_oss.h b/kernel/include/sound/pcm_oss.h
new file mode 100644
index 000000000..760c969d8
--- /dev/null
+++ b/kernel/include/sound/pcm_oss.h
@@ -0,0 +1,89 @@
+#ifndef __SOUND_PCM_OSS_H
+#define __SOUND_PCM_OSS_H
+
+/*
+ * Digital Audio (PCM) - OSS compatibility abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+struct snd_pcm_oss_setup {
+ char *task_name;
+ unsigned int disable:1,
+ direct:1,
+ block:1,
+ nonblock:1,
+ partialfrag:1,
+ nosilence:1,
+ buggyptr:1;
+ unsigned int periods;
+ unsigned int period_size;
+ struct snd_pcm_oss_setup *next;
+};
+
+struct snd_pcm_oss_runtime {
+ unsigned params: 1, /* format/parameter change */
+ prepare: 1, /* need to prepare the operation */
+ trigger: 1, /* trigger flag */
+ sync_trigger: 1; /* sync trigger flag */
+ int rate; /* requested rate */
+ int format; /* requested OSS format */
+ unsigned int channels; /* requested channels */
+ unsigned int fragshift;
+ unsigned int maxfrags;
+ unsigned int subdivision; /* requested subdivision */
+ size_t period_bytes; /* requested period size */
+ size_t period_frames; /* period frames for poll */
+ size_t period_ptr; /* actual write pointer to period */
+ unsigned int periods;
+ size_t buffer_bytes; /* requested buffer size */
+ size_t bytes; /* total # bytes processed */
+ size_t mmap_bytes;
+ char *buffer; /* vmallocated period */
+ size_t buffer_used; /* used length from period buffer */
+ struct mutex params_lock;
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+ struct snd_pcm_plugin *plugin_first;
+ struct snd_pcm_plugin *plugin_last;
+#endif
+ unsigned int prev_hw_ptr_period;
+};
+
+struct snd_pcm_oss_file {
+ struct snd_pcm_substream *streams[2];
+};
+
+struct snd_pcm_oss_substream {
+ unsigned oss: 1; /* oss mode */
+ struct snd_pcm_oss_setup setup; /* active setup */
+};
+
+struct snd_pcm_oss_stream {
+ struct snd_pcm_oss_setup *setup_list; /* setup list */
+ struct mutex setup_mutex;
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+ struct snd_info_entry *proc_entry;
+#endif
+};
+
+struct snd_pcm_oss {
+ int reg;
+ unsigned int reg_mask;
+};
+
+#endif /* __SOUND_PCM_OSS_H */
diff --git a/kernel/include/sound/pcm_params.h b/kernel/include/sound/pcm_params.h
new file mode 100644
index 000000000..c70435777
--- /dev/null
+++ b/kernel/include/sound/pcm_params.h
@@ -0,0 +1,376 @@
+#ifndef __SOUND_PCM_PARAMS_H
+#define __SOUND_PCM_PARAMS_H
+
+/*
+ * PCM params helpers
+ * Copyright (c) by Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/pcm.h>
+
+int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, int *dir);
+int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, int *dir);
+int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, int *dir);
+
+#define SNDRV_MASK_BITS 64 /* we use so far 64bits only */
+#define SNDRV_MASK_SIZE (SNDRV_MASK_BITS / 32)
+#define MASK_OFS(i) ((i) >> 5)
+#define MASK_BIT(i) (1U << ((i) & 31))
+
+static inline size_t snd_mask_sizeof(void)
+{
+ return sizeof(struct snd_mask);
+}
+
+static inline void snd_mask_none(struct snd_mask *mask)
+{
+ memset(mask, 0, sizeof(*mask));
+}
+
+static inline void snd_mask_any(struct snd_mask *mask)
+{
+ memset(mask, 0xff, SNDRV_MASK_SIZE * sizeof(u_int32_t));
+}
+
+static inline int snd_mask_empty(const struct snd_mask *mask)
+{
+ int i;
+ for (i = 0; i < SNDRV_MASK_SIZE; i++)
+ if (mask->bits[i])
+ return 0;
+ return 1;
+}
+
+static inline unsigned int snd_mask_min(const struct snd_mask *mask)
+{
+ int i;
+ for (i = 0; i < SNDRV_MASK_SIZE; i++) {
+ if (mask->bits[i])
+ return __ffs(mask->bits[i]) + (i << 5);
+ }
+ return 0;
+}
+
+static inline unsigned int snd_mask_max(const struct snd_mask *mask)
+{
+ int i;
+ for (i = SNDRV_MASK_SIZE - 1; i >= 0; i--) {
+ if (mask->bits[i])
+ return __fls(mask->bits[i]) + (i << 5);
+ }
+ return 0;
+}
+
+static inline void snd_mask_set(struct snd_mask *mask, unsigned int val)
+{
+ mask->bits[MASK_OFS(val)] |= MASK_BIT(val);
+}
+
+static inline void snd_mask_reset(struct snd_mask *mask, unsigned int val)
+{
+ mask->bits[MASK_OFS(val)] &= ~MASK_BIT(val);
+}
+
+static inline void snd_mask_set_range(struct snd_mask *mask,
+ unsigned int from, unsigned int to)
+{
+ unsigned int i;
+ for (i = from; i <= to; i++)
+ mask->bits[MASK_OFS(i)] |= MASK_BIT(i);
+}
+
+static inline void snd_mask_reset_range(struct snd_mask *mask,
+ unsigned int from, unsigned int to)
+{
+ unsigned int i;
+ for (i = from; i <= to; i++)
+ mask->bits[MASK_OFS(i)] &= ~MASK_BIT(i);
+}
+
+static inline void snd_mask_leave(struct snd_mask *mask, unsigned int val)
+{
+ unsigned int v;
+ v = mask->bits[MASK_OFS(val)] & MASK_BIT(val);
+ snd_mask_none(mask);
+ mask->bits[MASK_OFS(val)] = v;
+}
+
+static inline void snd_mask_intersect(struct snd_mask *mask,
+ const struct snd_mask *v)
+{
+ int i;
+ for (i = 0; i < SNDRV_MASK_SIZE; i++)
+ mask->bits[i] &= v->bits[i];
+}
+
+static inline int snd_mask_eq(const struct snd_mask *mask,
+ const struct snd_mask *v)
+{
+ return ! memcmp(mask, v, SNDRV_MASK_SIZE * sizeof(u_int32_t));
+}
+
+static inline void snd_mask_copy(struct snd_mask *mask,
+ const struct snd_mask *v)
+{
+ *mask = *v;
+}
+
+static inline int snd_mask_test(const struct snd_mask *mask, unsigned int val)
+{
+ return mask->bits[MASK_OFS(val)] & MASK_BIT(val);
+}
+
+static inline int snd_mask_single(const struct snd_mask *mask)
+{
+ int i, c = 0;
+ for (i = 0; i < SNDRV_MASK_SIZE; i++) {
+ if (! mask->bits[i])
+ continue;
+ if (mask->bits[i] & (mask->bits[i] - 1))
+ return 0;
+ if (c)
+ return 0;
+ c++;
+ }
+ return 1;
+}
+
+static inline int snd_mask_refine(struct snd_mask *mask,
+ const struct snd_mask *v)
+{
+ struct snd_mask old;
+ snd_mask_copy(&old, mask);
+ snd_mask_intersect(mask, v);
+ if (snd_mask_empty(mask))
+ return -EINVAL;
+ return !snd_mask_eq(mask, &old);
+}
+
+static inline int snd_mask_refine_first(struct snd_mask *mask)
+{
+ if (snd_mask_single(mask))
+ return 0;
+ snd_mask_leave(mask, snd_mask_min(mask));
+ return 1;
+}
+
+static inline int snd_mask_refine_last(struct snd_mask *mask)
+{
+ if (snd_mask_single(mask))
+ return 0;
+ snd_mask_leave(mask, snd_mask_max(mask));
+ return 1;
+}
+
+static inline int snd_mask_refine_min(struct snd_mask *mask, unsigned int val)
+{
+ if (snd_mask_min(mask) >= val)
+ return 0;
+ snd_mask_reset_range(mask, 0, val - 1);
+ if (snd_mask_empty(mask))
+ return -EINVAL;
+ return 1;
+}
+
+static inline int snd_mask_refine_max(struct snd_mask *mask, unsigned int val)
+{
+ if (snd_mask_max(mask) <= val)
+ return 0;
+ snd_mask_reset_range(mask, val + 1, SNDRV_MASK_BITS);
+ if (snd_mask_empty(mask))
+ return -EINVAL;
+ return 1;
+}
+
+static inline int snd_mask_refine_set(struct snd_mask *mask, unsigned int val)
+{
+ int changed;
+ changed = !snd_mask_single(mask);
+ snd_mask_leave(mask, val);
+ if (snd_mask_empty(mask))
+ return -EINVAL;
+ return changed;
+}
+
+static inline int snd_mask_value(const struct snd_mask *mask)
+{
+ return snd_mask_min(mask);
+}
+
+static inline void snd_interval_any(struct snd_interval *i)
+{
+ i->min = 0;
+ i->openmin = 0;
+ i->max = UINT_MAX;
+ i->openmax = 0;
+ i->integer = 0;
+ i->empty = 0;
+}
+
+static inline void snd_interval_none(struct snd_interval *i)
+{
+ i->empty = 1;
+}
+
+static inline int snd_interval_checkempty(const struct snd_interval *i)
+{
+ return (i->min > i->max ||
+ (i->min == i->max && (i->openmin || i->openmax)));
+}
+
+static inline int snd_interval_empty(const struct snd_interval *i)
+{
+ return i->empty;
+}
+
+static inline int snd_interval_single(const struct snd_interval *i)
+{
+ return (i->min == i->max ||
+ (i->min + 1 == i->max && i->openmax));
+}
+
+static inline int snd_interval_value(const struct snd_interval *i)
+{
+ return i->min;
+}
+
+static inline int snd_interval_min(const struct snd_interval *i)
+{
+ return i->min;
+}
+
+static inline int snd_interval_max(const struct snd_interval *i)
+{
+ unsigned int v;
+ v = i->max;
+ if (i->openmax)
+ v--;
+ return v;
+}
+
+static inline int snd_interval_test(const struct snd_interval *i, unsigned int val)
+{
+ return !((i->min > val || (i->min == val && i->openmin) ||
+ i->max < val || (i->max == val && i->openmax)));
+}
+
+static inline void snd_interval_copy(struct snd_interval *d, const struct snd_interval *s)
+{
+ *d = *s;
+}
+
+static inline int snd_interval_setinteger(struct snd_interval *i)
+{
+ if (i->integer)
+ return 0;
+ if (i->openmin && i->openmax && i->min == i->max)
+ return -EINVAL;
+ i->integer = 1;
+ return 1;
+}
+
+static inline int snd_interval_eq(const struct snd_interval *i1, const struct snd_interval *i2)
+{
+ if (i1->empty)
+ return i2->empty;
+ if (i2->empty)
+ return i1->empty;
+ return i1->min == i2->min && i1->openmin == i2->openmin &&
+ i1->max == i2->max && i1->openmax == i2->openmax;
+}
+
+/**
+ * params_access - get the access type from the hw params
+ * @p: hw params
+ */
+static inline snd_pcm_access_t params_access(const struct snd_pcm_hw_params *p)
+{
+ return (__force snd_pcm_access_t)snd_mask_min(hw_param_mask_c(p,
+ SNDRV_PCM_HW_PARAM_ACCESS));
+}
+
+/**
+ * params_format - get the sample format from the hw params
+ * @p: hw params
+ */
+static inline snd_pcm_format_t params_format(const struct snd_pcm_hw_params *p)
+{
+ return (__force snd_pcm_format_t)snd_mask_min(hw_param_mask_c(p,
+ SNDRV_PCM_HW_PARAM_FORMAT));
+}
+
+/**
+ * params_subformat - get the sample subformat from the hw params
+ * @p: hw params
+ */
+static inline snd_pcm_subformat_t
+params_subformat(const struct snd_pcm_hw_params *p)
+{
+ return (__force snd_pcm_subformat_t)snd_mask_min(hw_param_mask_c(p,
+ SNDRV_PCM_HW_PARAM_SUBFORMAT));
+}
+
+/**
+ * params_period_bytes - get the period size (in bytes) from the hw params
+ * @p: hw params
+ */
+static inline unsigned int
+params_period_bytes(const struct snd_pcm_hw_params *p)
+{
+ return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_PERIOD_BYTES)->min;
+}
+
+/**
+ * params_width - get the number of bits of the sample format from the hw params
+ * @p: hw params
+ *
+ * This function returns the number of bits per sample that the selected sample
+ * format of the hw params has.
+ */
+static inline int params_width(const struct snd_pcm_hw_params *p)
+{
+ return snd_pcm_format_width(params_format(p));
+}
+
+/*
+ * params_physical_width - get the storage size of the sample format from the hw params
+ * @p: hw params
+ *
+ * This functions returns the number of bits per sample that the selected sample
+ * format of the hw params takes up in memory. This will be equal or larger than
+ * params_width().
+ */
+static inline int params_physical_width(const struct snd_pcm_hw_params *p)
+{
+ return snd_pcm_format_physical_width(params_format(p));
+}
+
+static inline void
+params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt)
+{
+ snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT),
+ (__force int)fmt);
+}
+
+#endif /* __SOUND_PCM_PARAMS_H */
diff --git a/kernel/include/sound/pt2258.h b/kernel/include/sound/pt2258.h
new file mode 100644
index 000000000..160f812fa
--- /dev/null
+++ b/kernel/include/sound/pt2258.h
@@ -0,0 +1,37 @@
+/*
+ * ALSA Driver for the PT2258 volume controller.
+ *
+ * Copyright (c) 2006 Jochen Voss <voss@seehuhn.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifndef __SOUND_PT2258_H
+#define __SOUND_PT2258_H
+
+struct snd_pt2258 {
+ struct snd_card *card;
+ struct snd_i2c_bus *i2c_bus;
+ struct snd_i2c_device *i2c_dev;
+
+ unsigned char volume[6];
+ int mute;
+};
+
+extern int snd_pt2258_reset(struct snd_pt2258 *pt);
+extern int snd_pt2258_build_controls(struct snd_pt2258 *pt);
+
+#endif /* __SOUND_PT2258_H */
diff --git a/kernel/include/sound/pxa2xx-lib.h b/kernel/include/sound/pxa2xx-lib.h
new file mode 100644
index 000000000..56e818e4a
--- /dev/null
+++ b/kernel/include/sound/pxa2xx-lib.h
@@ -0,0 +1,38 @@
+#ifndef PXA2XX_LIB_H
+#define PXA2XX_LIB_H
+
+#include <linux/platform_device.h>
+#include <sound/ac97_codec.h>
+
+/* PCM */
+
+extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params);
+extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
+extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream);
+extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream);
+extern void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id);
+extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream);
+extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma);
+extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream);
+extern void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm);
+
+/* AC97 */
+
+extern unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg);
+extern void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val);
+
+extern bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97);
+extern bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97);
+extern void pxa2xx_ac97_finish_reset(struct snd_ac97 *ac97);
+
+extern int pxa2xx_ac97_hw_suspend(void);
+extern int pxa2xx_ac97_hw_resume(void);
+
+extern int pxa2xx_ac97_hw_probe(struct platform_device *dev);
+extern void pxa2xx_ac97_hw_remove(struct platform_device *dev);
+
+#endif
diff --git a/kernel/include/sound/rawmidi.h b/kernel/include/sound/rawmidi.h
new file mode 100644
index 000000000..f6cbef78d
--- /dev/null
+++ b/kernel/include/sound/rawmidi.h
@@ -0,0 +1,189 @@
+#ifndef __SOUND_RAWMIDI_H
+#define __SOUND_RAWMIDI_H
+
+/*
+ * Abstract layer for MIDI v1.0 stream
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/asound.h>
+#include <linux/interrupt.h>
+#include <linux/spinlock.h>
+#include <linux/wait.h>
+#include <linux/mutex.h>
+#include <linux/workqueue.h>
+#include <linux/device.h>
+
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+#include <sound/seq_device.h>
+#endif
+
+/*
+ * Raw MIDI interface
+ */
+
+#define SNDRV_RAWMIDI_DEVICES 8
+
+#define SNDRV_RAWMIDI_LFLG_OUTPUT (1<<0)
+#define SNDRV_RAWMIDI_LFLG_INPUT (1<<1)
+#define SNDRV_RAWMIDI_LFLG_OPEN (3<<0)
+#define SNDRV_RAWMIDI_LFLG_APPEND (1<<2)
+
+struct snd_rawmidi;
+struct snd_rawmidi_substream;
+struct snd_seq_port_info;
+struct pid;
+
+struct snd_rawmidi_ops {
+ int (*open) (struct snd_rawmidi_substream * substream);
+ int (*close) (struct snd_rawmidi_substream * substream);
+ void (*trigger) (struct snd_rawmidi_substream * substream, int up);
+ void (*drain) (struct snd_rawmidi_substream * substream);
+};
+
+struct snd_rawmidi_global_ops {
+ int (*dev_register) (struct snd_rawmidi * rmidi);
+ int (*dev_unregister) (struct snd_rawmidi * rmidi);
+ void (*get_port_info)(struct snd_rawmidi *rmidi, int number,
+ struct snd_seq_port_info *info);
+};
+
+struct snd_rawmidi_runtime {
+ struct snd_rawmidi_substream *substream;
+ unsigned int drain: 1, /* drain stage */
+ oss: 1; /* OSS compatible mode */
+ /* midi stream buffer */
+ unsigned char *buffer; /* buffer for MIDI data */
+ size_t buffer_size; /* size of buffer */
+ size_t appl_ptr; /* application pointer */
+ size_t hw_ptr; /* hardware pointer */
+ size_t avail_min; /* min avail for wakeup */
+ size_t avail; /* max used buffer for wakeup */
+ size_t xruns; /* over/underruns counter */
+ /* misc */
+ spinlock_t lock;
+ wait_queue_head_t sleep;
+ /* event handler (new bytes, input only) */
+ void (*event)(struct snd_rawmidi_substream *substream);
+ /* defers calls to event [input] or ops->trigger [output] */
+ struct work_struct event_work;
+ /* private data */
+ void *private_data;
+ void (*private_free)(struct snd_rawmidi_substream *substream);
+};
+
+struct snd_rawmidi_substream {
+ struct list_head list; /* list of all substream for given stream */
+ int stream; /* direction */
+ int number; /* substream number */
+ unsigned int opened: 1, /* open flag */
+ append: 1, /* append flag (merge more streams) */
+ active_sensing: 1; /* send active sensing when close */
+ int use_count; /* use counter (for output) */
+ size_t bytes;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_str *pstr;
+ char name[32];
+ struct snd_rawmidi_runtime *runtime;
+ struct pid *pid;
+ /* hardware layer */
+ struct snd_rawmidi_ops *ops;
+};
+
+struct snd_rawmidi_file {
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *input;
+ struct snd_rawmidi_substream *output;
+};
+
+struct snd_rawmidi_str {
+ unsigned int substream_count;
+ unsigned int substream_opened;
+ struct list_head substreams;
+};
+
+struct snd_rawmidi {
+ struct snd_card *card;
+ struct list_head list;
+ unsigned int device; /* device number */
+ unsigned int info_flags; /* SNDRV_RAWMIDI_INFO_XXXX */
+ char id[64];
+ char name[80];
+
+#ifdef CONFIG_SND_OSSEMUL
+ int ossreg;
+#endif
+
+ struct snd_rawmidi_global_ops *ops;
+
+ struct snd_rawmidi_str streams[2];
+
+ void *private_data;
+ void (*private_free) (struct snd_rawmidi *rmidi);
+
+ struct mutex open_mutex;
+ wait_queue_head_t open_wait;
+
+ struct device dev;
+
+ struct snd_info_entry *proc_entry;
+
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+ struct snd_seq_device *seq_dev;
+#endif
+};
+
+/* main rawmidi functions */
+
+int snd_rawmidi_new(struct snd_card *card, char *id, int device,
+ int output_count, int input_count,
+ struct snd_rawmidi **rmidi);
+void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream,
+ struct snd_rawmidi_ops *ops);
+
+/* callbacks */
+
+int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
+ const unsigned char *buffer, int count);
+int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream);
+int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
+ unsigned char *buffer, int count);
+int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count);
+int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
+ unsigned char *buffer, int count);
+
+/* main midi functions */
+
+int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info);
+int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice,
+ int mode, struct snd_rawmidi_file *rfile);
+int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile);
+int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream,
+ struct snd_rawmidi_params *params);
+int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream,
+ struct snd_rawmidi_params *params);
+int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream);
+int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream);
+int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream);
+long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream,
+ unsigned char *buf, long count);
+long snd_rawmidi_kernel_write(struct snd_rawmidi_substream *substream,
+ const unsigned char *buf, long count);
+
+#endif /* __SOUND_RAWMIDI_H */
diff --git a/kernel/include/sound/rcar_snd.h b/kernel/include/sound/rcar_snd.h
new file mode 100644
index 000000000..4cecd0c17
--- /dev/null
+++ b/kernel/include/sound/rcar_snd.h
@@ -0,0 +1,104 @@
+/*
+ * Renesas R-Car SRU/SCU/SSIU/SSI support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef RCAR_SND_H
+#define RCAR_SND_H
+
+#include <linux/sh_clk.h>
+
+#define RSND_GEN1_SRU 0
+#define RSND_GEN1_ADG 1
+#define RSND_GEN1_SSI 2
+
+#define RSND_GEN2_SCU 0
+#define RSND_GEN2_ADG 1
+#define RSND_GEN2_SSIU 2
+#define RSND_GEN2_SSI 3
+
+#define RSND_BASE_MAX 4
+
+/*
+ * flags
+ *
+ * 0xAB000000
+ *
+ * A : clock sharing settings
+ * B : SSI direction
+ */
+#define RSND_SSI_CLK_PIN_SHARE (1 << 31)
+#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */
+
+#define RSND_SSI(_dma_id, _irq, _flags) \
+{ .dma_id = _dma_id, .irq = _irq, .flags = _flags }
+#define RSND_SSI_UNUSED \
+{ .dma_id = -1, .irq = -1, .flags = 0 }
+
+struct rsnd_ssi_platform_info {
+ int dma_id;
+ int irq;
+ u32 flags;
+};
+
+#define RSND_SRC(rate, _dma_id) \
+{ .convert_rate = rate, .dma_id = _dma_id, }
+#define RSND_SRC_UNUSED \
+{ .convert_rate = 0, .dma_id = -1, }
+
+struct rsnd_src_platform_info {
+ u32 convert_rate; /* sampling rate convert */
+ int dma_id; /* for Gen2 SCU */
+ int irq;
+};
+
+/*
+ * flags
+ */
+struct rsnd_dvc_platform_info {
+ u32 flags;
+};
+
+struct rsnd_dai_path_info {
+ struct rsnd_ssi_platform_info *ssi;
+ struct rsnd_src_platform_info *src;
+ struct rsnd_dvc_platform_info *dvc;
+};
+
+struct rsnd_dai_platform_info {
+ struct rsnd_dai_path_info playback;
+ struct rsnd_dai_path_info capture;
+};
+
+/*
+ * flags
+ *
+ * 0x0000000A
+ *
+ * A : generation
+ */
+#define RSND_GEN_MASK (0xF << 0)
+#define RSND_GEN1 (1 << 0) /* fixme */
+#define RSND_GEN2 (2 << 0) /* fixme */
+
+struct rcar_snd_info {
+ u32 flags;
+ struct rsnd_ssi_platform_info *ssi_info;
+ int ssi_info_nr;
+ struct rsnd_src_platform_info *src_info;
+ int src_info_nr;
+ struct rsnd_dvc_platform_info *dvc_info;
+ int dvc_info_nr;
+ struct rsnd_dai_platform_info *dai_info;
+ int dai_info_nr;
+ int (*start)(int id);
+ int (*stop)(int id);
+};
+
+#endif
diff --git a/kernel/include/sound/rt286.h b/kernel/include/sound/rt286.h
new file mode 100644
index 000000000..eb773d148
--- /dev/null
+++ b/kernel/include/sound/rt286.h
@@ -0,0 +1,19 @@
+/*
+ * linux/sound/rt286.h -- Platform data for RT286
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT286_H
+#define __LINUX_SND_RT286_H
+
+struct rt286_platform_data {
+ bool cbj_en; /*combo jack enable*/
+ bool gpio2_en; /*GPIO2 enable*/
+};
+
+#endif
diff --git a/kernel/include/sound/rt5640.h b/kernel/include/sound/rt5640.h
new file mode 100644
index 000000000..59d26dd81
--- /dev/null
+++ b/kernel/include/sound/rt5640.h
@@ -0,0 +1,26 @@
+/*
+ * linux/sound/rt5640.h -- Platform data for RT5640
+ *
+ * Copyright 2011 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5640_H
+#define __LINUX_SND_RT5640_H
+
+struct rt5640_platform_data {
+ /* IN1 & IN2 can optionally be differential */
+ bool in1_diff;
+ bool in2_diff;
+
+ bool dmic_en;
+ bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */
+ bool dmic2_data_pin; /* 0 = IN1N; 1 = GPIO4 */
+
+ int ldo1_en; /* GPIO for LDO1_EN */
+};
+
+#endif
diff --git a/kernel/include/sound/rt5645.h b/kernel/include/sound/rt5645.h
new file mode 100644
index 000000000..120d96100
--- /dev/null
+++ b/kernel/include/sound/rt5645.h
@@ -0,0 +1,32 @@
+/*
+ * linux/sound/rt5645.h -- Platform data for RT5645
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5645_H
+#define __LINUX_SND_RT5645_H
+
+struct rt5645_platform_data {
+ /* IN2 can optionally be differential */
+ bool in2_diff;
+
+ bool dmic_en;
+ unsigned int dmic1_data_pin;
+ /* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */
+ unsigned int dmic2_data_pin;
+ /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */
+
+ unsigned int hp_det_gpio;
+ bool gpio_hp_det_active_high;
+
+ /* true if codec's jd function is used */
+ bool en_jd_func;
+ unsigned int jd_mode;
+};
+
+#endif
diff --git a/kernel/include/sound/rt5651.h b/kernel/include/sound/rt5651.h
new file mode 100644
index 000000000..d35de758d
--- /dev/null
+++ b/kernel/include/sound/rt5651.h
@@ -0,0 +1,21 @@
+/*
+ * linux/sound/rt286.h -- Platform data for RT286
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5651_H
+#define __LINUX_SND_RT5651_H
+
+struct rt5651_platform_data {
+ /* IN2 can optionally be differential */
+ bool in2_diff;
+
+ bool dmic_en;
+};
+
+#endif
diff --git a/kernel/include/sound/rt5670.h b/kernel/include/sound/rt5670.h
new file mode 100644
index 000000000..b7d605108
--- /dev/null
+++ b/kernel/include/sound/rt5670.h
@@ -0,0 +1,28 @@
+/*
+ * linux/sound/rt5670.h -- Platform data for RT5670
+ *
+ * Copyright 2014 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5670_H
+#define __LINUX_SND_RT5670_H
+
+struct rt5670_platform_data {
+ int jd_mode;
+ bool in2_diff;
+ bool dev_gpio;
+
+ bool dmic_en;
+ unsigned int dmic1_data_pin;
+ /* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/
+ unsigned int dmic2_data_pin;
+ /* 0 = GPIO8; 1 = IN3N; */
+ unsigned int dmic3_data_pin;
+ /* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/
+};
+
+#endif
diff --git a/kernel/include/sound/rt5677.h b/kernel/include/sound/rt5677.h
new file mode 100644
index 000000000..a6207043a
--- /dev/null
+++ b/kernel/include/sound/rt5677.h
@@ -0,0 +1,45 @@
+/*
+ * linux/sound/rt5677.h -- Platform data for RT5677
+ *
+ * Copyright 2013 Realtek Semiconductor Corp.
+ * Author: Oder Chiou <oder_chiou@realtek.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5677_H
+#define __LINUX_SND_RT5677_H
+
+enum rt5677_dmic2_clk {
+ RT5677_DMIC_CLK1 = 0,
+ RT5677_DMIC_CLK2 = 1,
+};
+
+
+struct rt5677_platform_data {
+ /* IN1/IN2/LOUT1/LOUT2/LOUT3 can optionally be differential */
+ bool in1_diff;
+ bool in2_diff;
+ bool lout1_diff;
+ bool lout2_diff;
+ bool lout3_diff;
+ /* DMIC2 clock source selection */
+ enum rt5677_dmic2_clk dmic2_clk_pin;
+
+ /* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */
+ u8 gpio_config[6];
+
+ /* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */
+ unsigned int jd1_gpio;
+ /* jd2 and jd3 can select 0 ~ 3 as
+ OFF, GPIO4, GPIO5 and GPIO6 respectively */
+ unsigned int jd2_gpio;
+ unsigned int jd3_gpio;
+
+ /* Set MICBIAS1 VDD 1v8 or 3v3 */
+ bool micbias1_vdd_3v3;
+};
+
+#endif
diff --git a/kernel/include/sound/s3c24xx_uda134x.h b/kernel/include/sound/s3c24xx_uda134x.h
new file mode 100644
index 000000000..33df4cb90
--- /dev/null
+++ b/kernel/include/sound/s3c24xx_uda134x.h
@@ -0,0 +1,14 @@
+#ifndef _S3C24XX_UDA134X_H_
+#define _S3C24XX_UDA134X_H_ 1
+
+#include <sound/uda134x.h>
+
+struct s3c24xx_uda134x_platform_data {
+ int l3_clk;
+ int l3_mode;
+ int l3_data;
+ void (*power) (int);
+ int model;
+};
+
+#endif
diff --git a/kernel/include/sound/sb.h b/kernel/include/sound/sb.h
new file mode 100644
index 000000000..bacefaee4
--- /dev/null
+++ b/kernel/include/sound/sb.h
@@ -0,0 +1,375 @@
+#ifndef __SOUND_SB_H
+#define __SOUND_SB_H
+
+/*
+ * Header file for SoundBlaster cards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+
+enum sb_hw_type {
+ SB_HW_AUTO,
+ SB_HW_10,
+ SB_HW_20,
+ SB_HW_201,
+ SB_HW_PRO,
+ SB_HW_JAZZ16, /* Media Vision Jazz16 */
+ SB_HW_16,
+ SB_HW_16CSP, /* SB16 with CSP chip */
+ SB_HW_ALS100, /* Avance Logic ALS100 chip */
+ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */
+ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */
+ SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */
+};
+
+#define SB_OPEN_PCM 0x01
+#define SB_OPEN_MIDI_INPUT 0x02
+#define SB_OPEN_MIDI_OUTPUT 0x04
+#define SB_OPEN_MIDI_INPUT_TRIGGER 0x08
+#define SB_OPEN_MIDI_OUTPUT_TRIGGER 0x10
+
+#define SB_MODE_HALT 0x00
+#define SB_MODE_PLAYBACK_8 0x01
+#define SB_MODE_PLAYBACK_16 0x02
+#define SB_MODE_PLAYBACK (SB_MODE_PLAYBACK_8 | SB_MODE_PLAYBACK_16)
+#define SB_MODE_CAPTURE_8 0x04
+#define SB_MODE_CAPTURE_16 0x08
+#define SB_MODE_CAPTURE (SB_MODE_CAPTURE_8 | SB_MODE_CAPTURE_16)
+
+#define SB_RATE_LOCK_PLAYBACK 0x10
+#define SB_RATE_LOCK_CAPTURE 0x20
+#define SB_RATE_LOCK (SB_RATE_LOCK_PLAYBACK | SB_RATE_LOCK_CAPTURE)
+
+#define SB_MPU_INPUT 1
+
+struct snd_sb {
+ unsigned long port; /* base port of DSP chip */
+ struct resource *res_port;
+ unsigned long mpu_port; /* MPU port for SB DSP 4.0+ */
+ int irq; /* IRQ number of DSP chip */
+ int dma8; /* 8-bit DMA */
+ int dma16; /* 16-bit DMA */
+ unsigned short version; /* version of DSP chip */
+ enum sb_hw_type hardware; /* see to SB_HW_XXXX */
+
+ unsigned long alt_port; /* alternate port (ALS4000) */
+ struct pci_dev *pci; /* ALS4000 */
+
+ unsigned int open; /* see to SB_OPEN_XXXX for sb8 */
+ /* also SNDRV_SB_CSP_MODE_XXX for sb16_csp */
+ unsigned int mode; /* current mode of stream */
+ unsigned int force_mode16; /* force 16-bit mode of streams */
+ unsigned int locked_rate; /* sb16 duplex */
+ unsigned int playback_format;
+ unsigned int capture_format;
+ struct timer_list midi_timer;
+ unsigned int p_dma_size;
+ unsigned int p_period_size;
+ unsigned int c_dma_size;
+ unsigned int c_period_size;
+
+ spinlock_t mixer_lock;
+
+ char name[32];
+
+ void *csp; /* used only when CONFIG_SND_SB16_CSP is set */
+
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *midi_substream_input;
+ struct snd_rawmidi_substream *midi_substream_output;
+ irq_handler_t rmidi_callback;
+
+ spinlock_t reg_lock;
+ spinlock_t open_lock;
+ spinlock_t midi_input_lock;
+
+ struct snd_info_entry *proc_entry;
+
+#ifdef CONFIG_PM
+ unsigned char saved_regs[0x20];
+#endif
+};
+
+/* I/O ports */
+
+#define SBP(chip, x) ((chip)->port + s_b_SB_##x)
+#define SBP1(port, x) ((port) + s_b_SB_##x)
+
+#define s_b_SB_RESET 0x6
+#define s_b_SB_READ 0xa
+#define s_b_SB_WRITE 0xc
+#define s_b_SB_COMMAND 0xc
+#define s_b_SB_STATUS 0xc
+#define s_b_SB_DATA_AVAIL 0xe
+#define s_b_SB_DATA_AVAIL_16 0xf
+#define s_b_SB_MIXER_ADDR 0x4
+#define s_b_SB_MIXER_DATA 0x5
+#define s_b_SB_OPL3_LEFT 0x0
+#define s_b_SB_OPL3_RIGHT 0x2
+#define s_b_SB_OPL3_BOTH 0x8
+
+#define SB_DSP_OUTPUT 0x14
+#define SB_DSP_INPUT 0x24
+#define SB_DSP_BLOCK_SIZE 0x48
+#define SB_DSP_HI_OUTPUT 0x91
+#define SB_DSP_HI_INPUT 0x99
+#define SB_DSP_LO_OUTPUT_AUTO 0x1c
+#define SB_DSP_LO_INPUT_AUTO 0x2c
+#define SB_DSP_HI_OUTPUT_AUTO 0x90
+#define SB_DSP_HI_INPUT_AUTO 0x98
+#define SB_DSP_IMMED_INT 0xf2
+#define SB_DSP_GET_VERSION 0xe1
+#define SB_DSP_SPEAKER_ON 0xd1
+#define SB_DSP_SPEAKER_OFF 0xd3
+#define SB_DSP_DMA8_OFF 0xd0
+#define SB_DSP_DMA8_ON 0xd4
+#define SB_DSP_DMA8_EXIT 0xda
+#define SB_DSP_DMA16_OFF 0xd5
+#define SB_DSP_DMA16_ON 0xd6
+#define SB_DSP_DMA16_EXIT 0xd9
+#define SB_DSP_SAMPLE_RATE 0x40
+#define SB_DSP_SAMPLE_RATE_OUT 0x41
+#define SB_DSP_SAMPLE_RATE_IN 0x42
+#define SB_DSP_MONO_8BIT 0xa0
+#define SB_DSP_MONO_16BIT 0xa4
+#define SB_DSP_STEREO_8BIT 0xa8
+#define SB_DSP_STEREO_16BIT 0xac
+
+#define SB_DSP_MIDI_INPUT_IRQ 0x31
+#define SB_DSP_MIDI_UART_IRQ 0x35
+#define SB_DSP_MIDI_OUTPUT 0x38
+
+#define SB_DSP4_OUT8_AI 0xc6
+#define SB_DSP4_IN8_AI 0xce
+#define SB_DSP4_OUT16_AI 0xb6
+#define SB_DSP4_IN16_AI 0xbe
+#define SB_DSP4_MODE_UNS_MONO 0x00
+#define SB_DSP4_MODE_SIGN_MONO 0x10
+#define SB_DSP4_MODE_UNS_STEREO 0x20
+#define SB_DSP4_MODE_SIGN_STEREO 0x30
+
+#define SB_DSP4_OUTPUT 0x3c
+#define SB_DSP4_INPUT_LEFT 0x3d
+#define SB_DSP4_INPUT_RIGHT 0x3e
+
+/* registers for SB 2.0 mixer */
+#define SB_DSP20_MASTER_DEV 0x02
+#define SB_DSP20_PCM_DEV 0x0A
+#define SB_DSP20_CD_DEV 0x08
+#define SB_DSP20_FM_DEV 0x06
+
+/* registers for SB PRO mixer */
+#define SB_DSP_MASTER_DEV 0x22
+#define SB_DSP_PCM_DEV 0x04
+#define SB_DSP_LINE_DEV 0x2e
+#define SB_DSP_CD_DEV 0x28
+#define SB_DSP_FM_DEV 0x26
+#define SB_DSP_MIC_DEV 0x0a
+#define SB_DSP_CAPTURE_SOURCE 0x0c
+#define SB_DSP_CAPTURE_FILT 0x0c
+#define SB_DSP_PLAYBACK_FILT 0x0e
+#define SB_DSP_STEREO_SW 0x0e
+
+#define SB_DSP_MIXS_MIC0 0x00 /* same as MIC */
+#define SB_DSP_MIXS_CD 0x01
+#define SB_DSP_MIXS_MIC 0x02
+#define SB_DSP_MIXS_LINE 0x03
+
+/* registers (only for left channel) for SB 16 mixer */
+#define SB_DSP4_MASTER_DEV 0x30
+#define SB_DSP4_BASS_DEV 0x46
+#define SB_DSP4_TREBLE_DEV 0x44
+#define SB_DSP4_SYNTH_DEV 0x34
+#define SB_DSP4_PCM_DEV 0x32
+#define SB_DSP4_SPEAKER_DEV 0x3b
+#define SB_DSP4_LINE_DEV 0x38
+#define SB_DSP4_MIC_DEV 0x3a
+#define SB_DSP4_OUTPUT_SW 0x3c
+#define SB_DSP4_CD_DEV 0x36
+#define SB_DSP4_IGAIN_DEV 0x3f
+#define SB_DSP4_OGAIN_DEV 0x41
+#define SB_DSP4_MIC_AGC 0x43
+
+/* additional registers for SB 16 mixer */
+#define SB_DSP4_IRQSETUP 0x80
+#define SB_DSP4_DMASETUP 0x81
+#define SB_DSP4_IRQSTATUS 0x82
+#define SB_DSP4_MPUSETUP 0x84
+
+#define SB_DSP4_3DSE 0x90
+
+/* Registers for DT-019x / ALS-007 mixer */
+#define SB_DT019X_MASTER_DEV 0x62
+#define SB_DT019X_PCM_DEV 0x64
+#define SB_DT019X_SYNTH_DEV 0x66
+#define SB_DT019X_CD_DEV 0x68
+#define SB_DT019X_MIC_DEV 0x6a
+#define SB_DT019X_SPKR_DEV 0x6a
+#define SB_DT019X_LINE_DEV 0x6e
+#define SB_DT019X_OUTPUT_SW2 0x4c
+#define SB_DT019X_CAPTURE_SW 0x6c
+
+#define SB_DT019X_CAP_CD 0x02
+#define SB_DT019X_CAP_MIC 0x04
+#define SB_DT019X_CAP_LINE 0x06
+#define SB_DT019X_CAP_SYNTH 0x07
+#define SB_DT019X_CAP_MAIN 0x07
+
+#define SB_ALS4000_MONO_IO_CTRL 0x4b
+#define SB_ALS4000_OUT_MIXER_CTRL_2 0x4c
+#define SB_ALS4000_MIC_IN_GAIN 0x4d
+#define SB_ALS4000_ANALOG_REFRNC_VOLT_CTRL 0x4e
+#define SB_ALS4000_FMDAC 0x4f
+#define SB_ALS4000_3D_SND_FX 0x50
+#define SB_ALS4000_3D_TIME_DELAY 0x51
+#define SB_ALS4000_3D_AUTO_MUTE 0x52
+#define SB_ALS4000_ANALOG_BLOCK_CTRL 0x53
+#define SB_ALS4000_3D_DELAYLINE_PATTERN 0x54
+#define SB_ALS4000_CR3_CONFIGURATION 0xc3 /* bit 7 is Digital Loop Enable */
+#define SB_ALS4000_QSOUND 0xdb
+
+/* IRQ setting bitmap */
+#define SB_IRQSETUP_IRQ9 0x01
+#define SB_IRQSETUP_IRQ5 0x02
+#define SB_IRQSETUP_IRQ7 0x04
+#define SB_IRQSETUP_IRQ10 0x08
+
+/* IRQ types */
+#define SB_IRQTYPE_8BIT 0x01
+#define SB_IRQTYPE_16BIT 0x02
+#define SB_IRQTYPE_MPUIN 0x04
+#define ALS4K_IRQTYPE_CR1E_DMA 0x20
+
+/* DMA setting bitmap */
+#define SB_DMASETUP_DMA0 0x01
+#define SB_DMASETUP_DMA1 0x02
+#define SB_DMASETUP_DMA3 0x08
+#define SB_DMASETUP_DMA5 0x20
+#define SB_DMASETUP_DMA6 0x40
+#define SB_DMASETUP_DMA7 0x80
+
+/*
+ *
+ */
+
+static inline void snd_sb_ack_8bit(struct snd_sb *chip)
+{
+ inb(SBP(chip, DATA_AVAIL));
+}
+
+static inline void snd_sb_ack_16bit(struct snd_sb *chip)
+{
+ inb(SBP(chip, DATA_AVAIL_16));
+}
+
+/* sb_common.c */
+int snd_sbdsp_command(struct snd_sb *chip, unsigned char val);
+int snd_sbdsp_get_byte(struct snd_sb *chip);
+int snd_sbdsp_reset(struct snd_sb *chip);
+int snd_sbdsp_create(struct snd_card *card,
+ unsigned long port,
+ int irq,
+ irq_handler_t irq_handler,
+ int dma8, int dma16,
+ unsigned short hardware,
+ struct snd_sb **r_chip);
+/* sb_mixer.c */
+void snd_sbmixer_write(struct snd_sb *chip, unsigned char reg, unsigned char data);
+unsigned char snd_sbmixer_read(struct snd_sb *chip, unsigned char reg);
+int snd_sbmixer_new(struct snd_sb *chip);
+#ifdef CONFIG_PM
+void snd_sbmixer_suspend(struct snd_sb *chip);
+void snd_sbmixer_resume(struct snd_sb *chip);
+#endif
+
+/* sb8_init.c */
+int snd_sb8dsp_pcm(struct snd_sb *chip, int device);
+/* sb8.c */
+irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip);
+int snd_sb8_playback_open(struct snd_pcm_substream *substream);
+int snd_sb8_capture_open(struct snd_pcm_substream *substream);
+int snd_sb8_playback_close(struct snd_pcm_substream *substream);
+int snd_sb8_capture_close(struct snd_pcm_substream *substream);
+/* midi8.c */
+irqreturn_t snd_sb8dsp_midi_interrupt(struct snd_sb *chip);
+int snd_sb8dsp_midi(struct snd_sb *chip, int device);
+
+/* sb16_init.c */
+int snd_sb16dsp_pcm(struct snd_sb *chip, int device);
+const struct snd_pcm_ops *snd_sb16dsp_get_pcm_ops(int direction);
+int snd_sb16dsp_configure(struct snd_sb *chip);
+/* sb16.c */
+irqreturn_t snd_sb16dsp_interrupt(int irq, void *dev_id);
+
+/* exported mixer stuffs */
+enum {
+ SB_MIX_SINGLE,
+ SB_MIX_DOUBLE,
+ SB_MIX_INPUT_SW,
+ SB_MIX_CAPTURE_PRO,
+ SB_MIX_CAPTURE_DT019X,
+ SB_MIX_MONO_CAPTURE_ALS4K
+};
+
+#define SB_MIXVAL_DOUBLE(left_reg, right_reg, left_shift, right_shift, mask) \
+ ((left_reg) | ((right_reg) << 8) | ((left_shift) << 16) | ((right_shift) << 19) | ((mask) << 24))
+#define SB_MIXVAL_SINGLE(reg, shift, mask) \
+ ((reg) | ((shift) << 16) | ((mask) << 24))
+#define SB_MIXVAL_INPUT_SW(reg1, reg2, left_shift, right_shift) \
+ ((reg1) | ((reg2) << 8) | ((left_shift) << 16) | ((right_shift) << 24))
+
+int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int type, unsigned long value);
+
+/* for ease of use */
+struct sbmix_elem {
+ const char *name;
+ int type;
+ unsigned long private_value;
+};
+
+#define SB_SINGLE(xname, reg, shift, mask) \
+{ .name = xname, \
+ .type = SB_MIX_SINGLE, \
+ .private_value = SB_MIXVAL_SINGLE(reg, shift, mask) }
+
+#define SB_DOUBLE(xname, left_reg, right_reg, left_shift, right_shift, mask) \
+{ .name = xname, \
+ .type = SB_MIX_DOUBLE, \
+ .private_value = SB_MIXVAL_DOUBLE(left_reg, right_reg, left_shift, right_shift, mask) }
+
+#define SB16_INPUT_SW(xname, reg1, reg2, left_shift, right_shift) \
+{ .name = xname, \
+ .type = SB_MIX_INPUT_SW, \
+ .private_value = SB_MIXVAL_INPUT_SW(reg1, reg2, left_shift, right_shift) }
+
+static inline int snd_sbmixer_add_ctl_elem(struct snd_sb *chip, const struct sbmix_elem *c)
+{
+ return snd_sbmixer_add_ctl(chip, c->name, 0, c->type, c->private_value);
+}
+
+#endif /* __SOUND_SB_H */
diff --git a/kernel/include/sound/sb16_csp.h b/kernel/include/sound/sb16_csp.h
new file mode 100644
index 000000000..c7c778800
--- /dev/null
+++ b/kernel/include/sound/sb16_csp.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (c) 1999 by Uros Bizjak <uros@kss-loka.si>
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ * SB16ASP/AWE32 CSP control
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SOUND_SB16_CSP_H
+#define __SOUND_SB16_CSP_H
+
+#include <sound/sb.h>
+#include <sound/hwdep.h>
+#include <linux/firmware.h>
+#include <uapi/sound/sb16_csp.h>
+
+struct snd_sb_csp;
+
+/* indices for the known CSP programs */
+enum {
+ CSP_PROGRAM_MULAW,
+ CSP_PROGRAM_ALAW,
+ CSP_PROGRAM_ADPCM_INIT,
+ CSP_PROGRAM_ADPCM_PLAYBACK,
+ CSP_PROGRAM_ADPCM_CAPTURE,
+
+ CSP_PROGRAM_COUNT
+};
+
+/*
+ * CSP operators
+ */
+struct snd_sb_csp_ops {
+ int (*csp_use) (struct snd_sb_csp * p);
+ int (*csp_unuse) (struct snd_sb_csp * p);
+ int (*csp_autoload) (struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode);
+ int (*csp_start) (struct snd_sb_csp * p, int sample_width, int channels);
+ int (*csp_stop) (struct snd_sb_csp * p);
+ int (*csp_qsound_transfer) (struct snd_sb_csp * p);
+};
+
+/*
+ * CSP private data
+ */
+struct snd_sb_csp {
+ struct snd_sb *chip; /* SB16 DSP */
+ int used; /* usage flag - exclusive */
+ char codec_name[16]; /* name of codec */
+ unsigned short func_nr; /* function number */
+ unsigned int acc_format; /* accepted PCM formats */
+ int acc_channels; /* accepted channels */
+ int acc_width; /* accepted sample width */
+ int acc_rates; /* accepted sample rates */
+ int mode; /* MODE */
+ int run_channels; /* current CSP channels */
+ int run_width; /* current sample width */
+ int version; /* CSP version (0x10 - 0x1f) */
+ int running; /* running state */
+
+ struct snd_sb_csp_ops ops; /* operators */
+
+ spinlock_t q_lock; /* locking */
+ int q_enabled; /* enabled flag */
+ int qpos_left; /* left position */
+ int qpos_right; /* right position */
+ int qpos_changed; /* position changed flag */
+
+ struct snd_kcontrol *qsound_switch;
+ struct snd_kcontrol *qsound_space;
+
+ struct mutex access_mutex; /* locking */
+
+ const struct firmware *csp_programs[CSP_PROGRAM_COUNT];
+};
+
+int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep);
+#endif /* __SOUND_SB16_CSP */
diff --git a/kernel/include/sound/seq_device.h b/kernel/include/sound/seq_device.h
new file mode 100644
index 000000000..ddc0d504c
--- /dev/null
+++ b/kernel/include/sound/seq_device.h
@@ -0,0 +1,96 @@
+#ifndef __SOUND_SEQ_DEVICE_H
+#define __SOUND_SEQ_DEVICE_H
+
+/*
+ * ALSA sequencer device management
+ * Copyright (c) 1999 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * registered device information
+ */
+
+struct snd_seq_device {
+ /* device info */
+ struct snd_card *card; /* sound card */
+ int device; /* device number */
+ const char *id; /* driver id */
+ char name[80]; /* device name */
+ int argsize; /* size of the argument */
+ void *driver_data; /* private data for driver */
+ void *private_data; /* private data for the caller */
+ void (*private_free)(struct snd_seq_device *device);
+ struct device dev;
+};
+
+#define to_seq_dev(_dev) \
+ container_of(_dev, struct snd_seq_device, dev)
+
+/* sequencer driver */
+
+/* driver operators
+ * probe:
+ * Initialize the device with given parameters.
+ * Typically,
+ * 1. call snd_hwdep_new
+ * 2. allocate private data and initialize it
+ * 3. call snd_hwdep_register
+ * 4. store the instance to dev->driver_data pointer.
+ *
+ * remove:
+ * Release the private data.
+ * Typically, call snd_device_free(dev->card, dev->driver_data)
+ */
+struct snd_seq_driver {
+ struct device_driver driver;
+ char *id;
+ int argsize;
+};
+
+#define to_seq_drv(_drv) \
+ container_of(_drv, struct snd_seq_driver, driver)
+
+/*
+ * prototypes
+ */
+#ifdef CONFIG_MODULES
+void snd_seq_device_load_drivers(void);
+#else
+#define snd_seq_device_load_drivers()
+#endif
+int snd_seq_device_new(struct snd_card *card, int device, const char *id,
+ int argsize, struct snd_seq_device **result);
+
+#define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device))
+
+int __must_check __snd_seq_driver_register(struct snd_seq_driver *drv,
+ struct module *mod);
+#define snd_seq_driver_register(drv) \
+ __snd_seq_driver_register(drv, THIS_MODULE)
+void snd_seq_driver_unregister(struct snd_seq_driver *drv);
+
+#define module_snd_seq_driver(drv) \
+ module_driver(drv, snd_seq_driver_register, snd_seq_driver_unregister)
+
+/*
+ * id strings for generic devices
+ */
+#define SNDRV_SEQ_DEV_ID_MIDISYNTH "seq-midi"
+#define SNDRV_SEQ_DEV_ID_OPL3 "opl3-synth"
+
+#endif /* __SOUND_SEQ_DEVICE_H */
diff --git a/kernel/include/sound/seq_kernel.h b/kernel/include/sound/seq_kernel.h
new file mode 100644
index 000000000..feb58d455
--- /dev/null
+++ b/kernel/include/sound/seq_kernel.h
@@ -0,0 +1,109 @@
+#ifndef __SOUND_SEQ_KERNEL_H
+#define __SOUND_SEQ_KERNEL_H
+
+/*
+ * Main kernel header file for the ALSA sequencer
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/time.h>
+#include <sound/asequencer.h>
+
+typedef struct snd_seq_real_time snd_seq_real_time_t;
+typedef union snd_seq_timestamp snd_seq_timestamp_t;
+
+/* maximum number of queues */
+#define SNDRV_SEQ_MAX_QUEUES 32
+
+/* max number of concurrent clients */
+#define SNDRV_SEQ_MAX_CLIENTS 192
+
+/* max number of concurrent ports */
+#define SNDRV_SEQ_MAX_PORTS 254
+
+/* max number of events in memory pool */
+#define SNDRV_SEQ_MAX_EVENTS 2000
+
+/* default number of events in memory pool */
+#define SNDRV_SEQ_DEFAULT_EVENTS 500
+
+/* max number of events in memory pool for one client (outqueue) */
+#define SNDRV_SEQ_MAX_CLIENT_EVENTS 2000
+
+/* default number of events in memory pool for one client (outqueue) */
+#define SNDRV_SEQ_DEFAULT_CLIENT_EVENTS 200
+
+/* max delivery path length */
+#define SNDRV_SEQ_MAX_HOPS 10
+
+/* max size of event size */
+#define SNDRV_SEQ_MAX_EVENT_LEN 0x3fffffff
+
+/* call-backs for kernel port */
+struct snd_seq_port_callback {
+ struct module *owner;
+ void *private_data;
+ int (*subscribe)(void *private_data, struct snd_seq_port_subscribe *info);
+ int (*unsubscribe)(void *private_data, struct snd_seq_port_subscribe *info);
+ int (*use)(void *private_data, struct snd_seq_port_subscribe *info);
+ int (*unuse)(void *private_data, struct snd_seq_port_subscribe *info);
+ int (*event_input)(struct snd_seq_event *ev, int direct, void *private_data, int atomic, int hop);
+ void (*private_free)(void *private_data);
+ /*...*/
+};
+
+/* interface for kernel client */
+__printf(3, 4)
+int snd_seq_create_kernel_client(struct snd_card *card, int client_index,
+ const char *name_fmt, ...);
+int snd_seq_delete_kernel_client(int client);
+int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, int atomic, int hop);
+int snd_seq_kernel_client_dispatch(int client, struct snd_seq_event *ev, int atomic, int hop);
+int snd_seq_kernel_client_ctl(int client, unsigned int cmd, void *arg);
+
+#define SNDRV_SEQ_EXT_MASK 0xc0000000
+#define SNDRV_SEQ_EXT_USRPTR 0x80000000
+#define SNDRV_SEQ_EXT_CHAINED 0x40000000
+
+typedef int (*snd_seq_dump_func_t)(void *ptr, void *buf, int count);
+int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char *buf,
+ int in_kernel, int size_aligned);
+int snd_seq_dump_var_event(const struct snd_seq_event *event,
+ snd_seq_dump_func_t func, void *private_data);
+
+/* interface for OSS emulation */
+int snd_seq_set_queue_tempo(int client, struct snd_seq_queue_tempo *tempo);
+
+/* port callback routines */
+void snd_port_init_callback(struct snd_seq_port_callback *p);
+struct snd_seq_port_callback *snd_port_alloc_callback(void);
+
+/* port attach/detach */
+int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp,
+ int cap, int type, int midi_channels, int midi_voices, char *portname);
+int snd_seq_event_port_detach(int client, int port);
+
+#ifdef CONFIG_MODULES
+void snd_seq_autoload_init(void);
+void snd_seq_autoload_exit(void);
+#else
+#define snd_seq_autoload_init()
+#define snd_seq_autoload_exit()
+#endif
+
+#endif /* __SOUND_SEQ_KERNEL_H */
diff --git a/kernel/include/sound/seq_midi_emul.h b/kernel/include/sound/seq_midi_emul.h
new file mode 100644
index 000000000..8139d8c19
--- /dev/null
+++ b/kernel/include/sound/seq_midi_emul.h
@@ -0,0 +1,197 @@
+#ifndef __SOUND_SEQ_MIDI_EMUL_H
+#define __SOUND_SEQ_MIDI_EMUL_H
+
+/*
+ * Midi channel definition for optional channel management.
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/seq_kernel.h>
+
+/*
+ * This structure is used to keep track of the current state on each
+ * channel. All drivers for hardware that does not understand midi
+ * directly will probably need to use this structure.
+ */
+struct snd_midi_channel {
+ void *private; /* A back pointer to driver data */
+ int number; /* The channel number */
+ int client; /* The client associated with this channel */
+ int port; /* The port associated with this channel */
+
+ unsigned char midi_mode; /* GM, GS, XG etc */
+ unsigned int
+ drum_channel:1, /* Drum channel */
+ param_type:1 /* RPN/NRPN */
+ ;
+
+ unsigned char midi_aftertouch; /* Aftertouch (key pressure) */
+ unsigned char midi_pressure; /* Channel pressure */
+ unsigned char midi_program; /* Instrument number */
+ short midi_pitchbend; /* Pitch bend amount */
+
+ unsigned char control[128]; /* Current value of all controls */
+ unsigned char note[128]; /* Current status for all notes */
+
+ short gm_rpn_pitch_bend_range; /* Pitch bend range */
+ short gm_rpn_fine_tuning; /* Master fine tuning */
+ short gm_rpn_coarse_tuning; /* Master coarse tuning */
+
+};
+
+/*
+ * A structure that represets a set of channels bound to a port. There
+ * would usually be 16 channels per port. But fewer could be used for
+ * particular cases.
+ * The channel set consists of information describing the client and
+ * port for this midi synth and an array of snd_midi_channel structures.
+ * A driver that had no need for snd_midi_channel could still use the
+ * channel set type if it wished with the channel array null.
+ */
+struct snd_midi_channel_set {
+ void *private_data; /* Driver data */
+ int client; /* Client for this port */
+ int port; /* The port number */
+
+ int max_channels; /* Size of the channels array */
+ struct snd_midi_channel *channels;
+
+ unsigned char midi_mode; /* MIDI operating mode */
+ unsigned char gs_master_volume; /* SYSEX master volume: 0-127 */
+ unsigned char gs_chorus_mode;
+ unsigned char gs_reverb_mode;
+
+};
+
+struct snd_midi_op {
+ void (*note_on)(void *private_data, int note, int vel, struct snd_midi_channel *chan);
+ void (*note_off)(void *private_data,int note, int vel, struct snd_midi_channel *chan); /* release note */
+ void (*key_press)(void *private_data, int note, int vel, struct snd_midi_channel *chan);
+ void (*note_terminate)(void *private_data, int note, struct snd_midi_channel *chan); /* terminate note immediately */
+ void (*control)(void *private_data, int type, struct snd_midi_channel *chan);
+ void (*nrpn)(void *private_data, struct snd_midi_channel *chan,
+ struct snd_midi_channel_set *chset);
+ void (*sysex)(void *private_data, unsigned char *buf, int len, int parsed,
+ struct snd_midi_channel_set *chset);
+};
+
+/*
+ * These defines are used so that pitchbend, aftertouch etc, can be
+ * distinguished from controller values.
+ */
+/* 0-127 controller values */
+#define MIDI_CTL_PITCHBEND 0x80
+#define MIDI_CTL_AFTERTOUCH 0x81
+#define MIDI_CTL_CHAN_PRESSURE 0x82
+
+/*
+ * These names exist to allow symbolic access to the controls array.
+ * The usage is eg: chan->gm_bank_select. Another implementation would
+ * be really have these members in the struct, and not the array.
+ */
+#define gm_bank_select control[0]
+#define gm_modulation control[1]
+#define gm_breath control[2]
+#define gm_foot_pedal control[4]
+#define gm_portamento_time control[5]
+#define gm_data_entry control[6]
+#define gm_volume control[7]
+#define gm_balance control[8]
+#define gm_pan control[10]
+#define gm_expression control[11]
+#define gm_effect_control1 control[12]
+#define gm_effect_control2 control[13]
+#define gm_slider1 control[16]
+#define gm_slider2 control[17]
+#define gm_slider3 control[18]
+#define gm_slider4 control[19]
+
+#define gm_bank_select_lsb control[32]
+#define gm_modulation_wheel_lsb control[33]
+#define gm_breath_lsb control[34]
+#define gm_foot_pedal_lsb control[36]
+#define gm_portamento_time_lsb control[37]
+#define gm_data_entry_lsb control[38]
+#define gm_volume_lsb control[39]
+#define gm_balance_lsb control[40]
+#define gm_pan_lsb control[42]
+#define gm_expression_lsb control[43]
+#define gm_effect_control1_lsb control[44]
+#define gm_effect_control2_lsb control[45]
+
+#define gm_sustain control[MIDI_CTL_SUSTAIN]
+#define gm_hold gm_sustain
+#define gm_portamento control[MIDI_CTL_PORTAMENTO]
+#define gm_sostenuto control[MIDI_CTL_SOSTENUTO]
+
+/*
+ * These macros give the complete value of the controls that consist
+ * of coarse and fine pairs. Of course the fine controls are seldom used
+ * but there is no harm in being complete.
+ */
+#define SNDRV_GM_BANK_SELECT(cp) (((cp)->control[0]<<7)|((cp)->control[32]))
+#define SNDRV_GM_MODULATION_WHEEL(cp) (((cp)->control[1]<<7)|((cp)->control[33]))
+#define SNDRV_GM_BREATH(cp) (((cp)->control[2]<<7)|((cp)->control[34]))
+#define SNDRV_GM_FOOT_PEDAL(cp) (((cp)->control[4]<<7)|((cp)->control[36]))
+#define SNDRV_GM_PORTAMENTO_TIME(cp) (((cp)->control[5]<<7)|((cp)->control[37]))
+#define SNDRV_GM_DATA_ENTRY(cp) (((cp)->control[6]<<7)|((cp)->control[38]))
+#define SNDRV_GM_VOLUME(cp) (((cp)->control[7]<<7)|((cp)->control[39]))
+#define SNDRV_GM_BALANCE(cp) (((cp)->control[8]<<7)|((cp)->control[40]))
+#define SNDRV_GM_PAN(cp) (((cp)->control[10]<<7)|((cp)->control[42]))
+#define SNDRV_GM_EXPRESSION(cp) (((cp)->control[11]<<7)|((cp)->control[43]))
+
+
+/* MIDI mode */
+#define SNDRV_MIDI_MODE_NONE 0 /* Generic midi */
+#define SNDRV_MIDI_MODE_GM 1
+#define SNDRV_MIDI_MODE_GS 2
+#define SNDRV_MIDI_MODE_XG 3
+#define SNDRV_MIDI_MODE_MT32 4
+
+/* MIDI note state */
+#define SNDRV_MIDI_NOTE_OFF 0x00
+#define SNDRV_MIDI_NOTE_ON 0x01
+#define SNDRV_MIDI_NOTE_RELEASED 0x02
+#define SNDRV_MIDI_NOTE_SOSTENUTO 0x04
+
+#define SNDRV_MIDI_PARAM_TYPE_REGISTERED 0
+#define SNDRV_MIDI_PARAM_TYPE_NONREGISTERED 1
+
+/* SYSEX parse flag */
+enum {
+ SNDRV_MIDI_SYSEX_NOT_PARSED = 0,
+ SNDRV_MIDI_SYSEX_GM_ON,
+ SNDRV_MIDI_SYSEX_GS_ON,
+ SNDRV_MIDI_SYSEX_GS_RESET,
+ SNDRV_MIDI_SYSEX_GS_CHORUS_MODE,
+ SNDRV_MIDI_SYSEX_GS_REVERB_MODE,
+ SNDRV_MIDI_SYSEX_GS_MASTER_VOLUME,
+ SNDRV_MIDI_SYSEX_GS_PROGRAM,
+ SNDRV_MIDI_SYSEX_GS_DRUM_CHANNEL,
+ SNDRV_MIDI_SYSEX_XG_ON,
+};
+
+/* Prototypes for midi_process.c */
+void snd_midi_process_event(struct snd_midi_op *ops, struct snd_seq_event *ev,
+ struct snd_midi_channel_set *chanset);
+void snd_midi_channel_set_clear(struct snd_midi_channel_set *chset);
+struct snd_midi_channel_set *snd_midi_channel_alloc_set(int n);
+void snd_midi_channel_free_set(struct snd_midi_channel_set *chset);
+
+#endif /* __SOUND_SEQ_MIDI_EMUL_H */
diff --git a/kernel/include/sound/seq_midi_event.h b/kernel/include/sound/seq_midi_event.h
new file mode 100644
index 000000000..e40f43e6f
--- /dev/null
+++ b/kernel/include/sound/seq_midi_event.h
@@ -0,0 +1,54 @@
+#ifndef __SOUND_SEQ_MIDI_EVENT_H
+#define __SOUND_SEQ_MIDI_EVENT_H
+
+/*
+ * MIDI byte <-> sequencer event coder
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>,
+ * Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/asequencer.h>
+
+#define MAX_MIDI_EVENT_BUF 256
+
+/* midi status */
+struct snd_midi_event {
+ int qlen; /* queue length */
+ int read; /* chars read */
+ int type; /* current event type */
+ unsigned char lastcmd; /* last command (for MIDI state handling) */
+ unsigned char nostat; /* no state flag */
+ int bufsize; /* allocated buffer size */
+ unsigned char *buf; /* input buffer */
+ spinlock_t lock;
+};
+
+int snd_midi_event_new(int bufsize, struct snd_midi_event **rdev);
+void snd_midi_event_free(struct snd_midi_event *dev);
+void snd_midi_event_reset_encode(struct snd_midi_event *dev);
+void snd_midi_event_reset_decode(struct snd_midi_event *dev);
+void snd_midi_event_no_status(struct snd_midi_event *dev, int on);
+/* encode from byte stream - return number of written bytes if success */
+long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count,
+ struct snd_seq_event *ev);
+int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, struct snd_seq_event *ev);
+/* decode from event to bytes - return number of written bytes if success */
+long snd_midi_event_decode(struct snd_midi_event *dev, unsigned char *buf, long count,
+ struct snd_seq_event *ev);
+
+#endif /* __SOUND_SEQ_MIDI_EVENT_H */
diff --git a/kernel/include/sound/seq_oss.h b/kernel/include/sound/seq_oss.h
new file mode 100644
index 000000000..d0b27ec6f
--- /dev/null
+++ b/kernel/include/sound/seq_oss.h
@@ -0,0 +1,96 @@
+#ifndef __SOUND_SEQ_OSS_H
+#define __SOUND_SEQ_OSS_H
+
+/*
+ * OSS compatible sequencer driver
+ *
+ * Copyright (C) 1998,99 Takashi Iwai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/asequencer.h>
+#include <sound/seq_kernel.h>
+
+/*
+ * argument structure for synthesizer operations
+ */
+struct snd_seq_oss_arg {
+ /* given by OSS sequencer */
+ int app_index; /* application unique index */
+ int file_mode; /* file mode - see below */
+ int seq_mode; /* sequencer mode - see below */
+
+ /* following must be initialized in open callback */
+ struct snd_seq_addr addr; /* opened port address */
+ void *private_data; /* private data for lowlevel drivers */
+
+ /* note-on event passing mode: initially given by OSS seq,
+ * but configurable by drivers - see below
+ */
+ int event_passing;
+};
+
+
+/*
+ * synthesizer operation callbacks
+ */
+struct snd_seq_oss_callback {
+ struct module *owner;
+ int (*open)(struct snd_seq_oss_arg *p, void *closure);
+ int (*close)(struct snd_seq_oss_arg *p);
+ int (*ioctl)(struct snd_seq_oss_arg *p, unsigned int cmd, unsigned long arg);
+ int (*load_patch)(struct snd_seq_oss_arg *p, int format, const char __user *buf, int offs, int count);
+ int (*reset)(struct snd_seq_oss_arg *p);
+ int (*raw_event)(struct snd_seq_oss_arg *p, unsigned char *data);
+};
+
+/* flag: file_mode */
+#define SNDRV_SEQ_OSS_FILE_ACMODE 3
+#define SNDRV_SEQ_OSS_FILE_READ 1
+#define SNDRV_SEQ_OSS_FILE_WRITE 2
+#define SNDRV_SEQ_OSS_FILE_NONBLOCK 4
+
+/* flag: seq_mode */
+#define SNDRV_SEQ_OSS_MODE_SYNTH 0
+#define SNDRV_SEQ_OSS_MODE_MUSIC 1
+
+/* flag: event_passing */
+#define SNDRV_SEQ_OSS_PROCESS_EVENTS 0 /* key == 255 is processed as velocity change */
+#define SNDRV_SEQ_OSS_PASS_EVENTS 1 /* pass all events to callback */
+#define SNDRV_SEQ_OSS_PROCESS_KEYPRESS 2 /* key >= 128 will be processed as key-pressure */
+
+/* default control rate: fixed */
+#define SNDRV_SEQ_OSS_CTRLRATE 100
+
+/* default max queue length: configurable by module option */
+#define SNDRV_SEQ_OSS_MAX_QLEN 1024
+
+
+/*
+ * data pointer to snd_seq_register_device
+ */
+struct snd_seq_oss_reg {
+ int type;
+ int subtype;
+ int nvoices;
+ struct snd_seq_oss_callback oper;
+ void *private_data;
+};
+
+/* device id */
+#define SNDRV_SEQ_DEV_ID_OSS "seq-oss"
+
+#endif /* __SOUND_SEQ_OSS_H */
diff --git a/kernel/include/sound/seq_oss_legacy.h b/kernel/include/sound/seq_oss_legacy.h
new file mode 100644
index 000000000..e66269ff9
--- /dev/null
+++ b/kernel/include/sound/seq_oss_legacy.h
@@ -0,0 +1,31 @@
+#ifndef __SOUND_SEQ_OSS_LEGACY_H
+#define __SOUND_SEQ_OSS_LEGACY_H
+
+/*
+ * OSS compatible macro definitions
+ *
+ * Copyright (C) 2000 Abramo Bagnara <abramo@alsa-project.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/soundcard.h>
+
+#ifndef SAMPLE_TYPE_AWE32
+#define SAMPLE_TYPE_AWE32 0x20
+#endif
+
+#endif /* __SOUND_SEQ_OSS_LEGACY_H */
+
diff --git a/kernel/include/sound/seq_virmidi.h b/kernel/include/sound/seq_virmidi.h
new file mode 100644
index 000000000..a03acd0d3
--- /dev/null
+++ b/kernel/include/sound/seq_virmidi.h
@@ -0,0 +1,81 @@
+#ifndef __SOUND_SEQ_VIRMIDI_H
+#define __SOUND_SEQ_VIRMIDI_H
+
+/*
+ * Virtual Raw MIDI client on Sequencer
+ * Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>,
+ * Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/rawmidi.h>
+#include <sound/seq_midi_event.h>
+
+/*
+ * device file instance:
+ * This instance is created at each time the midi device file is
+ * opened. Each instance has its own input buffer and MIDI parser
+ * (buffer), and is associated with the device instance.
+ */
+struct snd_virmidi {
+ struct list_head list;
+ int seq_mode;
+ int client;
+ int port;
+ unsigned int trigger: 1;
+ struct snd_midi_event *parser;
+ struct snd_seq_event event;
+ struct snd_virmidi_dev *rdev;
+ struct snd_rawmidi_substream *substream;
+};
+
+#define SNDRV_VIRMIDI_SUBSCRIBE (1<<0)
+#define SNDRV_VIRMIDI_USE (1<<1)
+
+/*
+ * device record:
+ * Each virtual midi device has one device instance. It contains
+ * common information and the linked-list of opened files,
+ */
+struct snd_virmidi_dev {
+ struct snd_card *card; /* associated card */
+ struct snd_rawmidi *rmidi; /* rawmidi device */
+ int seq_mode; /* SNDRV_VIRMIDI_XXX */
+ int device; /* sequencer device */
+ int client; /* created/attached client */
+ int port; /* created/attached port */
+ unsigned int flags; /* SNDRV_VIRMIDI_* */
+ rwlock_t filelist_lock;
+ struct list_head filelist;
+};
+
+/* sequencer mode:
+ * ATTACH = input/output events from midi device are routed to the
+ * attached sequencer port. sequencer port is not created
+ * by virmidi itself.
+ * the input to rawmidi must be processed by passing the
+ * incoming events via snd_virmidi_receive()
+ * DISPATCH = input/output events are routed to subscribers.
+ * sequencer port is created in virmidi.
+ */
+#define SNDRV_VIRMIDI_SEQ_NONE 0
+#define SNDRV_VIRMIDI_SEQ_ATTACH 1
+#define SNDRV_VIRMIDI_SEQ_DISPATCH 2
+
+int snd_virmidi_new(struct snd_card *card, int device, struct snd_rawmidi **rrmidi);
+
+#endif /* __SOUND_SEQ_VIRMIDI */
diff --git a/kernel/include/sound/sh_dac_audio.h b/kernel/include/sound/sh_dac_audio.h
new file mode 100644
index 000000000..f5deaf1dd
--- /dev/null
+++ b/kernel/include/sound/sh_dac_audio.h
@@ -0,0 +1,21 @@
+/*
+ * SH_DAC specific configuration, for the dac_audio platform_device
+ *
+ * Copyright (C) 2009 Rafael Ignacio Zurita <rizurita@yahoo.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#ifndef __INCLUDE_SH_DAC_AUDIO_H
+#define __INCLUDE_SH_DAC_AUDIO_H
+
+struct dac_audio_pdata {
+ int buffer_size;
+ int channel;
+ void (*start)(struct dac_audio_pdata *pd);
+ void (*stop)(struct dac_audio_pdata *pd);
+};
+
+#endif /* __INCLUDE_SH_DAC_AUDIO_H */
diff --git a/kernel/include/sound/sh_fsi.h b/kernel/include/sound/sh_fsi.h
new file mode 100644
index 000000000..7a9710b4b
--- /dev/null
+++ b/kernel/include/sound/sh_fsi.h
@@ -0,0 +1,35 @@
+#ifndef __SOUND_FSI_H
+#define __SOUND_FSI_H
+
+/*
+ * Fifo-attached Serial Interface (FSI) support for SH7724
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include <linux/clk.h>
+#include <sound/soc.h>
+
+/*
+ * flags
+ */
+#define SH_FSI_FMT_SPDIF (1 << 0) /* spdif for HDMI */
+#define SH_FSI_ENABLE_STREAM_MODE (1 << 1) /* for 16bit data */
+#define SH_FSI_CLK_CPG (1 << 2) /* FSIxCK + FSI-DIV */
+
+struct sh_fsi_port_info {
+ unsigned long flags;
+ int tx_id;
+ int rx_id;
+};
+
+struct sh_fsi_platform_info {
+ struct sh_fsi_port_info port_a;
+ struct sh_fsi_port_info port_b;
+};
+
+#endif /* __SOUND_FSI_H */
diff --git a/kernel/include/sound/simple_card.h b/kernel/include/sound/simple_card.h
new file mode 100644
index 000000000..b9b4f289f
--- /dev/null
+++ b/kernel/include/sound/simple_card.h
@@ -0,0 +1,36 @@
+/*
+ * ASoC simple sound card support
+ *
+ * Copyright (C) 2012 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __SIMPLE_CARD_H
+#define __SIMPLE_CARD_H
+
+#include <sound/soc.h>
+
+struct asoc_simple_dai {
+ const char *name;
+ unsigned int sysclk;
+ int slots;
+ int slot_width;
+ struct clk *clk;
+};
+
+struct asoc_simple_card_info {
+ const char *name;
+ const char *card;
+ const char *codec;
+ const char *platform;
+
+ unsigned int daifmt;
+ struct asoc_simple_dai cpu_dai;
+ struct asoc_simple_dai codec_dai;
+};
+
+#endif /* __SIMPLE_CARD_H */
diff --git a/kernel/include/sound/snd_wavefront.h b/kernel/include/sound/snd_wavefront.h
new file mode 100644
index 000000000..35e94b3d1
--- /dev/null
+++ b/kernel/include/sound/snd_wavefront.h
@@ -0,0 +1,142 @@
+#ifndef __SOUND_SND_WAVEFRONT_H__
+#define __SOUND_SND_WAVEFRONT_H__
+
+#include <sound/mpu401.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+#include <sound/wavefront.h> /* generic OSS/ALSA/user-level wavefront header */
+
+/* MIDI interface */
+
+struct _snd_wavefront_midi;
+struct _snd_wavefront_card;
+struct _snd_wavefront;
+
+typedef struct _snd_wavefront_midi snd_wavefront_midi_t;
+typedef struct _snd_wavefront_card snd_wavefront_card_t;
+typedef struct _snd_wavefront snd_wavefront_t;
+
+typedef enum { internal_mpu = 0, external_mpu = 1 } snd_wavefront_mpu_id;
+
+struct _snd_wavefront_midi {
+ unsigned long base; /* I/O port address */
+ char isvirtual; /* doing virtual MIDI stuff ? */
+ char istimer; /* timer is used */
+ snd_wavefront_mpu_id output_mpu; /* most-recently-used */
+ snd_wavefront_mpu_id input_mpu; /* most-recently-used */
+ unsigned int mode[2]; /* MPU401_MODE_XXX */
+ struct snd_rawmidi_substream *substream_output[2];
+ struct snd_rawmidi_substream *substream_input[2];
+ struct timer_list timer;
+ spinlock_t open;
+ spinlock_t virtual; /* protects isvirtual */
+};
+
+#define OUTPUT_READY 0x40
+#define INPUT_AVAIL 0x80
+#define MPU_ACK 0xFE
+#define UART_MODE_ON 0x3F
+
+extern struct snd_rawmidi_ops snd_wavefront_midi_output;
+extern struct snd_rawmidi_ops snd_wavefront_midi_input;
+
+extern void snd_wavefront_midi_enable_virtual (snd_wavefront_card_t *);
+extern void snd_wavefront_midi_disable_virtual (snd_wavefront_card_t *);
+extern void snd_wavefront_midi_interrupt (snd_wavefront_card_t *);
+extern int snd_wavefront_midi_start (snd_wavefront_card_t *);
+
+struct _snd_wavefront {
+ unsigned long irq; /* "you were one, one of the few ..." */
+ unsigned long base; /* low i/o port address */
+ struct resource *res_base; /* i/o port resource allocation */
+
+#define mpu_data_port base
+#define mpu_command_port base + 1 /* write semantics */
+#define mpu_status_port base + 1 /* read semantics */
+#define data_port base + 2
+#define status_port base + 3 /* read semantics */
+#define control_port base + 3 /* write semantics */
+#define block_port base + 4 /* 16 bit, writeonly */
+#define last_block_port base + 6 /* 16 bit, writeonly */
+
+ /* FX ports. These are mapped through the ICS2115 to the YS225.
+ The ICS2115 takes care of flipping the relevant pins on the
+ YS225 so that access to each of these ports does the right
+ thing. Note: these are NOT documented by Turtle Beach.
+ */
+
+#define fx_status base + 8
+#define fx_op base + 8
+#define fx_lcr base + 9
+#define fx_dsp_addr base + 0xa
+#define fx_dsp_page base + 0xb
+#define fx_dsp_lsb base + 0xc
+#define fx_dsp_msb base + 0xd
+#define fx_mod_addr base + 0xe
+#define fx_mod_data base + 0xf
+
+ volatile int irq_ok; /* set by interrupt handler */
+ volatile int irq_cnt; /* ditto */
+ char debug; /* debugging flags */
+ int freemem; /* installed RAM, in bytes */
+
+ char fw_version[2]; /* major = [0], minor = [1] */
+ char hw_version[2]; /* major = [0], minor = [1] */
+ char israw; /* needs Motorola microcode */
+ char has_fx; /* has FX processor (Tropez+) */
+ char fx_initialized; /* FX's register pages initialized */
+ char prog_status[WF_MAX_PROGRAM]; /* WF_SLOT_* */
+ char patch_status[WF_MAX_PATCH]; /* WF_SLOT_* */
+ char sample_status[WF_MAX_SAMPLE]; /* WF_ST_* | WF_SLOT_* */
+ int samples_used; /* how many */
+ char interrupts_are_midi; /* h/w MPU interrupts enabled ? */
+ char rom_samples_rdonly; /* can we write on ROM samples */
+ spinlock_t irq_lock;
+ wait_queue_head_t interrupt_sleeper;
+ snd_wavefront_midi_t midi; /* ICS2115 MIDI interface */
+ struct snd_card *card;
+};
+
+struct _snd_wavefront_card {
+ snd_wavefront_t wavefront;
+#ifdef CONFIG_PNP
+ struct pnp_dev *wss;
+ struct pnp_dev *ctrl;
+ struct pnp_dev *mpu;
+ struct pnp_dev *synth;
+#endif /* CONFIG_PNP */
+};
+
+extern void snd_wavefront_internal_interrupt (snd_wavefront_card_t *card);
+extern int snd_wavefront_detect_irq (snd_wavefront_t *dev) ;
+extern int snd_wavefront_check_irq (snd_wavefront_t *dev, int irq);
+extern int snd_wavefront_restart (snd_wavefront_t *dev);
+extern int snd_wavefront_start (snd_wavefront_t *dev);
+extern int snd_wavefront_detect (snd_wavefront_card_t *card);
+extern int snd_wavefront_config_midi (snd_wavefront_t *dev) ;
+extern int snd_wavefront_cmd (snd_wavefront_t *, int, unsigned char *,
+ unsigned char *);
+
+extern int snd_wavefront_synth_ioctl (struct snd_hwdep *,
+ struct file *,
+ unsigned int cmd,
+ unsigned long arg);
+extern int snd_wavefront_synth_open (struct snd_hwdep *, struct file *);
+extern int snd_wavefront_synth_release (struct snd_hwdep *, struct file *);
+
+/* FX processor - see also yss225.[ch] */
+
+extern int snd_wavefront_fx_start (snd_wavefront_t *);
+extern int snd_wavefront_fx_detect (snd_wavefront_t *);
+extern int snd_wavefront_fx_ioctl (struct snd_hwdep *,
+ struct file *,
+ unsigned int cmd,
+ unsigned long arg);
+extern int snd_wavefront_fx_open (struct snd_hwdep *, struct file *);
+extern int snd_wavefront_fx_release (struct snd_hwdep *, struct file *);
+
+/* prefix in all snd_printk() delivered messages */
+
+#define LOGNAME "WaveFront: "
+
+#endif /* __SOUND_SND_WAVEFRONT_H__ */
diff --git a/kernel/include/sound/soc-dai.h b/kernel/include/sound/soc-dai.h
new file mode 100644
index 000000000..2df96b138
--- /dev/null
+++ b/kernel/include/sound/soc-dai.h
@@ -0,0 +1,316 @@
+/*
+ * linux/sound/soc-dai.h -- ALSA SoC Layer
+ *
+ * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Digital Audio Interface (DAI) API.
+ */
+
+#ifndef __LINUX_SND_SOC_DAI_H
+#define __LINUX_SND_SOC_DAI_H
+
+
+#include <linux/list.h>
+
+struct snd_pcm_substream;
+struct snd_soc_dapm_widget;
+struct snd_compr_stream;
+
+/*
+ * DAI hardware audio formats.
+ *
+ * Describes the physical PCM data formating and clocking. Add new formats
+ * to the end.
+ */
+#define SND_SOC_DAIFMT_I2S 1 /* I2S mode */
+#define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */
+#define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */
+#define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */
+#define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */
+#define SND_SOC_DAIFMT_AC97 6 /* AC97 */
+#define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */
+
+/* left and right justified also known as MSB and LSB respectively */
+#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
+#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
+
+/*
+ * DAI Clock gating.
+ *
+ * DAI bit clocks can be be gated (disabled) when the DAI is not
+ * sending or receiving PCM data in a frame. This can be used to save power.
+ */
+#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
+#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
+
+/*
+ * DAI hardware signal inversions.
+ *
+ * Specifies whether the DAI can also support inverted clocks for the specified
+ * format.
+ */
+#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
+#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
+#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
+
+/*
+ * DAI hardware clock masters.
+ *
+ * This is wrt the codec, the inverse is true for the interface
+ * i.e. if the codec is clk and FRM master then the interface is
+ * clk and frame slave.
+ */
+#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
+#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
+#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
+#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
+
+#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
+#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
+#define SND_SOC_DAIFMT_INV_MASK 0x0f00
+#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
+
+/*
+ * Master Clock Directions
+ */
+#define SND_SOC_CLOCK_IN 0
+#define SND_SOC_CLOCK_OUT 1
+
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+ SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S16_BE |\
+ SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S20_3BE |\
+ SNDRV_PCM_FMTBIT_S24_3LE |\
+ SNDRV_PCM_FMTBIT_S24_3BE |\
+ SNDRV_PCM_FMTBIT_S32_LE |\
+ SNDRV_PCM_FMTBIT_S32_BE)
+
+struct snd_soc_dai_driver;
+struct snd_soc_dai;
+struct snd_ac97_bus_ops;
+
+/* Digital Audio Interface clocking API.*/
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir);
+
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+ int div_id, int div);
+
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
+ int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
+
+int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
+
+/* Digital Audio interface formatting */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
+
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
+
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot);
+
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
+
+/* Digital Audio Interface mute */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
+ int direction);
+
+int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
+
+struct snd_soc_dai_ops {
+ /*
+ * DAI clocking configuration, all optional.
+ * Called by soc_card drivers, normally in their hw_params.
+ */
+ int (*set_sysclk)(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir);
+ int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out);
+ int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
+ int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
+
+ /*
+ * DAI format configuration
+ * Called by soc_card drivers, normally in their hw_params.
+ */
+ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
+ int (*xlate_tdm_slot_mask)(unsigned int slots,
+ unsigned int *tx_mask, unsigned int *rx_mask);
+ int (*set_tdm_slot)(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width);
+ int (*set_channel_map)(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot);
+ int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
+
+ /*
+ * DAI digital mute - optional.
+ * Called by soc-core to minimise any pops.
+ */
+ int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+ int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
+
+ /*
+ * ALSA PCM audio operations - all optional.
+ * Called by soc-core during audio PCM operations.
+ */
+ int (*startup)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+ void (*shutdown)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+ int (*hw_params)(struct snd_pcm_substream *,
+ struct snd_pcm_hw_params *, struct snd_soc_dai *);
+ int (*hw_free)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+ int (*prepare)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+ /*
+ * NOTE: Commands passed to the trigger function are not necessarily
+ * compatible with the current state of the dai. For example this
+ * sequence of commands is possible: START STOP STOP.
+ * So do not unconditionally use refcounting functions in the trigger
+ * function, e.g. clk_enable/disable.
+ */
+ int (*trigger)(struct snd_pcm_substream *, int,
+ struct snd_soc_dai *);
+ int (*bespoke_trigger)(struct snd_pcm_substream *, int,
+ struct snd_soc_dai *);
+ /*
+ * For hardware based FIFO caused delay reporting.
+ * Optional.
+ */
+ snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+};
+
+/*
+ * Digital Audio Interface Driver.
+ *
+ * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
+ * operations and capabilities. Codec and platform drivers will register this
+ * structure for every DAI they have.
+ *
+ * This structure covers the clocking, formating and ALSA operations for each
+ * interface.
+ */
+struct snd_soc_dai_driver {
+ /* DAI description */
+ const char *name;
+ unsigned int id;
+ unsigned int base;
+
+ /* DAI driver callbacks */
+ int (*probe)(struct snd_soc_dai *dai);
+ int (*remove)(struct snd_soc_dai *dai);
+ int (*suspend)(struct snd_soc_dai *dai);
+ int (*resume)(struct snd_soc_dai *dai);
+ /* compress dai */
+ bool compress_dai;
+ /* DAI is also used for the control bus */
+ bool bus_control;
+
+ /* ops */
+ const struct snd_soc_dai_ops *ops;
+
+ /* DAI capabilities */
+ struct snd_soc_pcm_stream capture;
+ struct snd_soc_pcm_stream playback;
+ unsigned int symmetric_rates:1;
+ unsigned int symmetric_channels:1;
+ unsigned int symmetric_samplebits:1;
+
+ /* probe ordering - for components with runtime dependencies */
+ int probe_order;
+ int remove_order;
+};
+
+/*
+ * Digital Audio Interface runtime data.
+ *
+ * Holds runtime data for a DAI.
+ */
+struct snd_soc_dai {
+ const char *name;
+ int id;
+ struct device *dev;
+
+ /* driver ops */
+ struct snd_soc_dai_driver *driver;
+
+ /* DAI runtime info */
+ unsigned int capture_active:1; /* stream is in use */
+ unsigned int playback_active:1; /* stream is in use */
+ unsigned int symmetric_rates:1;
+ unsigned int symmetric_channels:1;
+ unsigned int symmetric_samplebits:1;
+ unsigned int active;
+ unsigned char probed:1;
+
+ struct snd_soc_dapm_widget *playback_widget;
+ struct snd_soc_dapm_widget *capture_widget;
+
+ /* DAI DMA data */
+ void *playback_dma_data;
+ void *capture_dma_data;
+
+ /* Symmetry data - only valid if symmetry is being enforced */
+ unsigned int rate;
+ unsigned int channels;
+ unsigned int sample_bits;
+
+ /* parent platform/codec */
+ struct snd_soc_codec *codec;
+ struct snd_soc_component *component;
+
+ /* CODEC TDM slot masks and params (for fixup) */
+ unsigned int tx_mask;
+ unsigned int rx_mask;
+
+ struct list_head list;
+};
+
+static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
+ const struct snd_pcm_substream *ss)
+{
+ return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ dai->playback_dma_data : dai->capture_dma_data;
+}
+
+static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
+ const struct snd_pcm_substream *ss,
+ void *data)
+{
+ if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->playback_dma_data = data;
+ else
+ dai->capture_dma_data = data;
+}
+
+static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
+ void *playback, void *capture)
+{
+ dai->playback_dma_data = playback;
+ dai->capture_dma_data = capture;
+}
+
+static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
+ void *data)
+{
+ dev_set_drvdata(dai->dev, data);
+}
+
+static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
+{
+ return dev_get_drvdata(dai->dev);
+}
+
+#endif
diff --git a/kernel/include/sound/soc-dapm.h b/kernel/include/sound/soc-dapm.h
new file mode 100644
index 000000000..1065095c6
--- /dev/null
+++ b/kernel/include/sound/soc-dapm.h
@@ -0,0 +1,626 @@
+/*
+ * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management
+ *
+ * Author: Liam Girdwood
+ * Created: Aug 11th 2005
+ * Copyright: Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_DAPM_H
+#define __LINUX_SND_SOC_DAPM_H
+
+#include <linux/types.h>
+#include <sound/control.h>
+
+struct device;
+
+/* widget has no PM register bit */
+#define SND_SOC_NOPM -1
+
+/*
+ * SoC dynamic audio power management
+ *
+ * We can have up to 4 power domains
+ * 1. Codec domain - VREF, VMID
+ * Usually controlled at codec probe/remove, although can be set
+ * at stream time if power is not needed for sidetone, etc.
+ * 2. Platform/Machine domain - physically connected inputs and outputs
+ * Is platform/machine and user action specific, is set in the machine
+ * driver and by userspace e.g when HP are inserted
+ * 3. Path domain - Internal codec path mixers
+ * Are automatically set when mixer and mux settings are
+ * changed by the user.
+ * 4. Stream domain - DAC's and ADC's.
+ * Enabled when stream playback/capture is started.
+ */
+
+/* codec domain */
+#define SND_SOC_DAPM_VMID(wname) \
+{ .id = snd_soc_dapm_vmid, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0}
+
+/* platform domain */
+#define SND_SOC_DAPM_SIGGEN(wname) \
+{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM }
+#define SND_SOC_DAPM_INPUT(wname) \
+{ .id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM }
+#define SND_SOC_DAPM_OUTPUT(wname) \
+{ .id = snd_soc_dapm_output, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM }
+#define SND_SOC_DAPM_MIC(wname, wevent) \
+{ .id = snd_soc_dapm_mic, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
+ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
+#define SND_SOC_DAPM_HP(wname, wevent) \
+{ .id = snd_soc_dapm_hp, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
+#define SND_SOC_DAPM_SPK(wname, wevent) \
+{ .id = snd_soc_dapm_spk, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
+#define SND_SOC_DAPM_LINE(wname, wevent) \
+{ .id = snd_soc_dapm_line, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD}
+
+#define SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert) \
+ .reg = wreg, .mask = 1, .shift = wshift, \
+ .on_val = winvert ? 0 : 1, .off_val = winvert ? 1 : 0
+
+/* path domain */
+#define SND_SOC_DAPM_PGA(wname, wreg, wshift, winvert,\
+ wcontrols, wncontrols) \
+{ .id = snd_soc_dapm_pga, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+#define SND_SOC_DAPM_OUT_DRV(wname, wreg, wshift, winvert,\
+ wcontrols, wncontrols) \
+{ .id = snd_soc_dapm_out_drv, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+#define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \
+ wcontrols, wncontrols)\
+{ .id = snd_soc_dapm_mixer, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+#define SND_SOC_DAPM_MIXER_NAMED_CTL(wname, wreg, wshift, winvert, \
+ wcontrols, wncontrols)\
+{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+#define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_micbias, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = NULL, .num_kcontrols = 0}
+#define SND_SOC_DAPM_SWITCH(wname, wreg, wshift, winvert, wcontrols) \
+{ .id = snd_soc_dapm_switch, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = 1}
+#define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \
+{ .id = snd_soc_dapm_mux, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = 1}
+
+/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
+#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
+ wcontrols) \
+{ .id = snd_soc_dapm_pga, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+#define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \
+ wcontrols)\
+{ .id = snd_soc_dapm_mixer, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+#define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \
+ wcontrols)\
+{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+
+/* path domain with event - event handler must return 0 for success */
+#define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \
+ wncontrols, wevent, wflags) \
+{ .id = snd_soc_dapm_pga, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
+ .event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_OUT_DRV_E(wname, wreg, wshift, winvert, wcontrols, \
+ wncontrols, wevent, wflags) \
+{ .id = snd_soc_dapm_out_drv, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
+ .event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \
+ wncontrols, wevent, wflags) \
+{ .id = snd_soc_dapm_mixer, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols, \
+ .event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_MIXER_NAMED_CTL_E(wname, wreg, wshift, winvert, \
+ wcontrols, wncontrols, wevent, wflags) \
+{ .id = snd_soc_dapm_mixer, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, \
+ .num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_switch, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = 1, \
+ .event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_mux, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = 1, \
+ .event = wevent, .event_flags = wflags}
+
+/* additional sequencing control within an event type */
+#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_pga, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .event = wevent, .event_flags = wflags, \
+ .subseq = wsubseq}
+#define SND_SOC_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, winvert, wevent, \
+ wflags) \
+{ .id = snd_soc_dapm_supply, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .event = wevent, .event_flags = wflags, .subseq = wsubseq}
+
+/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
+#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_pga, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+ .event = wevent, .event_flags = wflags}
+#define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_mixer, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+ .event = wevent, .event_flags = wflags}
+#define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \
+ wcontrols, wevent, wflags) \
+{ .id = snd_soc_dapm_mixer, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+ .event = wevent, .event_flags = wflags}
+
+/* events that are pre and post DAPM */
+#define SND_SOC_DAPM_PRE(wname, wevent) \
+{ .id = snd_soc_dapm_pre, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
+ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD}
+#define SND_SOC_DAPM_POST(wname, wevent) \
+{ .id = snd_soc_dapm_post, .name = wname, .kcontrol_news = NULL, \
+ .num_kcontrols = 0, .reg = SND_SOC_NOPM, .event = wevent, \
+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD}
+
+/* stream domain */
+#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
+#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .event = wevent, .event_flags = wflags }
+#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
+#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .event = wevent, .event_flags = wflags }
+#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert) }
+#define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .event = wevent, .event_flags = wflags}
+
+#define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
+#define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \
+{ .id = snd_soc_dapm_clock_supply, .name = wname, \
+ .reg = SND_SOC_NOPM, .event = dapm_clock_event, \
+ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
+
+/* generic widgets */
+#define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
+{ .id = wid, .name = wname, .kcontrol_news = NULL, .num_kcontrols = 0, \
+ .reg = wreg, .shift = wshift, .mask = wmask, \
+ .on_val = won_val, .off_val = woff_val, }
+#define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \
+{ .id = snd_soc_dapm_supply, .name = wname, \
+ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+ .event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags) \
+{ .id = snd_soc_dapm_regulator_supply, .name = wname, \
+ .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
+ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
+ .on_val = wflags}
+
+
+/* dapm kcontrol types */
+#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
+#define SOC_DAPM_SINGLE_AUTODISABLE(xname, reg, shift, max, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) }
+#define SOC_DAPM_SINGLE_VIRT(xname, max) \
+ SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0)
+#define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
+#define SOC_DAPM_SINGLE_TLV_AUTODISABLE(xname, reg, shift, max, invert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) }
+#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \
+ SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array)
+#define SOC_DAPM_ENUM(xname, xenum) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = snd_soc_dapm_get_enum_double, \
+ .put = snd_soc_dapm_put_enum_double, \
+ .private_value = (unsigned long)&xenum }
+#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = xget, \
+ .put = xput, \
+ .private_value = (unsigned long)&xenum }
+#define SOC_DAPM_PIN_SWITCH(xname) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname " Switch", \
+ .info = snd_soc_dapm_info_pin_switch, \
+ .get = snd_soc_dapm_get_pin_switch, \
+ .put = snd_soc_dapm_put_pin_switch, \
+ .private_value = (unsigned long)xname }
+
+/* dapm stream operations */
+#define SND_SOC_DAPM_STREAM_NOP 0x0
+#define SND_SOC_DAPM_STREAM_START 0x1
+#define SND_SOC_DAPM_STREAM_STOP 0x2
+#define SND_SOC_DAPM_STREAM_SUSPEND 0x4
+#define SND_SOC_DAPM_STREAM_RESUME 0x8
+#define SND_SOC_DAPM_STREAM_PAUSE_PUSH 0x10
+#define SND_SOC_DAPM_STREAM_PAUSE_RELEASE 0x20
+
+/* dapm event types */
+#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
+#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
+#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
+#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
+#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
+#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
+#define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */
+#define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */
+#define SND_SOC_DAPM_PRE_POST_PMD \
+ (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)
+
+/* convenience event type detection */
+#define SND_SOC_DAPM_EVENT_ON(e) \
+ (e & (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU))
+#define SND_SOC_DAPM_EVENT_OFF(e) \
+ (e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD))
+
+/* regulator widget flags */
+#define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */
+
+struct snd_soc_dapm_widget;
+enum snd_soc_dapm_type;
+struct snd_soc_dapm_path;
+struct snd_soc_dapm_pin;
+struct snd_soc_dapm_route;
+struct snd_soc_dapm_context;
+struct regulator;
+struct snd_soc_dapm_widget_list;
+struct snd_soc_dapm_update;
+
+int dapm_regulator_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event);
+int dapm_clock_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event);
+
+/* dapm controls */
+int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uncontrol);
+int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uncontrol);
+int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_widget *widget,
+ int num);
+int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
+ struct snd_soc_dai *dai);
+int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
+void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card);
+int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
+ const struct snd_soc_pcm_stream *params,
+ unsigned int num_params,
+ struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
+
+/* dapm path setup */
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
+void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route, int num);
+
+/* dapm events */
+void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
+ int event);
+void snd_soc_dapm_shutdown(struct snd_soc_card *card);
+
+/* external DAPM widget events */
+int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm,
+ struct snd_kcontrol *kcontrol, int connect,
+ struct snd_soc_dapm_update *update);
+int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm,
+ struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e,
+ struct snd_soc_dapm_update *update);
+
+/* dapm sys fs - used by the core */
+extern struct attribute *soc_dapm_dev_attrs[];
+void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
+ struct dentry *parent);
+
+/* dapm audio pin control and status */
+int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_disable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin);
+int snd_soc_dapm_nc_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_sync_unlocked(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol);
+
+/* Mostly internal - should not normally be used */
+void dapm_mark_endpoints_dirty(struct snd_soc_card *card);
+
+/* dapm path query */
+int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
+ struct snd_soc_dapm_widget_list **list);
+
+struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(
+ struct snd_kcontrol *kcontrol);
+
+/* dapm widget types */
+enum snd_soc_dapm_type {
+ snd_soc_dapm_input = 0, /* input pin */
+ snd_soc_dapm_output, /* output pin */
+ snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */
+ snd_soc_dapm_mixer, /* mixes several analog signals together */
+ snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */
+ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */
+ snd_soc_dapm_out_drv, /* output driver */
+ snd_soc_dapm_adc, /* analog to digital converter */
+ snd_soc_dapm_dac, /* digital to analog converter */
+ snd_soc_dapm_micbias, /* microphone bias (power) */
+ snd_soc_dapm_mic, /* microphone */
+ snd_soc_dapm_hp, /* headphones */
+ snd_soc_dapm_spk, /* speaker */
+ snd_soc_dapm_line, /* line input/output */
+ snd_soc_dapm_switch, /* analog switch */
+ snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */
+ snd_soc_dapm_pre, /* machine specific pre widget - exec first */
+ snd_soc_dapm_post, /* machine specific post widget - exec last */
+ snd_soc_dapm_supply, /* power/clock supply */
+ snd_soc_dapm_regulator_supply, /* external regulator */
+ snd_soc_dapm_clock_supply, /* external clock */
+ snd_soc_dapm_aif_in, /* audio interface input */
+ snd_soc_dapm_aif_out, /* audio interface output */
+ snd_soc_dapm_siggen, /* signal generator */
+ snd_soc_dapm_dai_in, /* link to DAI structure */
+ snd_soc_dapm_dai_out,
+ snd_soc_dapm_dai_link, /* link between two DAI structures */
+ snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */
+};
+
+enum snd_soc_dapm_subclass {
+ SND_SOC_DAPM_CLASS_INIT = 0,
+ SND_SOC_DAPM_CLASS_RUNTIME = 1,
+};
+
+/*
+ * DAPM audio route definition.
+ *
+ * Defines an audio route originating at source via control and finishing
+ * at sink.
+ */
+struct snd_soc_dapm_route {
+ const char *sink;
+ const char *control;
+ const char *source;
+
+ /* Note: currently only supported for links where source is a supply */
+ int (*connected)(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
+};
+
+/* dapm audio path between two widgets */
+struct snd_soc_dapm_path {
+ const char *name;
+
+ /* source (input) and sink (output) widgets */
+ struct snd_soc_dapm_widget *source;
+ struct snd_soc_dapm_widget *sink;
+
+ /* status */
+ u32 connect:1; /* source and sink widgets are connected */
+ u32 walking:1; /* path is in the process of being walked */
+ u32 weak:1; /* path ignored for power management */
+ u32 is_supply:1; /* At least one of the connected widgets is a supply */
+
+ int (*connected)(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
+
+ struct list_head list_source;
+ struct list_head list_sink;
+ struct list_head list_kcontrol;
+ struct list_head list;
+};
+
+/* dapm widget */
+struct snd_soc_dapm_widget {
+ enum snd_soc_dapm_type id;
+ const char *name; /* widget name */
+ const char *sname; /* stream name */
+ struct list_head list;
+ struct snd_soc_dapm_context *dapm;
+
+ void *priv; /* widget specific data */
+ struct regulator *regulator; /* attached regulator */
+ const struct snd_soc_pcm_stream *params; /* params for dai links */
+ unsigned int num_params; /* number of params for dai links */
+ unsigned int params_select; /* currently selected param for dai link */
+
+ /* dapm control */
+ int reg; /* negative reg = no direct dapm */
+ unsigned char shift; /* bits to shift */
+ unsigned int mask; /* non-shifted mask */
+ unsigned int on_val; /* on state value */
+ unsigned int off_val; /* off state value */
+ unsigned char power:1; /* block power status */
+ unsigned char active:1; /* active stream on DAC, ADC's */
+ unsigned char connected:1; /* connected codec pin */
+ unsigned char new:1; /* cnew complete */
+ unsigned char force:1; /* force state */
+ unsigned char ignore_suspend:1; /* kept enabled over suspend */
+ unsigned char new_power:1; /* power from this run */
+ unsigned char power_checked:1; /* power checked this run */
+ unsigned char is_supply:1; /* Widget is a supply type widget */
+ unsigned char is_sink:1; /* Widget is a sink type widget */
+ unsigned char is_source:1; /* Widget is a source type widget */
+ int subseq; /* sort within widget type */
+
+ int (*power_check)(struct snd_soc_dapm_widget *w);
+
+ /* external events */
+ unsigned short event_flags; /* flags to specify event types */
+ int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int);
+
+ /* kcontrols that relate to this widget */
+ int num_kcontrols;
+ const struct snd_kcontrol_new *kcontrol_news;
+ struct snd_kcontrol **kcontrols;
+
+ /* widget input and outputs */
+ struct list_head sources;
+ struct list_head sinks;
+
+ /* used during DAPM updates */
+ struct list_head work_list;
+ struct list_head power_list;
+ struct list_head dirty;
+ int inputs;
+ int outputs;
+
+ struct clk *clk;
+};
+
+struct snd_soc_dapm_update {
+ struct snd_kcontrol *kcontrol;
+ int reg;
+ int mask;
+ int val;
+};
+
+/* DAPM context */
+struct snd_soc_dapm_context {
+ enum snd_soc_bias_level bias_level;
+ unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
+ /* Go to BIAS_OFF in suspend if the DAPM context is idle */
+ unsigned int suspend_bias_off:1;
+ void (*seq_notifier)(struct snd_soc_dapm_context *,
+ enum snd_soc_dapm_type, int);
+
+ struct device *dev; /* from parent - for debug */
+ struct snd_soc_component *component; /* parent component */
+ struct snd_soc_card *card; /* parent card */
+
+ /* used during DAPM updates */
+ enum snd_soc_bias_level target_bias_level;
+ struct list_head list;
+
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+ int (*set_bias_level)(struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level);
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_dapm;
+#endif
+};
+
+/* A list of widgets associated with an object, typically a snd_kcontrol */
+struct snd_soc_dapm_widget_list {
+ int num_widgets;
+ struct snd_soc_dapm_widget *widgets[0];
+};
+
+struct snd_soc_dapm_stats {
+ int power_checks;
+ int path_checks;
+ int neighbour_checks;
+};
+
+#endif
diff --git a/kernel/include/sound/soc-dpcm.h b/kernel/include/sound/soc-dpcm.h
new file mode 100644
index 000000000..806059052
--- /dev/null
+++ b/kernel/include/sound/soc-dpcm.h
@@ -0,0 +1,162 @@
+/*
+ * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support
+ *
+ * Author: Liam Girdwood <lrg@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_DPCM_H
+#define __LINUX_SND_SOC_DPCM_H
+
+#include <linux/slab.h>
+#include <linux/list.h>
+#include <sound/pcm.h>
+
+struct snd_soc_pcm_runtime;
+
+/*
+ * Types of runtime_update to perform. e.g. originated from FE PCM ops
+ * or audio route changes triggered by muxes/mixers.
+ */
+enum snd_soc_dpcm_update {
+ SND_SOC_DPCM_UPDATE_NO = 0,
+ SND_SOC_DPCM_UPDATE_BE,
+ SND_SOC_DPCM_UPDATE_FE,
+};
+
+/*
+ * Dynamic PCM Frontend -> Backend link management states.
+ */
+enum snd_soc_dpcm_link_state {
+ SND_SOC_DPCM_LINK_STATE_NEW = 0, /* newly created link */
+ SND_SOC_DPCM_LINK_STATE_FREE, /* link to be dismantled */
+};
+
+/*
+ * Dynamic PCM Frontend -> Backend link PCM states.
+ */
+enum snd_soc_dpcm_state {
+ SND_SOC_DPCM_STATE_NEW = 0,
+ SND_SOC_DPCM_STATE_OPEN,
+ SND_SOC_DPCM_STATE_HW_PARAMS,
+ SND_SOC_DPCM_STATE_PREPARE,
+ SND_SOC_DPCM_STATE_START,
+ SND_SOC_DPCM_STATE_STOP,
+ SND_SOC_DPCM_STATE_PAUSED,
+ SND_SOC_DPCM_STATE_SUSPEND,
+ SND_SOC_DPCM_STATE_HW_FREE,
+ SND_SOC_DPCM_STATE_CLOSE,
+};
+
+/*
+ * Dynamic PCM trigger ordering. Triggering flexibility is required as some
+ * DSPs require triggering before/after their CPU platform and DAIs.
+ *
+ * i.e. some clients may want to manually order this call in their PCM
+ * trigger() whilst others will just use the regular core ordering.
+ */
+enum snd_soc_dpcm_trigger {
+ SND_SOC_DPCM_TRIGGER_PRE = 0,
+ SND_SOC_DPCM_TRIGGER_POST,
+ SND_SOC_DPCM_TRIGGER_BESPOKE,
+};
+
+/*
+ * Dynamic PCM link
+ * This links together a FE and BE DAI at runtime and stores the link
+ * state information and the hw_params configuration.
+ */
+struct snd_soc_dpcm {
+ /* FE and BE DAIs*/
+ struct snd_soc_pcm_runtime *be;
+ struct snd_soc_pcm_runtime *fe;
+
+ /* link state */
+ enum snd_soc_dpcm_link_state state;
+
+ /* list of BE and FE for this DPCM link */
+ struct list_head list_be;
+ struct list_head list_fe;
+
+ /* hw params for this link - may be different for each link */
+ struct snd_pcm_hw_params hw_params;
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_state;
+#endif
+};
+
+/*
+ * Dynamic PCM runtime data.
+ */
+struct snd_soc_dpcm_runtime {
+ struct list_head be_clients;
+ struct list_head fe_clients;
+
+ int users;
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcm_hw_params hw_params;
+
+ /* state and update */
+ enum snd_soc_dpcm_update runtime_update;
+ enum snd_soc_dpcm_state state;
+
+ int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */
+};
+
+/* can this BE stop and free */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream);
+
+/* can this BE perform a hw_params() */
+int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream);
+
+/* is the current PCM operation for this FE ? */
+int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream);
+
+/* is the current PCM operation for this BE ? */
+int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream);
+
+/* get the substream for this BE */
+struct snd_pcm_substream *
+ snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream);
+
+/* get the BE runtime state */
+enum snd_soc_dpcm_state
+ snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream);
+
+/* set the BE runtime state */
+void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
+ enum snd_soc_dpcm_state state);
+
+/* internal use only */
+int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
+void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd);
+int soc_dpcm_runtime_update(struct snd_soc_card *);
+
+int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
+ int stream, struct snd_soc_dapm_widget_list **list_);
+int dpcm_process_paths(struct snd_soc_pcm_runtime *fe,
+ int stream, struct snd_soc_dapm_widget_list **list, int new);
+int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream);
+int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream);
+void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream);
+void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream);
+int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream);
+int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream);
+int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd);
+int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream);
+int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
+ int event);
+
+static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list)
+{
+ kfree(*list);
+}
+
+
+#endif
diff --git a/kernel/include/sound/soc.h b/kernel/include/sound/soc.h
new file mode 100644
index 000000000..f6226914a
--- /dev/null
+++ b/kernel/include/sound/soc.h
@@ -0,0 +1,1562 @@
+/*
+ * linux/sound/soc.h -- ALSA SoC Layer
+ *
+ * Author: Liam Girdwood
+ * Created: Aug 11th 2005
+ * Copyright: Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_H
+#define __LINUX_SND_SOC_H
+
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <linux/types.h>
+#include <linux/notifier.h>
+#include <linux/workqueue.h>
+#include <linux/interrupt.h>
+#include <linux/kernel.h>
+#include <linux/regmap.h>
+#include <linux/log2.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/compress_driver.h>
+#include <sound/control.h>
+#include <sound/ac97_codec.h>
+
+/*
+ * Convenience kcontrol builders
+ */
+#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert, xautodisable) \
+ ((unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xreg, .shift = shift_left, \
+ .rshift = shift_right, .max = xmax, .platform_max = xmax, \
+ .invert = xinvert, .autodisable = xautodisable})
+#define SOC_DOUBLE_S_VALUE(xreg, shift_left, shift_right, xmin, xmax, xsign_bit, xinvert, xautodisable) \
+ ((unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xreg, .shift = shift_left, \
+ .rshift = shift_right, .min = xmin, .max = xmax, .platform_max = xmax, \
+ .sign_bit = xsign_bit, .invert = xinvert, .autodisable = xautodisable})
+#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \
+ SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable)
+#define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
+ ((unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert})
+#define SOC_DOUBLE_R_VALUE(xlreg, xrreg, xshift, xmax, xinvert) \
+ ((unsigned long)&(struct soc_mixer_control) \
+ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
+ .max = xmax, .platform_max = xmax, .invert = xinvert})
+#define SOC_DOUBLE_R_S_VALUE(xlreg, xrreg, xshift, xmin, xmax, xsign_bit, xinvert) \
+ ((unsigned long)&(struct soc_mixer_control) \
+ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
+ .max = xmax, .min = xmin, .platform_max = xmax, .sign_bit = xsign_bit, \
+ .invert = xinvert})
+#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \
+ ((unsigned long)&(struct soc_mixer_control) \
+ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
+ .min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert})
+#define SOC_SINGLE(xname, reg, shift, max, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
+ .put = snd_soc_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
+#define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \
+ .put = snd_soc_put_volsw_range, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xreg, .shift = xshift, \
+ .rshift = xshift, .min = xmin, .max = xmax, \
+ .platform_max = xmax, .invert = xinvert} }
+#define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
+ .put = snd_soc_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) }
+#define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array),\
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw_sx,\
+ .put = snd_soc_put_volsw_sx, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xreg, \
+ .shift = xshift, .rshift = xshift, \
+ .max = xmax, .min = xmin} }
+#define SOC_SINGLE_RANGE_TLV(xname, xreg, xshift, xmin, xmax, xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_range, \
+ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xreg, .shift = xshift, \
+ .rshift = xshift, .min = xmin, .max = xmax, \
+ .platform_max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
+ .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
+ max, invert, 0) }
+#define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
+ xmax, xinvert) }
+#define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \
+ xmax, xinvert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .info = snd_soc_info_volsw_range, \
+ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \
+ .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \
+ xshift, xmin, xmax, xinvert) }
+#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
+ .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
+ max, invert, 0) }
+#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
+ xmax, xinvert) }
+#define SOC_DOUBLE_R_RANGE_TLV(xname, reg_left, reg_right, xshift, xmin, \
+ xmax, xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_range, \
+ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \
+ .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \
+ xshift, xmin, xmax, xinvert) }
+#define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw_sx, \
+ .put = snd_soc_put_volsw_sx, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .rreg = xrreg, \
+ .shift = xshift, .rshift = xshift, \
+ .max = xmax, .min = xmin} }
+#define SOC_DOUBLE_R_S_TLV(xname, reg_left, reg_right, xshift, xmin, xmax, xsign_bit, xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_R_S_VALUE(reg_left, reg_right, xshift, \
+ xmin, xmax, xsign_bit, xinvert) }
+#define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
+ .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_S_VALUE(xreg, 0, 8, xmin, xmax, 7, 0, 0) }
+#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \
+{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
+ .items = xitems, .texts = xtexts, \
+ .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0}
+#define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \
+ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts)
+#define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \
+{ .items = xitems, .texts = xtexts }
+#define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \
+{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
+ .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues}
+#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \
+ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues)
+#define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts)
+#define SOC_ENUM(xname, xenum) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\
+ .info = snd_soc_info_enum_double, \
+ .get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \
+ .private_value = (unsigned long)&xenum }
+#define SOC_SINGLE_EXT(xname, xreg, xshift, xmax, xinvert,\
+ xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) }
+#define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\
+ xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = \
+ SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) }
+#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) }
+#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \
+ xmax, xinvert, 0) }
+#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
+ xmax, xinvert) }
+#define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_bool_ext, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = xdata }
+#define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&xenum }
+#define SOC_VALUE_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \
+ SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put)
+
+#define SND_SOC_BYTES(xname, xbase, xregs) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
+ .put = snd_soc_bytes_put, .private_value = \
+ ((unsigned long)&(struct soc_bytes) \
+ {.base = xbase, .num_regs = xregs }) }
+
+#define SND_SOC_BYTES_MASK(xname, xbase, xregs, xmask) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
+ .put = snd_soc_bytes_put, .private_value = \
+ ((unsigned long)&(struct soc_bytes) \
+ {.base = xbase, .num_regs = xregs, \
+ .mask = xmask }) }
+
+#define SND_SOC_BYTES_EXT(xname, xcount, xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_bytes_info_ext, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_bytes_ext) \
+ {.max = xcount} }
+#define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
+ .tlv.c = (snd_soc_bytes_tlv_callback), \
+ .info = snd_soc_bytes_info_ext, \
+ .private_value = (unsigned long)&(struct soc_bytes_ext) \
+ {.max = xcount, .get = xhandler_get, .put = xhandler_put, } }
+#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \
+ xmin, xmax, xinvert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .info = snd_soc_info_xr_sx, .get = snd_soc_get_xr_sx, \
+ .put = snd_soc_put_xr_sx, \
+ .private_value = (unsigned long)&(struct soc_mreg_control) \
+ {.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \
+ .invert = xinvert, .min = xmin, .max = xmax} }
+
+#define SOC_SINGLE_STROBE(xname, xreg, xshift, xinvert) \
+ SOC_SINGLE_EXT(xname, xreg, xshift, 1, xinvert, \
+ snd_soc_get_strobe, snd_soc_put_strobe)
+
+/*
+ * Simplified versions of above macros, declaring a struct and calculating
+ * ARRAY_SIZE internally
+ */
+#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \
+ const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
+ ARRAY_SIZE(xtexts), xtexts)
+#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \
+ SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts)
+#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \
+ const struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts)
+#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \
+ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \
+ ARRAY_SIZE(xtexts), xtexts, xvalues)
+#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
+ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
+#define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
+ const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
+
+/*
+ * Component probe and remove ordering levels for components with runtime
+ * dependencies.
+ */
+#define SND_SOC_COMP_ORDER_FIRST -2
+#define SND_SOC_COMP_ORDER_EARLY -1
+#define SND_SOC_COMP_ORDER_NORMAL 0
+#define SND_SOC_COMP_ORDER_LATE 1
+#define SND_SOC_COMP_ORDER_LAST 2
+
+/*
+ * Bias levels
+ *
+ * @ON: Bias is fully on for audio playback and capture operations.
+ * @PREPARE: Prepare for audio operations. Called before DAPM switching for
+ * stream start and stop operations.
+ * @STANDBY: Low power standby state when no playback/capture operations are
+ * in progress. NOTE: The transition time between STANDBY and ON
+ * should be as fast as possible and no longer than 10ms.
+ * @OFF: Power Off. No restrictions on transition times.
+ */
+enum snd_soc_bias_level {
+ SND_SOC_BIAS_OFF = 0,
+ SND_SOC_BIAS_STANDBY = 1,
+ SND_SOC_BIAS_PREPARE = 2,
+ SND_SOC_BIAS_ON = 3,
+};
+
+struct device_node;
+struct snd_jack;
+struct snd_soc_card;
+struct snd_soc_pcm_stream;
+struct snd_soc_ops;
+struct snd_soc_pcm_runtime;
+struct snd_soc_dai;
+struct snd_soc_dai_driver;
+struct snd_soc_platform;
+struct snd_soc_dai_link;
+struct snd_soc_platform_driver;
+struct snd_soc_codec;
+struct snd_soc_codec_driver;
+struct snd_soc_component;
+struct snd_soc_component_driver;
+struct soc_enum;
+struct snd_soc_jack;
+struct snd_soc_jack_zone;
+struct snd_soc_jack_pin;
+#include <sound/soc-dapm.h>
+#include <sound/soc-dpcm.h>
+
+struct snd_soc_jack_gpio;
+
+typedef int (*hw_write_t)(void *,const char* ,int);
+
+enum snd_soc_pcm_subclass {
+ SND_SOC_PCM_CLASS_PCM = 0,
+ SND_SOC_PCM_CLASS_BE = 1,
+};
+
+enum snd_soc_card_subclass {
+ SND_SOC_CARD_CLASS_INIT = 0,
+ SND_SOC_CARD_CLASS_RUNTIME = 1,
+};
+
+int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ int source, unsigned int freq, int dir);
+int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out);
+
+int snd_soc_register_card(struct snd_soc_card *card);
+int snd_soc_unregister_card(struct snd_soc_card *card);
+int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card);
+#ifdef CONFIG_PM_SLEEP
+int snd_soc_suspend(struct device *dev);
+int snd_soc_resume(struct device *dev);
+#else
+static inline int snd_soc_suspend(struct device *dev)
+{
+ return 0;
+}
+
+static inline int snd_soc_resume(struct device *dev)
+{
+ return 0;
+}
+#endif
+int snd_soc_poweroff(struct device *dev);
+int snd_soc_register_platform(struct device *dev,
+ const struct snd_soc_platform_driver *platform_drv);
+int devm_snd_soc_register_platform(struct device *dev,
+ const struct snd_soc_platform_driver *platform_drv);
+void snd_soc_unregister_platform(struct device *dev);
+int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform,
+ const struct snd_soc_platform_driver *platform_drv);
+void snd_soc_remove_platform(struct snd_soc_platform *platform);
+struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev);
+int snd_soc_register_codec(struct device *dev,
+ const struct snd_soc_codec_driver *codec_drv,
+ struct snd_soc_dai_driver *dai_drv, int num_dai);
+void snd_soc_unregister_codec(struct device *dev);
+int snd_soc_register_component(struct device *dev,
+ const struct snd_soc_component_driver *cmpnt_drv,
+ struct snd_soc_dai_driver *dai_drv, int num_dai);
+int devm_snd_soc_register_component(struct device *dev,
+ const struct snd_soc_component_driver *cmpnt_drv,
+ struct snd_soc_dai_driver *dai_drv, int num_dai);
+void snd_soc_unregister_component(struct device *dev);
+int snd_soc_cache_init(struct snd_soc_codec *codec);
+int snd_soc_cache_exit(struct snd_soc_codec *codec);
+
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+ unsigned int reg);
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+ unsigned int reg, unsigned int val);
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num);
+
+struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
+ const char *dai_link, int stream);
+struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
+ const char *dai_link);
+
+bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd);
+void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream);
+void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
+
+int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
+ unsigned int dai_fmt);
+
+/* Utility functions to get clock rates from various things */
+int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
+int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
+int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots);
+int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms);
+
+/* set runtime hw params */
+int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
+ const struct snd_pcm_hardware *hw);
+
+int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_platform *platform);
+
+int soc_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai);
+
+/* Jack reporting */
+int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type,
+ struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins,
+ unsigned int num_pins);
+
+void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask);
+int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_pin *pins);
+void snd_soc_jack_notifier_register(struct snd_soc_jack *jack,
+ struct notifier_block *nb);
+void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack,
+ struct notifier_block *nb);
+int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_zone *zones);
+int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage);
+#ifdef CONFIG_GPIOLIB
+int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_gpio *gpios);
+int snd_soc_jack_add_gpiods(struct device *gpiod_dev,
+ struct snd_soc_jack *jack,
+ int count, struct snd_soc_jack_gpio *gpios);
+void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_gpio *gpios);
+#else
+static inline int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_gpio *gpios)
+{
+ return 0;
+}
+
+static inline int snd_soc_jack_add_gpiods(struct device *gpiod_dev,
+ struct snd_soc_jack *jack,
+ int count,
+ struct snd_soc_jack_gpio *gpios)
+{
+ return 0;
+}
+
+static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_gpio *gpios)
+{
+}
+#endif
+
+/* codec register bit access */
+int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int mask, unsigned int value);
+int snd_soc_update_bits_locked(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int mask,
+ unsigned int value);
+int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int mask, unsigned int value);
+
+#ifdef CONFIG_SND_SOC_AC97_BUS
+struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec);
+struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec);
+void snd_soc_free_ac97_codec(struct snd_ac97 *ac97);
+
+int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops);
+int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops,
+ struct platform_device *pdev);
+
+extern struct snd_ac97_bus_ops *soc_ac97_ops;
+#else
+static inline int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops,
+ struct platform_device *pdev)
+{
+ return 0;
+}
+
+static inline int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops)
+{
+ return 0;
+}
+#endif
+
+/*
+ *Controls
+ */
+struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
+ void *data, const char *long_name,
+ const char *prefix);
+struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
+ const char *name);
+int snd_soc_add_component_controls(struct snd_soc_component *component,
+ const struct snd_kcontrol_new *controls, unsigned int num_controls);
+int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
+ const struct snd_kcontrol_new *controls, unsigned int num_controls);
+int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
+ const struct snd_kcontrol_new *controls, unsigned int num_controls);
+int snd_soc_add_card_controls(struct snd_soc_card *soc_card,
+ const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
+ const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+#define snd_soc_info_bool_ext snd_ctl_boolean_mono_info
+int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+#define snd_soc_get_volsw_2r snd_soc_get_volsw
+#define snd_soc_put_volsw_2r snd_soc_put_volsw
+int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_limit_volume(struct snd_soc_codec *codec,
+ const char *name, int max);
+int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *ucontrol);
+int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv);
+int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_get_strobe(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_strobe(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
+/**
+ * struct snd_soc_jack_pin - Describes a pin to update based on jack detection
+ *
+ * @pin: name of the pin to update
+ * @mask: bits to check for in reported jack status
+ * @invert: if non-zero then pin is enabled when status is not reported
+ */
+struct snd_soc_jack_pin {
+ struct list_head list;
+ const char *pin;
+ int mask;
+ bool invert;
+};
+
+/**
+ * struct snd_soc_jack_zone - Describes voltage zones of jack detection
+ *
+ * @min_mv: start voltage in mv
+ * @max_mv: end voltage in mv
+ * @jack_type: type of jack that is expected for this voltage
+ * @debounce_time: debounce_time for jack, codec driver should wait for this
+ * duration before reading the adc for voltages
+ * @:list: list container
+ */
+struct snd_soc_jack_zone {
+ unsigned int min_mv;
+ unsigned int max_mv;
+ unsigned int jack_type;
+ unsigned int debounce_time;
+ struct list_head list;
+};
+
+/**
+ * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection
+ *
+ * @gpio: legacy gpio number
+ * @idx: gpio descriptor index within the function of the GPIO
+ * consumer device
+ * @gpiod_dev GPIO consumer device
+ * @name: gpio name. Also as connection ID for the GPIO consumer
+ * device function name lookup
+ * @report: value to report when jack detected
+ * @invert: report presence in low state
+ * @debouce_time: debouce time in ms
+ * @wake: enable as wake source
+ * @jack_status_check: callback function which overrides the detection
+ * to provide more complex checks (eg, reading an
+ * ADC).
+ */
+struct snd_soc_jack_gpio {
+ unsigned int gpio;
+ unsigned int idx;
+ struct device *gpiod_dev;
+ const char *name;
+ int report;
+ int invert;
+ int debounce_time;
+ bool wake;
+
+ struct snd_soc_jack *jack;
+ struct delayed_work work;
+ struct gpio_desc *desc;
+
+ void *data;
+ int (*jack_status_check)(void *data);
+};
+
+struct snd_soc_jack {
+ struct mutex mutex;
+ struct snd_jack *jack;
+ struct snd_soc_card *card;
+ struct list_head pins;
+ int status;
+ struct blocking_notifier_head notifier;
+ struct list_head jack_zones;
+};
+
+/* SoC PCM stream information */
+struct snd_soc_pcm_stream {
+ const char *stream_name;
+ u64 formats; /* SNDRV_PCM_FMTBIT_* */
+ unsigned int rates; /* SNDRV_PCM_RATE_* */
+ unsigned int rate_min; /* min rate */
+ unsigned int rate_max; /* max rate */
+ unsigned int channels_min; /* min channels */
+ unsigned int channels_max; /* max channels */
+ unsigned int sig_bits; /* number of bits of content */
+};
+
+/* SoC audio ops */
+struct snd_soc_ops {
+ int (*startup)(struct snd_pcm_substream *);
+ void (*shutdown)(struct snd_pcm_substream *);
+ int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+ int (*hw_free)(struct snd_pcm_substream *);
+ int (*prepare)(struct snd_pcm_substream *);
+ int (*trigger)(struct snd_pcm_substream *, int);
+};
+
+struct snd_soc_compr_ops {
+ int (*startup)(struct snd_compr_stream *);
+ void (*shutdown)(struct snd_compr_stream *);
+ int (*set_params)(struct snd_compr_stream *);
+ int (*trigger)(struct snd_compr_stream *);
+};
+
+/* component interface */
+struct snd_soc_component_driver {
+ const char *name;
+
+ /* Default control and setup, added after probe() is run */
+ const struct snd_kcontrol_new *controls;
+ unsigned int num_controls;
+ const struct snd_soc_dapm_widget *dapm_widgets;
+ unsigned int num_dapm_widgets;
+ const struct snd_soc_dapm_route *dapm_routes;
+ unsigned int num_dapm_routes;
+
+ int (*probe)(struct snd_soc_component *);
+ void (*remove)(struct snd_soc_component *);
+
+ /* DT */
+ int (*of_xlate_dai_name)(struct snd_soc_component *component,
+ struct of_phandle_args *args,
+ const char **dai_name);
+ void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type,
+ int subseq);
+ int (*stream_event)(struct snd_soc_component *, int event);
+
+ /* probe ordering - for components with runtime dependencies */
+ int probe_order;
+ int remove_order;
+};
+
+struct snd_soc_component {
+ const char *name;
+ int id;
+ const char *name_prefix;
+ struct device *dev;
+ struct snd_soc_card *card;
+
+ unsigned int active;
+
+ unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */
+ unsigned int registered_as_component:1;
+ unsigned int probed:1;
+
+ struct list_head list;
+
+ struct snd_soc_dai_driver *dai_drv;
+ int num_dai;
+
+ const struct snd_soc_component_driver *driver;
+
+ struct list_head dai_list;
+
+ int (*read)(struct snd_soc_component *, unsigned int, unsigned int *);
+ int (*write)(struct snd_soc_component *, unsigned int, unsigned int);
+
+ struct regmap *regmap;
+ int val_bytes;
+
+ struct mutex io_mutex;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_root;
+#endif
+
+ /*
+ * DO NOT use any of the fields below in drivers, they are temporary and
+ * are going to be removed again soon. If you use them in driver code the
+ * driver will be marked as BROKEN when these fields are removed.
+ */
+
+ /* Don't use these, use snd_soc_component_get_dapm() */
+ struct snd_soc_dapm_context dapm;
+ struct snd_soc_dapm_context *dapm_ptr;
+
+ const struct snd_kcontrol_new *controls;
+ unsigned int num_controls;
+ const struct snd_soc_dapm_widget *dapm_widgets;
+ unsigned int num_dapm_widgets;
+ const struct snd_soc_dapm_route *dapm_routes;
+ unsigned int num_dapm_routes;
+ struct snd_soc_codec *codec;
+
+ int (*probe)(struct snd_soc_component *);
+ void (*remove)(struct snd_soc_component *);
+
+#ifdef CONFIG_DEBUG_FS
+ void (*init_debugfs)(struct snd_soc_component *component);
+ const char *debugfs_prefix;
+#endif
+};
+
+/* SoC Audio Codec device */
+struct snd_soc_codec {
+ struct device *dev;
+ const struct snd_soc_codec_driver *driver;
+
+ struct list_head list;
+ struct list_head card_list;
+
+ /* runtime */
+ unsigned int cache_bypass:1; /* Suppress access to the cache */
+ unsigned int suspended:1; /* Codec is in suspend PM state */
+ unsigned int cache_init:1; /* codec cache has been initialized */
+
+ /* codec IO */
+ void *control_data; /* codec control (i2c/3wire) data */
+ hw_write_t hw_write;
+ void *reg_cache;
+
+ /* component */
+ struct snd_soc_component component;
+
+ /* dapm */
+ struct snd_soc_dapm_context dapm;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_reg;
+#endif
+};
+
+/* codec driver */
+struct snd_soc_codec_driver {
+
+ /* driver ops */
+ int (*probe)(struct snd_soc_codec *);
+ int (*remove)(struct snd_soc_codec *);
+ int (*suspend)(struct snd_soc_codec *);
+ int (*resume)(struct snd_soc_codec *);
+ struct snd_soc_component_driver component_driver;
+
+ /* Default control and setup, added after probe() is run */
+ const struct snd_kcontrol_new *controls;
+ int num_controls;
+ const struct snd_soc_dapm_widget *dapm_widgets;
+ int num_dapm_widgets;
+ const struct snd_soc_dapm_route *dapm_routes;
+ int num_dapm_routes;
+
+ /* codec wide operations */
+ int (*set_sysclk)(struct snd_soc_codec *codec,
+ int clk_id, int source, unsigned int freq, int dir);
+ int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out);
+
+ /* codec IO */
+ struct regmap *(*get_regmap)(struct device *);
+ unsigned int (*read)(struct snd_soc_codec *, unsigned int);
+ int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
+ unsigned int reg_cache_size;
+ short reg_cache_step;
+ short reg_word_size;
+ const void *reg_cache_default;
+
+ /* codec bias level */
+ int (*set_bias_level)(struct snd_soc_codec *,
+ enum snd_soc_bias_level level);
+ bool idle_bias_off;
+ bool suspend_bias_off;
+
+ void (*seq_notifier)(struct snd_soc_dapm_context *,
+ enum snd_soc_dapm_type, int);
+
+ bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */
+};
+
+/* SoC platform interface */
+struct snd_soc_platform_driver {
+
+ int (*probe)(struct snd_soc_platform *);
+ int (*remove)(struct snd_soc_platform *);
+ struct snd_soc_component_driver component_driver;
+
+ /* pcm creation and destruction */
+ int (*pcm_new)(struct snd_soc_pcm_runtime *);
+ void (*pcm_free)(struct snd_pcm *);
+
+ /*
+ * For platform caused delay reporting.
+ * Optional.
+ */
+ snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+
+ /* platform stream pcm ops */
+ const struct snd_pcm_ops *ops;
+
+ /* platform stream compress ops */
+ const struct snd_compr_ops *compr_ops;
+
+ int (*bespoke_trigger)(struct snd_pcm_substream *, int);
+};
+
+struct snd_soc_dai_link_component {
+ const char *name;
+ struct device_node *of_node;
+ const char *dai_name;
+};
+
+struct snd_soc_platform {
+ struct device *dev;
+ const struct snd_soc_platform_driver *driver;
+
+ struct list_head list;
+
+ struct snd_soc_component component;
+};
+
+struct snd_soc_dai_link {
+ /* config - must be set by machine driver */
+ const char *name; /* Codec name */
+ const char *stream_name; /* Stream name */
+ /*
+ * You MAY specify the link's CPU-side device, either by device name,
+ * or by DT/OF node, but not both. If this information is omitted,
+ * the CPU-side DAI is matched using .cpu_dai_name only, which hence
+ * must be globally unique. These fields are currently typically used
+ * only for codec to codec links, or systems using device tree.
+ */
+ const char *cpu_name;
+ struct device_node *cpu_of_node;
+ /*
+ * You MAY specify the DAI name of the CPU DAI. If this information is
+ * omitted, the CPU-side DAI is matched using .cpu_name/.cpu_of_node
+ * only, which only works well when that device exposes a single DAI.
+ */
+ const char *cpu_dai_name;
+ /*
+ * You MUST specify the link's codec, either by device name, or by
+ * DT/OF node, but not both.
+ */
+ const char *codec_name;
+ struct device_node *codec_of_node;
+ /* You MUST specify the DAI name within the codec */
+ const char *codec_dai_name;
+
+ struct snd_soc_dai_link_component *codecs;
+ unsigned int num_codecs;
+
+ /*
+ * You MAY specify the link's platform/PCM/DMA driver, either by
+ * device name, or by DT/OF node, but not both. Some forms of link
+ * do not need a platform.
+ */
+ const char *platform_name;
+ struct device_node *platform_of_node;
+ int be_id; /* optional ID for machine driver BE identification */
+
+ const struct snd_soc_pcm_stream *params;
+ unsigned int num_params;
+
+ unsigned int dai_fmt; /* format to set on init */
+
+ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
+
+ /* Keep DAI active over suspend */
+ unsigned int ignore_suspend:1;
+
+ /* Symmetry requirements */
+ unsigned int symmetric_rates:1;
+ unsigned int symmetric_channels:1;
+ unsigned int symmetric_samplebits:1;
+
+ /* Mark this pcm with non atomic ops */
+ bool nonatomic;
+
+ /* Do not create a PCM for this DAI link (Backend link) */
+ unsigned int no_pcm:1;
+
+ /* This DAI link can route to other DAI links at runtime (Frontend)*/
+ unsigned int dynamic:1;
+
+ /* DPCM capture and Playback support */
+ unsigned int dpcm_capture:1;
+ unsigned int dpcm_playback:1;
+
+ /* pmdown_time is ignored at stop */
+ unsigned int ignore_pmdown_time:1;
+
+ /* codec/machine specific init - e.g. add machine controls */
+ int (*init)(struct snd_soc_pcm_runtime *rtd);
+
+ /* optional hw_params re-writing for BE and FE sync */
+ int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params);
+
+ /* machine stream operations */
+ const struct snd_soc_ops *ops;
+ const struct snd_soc_compr_ops *compr_ops;
+
+ /* For unidirectional dai links */
+ bool playback_only;
+ bool capture_only;
+};
+
+struct snd_soc_codec_conf {
+ /*
+ * specify device either by device name, or by
+ * DT/OF node, but not both.
+ */
+ const char *dev_name;
+ struct device_node *of_node;
+
+ /*
+ * optional map of kcontrol, widget and path name prefixes that are
+ * associated per device
+ */
+ const char *name_prefix;
+};
+
+struct snd_soc_aux_dev {
+ const char *name; /* Codec name */
+
+ /*
+ * specify multi-codec either by device name, or by
+ * DT/OF node, but not both.
+ */
+ const char *codec_name;
+ struct device_node *codec_of_node;
+
+ /* codec/machine specific init - e.g. add machine controls */
+ int (*init)(struct snd_soc_component *component);
+};
+
+/* SoC card */
+struct snd_soc_card {
+ const char *name;
+ const char *long_name;
+ const char *driver_name;
+ struct device *dev;
+ struct snd_card *snd_card;
+ struct module *owner;
+
+ struct mutex mutex;
+ struct mutex dapm_mutex;
+
+ bool instantiated;
+
+ int (*probe)(struct snd_soc_card *card);
+ int (*late_probe)(struct snd_soc_card *card);
+ int (*remove)(struct snd_soc_card *card);
+
+ /* the pre and post PM functions are used to do any PM work before and
+ * after the codec and DAI's do any PM work. */
+ int (*suspend_pre)(struct snd_soc_card *card);
+ int (*suspend_post)(struct snd_soc_card *card);
+ int (*resume_pre)(struct snd_soc_card *card);
+ int (*resume_post)(struct snd_soc_card *card);
+
+ /* callbacks */
+ int (*set_bias_level)(struct snd_soc_card *,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level);
+ int (*set_bias_level_post)(struct snd_soc_card *,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level);
+
+ long pmdown_time;
+
+ /* CPU <--> Codec DAI links */
+ struct snd_soc_dai_link *dai_link;
+ int num_links;
+ struct snd_soc_pcm_runtime *rtd;
+ int num_rtd;
+
+ /* optional codec specific configuration */
+ struct snd_soc_codec_conf *codec_conf;
+ int num_configs;
+
+ /*
+ * optional auxiliary devices such as amplifiers or codecs with DAI
+ * link unused
+ */
+ struct snd_soc_aux_dev *aux_dev;
+ int num_aux_devs;
+ struct snd_soc_pcm_runtime *rtd_aux;
+ int num_aux_rtd;
+
+ const struct snd_kcontrol_new *controls;
+ int num_controls;
+
+ /*
+ * Card-specific routes and widgets.
+ * Note: of_dapm_xxx for Device Tree; Otherwise for driver build-in.
+ */
+ const struct snd_soc_dapm_widget *dapm_widgets;
+ int num_dapm_widgets;
+ const struct snd_soc_dapm_route *dapm_routes;
+ int num_dapm_routes;
+ const struct snd_soc_dapm_widget *of_dapm_widgets;
+ int num_of_dapm_widgets;
+ const struct snd_soc_dapm_route *of_dapm_routes;
+ int num_of_dapm_routes;
+ bool fully_routed;
+
+ struct work_struct deferred_resume_work;
+
+ /* lists of probed devices belonging to this card */
+ struct list_head codec_dev_list;
+
+ struct list_head widgets;
+ struct list_head paths;
+ struct list_head dapm_list;
+ struct list_head dapm_dirty;
+
+ /* Generic DAPM context for the card */
+ struct snd_soc_dapm_context dapm;
+ struct snd_soc_dapm_stats dapm_stats;
+ struct snd_soc_dapm_update *update;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_card_root;
+ struct dentry *debugfs_pop_time;
+#endif
+ u32 pop_time;
+
+ void *drvdata;
+};
+
+/* SoC machine DAI configuration, glues a codec and cpu DAI together */
+struct snd_soc_pcm_runtime {
+ struct device *dev;
+ struct snd_soc_card *card;
+ struct snd_soc_dai_link *dai_link;
+ struct mutex pcm_mutex;
+ enum snd_soc_pcm_subclass pcm_subclass;
+ struct snd_pcm_ops ops;
+
+ unsigned int dev_registered:1;
+
+ /* Dynamic PCM BE runtime data */
+ struct snd_soc_dpcm_runtime dpcm[2];
+ int fe_compr;
+
+ long pmdown_time;
+ unsigned char pop_wait:1;
+
+ /* runtime devices */
+ struct snd_pcm *pcm;
+ struct snd_compr *compr;
+ struct snd_soc_codec *codec;
+ struct snd_soc_platform *platform;
+ struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *cpu_dai;
+ struct snd_soc_component *component; /* Only valid for AUX dev rtds */
+
+ struct snd_soc_dai **codec_dais;
+ unsigned int num_codecs;
+
+ struct delayed_work delayed_work;
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_dpcm_root;
+ struct dentry *debugfs_dpcm_state;
+#endif
+};
+
+/* mixer control */
+struct soc_mixer_control {
+ int min, max, platform_max;
+ int reg, rreg;
+ unsigned int shift, rshift;
+ unsigned int sign_bit;
+ unsigned int invert:1;
+ unsigned int autodisable:1;
+};
+
+struct soc_bytes {
+ int base;
+ int num_regs;
+ u32 mask;
+};
+
+struct soc_bytes_ext {
+ int max;
+ /* used for TLV byte control */
+ int (*get)(unsigned int __user *bytes, unsigned int size);
+ int (*put)(const unsigned int __user *bytes, unsigned int size);
+};
+
+/* multi register control */
+struct soc_mreg_control {
+ long min, max;
+ unsigned int regbase, regcount, nbits, invert;
+};
+
+/* enumerated kcontrol */
+struct soc_enum {
+ int reg;
+ unsigned char shift_l;
+ unsigned char shift_r;
+ unsigned int items;
+ unsigned int mask;
+ const char * const *texts;
+ const unsigned int *values;
+};
+
+/**
+ * snd_soc_component_to_codec() - Casts a component to the CODEC it is embedded in
+ * @component: The component to cast to a CODEC
+ *
+ * This function must only be used on components that are known to be CODECs.
+ * Otherwise the behavior is undefined.
+ */
+static inline struct snd_soc_codec *snd_soc_component_to_codec(
+ struct snd_soc_component *component)
+{
+ return container_of(component, struct snd_soc_codec, component);
+}
+
+/**
+ * snd_soc_component_to_platform() - Casts a component to the platform it is embedded in
+ * @component: The component to cast to a platform
+ *
+ * This function must only be used on components that are known to be platforms.
+ * Otherwise the behavior is undefined.
+ */
+static inline struct snd_soc_platform *snd_soc_component_to_platform(
+ struct snd_soc_component *component)
+{
+ return container_of(component, struct snd_soc_platform, component);
+}
+
+/**
+ * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is
+ * embedded in
+ * @dapm: The DAPM context to cast to the component
+ *
+ * This function must only be used on DAPM contexts that are known to be part of
+ * a component (e.g. in a component driver). Otherwise the behavior is
+ * undefined.
+ */
+static inline struct snd_soc_component *snd_soc_dapm_to_component(
+ struct snd_soc_dapm_context *dapm)
+{
+ return container_of(dapm, struct snd_soc_component, dapm);
+}
+
+/**
+ * snd_soc_dapm_to_codec() - Casts a DAPM context to the CODEC it is embedded in
+ * @dapm: The DAPM context to cast to the CODEC
+ *
+ * This function must only be used on DAPM contexts that are known to be part of
+ * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined.
+ */
+static inline struct snd_soc_codec *snd_soc_dapm_to_codec(
+ struct snd_soc_dapm_context *dapm)
+{
+ return container_of(dapm, struct snd_soc_codec, dapm);
+}
+
+/**
+ * snd_soc_dapm_to_platform() - Casts a DAPM context to the platform it is
+ * embedded in
+ * @dapm: The DAPM context to cast to the platform.
+ *
+ * This function must only be used on DAPM contexts that are known to be part of
+ * a platform (e.g. in a platform driver). Otherwise the behavior is undefined.
+ */
+static inline struct snd_soc_platform *snd_soc_dapm_to_platform(
+ struct snd_soc_dapm_context *dapm)
+{
+ return snd_soc_component_to_platform(snd_soc_dapm_to_component(dapm));
+}
+
+/**
+ * snd_soc_component_get_dapm() - Returns the DAPM context associated with a
+ * component
+ * @component: The component for which to get the DAPM context
+ */
+static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm(
+ struct snd_soc_component *component)
+{
+ return component->dapm_ptr;
+}
+
+/**
+ * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol
+ * @kcontrol: The kcontrol
+ *
+ * This function must only be used on DAPM contexts that are known to be part of
+ * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined.
+ */
+static inline struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(
+ struct snd_kcontrol *kcontrol)
+{
+ return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol));
+}
+
+/* codec IO */
+unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg);
+int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val);
+
+/**
+ * snd_soc_cache_sync() - Sync the register cache with the hardware
+ * @codec: CODEC to sync
+ *
+ * Note: This function will call regcache_sync()
+ */
+static inline int snd_soc_cache_sync(struct snd_soc_codec *codec)
+{
+ return regcache_sync(codec->component.regmap);
+}
+
+/* component IO */
+int snd_soc_component_read(struct snd_soc_component *component,
+ unsigned int reg, unsigned int *val);
+int snd_soc_component_write(struct snd_soc_component *component,
+ unsigned int reg, unsigned int val);
+int snd_soc_component_update_bits(struct snd_soc_component *component,
+ unsigned int reg, unsigned int mask, unsigned int val);
+int snd_soc_component_update_bits_async(struct snd_soc_component *component,
+ unsigned int reg, unsigned int mask, unsigned int val);
+void snd_soc_component_async_complete(struct snd_soc_component *component);
+int snd_soc_component_test_bits(struct snd_soc_component *component,
+ unsigned int reg, unsigned int mask, unsigned int value);
+
+#ifdef CONFIG_REGMAP
+
+void snd_soc_component_init_regmap(struct snd_soc_component *component,
+ struct regmap *regmap);
+void snd_soc_component_exit_regmap(struct snd_soc_component *component);
+
+/**
+ * snd_soc_codec_init_regmap() - Initialize regmap instance for the CODEC
+ * @codec: The CODEC for which to initialize the regmap instance
+ * @regmap: The regmap instance that should be used by the CODEC
+ *
+ * This function allows deferred assignment of the regmap instance that is
+ * associated with the CODEC. Only use this if the regmap instance is not yet
+ * ready when the CODEC is registered. The function must also be called before
+ * the first IO attempt of the CODEC.
+ */
+static inline void snd_soc_codec_init_regmap(struct snd_soc_codec *codec,
+ struct regmap *regmap)
+{
+ snd_soc_component_init_regmap(&codec->component, regmap);
+}
+
+/**
+ * snd_soc_codec_exit_regmap() - De-initialize regmap instance for the CODEC
+ * @codec: The CODEC for which to de-initialize the regmap instance
+ *
+ * Calls regmap_exit() on the regmap instance associated to the CODEC and
+ * removes the regmap instance from the CODEC.
+ *
+ * This function should only be used if snd_soc_codec_init_regmap() was used to
+ * initialize the regmap instance.
+ */
+static inline void snd_soc_codec_exit_regmap(struct snd_soc_codec *codec)
+{
+ snd_soc_component_exit_regmap(&codec->component);
+}
+
+#endif
+
+/* device driver data */
+
+static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card,
+ void *data)
+{
+ card->drvdata = data;
+}
+
+static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card)
+{
+ return card->drvdata;
+}
+
+static inline void snd_soc_component_set_drvdata(struct snd_soc_component *c,
+ void *data)
+{
+ dev_set_drvdata(c->dev, data);
+}
+
+static inline void *snd_soc_component_get_drvdata(struct snd_soc_component *c)
+{
+ return dev_get_drvdata(c->dev);
+}
+
+static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec,
+ void *data)
+{
+ snd_soc_component_set_drvdata(&codec->component, data);
+}
+
+static inline void *snd_soc_codec_get_drvdata(struct snd_soc_codec *codec)
+{
+ return snd_soc_component_get_drvdata(&codec->component);
+}
+
+static inline void snd_soc_platform_set_drvdata(struct snd_soc_platform *platform,
+ void *data)
+{
+ snd_soc_component_set_drvdata(&platform->component, data);
+}
+
+static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platform)
+{
+ return snd_soc_component_get_drvdata(&platform->component);
+}
+
+static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd,
+ void *data)
+{
+ dev_set_drvdata(rtd->dev, data);
+}
+
+static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd)
+{
+ return dev_get_drvdata(rtd->dev);
+}
+
+static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
+{
+ INIT_LIST_HEAD(&card->codec_dev_list);
+ INIT_LIST_HEAD(&card->widgets);
+ INIT_LIST_HEAD(&card->paths);
+ INIT_LIST_HEAD(&card->dapm_list);
+}
+
+static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
+{
+ if (mc->reg == mc->rreg && mc->shift == mc->rshift)
+ return 0;
+ /*
+ * mc->reg == mc->rreg && mc->shift != mc->rshift, or
+ * mc->reg != mc->rreg means that the control is
+ * stereo (bits in one register or in two registers)
+ */
+ return 1;
+}
+
+static inline unsigned int snd_soc_enum_val_to_item(struct soc_enum *e,
+ unsigned int val)
+{
+ unsigned int i;
+
+ if (!e->values)
+ return val;
+
+ for (i = 0; i < e->items; i++)
+ if (val == e->values[i])
+ return i;
+
+ return 0;
+}
+
+static inline unsigned int snd_soc_enum_item_to_val(struct soc_enum *e,
+ unsigned int item)
+{
+ if (!e->values)
+ return item;
+
+ return e->values[item];
+}
+
+static inline bool snd_soc_component_is_active(
+ struct snd_soc_component *component)
+{
+ return component->active != 0;
+}
+
+static inline bool snd_soc_codec_is_active(struct snd_soc_codec *codec)
+{
+ return snd_soc_component_is_active(&codec->component);
+}
+
+/**
+ * snd_soc_kcontrol_component() - Returns the component that registered the
+ * control
+ * @kcontrol: The control for which to get the component
+ *
+ * Note: This function will work correctly if the control has been registered
+ * for a component. Either with snd_soc_add_codec_controls() or
+ * snd_soc_add_platform_controls() or via table based setup for either a
+ * CODEC, a platform or component driver. Otherwise the behavior is undefined.
+ */
+static inline struct snd_soc_component *snd_soc_kcontrol_component(
+ struct snd_kcontrol *kcontrol)
+{
+ return snd_kcontrol_chip(kcontrol);
+}
+
+/**
+ * snd_soc_kcontrol_codec() - Returns the CODEC that registered the control
+ * @kcontrol: The control for which to get the CODEC
+ *
+ * Note: This function will only work correctly if the control has been
+ * registered with snd_soc_add_codec_controls() or via table based setup of
+ * snd_soc_codec_driver. Otherwise the behavior is undefined.
+ */
+static inline struct snd_soc_codec *snd_soc_kcontrol_codec(
+ struct snd_kcontrol *kcontrol)
+{
+ return snd_soc_component_to_codec(snd_soc_kcontrol_component(kcontrol));
+}
+
+/**
+ * snd_soc_kcontrol_platform() - Returns the platform that registered the control
+ * @kcontrol: The control for which to get the platform
+ *
+ * Note: This function will only work correctly if the control has been
+ * registered with snd_soc_add_platform_controls() or via table based setup of
+ * a snd_soc_platform_driver. Otherwise the behavior is undefined.
+ */
+static inline struct snd_soc_platform *snd_soc_kcontrol_platform(
+ struct snd_kcontrol *kcontrol)
+{
+ return snd_soc_component_to_platform(snd_soc_kcontrol_component(kcontrol));
+}
+
+int snd_soc_util_init(void);
+void snd_soc_util_exit(void);
+
+int snd_soc_of_parse_card_name(struct snd_soc_card *card,
+ const char *propname);
+int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
+ const char *propname);
+int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *slots,
+ unsigned int *slot_width);
+int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
+ const char *propname);
+unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
+ const char *prefix,
+ struct device_node **bitclkmaster,
+ struct device_node **framemaster);
+int snd_soc_of_get_dai_name(struct device_node *of_node,
+ const char **dai_name);
+int snd_soc_of_get_dai_link_codecs(struct device *dev,
+ struct device_node *of_node,
+ struct snd_soc_dai_link *dai_link);
+
+#include <sound/soc-dai.h>
+
+#ifdef CONFIG_DEBUG_FS
+extern struct dentry *snd_soc_debugfs_root;
+#endif
+
+extern const struct dev_pm_ops snd_soc_pm_ops;
+
+/* Helper functions */
+static inline void snd_soc_dapm_mutex_lock(struct snd_soc_dapm_context *dapm)
+{
+ mutex_lock(&dapm->card->dapm_mutex);
+}
+
+static inline void snd_soc_dapm_mutex_unlock(struct snd_soc_dapm_context *dapm)
+{
+ mutex_unlock(&dapm->card->dapm_mutex);
+}
+
+#endif
diff --git a/kernel/include/sound/soundfont.h b/kernel/include/sound/soundfont.h
new file mode 100644
index 000000000..7c93efdba
--- /dev/null
+++ b/kernel/include/sound/soundfont.h
@@ -0,0 +1,129 @@
+#ifndef __SOUND_SOUNDFONT_H
+#define __SOUND_SOUNDFONT_H
+
+/*
+ * Soundfont defines and definitions.
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (c) 1999-2000 Takashi iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/sfnt_info.h>
+#include <sound/util_mem.h>
+
+#define SF_MAX_INSTRUMENTS 128 /* maximum instrument number */
+#define SF_MAX_PRESETS 256 /* drums are mapped from 128 to 256 */
+#define SF_IS_DRUM_BANK(z) ((z) == 128)
+
+struct snd_sf_zone {
+ struct snd_sf_zone *next; /* Link to next */
+ unsigned char bank; /* Midi bank for this zone */
+ unsigned char instr; /* Midi program for this zone */
+ unsigned char mapped; /* True if mapped to something else */
+
+ struct soundfont_voice_info v; /* All the soundfont parameters */
+ int counter;
+ struct snd_sf_sample *sample; /* Link to sample */
+
+ /* The following deals with preset numbers (programs) */
+ struct snd_sf_zone *next_instr; /* Next zone of this instrument */
+ struct snd_sf_zone *next_zone; /* Next zone in play list */
+};
+
+struct snd_sf_sample {
+ struct soundfont_sample_info v;
+ int counter;
+ struct snd_util_memblk *block; /* allocated data block */
+ struct snd_sf_sample *next;
+};
+
+/*
+ * This represents all the information relating to a soundfont.
+ */
+struct snd_soundfont {
+ struct snd_soundfont *next; /* Link to next */
+ /*struct snd_soundfont *prev;*/ /* Link to previous */
+ short id; /* file id */
+ short type; /* font type */
+ unsigned char name[SNDRV_SFNT_PATCH_NAME_LEN]; /* identifier */
+ struct snd_sf_zone *zones; /* Font information */
+ struct snd_sf_sample *samples; /* The sample headers */
+};
+
+/*
+ * Type of the sample access callback
+ */
+struct snd_sf_callback {
+ void *private_data;
+ int (*sample_new)(void *private_data, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr,
+ const void __user *buf, long count);
+ int (*sample_free)(void *private_data, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr);
+ void (*sample_reset)(void *private);
+};
+
+/*
+ * List of soundfonts.
+ */
+struct snd_sf_list {
+ struct snd_soundfont *currsf; /* The currently open soundfont */
+ int open_client; /* client pointer for lock */
+ int mem_used; /* used memory size */
+ struct snd_sf_zone *presets[SF_MAX_PRESETS];
+ struct snd_soundfont *fonts; /* The list of soundfonts */
+ int fonts_size; /* number of fonts allocated */
+ int zone_counter; /* last allocated time for zone */
+ int sample_counter; /* last allocated time for sample */
+ int zone_locked; /* locked time for zone */
+ int sample_locked; /* locked time for sample */
+ struct snd_sf_callback callback; /* callback functions */
+ int presets_locked;
+ struct mutex presets_mutex;
+ spinlock_t lock;
+ struct snd_util_memhdr *memhdr;
+};
+
+/* Prototypes for soundfont.c */
+int snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data,
+ long count, int client);
+int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data,
+ long count, int client);
+int snd_soundfont_close_check(struct snd_sf_list *sflist, int client);
+
+struct snd_sf_list *snd_sf_new(struct snd_sf_callback *callback,
+ struct snd_util_memhdr *hdr);
+void snd_sf_free(struct snd_sf_list *sflist);
+
+int snd_soundfont_remove_samples(struct snd_sf_list *sflist);
+int snd_soundfont_remove_unlocked(struct snd_sf_list *sflist);
+
+int snd_soundfont_search_zone(struct snd_sf_list *sflist, int *notep, int vel,
+ int preset, int bank,
+ int def_preset, int def_bank,
+ struct snd_sf_zone **table, int max_layers);
+
+/* Parameter conversions */
+int snd_sf_calc_parm_hold(int msec);
+int snd_sf_calc_parm_attack(int msec);
+int snd_sf_calc_parm_decay(int msec);
+#define snd_sf_calc_parm_delay(msec) (0x8000 - (msec) * 1000 / 725)
+extern int snd_sf_vol_table[128];
+int snd_sf_linear_to_log(unsigned int amount, int offset, int ratio);
+
+
+#endif /* __SOUND_SOUNDFONT_H */
diff --git a/kernel/include/sound/spear_dma.h b/kernel/include/sound/spear_dma.h
new file mode 100644
index 000000000..e290de4e7
--- /dev/null
+++ b/kernel/include/sound/spear_dma.h
@@ -0,0 +1,34 @@
+/*
+* linux/spear_dma.h
+*
+* Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux@gmail.com)
+*
+* This program is free software; you can redistribute it and/or modify
+* it under the terms of the GNU General Public License as published by
+* the Free Software Foundation; either version 2 of the License, or
+* (at your option) any later version.
+*
+* This program is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+* GNU General Public License for more details.
+*
+* You should have received a copy of the GNU General Public License
+* along with this program; if not, write to the Free Software
+* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*
+*/
+
+#ifndef SPEAR_DMA_H
+#define SPEAR_DMA_H
+
+#include <linux/dmaengine.h>
+
+struct spear_dma_data {
+ void *data;
+ dma_addr_t addr;
+ u32 max_burst;
+ enum dma_slave_buswidth addr_width;
+};
+
+#endif /* SPEAR_DMA_H */
diff --git a/kernel/include/sound/spear_spdif.h b/kernel/include/sound/spear_spdif.h
new file mode 100644
index 000000000..a12f39695
--- /dev/null
+++ b/kernel/include/sound/spear_spdif.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (ST) 2012 Vipin Kumar (vipin.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_SPDIF_H
+#define __SOUND_SPDIF_H
+
+struct spear_spdif_platform_data {
+ /* DMA params */
+ void *dma_params;
+ bool (*filter)(struct dma_chan *chan, void *slave);
+ void (*reset_perip)(void);
+};
+
+#endif /* SOUND_SPDIF_H */
diff --git a/kernel/include/sound/sta32x.h b/kernel/include/sound/sta32x.h
new file mode 100644
index 000000000..a894f7d17
--- /dev/null
+++ b/kernel/include/sound/sta32x.h
@@ -0,0 +1,43 @@
+/*
+ * Platform data for ST STA32x ASoC codec driver.
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+#ifndef __LINUX_SND__STA32X_H
+#define __LINUX_SND__STA32X_H
+
+#define STA32X_OCFG_2CH 0
+#define STA32X_OCFG_2_1CH 1
+#define STA32X_OCFG_1CH 3
+
+#define STA32X_OM_CH1 0
+#define STA32X_OM_CH2 1
+#define STA32X_OM_CH3 2
+
+#define STA32X_THERMAL_ADJUSTMENT_ENABLE 1
+#define STA32X_THERMAL_RECOVERY_ENABLE 2
+
+struct sta32x_platform_data {
+ u8 output_conf;
+ u8 ch1_output_mapping;
+ u8 ch2_output_mapping;
+ u8 ch3_output_mapping;
+ int needs_esd_watchdog;
+ u8 drop_compensation_ns;
+ unsigned int thermal_warning_recovery:1;
+ unsigned int thermal_warning_adjustment:1;
+ unsigned int fault_detect_recovery:1;
+ unsigned int max_power_use_mpcc:1;
+ unsigned int max_power_correction:1;
+ unsigned int am_reduction_mode:1;
+ unsigned int odd_pwm_speed_mode:1;
+ unsigned int invalid_input_detect_mute:1;
+};
+
+#endif /* __LINUX_SND__STA32X_H */
diff --git a/kernel/include/sound/sta350.h b/kernel/include/sound/sta350.h
new file mode 100644
index 000000000..42edceb09
--- /dev/null
+++ b/kernel/include/sound/sta350.h
@@ -0,0 +1,57 @@
+/*
+ * Platform data for ST STA350 ASoC codec driver.
+ *
+ * Copyright: 2014 Raumfeld GmbH
+ * Author: Sven Brandau <info@brandau.biz>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+#ifndef __LINUX_SND__STA350_H
+#define __LINUX_SND__STA350_H
+
+#define STA350_OCFG_2CH 0
+#define STA350_OCFG_2_1CH 1
+#define STA350_OCFG_1CH 3
+
+#define STA350_OM_CH1 0
+#define STA350_OM_CH2 1
+#define STA350_OM_CH3 2
+
+#define STA350_THERMAL_ADJUSTMENT_ENABLE 1
+#define STA350_THERMAL_RECOVERY_ENABLE 2
+#define STA350_FAULT_DETECT_RECOVERY_BYPASS 1
+
+#define STA350_FFX_PM_DROP_COMP 0
+#define STA350_FFX_PM_TAPERED_COMP 1
+#define STA350_FFX_PM_FULL_POWER 2
+#define STA350_FFX_PM_VARIABLE_DROP_COMP 3
+
+
+struct sta350_platform_data {
+ u8 output_conf;
+ u8 ch1_output_mapping;
+ u8 ch2_output_mapping;
+ u8 ch3_output_mapping;
+ u8 ffx_power_output_mode;
+ u8 drop_compensation_ns;
+ u8 powerdown_delay_divider;
+ unsigned int thermal_warning_recovery:1;
+ unsigned int thermal_warning_adjustment:1;
+ unsigned int fault_detect_recovery:1;
+ unsigned int oc_warning_adjustment:1;
+ unsigned int max_power_use_mpcc:1;
+ unsigned int max_power_correction:1;
+ unsigned int am_reduction_mode:1;
+ unsigned int odd_pwm_speed_mode:1;
+ unsigned int distortion_compensation:1;
+ unsigned int invalid_input_detect_mute:1;
+ unsigned int activate_mute_output:1;
+ unsigned int bridge_immediate_off:1;
+ unsigned int noise_shape_dc_cut:1;
+ unsigned int powerdown_master_vol:1;
+};
+
+#endif /* __LINUX_SND__STA350_H */
diff --git a/kernel/include/sound/tas2552-plat.h b/kernel/include/sound/tas2552-plat.h
new file mode 100644
index 000000000..65e7627ba
--- /dev/null
+++ b/kernel/include/sound/tas2552-plat.h
@@ -0,0 +1,25 @@
+/*
+ * TAS2552 driver platform header
+ *
+ * Copyright (C) 2014 Texas Instruments Inc.
+ *
+ * Author: Dan Murphy <dmurphy@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef TAS2552_PLAT_H
+#define TAS2552_PLAT_H
+
+struct tas2552_platform_data {
+ int enable_gpio;
+};
+
+#endif
diff --git a/kernel/include/sound/tas5086.h b/kernel/include/sound/tas5086.h
new file mode 100644
index 000000000..aac481b7d
--- /dev/null
+++ b/kernel/include/sound/tas5086.h
@@ -0,0 +1,7 @@
+#ifndef _SND_SOC_CODEC_TAS5086_H_
+#define _SND_SOC_CODEC_TAS5086_H_
+
+#define TAS5086_CLK_IDX_MCLK 0
+#define TAS5086_CLK_IDX_SCLK 1
+
+#endif /* _SND_SOC_CODEC_TAS5086_H_ */
diff --git a/kernel/include/sound/tea6330t.h b/kernel/include/sound/tea6330t.h
new file mode 100644
index 000000000..e6beec23d
--- /dev/null
+++ b/kernel/include/sound/tea6330t.h
@@ -0,0 +1,31 @@
+#ifndef __SOUND_TEA6330T_H
+#define __SOUND_TEA6330T_H
+
+/*
+ * Routines for control of TEA6330T circuit.
+ * Sound fader control circuit for car radios.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ */
+
+#include <sound/i2c.h> /* generic i2c support */
+
+int snd_tea6330t_detect(struct snd_i2c_bus *bus, int equalizer);
+int snd_tea6330t_update_mixer(struct snd_card *card, struct snd_i2c_bus *bus,
+ int equalizer, int fader);
+
+#endif /* __SOUND_TEA6330T_H */
diff --git a/kernel/include/sound/timer.h b/kernel/include/sound/timer.h
new file mode 100644
index 000000000..7990469a4
--- /dev/null
+++ b/kernel/include/sound/timer.h
@@ -0,0 +1,143 @@
+#ifndef __SOUND_TIMER_H
+#define __SOUND_TIMER_H
+
+/*
+ * Timer abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/asound.h>
+#include <linux/interrupt.h>
+
+#define snd_timer_chip(timer) ((timer)->private_data)
+
+#define SNDRV_TIMER_DEVICES 16
+
+#define SNDRV_TIMER_DEV_FLG_PCM 0x10000000
+
+#define SNDRV_TIMER_HW_AUTO 0x00000001 /* auto trigger is supported */
+#define SNDRV_TIMER_HW_STOP 0x00000002 /* call stop before start */
+#define SNDRV_TIMER_HW_SLAVE 0x00000004 /* only slave timer (variable resolution) */
+#define SNDRV_TIMER_HW_FIRST 0x00000008 /* first tick can be incomplete */
+#define SNDRV_TIMER_HW_TASKLET 0x00000010 /* timer is called from tasklet */
+
+#define SNDRV_TIMER_IFLG_SLAVE 0x00000001
+#define SNDRV_TIMER_IFLG_RUNNING 0x00000002
+#define SNDRV_TIMER_IFLG_START 0x00000004
+#define SNDRV_TIMER_IFLG_AUTO 0x00000008 /* auto restart */
+#define SNDRV_TIMER_IFLG_FAST 0x00000010 /* fast callback (do not use tasklet) */
+#define SNDRV_TIMER_IFLG_CALLBACK 0x00000020 /* timer callback is active */
+#define SNDRV_TIMER_IFLG_EXCLUSIVE 0x00000040 /* exclusive owner - no more instances */
+#define SNDRV_TIMER_IFLG_EARLY_EVENT 0x00000080 /* write early event to the poll queue */
+
+#define SNDRV_TIMER_FLG_CHANGE 0x00000001
+#define SNDRV_TIMER_FLG_RESCHED 0x00000002 /* need reschedule */
+
+struct snd_timer;
+
+struct snd_timer_hardware {
+ /* -- must be filled with low-level driver */
+ unsigned int flags; /* various flags */
+ unsigned long resolution; /* average timer resolution for one tick in nsec */
+ unsigned long resolution_min; /* minimal resolution */
+ unsigned long resolution_max; /* maximal resolution */
+ unsigned long ticks; /* max timer ticks per interrupt */
+ /* -- low-level functions -- */
+ int (*open) (struct snd_timer * timer);
+ int (*close) (struct snd_timer * timer);
+ unsigned long (*c_resolution) (struct snd_timer * timer);
+ int (*start) (struct snd_timer * timer);
+ int (*stop) (struct snd_timer * timer);
+ int (*set_period) (struct snd_timer * timer, unsigned long period_num, unsigned long period_den);
+ int (*precise_resolution) (struct snd_timer * timer, unsigned long *num, unsigned long *den);
+};
+
+struct snd_timer {
+ int tmr_class;
+ struct snd_card *card;
+ struct module *module;
+ int tmr_device;
+ int tmr_subdevice;
+ char id[64];
+ char name[80];
+ unsigned int flags;
+ int running; /* running instances */
+ unsigned long sticks; /* schedule ticks */
+ void *private_data;
+ void (*private_free) (struct snd_timer *timer);
+ struct snd_timer_hardware hw;
+ spinlock_t lock;
+ struct list_head device_list;
+ struct list_head open_list_head;
+ struct list_head active_list_head;
+ struct list_head ack_list_head;
+ struct list_head sack_list_head; /* slow ack list head */
+ struct tasklet_struct task_queue;
+};
+
+struct snd_timer_instance {
+ struct snd_timer *timer;
+ char *owner;
+ unsigned int flags;
+ void *private_data;
+ void (*private_free) (struct snd_timer_instance *ti);
+ void (*callback) (struct snd_timer_instance *timeri,
+ unsigned long ticks, unsigned long resolution);
+ void (*ccallback) (struct snd_timer_instance * timeri,
+ int event,
+ struct timespec * tstamp,
+ unsigned long resolution);
+ void *callback_data;
+ unsigned long ticks; /* auto-load ticks when expired */
+ unsigned long cticks; /* current ticks */
+ unsigned long pticks; /* accumulated ticks for callback */
+ unsigned long resolution; /* current resolution for tasklet */
+ unsigned long lost; /* lost ticks */
+ int slave_class;
+ unsigned int slave_id;
+ struct list_head open_list;
+ struct list_head active_list;
+ struct list_head ack_list;
+ struct list_head slave_list_head;
+ struct list_head slave_active_head;
+ struct snd_timer_instance *master;
+};
+
+/*
+ * Registering
+ */
+
+int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, struct snd_timer **rtimer);
+void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstamp);
+int snd_timer_global_new(char *id, int device, struct snd_timer **rtimer);
+int snd_timer_global_free(struct snd_timer *timer);
+int snd_timer_global_register(struct snd_timer *timer);
+
+int snd_timer_open(struct snd_timer_instance **ti, char *owner, struct snd_timer_id *tid, unsigned int slave_id);
+int snd_timer_close(struct snd_timer_instance *timeri);
+unsigned long snd_timer_resolution(struct snd_timer_instance *timeri);
+int snd_timer_start(struct snd_timer_instance *timeri, unsigned int ticks);
+int snd_timer_stop(struct snd_timer_instance *timeri);
+int snd_timer_continue(struct snd_timer_instance *timeri);
+int snd_timer_pause(struct snd_timer_instance *timeri);
+
+void snd_timer_interrupt(struct snd_timer *timer, unsigned long ticks_left);
+
+#endif /* __SOUND_TIMER_H */
diff --git a/kernel/include/sound/tlv.h b/kernel/include/sound/tlv.h
new file mode 100644
index 000000000..e11e17942
--- /dev/null
+++ b/kernel/include/sound/tlv.h
@@ -0,0 +1,101 @@
+#ifndef __SOUND_TLV_H
+#define __SOUND_TLV_H
+
+/*
+ * Advanced Linux Sound Architecture - ALSA - Driver
+ * Copyright (c) 2006 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * TLV structure is right behind the struct snd_ctl_tlv:
+ * unsigned int type - see SNDRV_CTL_TLVT_*
+ * unsigned int length
+ * .... data aligned to sizeof(unsigned int), use
+ * block_length = (length + (sizeof(unsigned int) - 1)) &
+ * ~(sizeof(unsigned int) - 1)) ....
+ */
+
+#define SNDRV_CTL_TLVT_CONTAINER 0 /* one level down - group of TLVs */
+#define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */
+#define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */
+#define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */
+#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */
+#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */
+
+#define TLV_ITEM(type, ...) \
+ (type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__
+#define TLV_LENGTH(...) \
+ ((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
+
+#define TLV_CONTAINER_ITEM(...) \
+ TLV_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
+#define DECLARE_TLV_CONTAINER(name, ...) \
+ unsigned int name[] = { TLV_CONTAINER_ITEM(__VA_ARGS__) }
+
+#define TLV_DB_SCALE_MASK 0xffff
+#define TLV_DB_SCALE_MUTE 0x10000
+#define TLV_DB_SCALE_ITEM(min, step, mute) \
+ TLV_ITEM(SNDRV_CTL_TLVT_DB_SCALE, \
+ (min), \
+ ((step) & TLV_DB_SCALE_MASK) | \
+ ((mute) ? TLV_DB_SCALE_MUTE : 0))
+#define DECLARE_TLV_DB_SCALE(name, min, step, mute) \
+ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) }
+
+/* dB scale specified with min/max values instead of step */
+#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \
+ TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
+#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
+ TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB))
+#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \
+ unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) }
+#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \
+ unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) }
+
+/* linear volume between min_dB and max_dB (.01dB unit) */
+#define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \
+ TLV_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
+#define DECLARE_TLV_DB_LINEAR(name, min_dB, max_dB) \
+ unsigned int name[] = { TLV_DB_LINEAR_ITEM(min_dB, max_dB) }
+
+/* dB range container:
+ * Items in dB range container must be ordered by their values and by their
+ * dB values. This implies that larger values must correspond with larger
+ * dB values (which is also required for all other mixer controls).
+ */
+/* Each item is: <min> <max> <TLV> */
+#define TLV_DB_RANGE_ITEM(...) \
+ TLV_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__)
+#define DECLARE_TLV_DB_RANGE(name, ...) \
+ unsigned int name[] = { TLV_DB_RANGE_ITEM(__VA_ARGS__) }
+/* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */
+#define TLV_DB_RANGE_HEAD(num) \
+ SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int)
+
+#define TLV_DB_GAIN_MUTE -9999999
+
+/*
+ * channel-mapping TLV items
+ * TLV length must match with num_channels
+ */
+#define SNDRV_CTL_TLVT_CHMAP_FIXED 0x101 /* fixed channel position */
+#define SNDRV_CTL_TLVT_CHMAP_VAR 0x102 /* channels freely swappable */
+#define SNDRV_CTL_TLVT_CHMAP_PAIRED 0x103 /* pair-wise swappable */
+
+#endif /* __SOUND_TLV_H */
diff --git a/kernel/include/sound/tlv320aic32x4.h b/kernel/include/sound/tlv320aic32x4.h
new file mode 100644
index 000000000..24e5d991f
--- /dev/null
+++ b/kernel/include/sound/tlv320aic32x4.h
@@ -0,0 +1,32 @@
+/*
+ * tlv320aic32x4.h -- TLV320AIC32X4 Soc Audio driver platform data
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AIC32X4_PDATA_H
+#define _AIC32X4_PDATA_H
+
+#define AIC32X4_PWR_MICBIAS_2075_LDOIN 0x00000001
+#define AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE 0x00000002
+#define AIC32X4_PWR_AIC32X4_LDO_ENABLE 0x00000004
+#define AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36 0x00000008
+#define AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED 0x00000010
+
+#define AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K 0x00000001
+#define AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K 0x00000002
+
+struct aic32x4_pdata {
+ u32 power_cfg;
+ u32 micpga_routing;
+ bool swapdacs;
+ int rstn_gpio;
+};
+
+#endif
diff --git a/kernel/include/sound/tlv320aic3x.h b/kernel/include/sound/tlv320aic3x.h
new file mode 100644
index 000000000..9407fd003
--- /dev/null
+++ b/kernel/include/sound/tlv320aic3x.h
@@ -0,0 +1,68 @@
+/*
+ * Platform data for Texas Instruments TLV320AIC3x codec
+ *
+ * Author: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#ifndef __TLV320AIC3x_H__
+#define __TLV320AIC3x_H__
+
+/* GPIO API */
+enum {
+ AIC3X_GPIO1_FUNC_DISABLED = 0,
+ AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC = 1,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX = 2,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2 = 3,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4 = 4,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8 = 5,
+ AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ = 6,
+ AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ = 7,
+ AIC3X_GPIO1_FUNC_INPUT = 8,
+ AIC3X_GPIO1_FUNC_OUTPUT = 9,
+ AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK = 10,
+ AIC3X_GPIO1_FUNC_AUDIO_WORDCLK = 11,
+ AIC3X_GPIO1_FUNC_BUTTON_IRQ = 12,
+ AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ = 13,
+ AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 14,
+ AIC3X_GPIO1_FUNC_ALL_IRQ = 16
+};
+
+enum {
+ AIC3X_GPIO2_FUNC_DISABLED = 0,
+ AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ = 2,
+ AIC3X_GPIO2_FUNC_INPUT = 3,
+ AIC3X_GPIO2_FUNC_OUTPUT = 4,
+ AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT = 5,
+ AIC3X_GPIO2_FUNC_AUDIO_BITCLK = 8,
+ AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9,
+ AIC3X_GPIO2_FUNC_ALL_IRQ = 10,
+ AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11,
+ AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12,
+ AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ = 13,
+ AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ = 14,
+ AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15
+};
+
+enum aic3x_micbias_voltage {
+ AIC3X_MICBIAS_OFF = 0,
+ AIC3X_MICBIAS_2_0V = 1,
+ AIC3X_MICBIAS_2_5V = 2,
+ AIC3X_MICBIAS_AVDDV = 3,
+};
+
+struct aic3x_setup_data {
+ unsigned int gpio_func[2];
+};
+
+struct aic3x_pdata {
+ int gpio_reset; /* < 0 if not used */
+ struct aic3x_setup_data *setup;
+
+ /* Selects the micbias voltage */
+ enum aic3x_micbias_voltage micbias_vg;
+};
+
+#endif
diff --git a/kernel/include/sound/tlv320dac33-plat.h b/kernel/include/sound/tlv320dac33-plat.h
new file mode 100644
index 000000000..0b94192a8
--- /dev/null
+++ b/kernel/include/sound/tlv320dac33-plat.h
@@ -0,0 +1,24 @@
+/*
+ * Platform header for Texas Instruments TLV320DAC33 codec driver
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * Copyright: (C) 2009 Nokia Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __TLV320DAC33_PLAT_H
+#define __TLV320DAC33_PLAT_H
+
+struct tlv320dac33_platform_data {
+ int power_gpio;
+ int mode1_latency; /* latency caused by the i2c writes in us */
+ int auto_fifo_config; /* FIFO config based on the period size */
+ int keep_bclk; /* Keep the BCLK running in FIFO modes */
+ u8 burst_bclkdiv;
+};
+
+#endif /* __TLV320DAC33_PLAT_H */
diff --git a/kernel/include/sound/tpa6130a2-plat.h b/kernel/include/sound/tpa6130a2-plat.h
new file mode 100644
index 000000000..4cc109384
--- /dev/null
+++ b/kernel/include/sound/tpa6130a2-plat.h
@@ -0,0 +1,30 @@
+/*
+ * TPA6130A2 driver platform header
+ *
+ * Copyright (C) Nokia Corporation
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef TPA6130A2_PLAT_H
+#define TPA6130A2_PLAT_H
+
+struct tpa6130a2_platform_data {
+ int power_gpio;
+};
+
+#endif
diff --git a/kernel/include/sound/uda134x.h b/kernel/include/sound/uda134x.h
new file mode 100644
index 000000000..509efb050
--- /dev/null
+++ b/kernel/include/sound/uda134x.h
@@ -0,0 +1,27 @@
+/*
+ * uda134x.h -- UDA134x ALSA SoC Codec driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _UDA134X_H
+#define _UDA134X_H
+
+#include <sound/l3.h>
+
+struct uda134x_platform_data {
+ struct l3_pins l3;
+ void (*power) (int);
+ int model;
+#define UDA134X_UDA1340 1
+#define UDA134X_UDA1341 2
+#define UDA134X_UDA1344 3
+#define UDA134X_UDA1345 4
+};
+
+#endif /* _UDA134X_H */
diff --git a/kernel/include/sound/uda1380.h b/kernel/include/sound/uda1380.h
new file mode 100644
index 000000000..381319c70
--- /dev/null
+++ b/kernel/include/sound/uda1380.h
@@ -0,0 +1,22 @@
+/*
+ * UDA1380 ALSA SoC Codec driver
+ *
+ * Copyright 2009 Philipp Zabel
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __UDA1380_H
+#define __UDA1380_H
+
+struct uda1380_platform_data {
+ int gpio_power;
+ int gpio_reset;
+ int dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL 1
+};
+
+#endif /* __UDA1380_H */
diff --git a/kernel/include/sound/util_mem.h b/kernel/include/sound/util_mem.h
new file mode 100644
index 000000000..a1fb706b5
--- /dev/null
+++ b/kernel/include/sound/util_mem.h
@@ -0,0 +1,64 @@
+#ifndef __SOUND_UTIL_MEM_H
+#define __SOUND_UTIL_MEM_H
+
+#include <linux/mutex.h>
+/*
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Generic memory management routines for soundcard memory allocation
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/*
+ * memory block
+ */
+struct snd_util_memblk {
+ unsigned int size; /* size of this block */
+ unsigned int offset; /* zero-offset of this block */
+ struct list_head list; /* link */
+};
+
+#define snd_util_memblk_argptr(blk) (void*)((char*)(blk) + sizeof(struct snd_util_memblk))
+
+/*
+ * memory management information
+ */
+struct snd_util_memhdr {
+ unsigned int size; /* size of whole data */
+ struct list_head block; /* block linked-list header */
+ int nblocks; /* # of allocated blocks */
+ unsigned int used; /* used memory size */
+ int block_extra_size; /* extra data size of chunk */
+ struct mutex block_mutex; /* lock */
+};
+
+/*
+ * prototypes
+ */
+struct snd_util_memhdr *snd_util_memhdr_new(int memsize);
+void snd_util_memhdr_free(struct snd_util_memhdr *hdr);
+struct snd_util_memblk *snd_util_mem_alloc(struct snd_util_memhdr *hdr, int size);
+int snd_util_mem_free(struct snd_util_memhdr *hdr, struct snd_util_memblk *blk);
+int snd_util_mem_avail(struct snd_util_memhdr *hdr);
+
+/* functions without mutex */
+struct snd_util_memblk *__snd_util_mem_alloc(struct snd_util_memhdr *hdr, int size);
+void __snd_util_mem_free(struct snd_util_memhdr *hdr, struct snd_util_memblk *blk);
+struct snd_util_memblk *__snd_util_memblk_new(struct snd_util_memhdr *hdr,
+ unsigned int units,
+ struct list_head *prev);
+
+#endif /* __SOUND_UTIL_MEM_H */
diff --git a/kernel/include/sound/vx_core.h b/kernel/include/sound/vx_core.h
new file mode 100644
index 000000000..cae9c9d4e
--- /dev/null
+++ b/kernel/include/sound/vx_core.h
@@ -0,0 +1,548 @@
+/*
+ * Driver for Digigram VX soundcards
+ *
+ * Hardware core part
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_VX_COMMON_H
+#define __SOUND_VX_COMMON_H
+
+#include <sound/pcm.h>
+#include <sound/hwdep.h>
+#include <linux/interrupt.h>
+
+struct firmware;
+struct device;
+
+#define VX_DRIVER_VERSION 0x010000 /* 1.0.0 */
+
+/*
+ */
+#define SIZE_MAX_CMD 0x10
+#define SIZE_MAX_STATUS 0x10
+
+struct vx_rmh {
+ u16 LgCmd; /* length of the command to send (WORDs) */
+ u16 LgStat; /* length of the status received (WORDs) */
+ u32 Cmd[SIZE_MAX_CMD];
+ u32 Stat[SIZE_MAX_STATUS];
+ u16 DspStat; /* status type, RMP_SSIZE_XXX */
+};
+
+typedef u64 pcx_time_t;
+
+#define VX_MAX_PIPES 16
+#define VX_MAX_PERIODS 32
+#define VX_MAX_CODECS 2
+
+struct vx_ibl_info {
+ int size; /* the current IBL size (0 = query) in bytes */
+ int max_size; /* max. IBL size in bytes */
+ int min_size; /* min. IBL size in bytes */
+ int granularity; /* granularity */
+};
+
+struct vx_pipe {
+ int number;
+ unsigned int is_capture: 1;
+ unsigned int data_mode: 1;
+ unsigned int running: 1;
+ unsigned int prepared: 1;
+ int channels;
+ unsigned int differed_type;
+ pcx_time_t pcx_time;
+ struct snd_pcm_substream *substream;
+
+ int hbuf_size; /* H-buffer size in bytes */
+ int buffer_bytes; /* the ALSA pcm buffer size in bytes */
+ int period_bytes; /* the ALSA pcm period size in bytes */
+ int hw_ptr; /* the current hardware pointer in bytes */
+ int position; /* the current position in frames (playback only) */
+ int transferred; /* the transferred size (per period) in frames */
+ int align; /* size of alignment */
+ u64 cur_count; /* current sample position (for playback) */
+
+ unsigned int references; /* an output pipe may be used for monitoring and/or playback */
+ struct vx_pipe *monitoring_pipe; /* pointer to the monitoring pipe (capture pipe only)*/
+};
+
+struct vx_core;
+
+struct snd_vx_ops {
+ /* low-level i/o */
+ unsigned char (*in8)(struct vx_core *chip, int reg);
+ unsigned int (*in32)(struct vx_core *chip, int reg);
+ void (*out8)(struct vx_core *chip, int reg, unsigned char val);
+ void (*out32)(struct vx_core *chip, int reg, unsigned int val);
+ /* irq */
+ int (*test_and_ack)(struct vx_core *chip);
+ void (*validate_irq)(struct vx_core *chip, int enable);
+ /* codec */
+ void (*write_codec)(struct vx_core *chip, int codec, unsigned int data);
+ void (*akm_write)(struct vx_core *chip, int reg, unsigned int data);
+ void (*reset_codec)(struct vx_core *chip);
+ void (*change_audio_source)(struct vx_core *chip, int src);
+ void (*set_clock_source)(struct vx_core *chp, int src);
+ /* chip init */
+ int (*load_dsp)(struct vx_core *chip, int idx, const struct firmware *fw);
+ void (*reset_dsp)(struct vx_core *chip);
+ void (*reset_board)(struct vx_core *chip, int cold_reset);
+ int (*add_controls)(struct vx_core *chip);
+ /* pcm */
+ void (*dma_write)(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int count);
+ void (*dma_read)(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int count);
+};
+
+struct snd_vx_hardware {
+ const char *name;
+ int type; /* VX_TYPE_XXX */
+
+ /* hardware specs */
+ unsigned int num_codecs;
+ unsigned int num_ins;
+ unsigned int num_outs;
+ unsigned int output_level_max;
+ const unsigned int *output_level_db_scale;
+};
+
+/* hwdep id string */
+#define SND_VX_HWDEP_ID "VX Loader"
+
+/* hardware type */
+enum {
+ /* VX222 PCI */
+ VX_TYPE_BOARD, /* old VX222 PCI */
+ VX_TYPE_V2, /* VX222 V2 PCI */
+ VX_TYPE_MIC, /* VX222 Mic PCI */
+ /* VX-pocket */
+ VX_TYPE_VXPOCKET, /* VXpocket V2 */
+ VX_TYPE_VXP440, /* VXpocket 440 */
+ VX_TYPE_NUMS
+};
+
+/* chip status */
+enum {
+ VX_STAT_XILINX_LOADED = (1 << 0), /* devices are registered */
+ VX_STAT_DEVICE_INIT = (1 << 1), /* devices are registered */
+ VX_STAT_CHIP_INIT = (1 << 2), /* all operational */
+ VX_STAT_IN_SUSPEND = (1 << 10), /* in suspend phase */
+ VX_STAT_IS_STALE = (1 << 15) /* device is stale */
+};
+
+/* min/max values for analog output for old codecs */
+#define VX_ANALOG_OUT_LEVEL_MAX 0xe3
+
+struct vx_core {
+ /* ALSA stuff */
+ struct snd_card *card;
+ struct snd_pcm *pcm[VX_MAX_CODECS];
+ int type; /* VX_TYPE_XXX */
+
+ int irq;
+ /* ports are defined externally */
+
+ /* low-level functions */
+ struct snd_vx_hardware *hw;
+ struct snd_vx_ops *ops;
+
+ struct mutex lock;
+
+ unsigned int chip_status;
+ unsigned int pcm_running;
+
+ struct device *dev;
+ struct snd_hwdep *hwdep;
+
+ struct vx_rmh irq_rmh; /* RMH used in interrupts */
+
+ unsigned int audio_info; /* see VX_AUDIO_INFO */
+ unsigned int audio_ins;
+ unsigned int audio_outs;
+ struct vx_pipe **playback_pipes;
+ struct vx_pipe **capture_pipes;
+
+ /* clock and audio sources */
+ unsigned int audio_source; /* current audio input source */
+ unsigned int audio_source_target;
+ unsigned int clock_mode; /* clock mode (VX_CLOCK_MODE_XXX) */
+ unsigned int clock_source; /* current clock source (INTERNAL_QUARTZ or UER_SYNC) */
+ unsigned int freq; /* current frequency */
+ unsigned int freq_detected; /* detected frequency from digital in */
+ unsigned int uer_detected; /* VX_UER_MODE_XXX */
+ unsigned int uer_bits; /* IEC958 status bits */
+ struct vx_ibl_info ibl; /* IBL information */
+
+ /* mixer setting */
+ int output_level[VX_MAX_CODECS][2]; /* analog output level */
+ int audio_gain[2][4]; /* digital audio level (playback/capture) */
+ unsigned char audio_active[4]; /* mute/unmute on digital playback */
+ int audio_monitor[4]; /* playback hw-monitor level */
+ unsigned char audio_monitor_active[4]; /* playback hw-monitor mute/unmute */
+
+ struct mutex mixer_mutex;
+
+ const struct firmware *firmware[4]; /* loaded firmware data */
+};
+
+
+/*
+ * constructor
+ */
+struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw,
+ struct snd_vx_ops *ops, int extra_size);
+int snd_vx_setup_firmware(struct vx_core *chip);
+int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *dsp);
+int snd_vx_dsp_boot(struct vx_core *chip, const struct firmware *dsp);
+int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp);
+
+void snd_vx_free_firmware(struct vx_core *chip);
+
+/*
+ * interrupt handler; exported for pcmcia
+ */
+irqreturn_t snd_vx_irq_handler(int irq, void *dev);
+irqreturn_t snd_vx_threaded_irq_handler(int irq, void *dev);
+
+/*
+ * lowlevel functions
+ */
+static inline int vx_test_and_ack(struct vx_core *chip)
+{
+ return chip->ops->test_and_ack(chip);
+}
+
+static inline void vx_validate_irq(struct vx_core *chip, int enable)
+{
+ chip->ops->validate_irq(chip, enable);
+}
+
+static inline unsigned char snd_vx_inb(struct vx_core *chip, int reg)
+{
+ return chip->ops->in8(chip, reg);
+}
+
+static inline unsigned int snd_vx_inl(struct vx_core *chip, int reg)
+{
+ return chip->ops->in32(chip, reg);
+}
+
+static inline void snd_vx_outb(struct vx_core *chip, int reg, unsigned char val)
+{
+ chip->ops->out8(chip, reg, val);
+}
+
+static inline void snd_vx_outl(struct vx_core *chip, int reg, unsigned int val)
+{
+ chip->ops->out32(chip, reg, val);
+}
+
+#define vx_inb(chip,reg) snd_vx_inb(chip, VX_##reg)
+#define vx_outb(chip,reg,val) snd_vx_outb(chip, VX_##reg,val)
+#define vx_inl(chip,reg) snd_vx_inl(chip, VX_##reg)
+#define vx_outl(chip,reg,val) snd_vx_outl(chip, VX_##reg,val)
+
+static inline void vx_reset_dsp(struct vx_core *chip)
+{
+ chip->ops->reset_dsp(chip);
+}
+
+int vx_send_msg(struct vx_core *chip, struct vx_rmh *rmh);
+int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh);
+int vx_send_rih(struct vx_core *chip, int cmd);
+int vx_send_rih_nolock(struct vx_core *chip, int cmd);
+
+void vx_reset_codec(struct vx_core *chip, int cold_reset);
+
+/*
+ * check the bit on the specified register
+ * returns zero if a bit matches, or a negative error code.
+ * exported for vxpocket driver
+ */
+int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int time);
+#define vx_check_isr(chip,mask,bit,time) snd_vx_check_reg_bit(chip, VX_ISR, mask, bit, time)
+#define vx_wait_isr_bit(chip,bit) vx_check_isr(chip, bit, bit, 200)
+#define vx_wait_for_rx_full(chip) vx_wait_isr_bit(chip, ISR_RX_FULL)
+
+
+/*
+ * pseudo-DMA transfer
+ */
+static inline void vx_pseudo_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int count)
+{
+ chip->ops->dma_write(chip, runtime, pipe, count);
+}
+
+static inline void vx_pseudo_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int count)
+{
+ chip->ops->dma_read(chip, runtime, pipe, count);
+}
+
+
+
+/* error with hardware code,
+ * the return value is -(VX_ERR_MASK | actual-hw-error-code)
+ */
+#define VX_ERR_MASK 0x1000000
+#define vx_get_error(err) (-(err) & ~VX_ERR_MASK)
+
+
+/*
+ * pcm stuff
+ */
+int snd_vx_pcm_new(struct vx_core *chip);
+void vx_pcm_update_intr(struct vx_core *chip, unsigned int events);
+
+/*
+ * mixer stuff
+ */
+int snd_vx_mixer_new(struct vx_core *chip);
+void vx_toggle_dac_mute(struct vx_core *chip, int mute);
+int vx_sync_audio_source(struct vx_core *chip);
+int vx_set_monitor_level(struct vx_core *chip, int audio, int level, int active);
+
+/*
+ * IEC958 & clock stuff
+ */
+void vx_set_iec958_status(struct vx_core *chip, unsigned int bits);
+int vx_set_clock(struct vx_core *chip, unsigned int freq);
+void vx_set_internal_clock(struct vx_core *chip, unsigned int freq);
+int vx_change_frequency(struct vx_core *chip);
+
+
+/*
+ * PM
+ */
+int snd_vx_suspend(struct vx_core *card);
+int snd_vx_resume(struct vx_core *card);
+
+/*
+ * hardware constants
+ */
+
+#define vx_has_new_dsp(chip) ((chip)->type != VX_TYPE_BOARD)
+#define vx_is_pcmcia(chip) ((chip)->type >= VX_TYPE_VXPOCKET)
+
+/* audio input source */
+enum {
+ VX_AUDIO_SRC_DIGITAL,
+ VX_AUDIO_SRC_LINE,
+ VX_AUDIO_SRC_MIC
+};
+
+/* clock source */
+enum {
+ INTERNAL_QUARTZ,
+ UER_SYNC
+};
+
+/* clock mode */
+enum {
+ VX_CLOCK_MODE_AUTO, /* depending on the current audio source */
+ VX_CLOCK_MODE_INTERNAL, /* fixed to internal quartz */
+ VX_CLOCK_MODE_EXTERNAL /* fixed to UER sync */
+};
+
+/* SPDIF/UER type */
+enum {
+ VX_UER_MODE_CONSUMER,
+ VX_UER_MODE_PROFESSIONAL,
+ VX_UER_MODE_NOT_PRESENT,
+};
+
+/* register indices */
+enum {
+ VX_ICR,
+ VX_CVR,
+ VX_ISR,
+ VX_IVR,
+ VX_RXH,
+ VX_TXH = VX_RXH,
+ VX_RXM,
+ VX_TXM = VX_RXM,
+ VX_RXL,
+ VX_TXL = VX_RXL,
+ VX_DMA,
+ VX_CDSP,
+ VX_RFREQ,
+ VX_RUER_V2,
+ VX_GAIN,
+ VX_DATA = VX_GAIN,
+ VX_MEMIRQ,
+ VX_ACQ,
+ VX_BIT0,
+ VX_BIT1,
+ VX_MIC0,
+ VX_MIC1,
+ VX_MIC2,
+ VX_MIC3,
+ VX_PLX0,
+ VX_PLX1,
+ VX_PLX2,
+
+ VX_LOFREQ, // V2: ACQ, VP: RFREQ
+ VX_HIFREQ, // V2: BIT0, VP: RUER_V2
+ VX_CSUER, // V2: BIT1, VP: BIT0
+ VX_RUER, // V2: RUER_V2, VP: BIT1
+
+ VX_REG_MAX,
+
+ /* aliases for VX board */
+ VX_RESET_DMA = VX_ISR,
+ VX_CFG = VX_RFREQ,
+ VX_STATUS = VX_MEMIRQ,
+ VX_SELMIC = VX_MIC0,
+ VX_COMPOT = VX_MIC1,
+ VX_SCOMPR = VX_MIC2,
+ VX_GLIMIT = VX_MIC3,
+ VX_INTCSR = VX_PLX0,
+ VX_CNTRL = VX_PLX1,
+ VX_GPIOC = VX_PLX2,
+
+ /* aliases for VXPOCKET board */
+ VX_MICRO = VX_MEMIRQ,
+ VX_CODEC2 = VX_MEMIRQ,
+ VX_DIALOG = VX_ACQ,
+
+};
+
+/* RMH status type */
+enum {
+ RMH_SSIZE_FIXED = 0, /* status size given by the driver (in LgStat) */
+ RMH_SSIZE_ARG = 1, /* status size given in the LSB byte */
+ RMH_SSIZE_MASK = 2, /* status size given in bitmask */
+};
+
+
+/* bits for ICR register */
+#define ICR_HF1 0x10
+#define ICR_HF0 0x08
+#define ICR_TREQ 0x02 /* Interrupt mode + HREQ set on for transfer (->DSP) request */
+#define ICR_RREQ 0x01 /* Interrupt mode + RREQ set on for transfer (->PC) request */
+
+/* bits for CVR register */
+#define CVR_HC 0x80
+
+/* bits for ISR register */
+#define ISR_HF3 0x10
+#define ISR_HF2 0x08
+#define ISR_CHK 0x10
+#define ISR_ERR 0x08
+#define ISR_TX_READY 0x04
+#define ISR_TX_EMPTY 0x02
+#define ISR_RX_FULL 0x01
+
+/* Constants used to access the DATA register */
+#define VX_DATA_CODEC_MASK 0x80
+#define VX_DATA_XICOR_MASK 0x80
+
+/* Constants used to access the CSUER register (both for VX2 and VXP) */
+#define VX_SUER_FREQ_MASK 0x0c
+#define VX_SUER_FREQ_32KHz_MASK 0x0c
+#define VX_SUER_FREQ_44KHz_MASK 0x00
+#define VX_SUER_FREQ_48KHz_MASK 0x04
+#define VX_SUER_DATA_PRESENT_MASK 0x02
+#define VX_SUER_CLOCK_PRESENT_MASK 0x01
+
+#define VX_CUER_HH_BITC_SEL_MASK 0x08
+#define VX_CUER_MH_BITC_SEL_MASK 0x04
+#define VX_CUER_ML_BITC_SEL_MASK 0x02
+#define VX_CUER_LL_BITC_SEL_MASK 0x01
+
+#define XX_UER_CBITS_OFFSET_MASK 0x1f
+
+
+/* bits for audio_info */
+#define VX_AUDIO_INFO_REAL_TIME (1<<0) /* real-time processing available */
+#define VX_AUDIO_INFO_OFFLINE (1<<1) /* offline processing available */
+#define VX_AUDIO_INFO_MPEG1 (1<<5)
+#define VX_AUDIO_INFO_MPEG2 (1<<6)
+#define VX_AUDIO_INFO_LINEAR_8 (1<<7)
+#define VX_AUDIO_INFO_LINEAR_16 (1<<8)
+#define VX_AUDIO_INFO_LINEAR_24 (1<<9)
+
+/* DSP Interrupt Request values */
+#define VXP_IRQ_OFFSET 0x40 /* add 0x40 offset for vxpocket and vx222/v2 */
+/* call with vx_send_irq_dsp() */
+#define IRQ_MESS_WRITE_END 0x30
+#define IRQ_MESS_WRITE_NEXT 0x32
+#define IRQ_MESS_READ_NEXT 0x34
+#define IRQ_MESS_READ_END 0x36
+#define IRQ_MESSAGE 0x38
+#define IRQ_RESET_CHK 0x3A
+#define IRQ_CONNECT_STREAM_NEXT 0x26
+#define IRQ_CONNECT_STREAM_END 0x28
+#define IRQ_PAUSE_START_CONNECT 0x2A
+#define IRQ_END_CONNECTION 0x2C
+
+/* Is there async. events pending ( IT Source Test ) */
+#define ASYNC_EVENTS_PENDING 0x008000
+#define HBUFFER_EVENTS_PENDING 0x004000 // Not always accurate
+#define NOTIF_EVENTS_PENDING 0x002000
+#define TIME_CODE_EVENT_PENDING 0x001000
+#define FREQUENCY_CHANGE_EVENT_PENDING 0x000800
+#define END_OF_BUFFER_EVENTS_PENDING 0x000400
+#define FATAL_DSP_ERROR 0xff0000
+
+/* Stream Format Header Defines */
+#define HEADER_FMT_BASE 0xFED00000
+#define HEADER_FMT_MONO 0x000000C0
+#define HEADER_FMT_INTEL 0x00008000
+#define HEADER_FMT_16BITS 0x00002000
+#define HEADER_FMT_24BITS 0x00004000
+#define HEADER_FMT_UPTO11 0x00000200 /* frequency is less or equ. to 11k.*/
+#define HEADER_FMT_UPTO32 0x00000100 /* frequency is over 11k and less then 32k.*/
+
+/* Constants used to access the Codec */
+#define XX_CODEC_SELECTOR 0x20
+/* codec commands */
+#define XX_CODEC_ADC_CONTROL_REGISTER 0x01
+#define XX_CODEC_DAC_CONTROL_REGISTER 0x02
+#define XX_CODEC_LEVEL_LEFT_REGISTER 0x03
+#define XX_CODEC_LEVEL_RIGHT_REGISTER 0x04
+#define XX_CODEC_PORT_MODE_REGISTER 0x05
+#define XX_CODEC_STATUS_REPORT_REGISTER 0x06
+#define XX_CODEC_CLOCK_CONTROL_REGISTER 0x07
+
+/*
+ * Audio-level control values
+ */
+#define CVAL_M110DB 0x000 /* -110dB */
+#define CVAL_M99DB 0x02C
+#define CVAL_M21DB 0x163
+#define CVAL_M18DB 0x16F
+#define CVAL_M10DB 0x18F
+#define CVAL_0DB 0x1B7
+#define CVAL_18DB 0x1FF /* +18dB */
+#define CVAL_MAX 0x1FF
+
+#define AUDIO_IO_HAS_MUTE_LEVEL 0x400000
+#define AUDIO_IO_HAS_MUTE_MONITORING_1 0x200000
+#define AUDIO_IO_HAS_MUTE_MONITORING_2 0x100000
+#define VALID_AUDIO_IO_DIGITAL_LEVEL 0x01
+#define VALID_AUDIO_IO_MONITORING_LEVEL 0x02
+#define VALID_AUDIO_IO_MUTE_LEVEL 0x04
+#define VALID_AUDIO_IO_MUTE_MONITORING_1 0x08
+#define VALID_AUDIO_IO_MUTE_MONITORING_2 0x10
+
+
+#endif /* __SOUND_VX_COMMON_H */
diff --git a/kernel/include/sound/wavefront.h b/kernel/include/sound/wavefront.h
new file mode 100644
index 000000000..15d82e594
--- /dev/null
+++ b/kernel/include/sound/wavefront.h
@@ -0,0 +1,695 @@
+#ifndef __SOUND_WAVEFRONT_H__
+#define __SOUND_WAVEFRONT_H__
+
+/*
+ * Driver for Turtle Beach Wavefront cards (Maui,Tropez,Tropez+)
+ *
+ * Copyright (c) by Paul Barton-Davis <pbd@op.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#if (!defined(__GNUC__) && !defined(__GNUG__))
+
+ You will not be able to compile this file correctly without gcc, because
+ it is necessary to pack the "wavefront_alias" structure to a size
+ of 22 bytes, corresponding to 16-bit alignment (as would have been
+ the case on the original platform, MS-DOS). If this is not done,
+ then WavePatch-format files cannot be read/written correctly.
+ The method used to do this here ("__attribute__((packed)") is
+ completely compiler dependent.
+
+ All other wavefront_* types end up aligned to 32 bit values and
+ still have the same (correct) size.
+
+#else
+
+ /* However, note that as of G++ 2.7.3.2, g++ was unable to
+ correctly parse *type* __attribute__ tags. It will do the
+ right thing if we use the "packed" attribute on each struct
+ member, which has the same semantics anyway.
+ */
+
+#endif /* __GNUC__ */
+
+/***************************** WARNING ********************************
+ PLEASE DO NOT MODIFY THIS FILE IN ANY WAY THAT AFFECTS ITS ABILITY TO
+ BE USED WITH EITHER C *OR* C++.
+ **********************************************************************/
+
+#ifndef NUM_MIDIKEYS
+#define NUM_MIDIKEYS 128
+#endif /* NUM_MIDIKEYS */
+
+#ifndef NUM_MIDICHANNELS
+#define NUM_MIDICHANNELS 16
+#endif /* NUM_MIDICHANNELS */
+
+/* These are very useful/important. the original wavefront interface
+ was developed on a 16 bit system, where sizeof(int) = 2
+ bytes. Defining things like this makes the code much more portable, and
+ easier to understand without having to toggle back and forth
+ between a 16-bit view of the world and a 32-bit one.
+ */
+
+#ifndef __KERNEL__
+/* keep them for compatibility */
+typedef short s16;
+typedef unsigned short u16;
+typedef int s32;
+typedef unsigned int u32;
+typedef char s8;
+typedef unsigned char u8;
+typedef s16 INT16;
+typedef u16 UINT16;
+typedef s32 INT32;
+typedef u32 UINT32;
+typedef s8 CHAR8;
+typedef u8 UCHAR8;
+#endif
+
+/* Pseudo-commands not part of the WaveFront command set.
+ These are used for various driver controls and direct
+ hardware control.
+ */
+
+#define WFC_DEBUG_DRIVER 0
+#define WFC_FX_IOCTL 1
+#define WFC_PATCH_STATUS 2
+#define WFC_PROGRAM_STATUS 3
+#define WFC_SAMPLE_STATUS 4
+#define WFC_DISABLE_INTERRUPTS 5
+#define WFC_ENABLE_INTERRUPTS 6
+#define WFC_INTERRUPT_STATUS 7
+#define WFC_ROMSAMPLES_RDONLY 8
+#define WFC_IDENTIFY_SLOT_TYPE 9
+
+/* Wavefront synth commands
+ */
+
+#define WFC_DOWNLOAD_SAMPLE 0x80
+#define WFC_DOWNLOAD_BLOCK 0x81
+#define WFC_DOWNLOAD_MULTISAMPLE 0x82
+#define WFC_DOWNLOAD_SAMPLE_ALIAS 0x83
+#define WFC_DELETE_SAMPLE 0x84
+#define WFC_REPORT_FREE_MEMORY 0x85
+#define WFC_DOWNLOAD_PATCH 0x86
+#define WFC_DOWNLOAD_PROGRAM 0x87
+#define WFC_SET_SYNTHVOL 0x89
+#define WFC_SET_NVOICES 0x8B
+#define WFC_DOWNLOAD_DRUM 0x90
+#define WFC_GET_SYNTHVOL 0x92
+#define WFC_GET_NVOICES 0x94
+#define WFC_DISABLE_CHANNEL 0x9A
+#define WFC_ENABLE_CHANNEL 0x9B
+#define WFC_MISYNTH_OFF 0x9D
+#define WFC_MISYNTH_ON 0x9E
+#define WFC_FIRMWARE_VERSION 0x9F
+#define WFC_GET_NSAMPLES 0xA0
+#define WFC_DISABLE_DRUM_PROGRAM 0xA2
+#define WFC_UPLOAD_PATCH 0xA3
+#define WFC_UPLOAD_PROGRAM 0xA4
+#define WFC_SET_TUNING 0xA6
+#define WFC_GET_TUNING 0xA7
+#define WFC_VMIDI_ON 0xA8
+#define WFC_VMIDI_OFF 0xA9
+#define WFC_MIDI_STATUS 0xAA
+#define WFC_GET_CHANNEL_STATUS 0xAB
+#define WFC_DOWNLOAD_SAMPLE_HEADER 0xAC
+#define WFC_UPLOAD_SAMPLE_HEADER 0xAD
+#define WFC_UPLOAD_MULTISAMPLE 0xAE
+#define WFC_UPLOAD_SAMPLE_ALIAS 0xAF
+#define WFC_IDENTIFY_SAMPLE_TYPE 0xB0
+#define WFC_DOWNLOAD_EDRUM_PROGRAM 0xB1
+#define WFC_UPLOAD_EDRUM_PROGRAM 0xB2
+#define WFC_SET_EDRUM_CHANNEL 0xB3
+#define WFC_INSTOUT_LEVELS 0xB4
+#define WFC_PEAKOUT_LEVELS 0xB5
+#define WFC_REPORT_CHANNEL_PROGRAMS 0xB6
+#define WFC_HARDWARE_VERSION 0xCF
+#define WFC_UPLOAD_SAMPLE_PARAMS 0xD7
+#define WFC_DOWNLOAD_OS 0xF1
+#define WFC_NOOP 0xFF
+
+#define WF_MAX_SAMPLE 512
+#define WF_MAX_PATCH 256
+#define WF_MAX_PROGRAM 128
+
+#define WF_SECTION_MAX 44 /* longest OS section length */
+
+/* # of bytes we send to the board when sending it various kinds of
+ substantive data, such as samples, patches and programs.
+*/
+
+#define WF_PROGRAM_BYTES 32
+#define WF_PATCH_BYTES 132
+#define WF_SAMPLE_BYTES 27
+#define WF_SAMPLE_HDR_BYTES 25
+#define WF_ALIAS_BYTES 25
+#define WF_DRUM_BYTES 9
+#define WF_MSAMPLE_BYTES 259 /* (MIDI_KEYS * 2) + 3 */
+
+#define WF_ACK 0x80
+#define WF_DMA_ACK 0x81
+
+/* OR-values for MIDI status bits */
+
+#define WF_MIDI_VIRTUAL_ENABLED 0x1
+#define WF_MIDI_VIRTUAL_IS_EXTERNAL 0x2
+#define WF_MIDI_IN_TO_SYNTH_DISABLED 0x4
+
+/* slot indexes for struct address_info: makes code a little more mnemonic */
+
+#define WF_SYNTH_SLOT 0
+#define WF_INTERNAL_MIDI_SLOT 1
+#define WF_EXTERNAL_MIDI_SLOT 2
+
+/* Magic MIDI bytes used to switch I/O streams on the ICS2115 MPU401
+ emulation. Note these NEVER show up in output from the device and
+ should NEVER be used in input unless Virtual MIDI mode has been
+ disabled. If they do show up as input, the results are unpredictable.
+*/
+
+#define WF_EXTERNAL_SWITCH 0xFD
+#define WF_INTERNAL_SWITCH 0xF9
+
+/* Debugging flags */
+
+#define WF_DEBUG_CMD 0x1
+#define WF_DEBUG_DATA 0x2
+#define WF_DEBUG_LOAD_PATCH 0x4
+#define WF_DEBUG_IO 0x8
+
+/* WavePatch file format stuff */
+
+#define WF_WAVEPATCH_VERSION 120; /* Current version number (1.2) */
+#define WF_MAX_COMMENT 64 /* Comment length */
+#define WF_NUM_LAYERS 4
+#define WF_NAME_LENGTH 32
+#define WF_SOURCE_LENGTH 260
+
+#define BankFileID "Bank"
+#define DrumkitFileID "DrumKit"
+#define ProgramFileID "Program"
+
+struct wf_envelope
+{
+ u8 attack_time:7;
+ u8 Unused1:1;
+
+ u8 decay1_time:7;
+ u8 Unused2:1;
+
+ u8 decay2_time:7;
+ u8 Unused3:1;
+
+ u8 sustain_time:7;
+ u8 Unused4:1;
+
+ u8 release_time:7;
+ u8 Unused5:1;
+
+ u8 release2_time:7;
+ u8 Unused6:1;
+
+ s8 attack_level;
+ s8 decay1_level;
+ s8 decay2_level;
+ s8 sustain_level;
+ s8 release_level;
+
+ u8 attack_velocity:7;
+ u8 Unused7:1;
+
+ u8 volume_velocity:7;
+ u8 Unused8:1;
+
+ u8 keyboard_scaling:7;
+ u8 Unused9:1;
+};
+typedef struct wf_envelope wavefront_envelope;
+
+struct wf_lfo
+{
+ u8 sample_number;
+
+ u8 frequency:7;
+ u8 Unused1:1;
+
+ u8 am_src:4;
+ u8 fm_src:4;
+
+ s8 fm_amount;
+ s8 am_amount;
+ s8 start_level;
+ s8 end_level;
+
+ u8 ramp_delay:7;
+ u8 wave_restart:1; /* for LFO2 only */
+
+ u8 ramp_time:7;
+ u8 Unused2:1;
+};
+typedef struct wf_lfo wavefront_lfo;
+
+struct wf_patch
+{
+ s16 frequency_bias; /* ** THIS IS IN MOTOROLA FORMAT!! ** */
+
+ u8 amplitude_bias:7;
+ u8 Unused1:1;
+
+ u8 portamento:7;
+ u8 Unused2:1;
+
+ u8 sample_number;
+
+ u8 pitch_bend:4;
+ u8 sample_msb:1;
+ u8 Unused3:3;
+
+ u8 mono:1;
+ u8 retrigger:1;
+ u8 nohold:1;
+ u8 restart:1;
+ u8 filterconfig:2; /* SDK says "not used" */
+ u8 reuse:1;
+ u8 reset_lfo:1;
+
+ u8 fm_src2:4;
+ u8 fm_src1:4;
+
+ s8 fm_amount1;
+ s8 fm_amount2;
+
+ u8 am_src:4;
+ u8 Unused4:4;
+
+ s8 am_amount;
+
+ u8 fc1_mode:4;
+ u8 fc2_mode:4;
+
+ s8 fc1_mod_amount;
+ s8 fc1_keyboard_scaling;
+ s8 fc1_bias;
+ s8 fc2_mod_amount;
+ s8 fc2_keyboard_scaling;
+ s8 fc2_bias;
+
+ u8 randomizer:7;
+ u8 Unused5:1;
+
+ struct wf_envelope envelope1;
+ struct wf_envelope envelope2;
+ struct wf_lfo lfo1;
+ struct wf_lfo lfo2;
+};
+typedef struct wf_patch wavefront_patch;
+
+struct wf_layer
+{
+ u8 patch_number;
+
+ u8 mix_level:7;
+ u8 mute:1;
+
+ u8 split_point:7;
+ u8 play_below:1;
+
+ u8 pan_mod_src:2;
+ u8 pan_or_mod:1;
+ u8 pan:4;
+ u8 split_type:1;
+};
+typedef struct wf_layer wavefront_layer;
+
+struct wf_program
+{
+ struct wf_layer layer[WF_NUM_LAYERS];
+};
+typedef struct wf_program wavefront_program;
+
+struct wf_sample_offset
+{
+ s32 Fraction:4;
+ s32 Integer:20;
+ s32 Unused:8;
+};
+typedef struct wf_sample_offset wavefront_sample_offset;
+
+/* Sample slot types */
+
+#define WF_ST_SAMPLE 0
+#define WF_ST_MULTISAMPLE 1
+#define WF_ST_ALIAS 2
+#define WF_ST_EMPTY 3
+
+/* pseudo's */
+
+#define WF_ST_DRUM 4
+#define WF_ST_PROGRAM 5
+#define WF_ST_PATCH 6
+#define WF_ST_SAMPLEHDR 7
+
+#define WF_ST_MASK 0xf
+
+/* Flags for slot status. These occupy the upper bits of the same byte
+ as a sample type.
+*/
+
+#define WF_SLOT_USED 0x80 /* XXX don't rely on this being accurate */
+#define WF_SLOT_FILLED 0x40
+#define WF_SLOT_ROM 0x20
+
+#define WF_SLOT_MASK 0xf0
+
+/* channel constants */
+
+#define WF_CH_MONO 0
+#define WF_CH_LEFT 1
+#define WF_CH_RIGHT 2
+
+/* Sample formats */
+
+#define LINEAR_16BIT 0
+#define WHITE_NOISE 1
+#define LINEAR_8BIT 2
+#define MULAW_8BIT 3
+
+#define WF_SAMPLE_IS_8BIT(smpl) ((smpl)->SampleResolution&2)
+
+
+/*
+
+ Because most/all of the sample data we pass in via pointers has
+ never been copied (just mmap-ed into user space straight from the
+ disk), it would be nice to allow handling of multi-channel sample
+ data without forcing user-level extraction of the relevant bytes.
+
+ So, we need a way of specifying which channel to use (the WaveFront
+ only handles mono samples in a given slot), and the only way to do
+ this without using some struct other than wavefront_sample as the
+ interface is the awful hack of using the unused bits in a
+ wavefront_sample:
+
+ Val Meaning
+ --- -------
+ 0 no channel selection (use channel 1, sample is MONO)
+ 1 use first channel, and skip one
+ 2 use second channel, and skip one
+ 3 use third channel, and skip two
+ 4 use fourth channel, skip three
+ 5 use fifth channel, skip four
+ 6 use six channel, skip five
+
+
+ This can handle up to 4 channels, and anyone downloading >4 channels
+ of sample data just to select one of them needs to find some tools
+ like sox ...
+
+ NOTE: values 0, 1 and 2 correspond to WF_CH_* above. This is
+ important.
+
+*/
+
+#define WF_SET_CHANNEL(samp,chn) \
+ (samp)->Unused1 = chn & 0x1; \
+ (samp)->Unused2 = chn & 0x2; \
+ (samp)->Unused3 = chn & 0x4
+
+#define WF_GET_CHANNEL(samp) \
+ (((samp)->Unused3 << 2)|((samp)->Unused2<<1)|(samp)->Unused1)
+
+typedef struct wf_sample {
+ struct wf_sample_offset sampleStartOffset;
+ struct wf_sample_offset loopStartOffset;
+ struct wf_sample_offset loopEndOffset;
+ struct wf_sample_offset sampleEndOffset;
+ s16 FrequencyBias;
+ u8 SampleResolution:2; /* sample_format */
+ u8 Unused1:1;
+ u8 Loop:1;
+ u8 Bidirectional:1;
+ u8 Unused2:1;
+ u8 Reverse:1;
+ u8 Unused3:1;
+} wavefront_sample;
+
+typedef struct wf_multisample {
+ s16 NumberOfSamples; /* log2 of the number of samples */
+ s16 SampleNumber[NUM_MIDIKEYS];
+} wavefront_multisample;
+
+typedef struct wf_alias {
+ s16 OriginalSample;
+
+ struct wf_sample_offset sampleStartOffset;
+ struct wf_sample_offset loopStartOffset;
+ struct wf_sample_offset sampleEndOffset;
+ struct wf_sample_offset loopEndOffset;
+
+ s16 FrequencyBias;
+
+ u8 SampleResolution:2;
+ u8 Unused1:1;
+ u8 Loop:1;
+ u8 Bidirectional:1;
+ u8 Unused2:1;
+ u8 Reverse:1;
+ u8 Unused3:1;
+
+ /* This structure is meant to be padded only to 16 bits on their
+ original. Of course, whoever wrote their documentation didn't
+ realize that sizeof(struct) can be >=
+ sum(sizeof(struct-fields)) and so thought that giving a C level
+ description of the structs used in WavePatch files was
+ sufficient. I suppose it was, as long as you remember the
+ standard 16->32 bit issues.
+ */
+
+ u8 sixteen_bit_padding;
+} __attribute__((packed)) wavefront_alias;
+
+typedef struct wf_drum {
+ u8 PatchNumber;
+ u8 MixLevel:7;
+ u8 Unmute:1;
+ u8 Group:4;
+ u8 Unused1:4;
+ u8 PanModSource:2;
+ u8 PanModulated:1;
+ u8 PanAmount:4;
+ u8 Unused2:1;
+} wavefront_drum;
+
+typedef struct wf_drumkit {
+ struct wf_drum drum[NUM_MIDIKEYS];
+} wavefront_drumkit;
+
+typedef struct wf_channel_programs {
+ u8 Program[NUM_MIDICHANNELS];
+} wavefront_channel_programs;
+
+/* How to get MIDI channel status from the data returned by
+ a WFC_GET_CHANNEL_STATUS command (a struct wf_channel_programs)
+*/
+
+#define WF_CHANNEL_STATUS(ch,wcp) (wcp)[(ch/7)] & (1<<((ch)%7))
+
+typedef union wf_any {
+ wavefront_sample s;
+ wavefront_multisample ms;
+ wavefront_alias a;
+ wavefront_program pr;
+ wavefront_patch p;
+ wavefront_drum d;
+} wavefront_any;
+
+/* Hannu Solvainen hoped that his "patch_info" struct in soundcard.h
+ might work for other wave-table based patch loading situations.
+ Alas, his fears were correct. The WaveFront doesn't even come with
+ just "patches", but several different kind of structures that
+ control the sound generation process.
+ */
+
+typedef struct wf_patch_info {
+
+ /* the first two fields are used by the OSS "patch loading" interface
+ only, and are unused by the current user-level library.
+ */
+
+ s16 key; /* Use WAVEFRONT_PATCH here */
+ u16 devno; /* fill in when sending */
+ u8 subkey; /* WF_ST_{SAMPLE,ALIAS,etc.} */
+
+#define WAVEFRONT_FIND_FREE_SAMPLE_SLOT 999
+
+ u16 number; /* patch/sample/prog number */
+
+ u32 size; /* size of any data included in
+ one of the fields in `hdrptr', or
+ as `dataptr'.
+
+ NOTE: for actual samples, this is
+ the size of the *SELECTED CHANNEL*
+ even if more data is actually available.
+
+ So, a stereo sample (2 channels) of
+ 6000 bytes total has `size' = 3000.
+
+ See the macros and comments for
+ WF_{GET,SET}_CHANNEL above.
+
+ */
+ wavefront_any __user *hdrptr; /* user-space ptr to hdr bytes */
+ u16 __user *dataptr; /* actual sample data */
+
+ wavefront_any hdr; /* kernel-space copy of hdr bytes */
+} wavefront_patch_info;
+
+/* The maximum number of bytes we will ever move to or from user space
+ in response to a WFC_* command. This obviously doesn't cover
+ actual sample data.
+*/
+
+#define WF_MAX_READ sizeof(wavefront_multisample)
+#define WF_MAX_WRITE sizeof(wavefront_multisample)
+
+/*
+ This allows us to execute any WF command except the download/upload
+ ones, which are handled differently due to copyin/copyout issues as
+ well as data-nybbling to/from the card.
+ */
+
+typedef struct wavefront_control {
+ int cmd; /* WFC_* */
+ char status; /* return status to user-space */
+ unsigned char rbuf[WF_MAX_READ]; /* bytes read from card */
+ unsigned char wbuf[WF_MAX_WRITE]; /* bytes written to card */
+} wavefront_control;
+
+#define WFCTL_WFCMD 0x1
+#define WFCTL_LOAD_SPP 0x2
+
+/* Modulator table */
+
+#define WF_MOD_LFO1 0
+#define WF_MOD_LFO2 1
+#define WF_MOD_ENV1 2
+#define WF_MOD_ENV2 3
+#define WF_MOD_KEYBOARD 4
+#define WF_MOD_LOGKEY 5
+#define WF_MOD_VELOCITY 6
+#define WF_MOD_LOGVEL 7
+#define WF_MOD_RANDOM 8
+#define WF_MOD_PRESSURE 9
+#define WF_MOD_MOD_WHEEL 10
+#define WF_MOD_1 WF_MOD_MOD_WHEEL
+#define WF_MOD_BREATH 11
+#define WF_MOD_2 WF_MOD_BREATH
+#define WF_MOD_FOOT 12
+#define WF_MOD_4 WF_MOD_FOOT
+#define WF_MOD_VOLUME 13
+#define WF_MOD_7 WF_MOD_VOLUME
+#define WF_MOD_PAN 14
+#define WF_MOD_10 WF_MOD_PAN
+#define WF_MOD_EXPR 15
+#define WF_MOD_11 WF_MOD_EXPR
+
+/* FX-related material */
+
+typedef struct wf_fx_info {
+ int request; /* see list below */
+ long data[4]; /* we don't need much */
+} wavefront_fx_info;
+
+/* support for each of these will be forthcoming once I or someone
+ else has figured out which of the addresses on page 6 and page 7 of
+ the YSS225 control each parameter. Incidentally, these come from
+ the Windows driver interface, but again, Turtle Beach didn't
+ document the API to use them.
+*/
+
+#define WFFX_SETOUTGAIN 0
+#define WFFX_SETSTEREOOUTGAIN 1
+#define WFFX_SETREVERBIN1GAIN 2
+#define WFFX_SETREVERBIN2GAIN 3
+#define WFFX_SETREVERBIN3GAIN 4
+#define WFFX_SETCHORUSINPORT 5
+#define WFFX_SETREVERBIN1PORT 6
+#define WFFX_SETREVERBIN2PORT 7
+#define WFFX_SETREVERBIN3PORT 8
+#define WFFX_SETEFFECTPORT 9
+#define WFFX_SETAUXPORT 10
+#define WFFX_SETREVERBTYPE 11
+#define WFFX_SETREVERBDELAY 12
+#define WFFX_SETCHORUSLFO 13
+#define WFFX_SETCHORUSPMD 14
+#define WFFX_SETCHORUSAMD 15
+#define WFFX_SETEFFECT 16
+#define WFFX_SETBASEALL 17
+#define WFFX_SETREVERBALL 18
+#define WFFX_SETCHORUSALL 20
+#define WFFX_SETREVERBDEF 22
+#define WFFX_SETCHORUSDEF 23
+#define WFFX_DELAYSETINGAIN 24
+#define WFFX_DELAYSETFBGAIN 25
+#define WFFX_DELAYSETFBLPF 26
+#define WFFX_DELAYSETGAIN 27
+#define WFFX_DELAYSETTIME 28
+#define WFFX_DELAYSETFBTIME 29
+#define WFFX_DELAYSETALL 30
+#define WFFX_DELAYSETDEF 32
+#define WFFX_SDELAYSETINGAIN 33
+#define WFFX_SDELAYSETFBGAIN 34
+#define WFFX_SDELAYSETFBLPF 35
+#define WFFX_SDELAYSETGAIN 36
+#define WFFX_SDELAYSETTIME 37
+#define WFFX_SDELAYSETFBTIME 38
+#define WFFX_SDELAYSETALL 39
+#define WFFX_SDELAYSETDEF 41
+#define WFFX_DEQSETINGAIN 42
+#define WFFX_DEQSETFILTER 43
+#define WFFX_DEQSETALL 44
+#define WFFX_DEQSETDEF 46
+#define WFFX_MUTE 47
+#define WFFX_FLANGESETBALANCE 48
+#define WFFX_FLANGESETDELAY 49
+#define WFFX_FLANGESETDWFFX_TH 50
+#define WFFX_FLANGESETFBGAIN 51
+#define WFFX_FLANGESETINGAIN 52
+#define WFFX_FLANGESETLFO 53
+#define WFFX_FLANGESETALL 54
+#define WFFX_FLANGESETDEF 56
+#define WFFX_PITCHSETSHIFT 57
+#define WFFX_PITCHSETBALANCE 58
+#define WFFX_PITCHSETALL 59
+#define WFFX_PITCHSETDEF 61
+#define WFFX_SRSSETINGAIN 62
+#define WFFX_SRSSETSPACE 63
+#define WFFX_SRSSETCENTER 64
+#define WFFX_SRSSETGAIN 65
+#define WFFX_SRSSETMODE 66
+#define WFFX_SRSSETDEF 68
+
+/* Allow direct user-space control over FX memory/coefficient data.
+ In theory this could be used to download the FX microprogram,
+ but it would be a little slower, and involve some weird code.
+ */
+
+#define WFFX_MEMSET 69
+
+#endif /* __SOUND_WAVEFRONT_H__ */
diff --git a/kernel/include/sound/wm0010.h b/kernel/include/sound/wm0010.h
new file mode 100644
index 000000000..3261e9081
--- /dev/null
+++ b/kernel/include/sound/wm0010.h
@@ -0,0 +1,27 @@
+/*
+ * wm0010.h -- Platform data for WM0010 DSP Driver
+ *
+ * Copyright 2012 Wolfson Microelectronics PLC.
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef WM0010_PDATA_H
+#define WM0010_PDATA_H
+
+struct wm0010_pdata {
+ int gpio_reset;
+
+ /* Set if there is an inverter between the GPIO controlling
+ * the reset signal and the device.
+ */
+ int reset_active_high;
+ int irq_flags;
+};
+
+#endif
diff --git a/kernel/include/sound/wm1250-ev1.h b/kernel/include/sound/wm1250-ev1.h
new file mode 100644
index 000000000..7dff82834
--- /dev/null
+++ b/kernel/include/sound/wm1250-ev1.h
@@ -0,0 +1,27 @@
+/*
+ * linux/sound/wm1250-ev1.h - Platform data for WM1250-EV1
+ *
+ * Copyright 2011 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM1250_EV1_H
+#define __LINUX_SND_WM1250_EV1_H
+
+#define WM1250_EV1_NUM_GPIOS 5
+
+#define WM1250_EV1_GPIO_CLK_ENA 0
+#define WM1250_EV1_GPIO_CLK_SEL0 1
+#define WM1250_EV1_GPIO_CLK_SEL1 2
+#define WM1250_EV1_GPIO_OSR 3
+#define WM1250_EV1_GPIO_MASTER 4
+
+
+struct wm1250_ev1_pdata {
+ int gpios[WM1250_EV1_NUM_GPIOS];
+};
+
+#endif
diff --git a/kernel/include/sound/wm2000.h b/kernel/include/sound/wm2000.h
new file mode 100644
index 000000000..4de81f41c
--- /dev/null
+++ b/kernel/include/sound/wm2000.h
@@ -0,0 +1,23 @@
+/*
+ * linux/sound/wm2000.h -- Platform data for WM2000
+ *
+ * Copyright 2010 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM2000_H
+#define __LINUX_SND_WM2000_H
+
+struct wm2000_platform_data {
+ /** Filename for system-specific image to download to device. */
+ const char *download_file;
+
+ /** Disable speech clarity enhancement, for use when an
+ * external algorithm is used. */
+ unsigned int speech_enh_disable:1;
+};
+
+#endif
diff --git a/kernel/include/sound/wm2200.h b/kernel/include/sound/wm2200.h
new file mode 100644
index 000000000..bc7ab1a4b
--- /dev/null
+++ b/kernel/include/sound/wm2200.h
@@ -0,0 +1,61 @@
+/*
+ * linux/sound/wm2200.h -- Platform data for WM2200
+ *
+ * Copyright 2012 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM2200_H
+#define __LINUX_SND_WM2200_H
+
+#define WM2200_GPIO_SET 0x10000
+#define WM2200_MAX_MICBIAS 2
+
+enum wm2200_in_mode {
+ WM2200_IN_SE = 0,
+ WM2200_IN_DIFF = 1,
+ WM2200_IN_DMIC = 2,
+};
+
+enum wm2200_dmic_sup {
+ WM2200_DMIC_SUP_MICVDD = 0,
+ WM2200_DMIC_SUP_MICBIAS1 = 1,
+ WM2200_DMIC_SUP_MICBIAS2 = 2,
+};
+
+enum wm2200_mbias_lvl {
+ WM2200_MBIAS_LVL_1V5 = 1,
+ WM2200_MBIAS_LVL_1V8 = 2,
+ WM2200_MBIAS_LVL_1V9 = 3,
+ WM2200_MBIAS_LVL_2V0 = 4,
+ WM2200_MBIAS_LVL_2V2 = 5,
+ WM2200_MBIAS_LVL_2V4 = 6,
+ WM2200_MBIAS_LVL_2V5 = 7,
+ WM2200_MBIAS_LVL_2V6 = 8,
+};
+
+struct wm2200_micbias {
+ enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */
+ unsigned int discharge:1; /** Actively discharge */
+ unsigned int fast_start:1; /** Enable aggressive startup ramp rate */
+ unsigned int bypass:1; /** Use bypass mode */
+};
+
+struct wm2200_pdata {
+ int reset; /** GPIO controlling /RESET, if any */
+ int ldo_ena; /** GPIO controlling LODENA, if any */
+ int irq_flags;
+
+ int gpio_defaults[4];
+
+ enum wm2200_in_mode in_mode[3];
+ enum wm2200_dmic_sup dmic_sup[3];
+
+ /** MICBIAS configurations */
+ struct wm2200_micbias micbias[WM2200_MAX_MICBIAS];
+};
+
+#endif
diff --git a/kernel/include/sound/wm5100.h b/kernel/include/sound/wm5100.h
new file mode 100644
index 000000000..617d0c4a1
--- /dev/null
+++ b/kernel/include/sound/wm5100.h
@@ -0,0 +1,59 @@
+/*
+ * linux/sound/wm5100.h -- Platform data for WM5100
+ *
+ * Copyright 2011 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM5100_H
+#define __LINUX_SND_WM5100_H
+
+enum wm5100_in_mode {
+ WM5100_IN_SE = 0,
+ WM5100_IN_DIFF = 1,
+ WM5100_IN_DMIC = 2,
+};
+
+enum wm5100_dmic_sup {
+ WM5100_DMIC_SUP_MICVDD = 0,
+ WM5100_DMIC_SUP_MICBIAS1 = 1,
+ WM5100_DMIC_SUP_MICBIAS2 = 2,
+ WM5100_DMIC_SUP_MICBIAS3 = 3,
+};
+
+enum wm5100_micdet_bias {
+ WM5100_MICDET_MICBIAS1 = 0,
+ WM5100_MICDET_MICBIAS2 = 1,
+ WM5100_MICDET_MICBIAS3 = 2,
+};
+
+struct wm5100_jack_mode {
+ enum wm5100_micdet_bias bias;
+ int hp_pol;
+ int micd_src;
+};
+
+#define WM5100_GPIO_SET 0x10000
+
+struct wm5100_pdata {
+ int reset; /** GPIO controlling /RESET, if any */
+ int ldo_ena; /** GPIO controlling LODENA, if any */
+ int hp_pol; /** GPIO controlling headset polarity, if any */
+ int irq_flags;
+ int gpio_base;
+
+ struct wm5100_jack_mode jack_modes[2];
+
+ /* Input pin mode selection */
+ enum wm5100_in_mode in_mode[4];
+
+ /* DMIC supply selection */
+ enum wm5100_dmic_sup dmic_sup[4];
+
+ int gpio_defaults[6];
+};
+
+#endif
diff --git a/kernel/include/sound/wm8903.h b/kernel/include/sound/wm8903.h
new file mode 100644
index 000000000..b310c5a3a
--- /dev/null
+++ b/kernel/include/sound/wm8903.h
@@ -0,0 +1,266 @@
+/*
+ * linux/sound/wm8903.h -- Platform data for WM8903
+ *
+ * Copyright 2010 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM8903_H
+#define __LINUX_SND_WM8903_H
+
+/*
+ * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of
+ * zero in platform data means "don't touch this pin".
+ */
+#define WM8903_GPIO_CONFIG_ZERO 0x8000
+
+/*
+ * R6 (0x06) - Mic Bias Control 0
+ */
+#define WM8903_MICDET_THR_MASK 0x0030 /* MICDET_THR - [5:4] */
+#define WM8903_MICDET_THR_SHIFT 4 /* MICDET_THR - [5:4] */
+#define WM8903_MICDET_THR_WIDTH 2 /* MICDET_THR - [5:4] */
+#define WM8903_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */
+#define WM8903_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */
+#define WM8903_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */
+#define WM8903_MICDET_ENA 0x0002 /* MICDET_ENA */
+#define WM8903_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */
+#define WM8903_MICDET_ENA_SHIFT 1 /* MICDET_ENA */
+#define WM8903_MICDET_ENA_WIDTH 1 /* MICDET_ENA */
+#define WM8903_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */
+
+/*
+ * WM8903_GPn_FN values
+ *
+ * See datasheets for list of valid values per pin
+ */
+#define WM8903_GPn_FN_GPIO_OUTPUT 0
+#define WM8903_GPn_FN_BCLK 1
+#define WM8903_GPn_FN_IRQ_OUTPT 2
+#define WM8903_GPn_FN_GPIO_INPUT 3
+#define WM8903_GPn_FN_MICBIAS_CURRENT_DETECT 4
+#define WM8903_GPn_FN_MICBIAS_SHORT_DETECT 5
+#define WM8903_GPn_FN_DMIC_LR_CLK_OUTPUT 6
+#define WM8903_GPn_FN_FLL_LOCK_OUTPUT 8
+#define WM8903_GPn_FN_FLL_CLOCK_OUTPUT 9
+
+/*
+ * R116 (0x74) - GPIO Control 1
+ */
+#define WM8903_GP1_FN_MASK 0x1F00 /* GP1_FN - [12:8] */
+#define WM8903_GP1_FN_SHIFT 8 /* GP1_FN - [12:8] */
+#define WM8903_GP1_FN_WIDTH 5 /* GP1_FN - [12:8] */
+#define WM8903_GP1_DIR 0x0080 /* GP1_DIR */
+#define WM8903_GP1_DIR_MASK 0x0080 /* GP1_DIR */
+#define WM8903_GP1_DIR_SHIFT 7 /* GP1_DIR */
+#define WM8903_GP1_DIR_WIDTH 1 /* GP1_DIR */
+#define WM8903_GP1_OP_CFG 0x0040 /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_MASK 0x0040 /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_SHIFT 6 /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_WIDTH 1 /* GP1_OP_CFG */
+#define WM8903_GP1_IP_CFG 0x0020 /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_MASK 0x0020 /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_SHIFT 5 /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_WIDTH 1 /* GP1_IP_CFG */
+#define WM8903_GP1_LVL 0x0010 /* GP1_LVL */
+#define WM8903_GP1_LVL_MASK 0x0010 /* GP1_LVL */
+#define WM8903_GP1_LVL_SHIFT 4 /* GP1_LVL */
+#define WM8903_GP1_LVL_WIDTH 1 /* GP1_LVL */
+#define WM8903_GP1_PD 0x0008 /* GP1_PD */
+#define WM8903_GP1_PD_MASK 0x0008 /* GP1_PD */
+#define WM8903_GP1_PD_SHIFT 3 /* GP1_PD */
+#define WM8903_GP1_PD_WIDTH 1 /* GP1_PD */
+#define WM8903_GP1_PU 0x0004 /* GP1_PU */
+#define WM8903_GP1_PU_MASK 0x0004 /* GP1_PU */
+#define WM8903_GP1_PU_SHIFT 2 /* GP1_PU */
+#define WM8903_GP1_PU_WIDTH 1 /* GP1_PU */
+#define WM8903_GP1_INTMODE 0x0002 /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_MASK 0x0002 /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_SHIFT 1 /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_WIDTH 1 /* GP1_INTMODE */
+#define WM8903_GP1_DB 0x0001 /* GP1_DB */
+#define WM8903_GP1_DB_MASK 0x0001 /* GP1_DB */
+#define WM8903_GP1_DB_SHIFT 0 /* GP1_DB */
+#define WM8903_GP1_DB_WIDTH 1 /* GP1_DB */
+
+/*
+ * R117 (0x75) - GPIO Control 2
+ */
+#define WM8903_GP2_FN_MASK 0x1F00 /* GP2_FN - [12:8] */
+#define WM8903_GP2_FN_SHIFT 8 /* GP2_FN - [12:8] */
+#define WM8903_GP2_FN_WIDTH 5 /* GP2_FN - [12:8] */
+#define WM8903_GP2_DIR 0x0080 /* GP2_DIR */
+#define WM8903_GP2_DIR_MASK 0x0080 /* GP2_DIR */
+#define WM8903_GP2_DIR_SHIFT 7 /* GP2_DIR */
+#define WM8903_GP2_DIR_WIDTH 1 /* GP2_DIR */
+#define WM8903_GP2_OP_CFG 0x0040 /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_MASK 0x0040 /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_SHIFT 6 /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_WIDTH 1 /* GP2_OP_CFG */
+#define WM8903_GP2_IP_CFG 0x0020 /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_MASK 0x0020 /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_SHIFT 5 /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_WIDTH 1 /* GP2_IP_CFG */
+#define WM8903_GP2_LVL 0x0010 /* GP2_LVL */
+#define WM8903_GP2_LVL_MASK 0x0010 /* GP2_LVL */
+#define WM8903_GP2_LVL_SHIFT 4 /* GP2_LVL */
+#define WM8903_GP2_LVL_WIDTH 1 /* GP2_LVL */
+#define WM8903_GP2_PD 0x0008 /* GP2_PD */
+#define WM8903_GP2_PD_MASK 0x0008 /* GP2_PD */
+#define WM8903_GP2_PD_SHIFT 3 /* GP2_PD */
+#define WM8903_GP2_PD_WIDTH 1 /* GP2_PD */
+#define WM8903_GP2_PU 0x0004 /* GP2_PU */
+#define WM8903_GP2_PU_MASK 0x0004 /* GP2_PU */
+#define WM8903_GP2_PU_SHIFT 2 /* GP2_PU */
+#define WM8903_GP2_PU_WIDTH 1 /* GP2_PU */
+#define WM8903_GP2_INTMODE 0x0002 /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_MASK 0x0002 /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_SHIFT 1 /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_WIDTH 1 /* GP2_INTMODE */
+#define WM8903_GP2_DB 0x0001 /* GP2_DB */
+#define WM8903_GP2_DB_MASK 0x0001 /* GP2_DB */
+#define WM8903_GP2_DB_SHIFT 0 /* GP2_DB */
+#define WM8903_GP2_DB_WIDTH 1 /* GP2_DB */
+
+/*
+ * R118 (0x76) - GPIO Control 3
+ */
+#define WM8903_GP3_FN_MASK 0x1F00 /* GP3_FN - [12:8] */
+#define WM8903_GP3_FN_SHIFT 8 /* GP3_FN - [12:8] */
+#define WM8903_GP3_FN_WIDTH 5 /* GP3_FN - [12:8] */
+#define WM8903_GP3_DIR 0x0080 /* GP3_DIR */
+#define WM8903_GP3_DIR_MASK 0x0080 /* GP3_DIR */
+#define WM8903_GP3_DIR_SHIFT 7 /* GP3_DIR */
+#define WM8903_GP3_DIR_WIDTH 1 /* GP3_DIR */
+#define WM8903_GP3_OP_CFG 0x0040 /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_MASK 0x0040 /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_SHIFT 6 /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_WIDTH 1 /* GP3_OP_CFG */
+#define WM8903_GP3_IP_CFG 0x0020 /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_MASK 0x0020 /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_SHIFT 5 /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_WIDTH 1 /* GP3_IP_CFG */
+#define WM8903_GP3_LVL 0x0010 /* GP3_LVL */
+#define WM8903_GP3_LVL_MASK 0x0010 /* GP3_LVL */
+#define WM8903_GP3_LVL_SHIFT 4 /* GP3_LVL */
+#define WM8903_GP3_LVL_WIDTH 1 /* GP3_LVL */
+#define WM8903_GP3_PD 0x0008 /* GP3_PD */
+#define WM8903_GP3_PD_MASK 0x0008 /* GP3_PD */
+#define WM8903_GP3_PD_SHIFT 3 /* GP3_PD */
+#define WM8903_GP3_PD_WIDTH 1 /* GP3_PD */
+#define WM8903_GP3_PU 0x0004 /* GP3_PU */
+#define WM8903_GP3_PU_MASK 0x0004 /* GP3_PU */
+#define WM8903_GP3_PU_SHIFT 2 /* GP3_PU */
+#define WM8903_GP3_PU_WIDTH 1 /* GP3_PU */
+#define WM8903_GP3_INTMODE 0x0002 /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_MASK 0x0002 /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_SHIFT 1 /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_WIDTH 1 /* GP3_INTMODE */
+#define WM8903_GP3_DB 0x0001 /* GP3_DB */
+#define WM8903_GP3_DB_MASK 0x0001 /* GP3_DB */
+#define WM8903_GP3_DB_SHIFT 0 /* GP3_DB */
+#define WM8903_GP3_DB_WIDTH 1 /* GP3_DB */
+
+/*
+ * R119 (0x77) - GPIO Control 4
+ */
+#define WM8903_GP4_FN_MASK 0x1F00 /* GP4_FN - [12:8] */
+#define WM8903_GP4_FN_SHIFT 8 /* GP4_FN - [12:8] */
+#define WM8903_GP4_FN_WIDTH 5 /* GP4_FN - [12:8] */
+#define WM8903_GP4_DIR 0x0080 /* GP4_DIR */
+#define WM8903_GP4_DIR_MASK 0x0080 /* GP4_DIR */
+#define WM8903_GP4_DIR_SHIFT 7 /* GP4_DIR */
+#define WM8903_GP4_DIR_WIDTH 1 /* GP4_DIR */
+#define WM8903_GP4_OP_CFG 0x0040 /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_MASK 0x0040 /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_SHIFT 6 /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_WIDTH 1 /* GP4_OP_CFG */
+#define WM8903_GP4_IP_CFG 0x0020 /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_MASK 0x0020 /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_SHIFT 5 /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_WIDTH 1 /* GP4_IP_CFG */
+#define WM8903_GP4_LVL 0x0010 /* GP4_LVL */
+#define WM8903_GP4_LVL_MASK 0x0010 /* GP4_LVL */
+#define WM8903_GP4_LVL_SHIFT 4 /* GP4_LVL */
+#define WM8903_GP4_LVL_WIDTH 1 /* GP4_LVL */
+#define WM8903_GP4_PD 0x0008 /* GP4_PD */
+#define WM8903_GP4_PD_MASK 0x0008 /* GP4_PD */
+#define WM8903_GP4_PD_SHIFT 3 /* GP4_PD */
+#define WM8903_GP4_PD_WIDTH 1 /* GP4_PD */
+#define WM8903_GP4_PU 0x0004 /* GP4_PU */
+#define WM8903_GP4_PU_MASK 0x0004 /* GP4_PU */
+#define WM8903_GP4_PU_SHIFT 2 /* GP4_PU */
+#define WM8903_GP4_PU_WIDTH 1 /* GP4_PU */
+#define WM8903_GP4_INTMODE 0x0002 /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_MASK 0x0002 /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_SHIFT 1 /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_WIDTH 1 /* GP4_INTMODE */
+#define WM8903_GP4_DB 0x0001 /* GP4_DB */
+#define WM8903_GP4_DB_MASK 0x0001 /* GP4_DB */
+#define WM8903_GP4_DB_SHIFT 0 /* GP4_DB */
+#define WM8903_GP4_DB_WIDTH 1 /* GP4_DB */
+
+/*
+ * R120 (0x78) - GPIO Control 5
+ */
+#define WM8903_GP5_FN_MASK 0x1F00 /* GP5_FN - [12:8] */
+#define WM8903_GP5_FN_SHIFT 8 /* GP5_FN - [12:8] */
+#define WM8903_GP5_FN_WIDTH 5 /* GP5_FN - [12:8] */
+#define WM8903_GP5_DIR 0x0080 /* GP5_DIR */
+#define WM8903_GP5_DIR_MASK 0x0080 /* GP5_DIR */
+#define WM8903_GP5_DIR_SHIFT 7 /* GP5_DIR */
+#define WM8903_GP5_DIR_WIDTH 1 /* GP5_DIR */
+#define WM8903_GP5_OP_CFG 0x0040 /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_MASK 0x0040 /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_SHIFT 6 /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_WIDTH 1 /* GP5_OP_CFG */
+#define WM8903_GP5_IP_CFG 0x0020 /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_MASK 0x0020 /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_SHIFT 5 /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_WIDTH 1 /* GP5_IP_CFG */
+#define WM8903_GP5_LVL 0x0010 /* GP5_LVL */
+#define WM8903_GP5_LVL_MASK 0x0010 /* GP5_LVL */
+#define WM8903_GP5_LVL_SHIFT 4 /* GP5_LVL */
+#define WM8903_GP5_LVL_WIDTH 1 /* GP5_LVL */
+#define WM8903_GP5_PD 0x0008 /* GP5_PD */
+#define WM8903_GP5_PD_MASK 0x0008 /* GP5_PD */
+#define WM8903_GP5_PD_SHIFT 3 /* GP5_PD */
+#define WM8903_GP5_PD_WIDTH 1 /* GP5_PD */
+#define WM8903_GP5_PU 0x0004 /* GP5_PU */
+#define WM8903_GP5_PU_MASK 0x0004 /* GP5_PU */
+#define WM8903_GP5_PU_SHIFT 2 /* GP5_PU */
+#define WM8903_GP5_PU_WIDTH 1 /* GP5_PU */
+#define WM8903_GP5_INTMODE 0x0002 /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_MASK 0x0002 /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_SHIFT 1 /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_WIDTH 1 /* GP5_INTMODE */
+#define WM8903_GP5_DB 0x0001 /* GP5_DB */
+#define WM8903_GP5_DB_MASK 0x0001 /* GP5_DB */
+#define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */
+#define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */
+
+#define WM8903_NUM_GPIO 5
+
+struct wm8903_platform_data {
+ bool irq_active_low; /* Set if IRQ active low, default high */
+
+ /* Default register value for R6 (Mic bias), used to configure
+ * microphone detection. In conjunction with gpio_cfg this
+ * can be used to route the microphone status signals out onto
+ * the GPIOs for use with snd_soc_jack_add_gpios().
+ */
+ u16 micdet_cfg;
+
+ int micdet_delay; /* Delay after microphone detection (ms) */
+
+ int gpio_base;
+ u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */
+};
+
+#endif
diff --git a/kernel/include/sound/wm8904.h b/kernel/include/sound/wm8904.h
new file mode 100644
index 000000000..898be3a8d
--- /dev/null
+++ b/kernel/include/sound/wm8904.h
@@ -0,0 +1,163 @@
+/*
+ * Platform data for WM8904
+ *
+ * Copyright 2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef __MFD_WM8994_PDATA_H__
+#define __MFD_WM8994_PDATA_H__
+
+/* Used to enable configuration of a GPIO to all zeros */
+#define WM8904_GPIO_NO_CONFIG 0x8000
+
+/*
+ * R6 (0x06) - Mic Bias Control 0
+ */
+#define WM8904_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */
+#define WM8904_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */
+#define WM8904_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */
+#define WM8904_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */
+#define WM8904_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */
+#define WM8904_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */
+#define WM8904_MICDET_ENA 0x0002 /* MICDET_ENA */
+#define WM8904_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */
+#define WM8904_MICDET_ENA_SHIFT 1 /* MICDET_ENA */
+#define WM8904_MICDET_ENA_WIDTH 1 /* MICDET_ENA */
+#define WM8904_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */
+#define WM8904_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */
+#define WM8904_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */
+#define WM8904_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */
+
+/*
+ * R7 (0x07) - Mic Bias Control 1
+ */
+#define WM8904_MIC_DET_FILTER_ENA 0x8000 /* MIC_DET_FILTER_ENA */
+#define WM8904_MIC_DET_FILTER_ENA_MASK 0x8000 /* MIC_DET_FILTER_ENA */
+#define WM8904_MIC_DET_FILTER_ENA_SHIFT 15 /* MIC_DET_FILTER_ENA */
+#define WM8904_MIC_DET_FILTER_ENA_WIDTH 1 /* MIC_DET_FILTER_ENA */
+#define WM8904_MIC_SHORT_FILTER_ENA 0x4000 /* MIC_SHORT_FILTER_ENA */
+#define WM8904_MIC_SHORT_FILTER_ENA_MASK 0x4000 /* MIC_SHORT_FILTER_ENA */
+#define WM8904_MIC_SHORT_FILTER_ENA_SHIFT 14 /* MIC_SHORT_FILTER_ENA */
+#define WM8904_MIC_SHORT_FILTER_ENA_WIDTH 1 /* MIC_SHORT_FILTER_ENA */
+#define WM8904_MICBIAS_SEL_MASK 0x0007 /* MICBIAS_SEL - [2:0] */
+#define WM8904_MICBIAS_SEL_SHIFT 0 /* MICBIAS_SEL - [2:0] */
+#define WM8904_MICBIAS_SEL_WIDTH 3 /* MICBIAS_SEL - [2:0] */
+
+
+/*
+ * R121 (0x79) - GPIO Control 1
+ */
+#define WM8904_GPIO1_PU 0x0020 /* GPIO1_PU */
+#define WM8904_GPIO1_PU_MASK 0x0020 /* GPIO1_PU */
+#define WM8904_GPIO1_PU_SHIFT 5 /* GPIO1_PU */
+#define WM8904_GPIO1_PU_WIDTH 1 /* GPIO1_PU */
+#define WM8904_GPIO1_PD 0x0010 /* GPIO1_PD */
+#define WM8904_GPIO1_PD_MASK 0x0010 /* GPIO1_PD */
+#define WM8904_GPIO1_PD_SHIFT 4 /* GPIO1_PD */
+#define WM8904_GPIO1_PD_WIDTH 1 /* GPIO1_PD */
+#define WM8904_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */
+#define WM8904_GPIO1_SEL_SHIFT 0 /* GPIO1_SEL - [3:0] */
+#define WM8904_GPIO1_SEL_WIDTH 4 /* GPIO1_SEL - [3:0] */
+
+/*
+ * R122 (0x7A) - GPIO Control 2
+ */
+#define WM8904_GPIO2_PU 0x0020 /* GPIO2_PU */
+#define WM8904_GPIO2_PU_MASK 0x0020 /* GPIO2_PU */
+#define WM8904_GPIO2_PU_SHIFT 5 /* GPIO2_PU */
+#define WM8904_GPIO2_PU_WIDTH 1 /* GPIO2_PU */
+#define WM8904_GPIO2_PD 0x0010 /* GPIO2_PD */
+#define WM8904_GPIO2_PD_MASK 0x0010 /* GPIO2_PD */
+#define WM8904_GPIO2_PD_SHIFT 4 /* GPIO2_PD */
+#define WM8904_GPIO2_PD_WIDTH 1 /* GPIO2_PD */
+#define WM8904_GPIO2_SEL_MASK 0x000F /* GPIO2_SEL - [3:0] */
+#define WM8904_GPIO2_SEL_SHIFT 0 /* GPIO2_SEL - [3:0] */
+#define WM8904_GPIO2_SEL_WIDTH 4 /* GPIO2_SEL - [3:0] */
+
+/*
+ * R123 (0x7B) - GPIO Control 3
+ */
+#define WM8904_GPIO3_PU 0x0020 /* GPIO3_PU */
+#define WM8904_GPIO3_PU_MASK 0x0020 /* GPIO3_PU */
+#define WM8904_GPIO3_PU_SHIFT 5 /* GPIO3_PU */
+#define WM8904_GPIO3_PU_WIDTH 1 /* GPIO3_PU */
+#define WM8904_GPIO3_PD 0x0010 /* GPIO3_PD */
+#define WM8904_GPIO3_PD_MASK 0x0010 /* GPIO3_PD */
+#define WM8904_GPIO3_PD_SHIFT 4 /* GPIO3_PD */
+#define WM8904_GPIO3_PD_WIDTH 1 /* GPIO3_PD */
+#define WM8904_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */
+#define WM8904_GPIO3_SEL_SHIFT 0 /* GPIO3_SEL - [3:0] */
+#define WM8904_GPIO3_SEL_WIDTH 4 /* GPIO3_SEL - [3:0] */
+
+/*
+ * R124 (0x7C) - GPIO Control 4
+ */
+#define WM8904_GPI7_ENA 0x0200 /* GPI7_ENA */
+#define WM8904_GPI7_ENA_MASK 0x0200 /* GPI7_ENA */
+#define WM8904_GPI7_ENA_SHIFT 9 /* GPI7_ENA */
+#define WM8904_GPI7_ENA_WIDTH 1 /* GPI7_ENA */
+#define WM8904_GPI8_ENA 0x0100 /* GPI8_ENA */
+#define WM8904_GPI8_ENA_MASK 0x0100 /* GPI8_ENA */
+#define WM8904_GPI8_ENA_SHIFT 8 /* GPI8_ENA */
+#define WM8904_GPI8_ENA_WIDTH 1 /* GPI8_ENA */
+#define WM8904_GPIO_BCLK_MODE_ENA 0x0080 /* GPIO_BCLK_MODE_ENA */
+#define WM8904_GPIO_BCLK_MODE_ENA_MASK 0x0080 /* GPIO_BCLK_MODE_ENA */
+#define WM8904_GPIO_BCLK_MODE_ENA_SHIFT 7 /* GPIO_BCLK_MODE_ENA */
+#define WM8904_GPIO_BCLK_MODE_ENA_WIDTH 1 /* GPIO_BCLK_MODE_ENA */
+#define WM8904_GPIO_BCLK_SEL_MASK 0x000F /* GPIO_BCLK_SEL - [3:0] */
+#define WM8904_GPIO_BCLK_SEL_SHIFT 0 /* GPIO_BCLK_SEL - [3:0] */
+#define WM8904_GPIO_BCLK_SEL_WIDTH 4 /* GPIO_BCLK_SEL - [3:0] */
+
+#define WM8904_MIC_REGS 2
+#define WM8904_GPIO_REGS 4
+#define WM8904_DRC_REGS 4
+#define WM8904_EQ_REGS 25
+
+/**
+ * DRC configurations are specified with a label and a set of register
+ * values to write (the enable bits will be ignored). At runtime an
+ * enumerated control will be presented for each DRC block allowing
+ * the user to choose the configration to use.
+ *
+ * Configurations may be generated by hand or by using the DRC control
+ * panel provided by the WISCE - see http://www.wolfsonmicro.com/wisce/
+ * for details.
+ */
+struct wm8904_drc_cfg {
+ const char *name;
+ u16 regs[WM8904_DRC_REGS];
+};
+
+/**
+ * ReTune Mobile configurations are specified with a label, sample
+ * rate and set of values to write (the enable bits will be ignored).
+ *
+ * Configurations are expected to be generated using the ReTune Mobile
+ * control panel in WISCE - see http://www.wolfsonmicro.com/wisce/
+ */
+struct wm8904_retune_mobile_cfg {
+ const char *name;
+ unsigned int rate;
+ u16 regs[WM8904_EQ_REGS];
+};
+
+struct wm8904_pdata {
+ int num_drc_cfgs;
+ struct wm8904_drc_cfg *drc_cfgs;
+
+ int num_retune_mobile_cfgs;
+ struct wm8904_retune_mobile_cfg *retune_mobile_cfgs;
+
+ u32 gpio_cfg[WM8904_GPIO_REGS];
+ u32 mic_cfg[WM8904_MIC_REGS];
+};
+
+#endif
diff --git a/kernel/include/sound/wm8955.h b/kernel/include/sound/wm8955.h
new file mode 100644
index 000000000..5074ef499
--- /dev/null
+++ b/kernel/include/sound/wm8955.h
@@ -0,0 +1,26 @@
+/*
+ * Platform data for WM8955
+ *
+ * Copyright 2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef __WM8955_PDATA_H__
+#define __WM8955_PDATA_H__
+
+struct wm8955_pdata {
+ /* Configure LOUT2/ROUT2 to drive a speaker */
+ unsigned int out2_speaker:1;
+
+ /* Configure MONOIN+/- in differential mode */
+ unsigned int monoin_diff:1;
+};
+
+#endif
diff --git a/kernel/include/sound/wm8960.h b/kernel/include/sound/wm8960.h
new file mode 100644
index 000000000..e8ce8ee7d
--- /dev/null
+++ b/kernel/include/sound/wm8960.h
@@ -0,0 +1,24 @@
+/*
+ * wm8960.h -- WM8960 Soc Audio driver platform data
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8960_PDATA_H
+#define _WM8960_PDATA_H
+
+#define WM8960_DRES_400R 0
+#define WM8960_DRES_200R 1
+#define WM8960_DRES_600R 2
+#define WM8960_DRES_150R 3
+#define WM8960_DRES_MAX 3
+
+struct wm8960_data {
+ bool capless; /* Headphone outputs configured in capless mode */
+
+ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */
+};
+
+#endif
diff --git a/kernel/include/sound/wm8962.h b/kernel/include/sound/wm8962.h
new file mode 100644
index 000000000..0af7c1674
--- /dev/null
+++ b/kernel/include/sound/wm8962.h
@@ -0,0 +1,61 @@
+/*
+ * wm8962.h -- WM8962 Soc Audio driver platform data
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8962_PDATA_H
+#define _WM8962_PDATA_H
+
+#define WM8962_MAX_GPIO 6
+
+/* Use to set GPIO default values to zero */
+#define WM8962_GPIO_SET 0x10000
+
+#define WM8962_GPIO_FN_CLKOUT 0
+#define WM8962_GPIO_FN_LOGIC 1
+#define WM8962_GPIO_FN_SDOUT 2
+#define WM8962_GPIO_FN_IRQ 3
+#define WM8962_GPIO_FN_THERMAL 4
+#define WM8962_GPIO_FN_PLL2_LOCK 6
+#define WM8962_GPIO_FN_PLL3_LOCK 7
+#define WM8962_GPIO_FN_FLL_LOCK 9
+#define WM8962_GPIO_FN_DRC_ACT 10
+#define WM8962_GPIO_FN_WSEQ_DONE 11
+#define WM8962_GPIO_FN_ALC_NG_ACT 12
+#define WM8962_GPIO_FN_ALC_PEAK_LIMIT 13
+#define WM8962_GPIO_FN_ALC_SATURATION 14
+#define WM8962_GPIO_FN_ALC_LEVEL_THR 15
+#define WM8962_GPIO_FN_ALC_LEVEL_LOCK 16
+#define WM8962_GPIO_FN_FIFO_ERR 17
+#define WM8962_GPIO_FN_OPCLK 18
+#define WM8962_GPIO_FN_DMICCLK 19
+#define WM8962_GPIO_FN_DMICDAT 20
+#define WM8962_GPIO_FN_MICD 21
+#define WM8962_GPIO_FN_MICSCD 22
+
+struct wm8962_pdata {
+ struct clk *mclk;
+ int gpio_base;
+ u32 gpio_init[WM8962_MAX_GPIO];
+
+ /* Setup for microphone detection, raw value to be written to
+ * R48(0x30) - only microphone related bits will be updated.
+ * Detection may be enabled here for use with signals brought
+ * out on the GPIOs. */
+ u32 mic_cfg;
+
+ bool irq_active_low;
+
+ bool spk_mono; /* Speaker outputs tied together as mono */
+
+ /**
+ * This flag should be set if one or both IN4 inputs is wired
+ * in a DC measurement configuration.
+ */
+ bool in4_dc_measure;
+};
+
+#endif
diff --git a/kernel/include/sound/wm8993.h b/kernel/include/sound/wm8993.h
new file mode 100644
index 000000000..8016fd826
--- /dev/null
+++ b/kernel/include/sound/wm8993.h
@@ -0,0 +1,48 @@
+/*
+ * linux/sound/wm8993.h -- Platform data for WM8993
+ *
+ * Copyright 2009 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM8993_H
+#define __LINUX_SND_WM8993_H
+
+/* Note that EQ1 only contains the enable/disable bit so will be
+ ignored but is included for simplicity.
+ */
+struct wm8993_retune_mobile_setting {
+ const char *name;
+ unsigned int rate;
+ u16 config[24];
+};
+
+struct wm8993_platform_data {
+ struct wm8993_retune_mobile_setting *retune_configs;
+ int num_retune_configs;
+
+ /* LINEOUT can be differential or single ended */
+ unsigned int lineout1_diff:1;
+ unsigned int lineout2_diff:1;
+
+ /* Common mode feedback */
+ unsigned int lineout1fb:1;
+ unsigned int lineout2fb:1;
+
+ /* Delay to add for microphones to stabalise after power up */
+ int micbias1_delay;
+ int micbias2_delay;
+
+ /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
+ unsigned int micbias1_lvl:1;
+ unsigned int micbias2_lvl:1;
+
+ /* Jack detect threshold levels, see datasheet for values */
+ unsigned int jd_scthr:2;
+ unsigned int jd_thr:2;
+};
+
+#endif
diff --git a/kernel/include/sound/wm8996.h b/kernel/include/sound/wm8996.h
new file mode 100644
index 000000000..ea4d88f43
--- /dev/null
+++ b/kernel/include/sound/wm8996.h
@@ -0,0 +1,55 @@
+/*
+ * linux/sound/wm8996.h -- Platform data for WM8996
+ *
+ * Copyright 2011 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM8996_H
+#define __LINUX_SND_WM8996_H
+
+enum wm8996_inmode {
+ WM8996_DIFFERRENTIAL_1 = 0, /* IN1xP - IN1xN */
+ WM8996_INVERTING = 1, /* IN1xN */
+ WM8996_NON_INVERTING = 2, /* IN1xP */
+ WM8996_DIFFERENTIAL_2 = 3, /* IN2xP - IN2xP */
+};
+
+/**
+ * ReTune Mobile configurations are specified with a label, sample
+ * rate and set of values to write (the enable bits will be ignored).
+ *
+ * Configurations are expected to be generated using the ReTune Mobile
+ * control panel in WISCE - see http://www.wolfsonmicro.com/wisce/
+ */
+struct wm8996_retune_mobile_config {
+ const char *name;
+ int rate;
+ u16 regs[20];
+};
+
+#define WM8996_SET_DEFAULT 0x10000
+
+struct wm8996_pdata {
+ int irq_flags; /** Set IRQ trigger flags; default active low */
+
+ int ldo_ena; /** GPIO for LDO1; -1 for none */
+
+ int micdet_def; /** Default MICDET_SRC/HP1FB_SRC/MICD_BIAS */
+
+ enum wm8996_inmode inl_mode;
+ enum wm8996_inmode inr_mode;
+
+ u32 spkmute_seq; /** Value for register 0x802 */
+
+ int gpio_base;
+ u32 gpio_default[5];
+
+ int num_retune_mobile_cfgs;
+ struct wm8996_retune_mobile_config *retune_mobile_cfgs;
+};
+
+#endif
diff --git a/kernel/include/sound/wm9081.h b/kernel/include/sound/wm9081.h
new file mode 100644
index 000000000..f34b0b171
--- /dev/null
+++ b/kernel/include/sound/wm9081.h
@@ -0,0 +1,28 @@
+/*
+ * linux/sound/wm9081.h -- Platform data for WM9081
+ *
+ * Copyright 2009 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM_9081_H
+#define __LINUX_SND_WM_9081_H
+
+struct wm9081_retune_mobile_setting {
+ const char *name;
+ unsigned int rate;
+ u16 config[20];
+};
+
+struct wm9081_pdata {
+ bool irq_high; /* IRQ is active high */
+ bool irq_cmos; /* IRQ is in CMOS mode */
+
+ struct wm9081_retune_mobile_setting *retune_configs;
+ int num_retune_configs;
+};
+
+#endif
diff --git a/kernel/include/sound/wm9090.h b/kernel/include/sound/wm9090.h
new file mode 100644
index 000000000..3718928cd
--- /dev/null
+++ b/kernel/include/sound/wm9090.h
@@ -0,0 +1,28 @@
+/*
+ * linux/sound/wm9090.h -- Platform data for WM9090
+ *
+ * Copyright 2009, 2010 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM9090_H
+#define __LINUX_SND_WM9090_H
+
+struct wm9090_platform_data {
+ /* Line inputs 1 & 2 can optionally be differential */
+ unsigned int lin1_diff:1;
+ unsigned int lin2_diff:1;
+
+ /* AGC configuration. This is intended to protect the speaker
+ * against overdriving and will therefore depend on the
+ * hardware setup with incorrect runtime configuration
+ * potentially causing hardware damage.
+ */
+ unsigned int agc_ena:1;
+ u16 agc[3];
+};
+
+#endif
diff --git a/kernel/include/sound/wss.h b/kernel/include/sound/wss.h
new file mode 100644
index 000000000..1823e3a96
--- /dev/null
+++ b/kernel/include/sound/wss.h
@@ -0,0 +1,235 @@
+#ifndef __SOUND_WSS_H
+#define __SOUND_WSS_H
+
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Definitions for CS4231 & InterWave chips & compatible chips
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/timer.h>
+
+#include <sound/cs4231-regs.h>
+
+/* defines for codec.mode */
+
+#define WSS_MODE_NONE 0x0000
+#define WSS_MODE_PLAY 0x0001
+#define WSS_MODE_RECORD 0x0002
+#define WSS_MODE_TIMER 0x0004
+#define WSS_MODE_OPEN (WSS_MODE_PLAY|WSS_MODE_RECORD|WSS_MODE_TIMER)
+
+/* defines for codec.hardware */
+
+#define WSS_HW_DETECT 0x0000 /* let CS4231 driver detect chip */
+#define WSS_HW_DETECT3 0x0001 /* allow mode 3 */
+#define WSS_HW_TYPE_MASK 0xff00 /* type mask */
+#define WSS_HW_CS4231_MASK 0x0100 /* CS4231 serie */
+#define WSS_HW_CS4231 0x0100 /* CS4231 chip */
+#define WSS_HW_CS4231A 0x0101 /* CS4231A chip */
+#define WSS_HW_AD1845 0x0102 /* AD1845 chip */
+#define WSS_HW_CS4232_MASK 0x0200 /* CS4232 serie (has control ports) */
+#define WSS_HW_CS4232 0x0200 /* CS4232 */
+#define WSS_HW_CS4232A 0x0201 /* CS4232A */
+#define WSS_HW_CS4236 0x0202 /* CS4236 */
+#define WSS_HW_CS4236B_MASK 0x0400 /* CS4236B serie (has extended control regs) */
+#define WSS_HW_CS4235 0x0400 /* CS4235 - Crystal Clear (tm) stereo enhancement */
+#define WSS_HW_CS4236B 0x0401 /* CS4236B */
+#define WSS_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */
+#define WSS_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */
+#define WSS_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */
+#define WSS_HW_AD1848_MASK 0x0800 /* AD1848 serie (half duplex) */
+#define WSS_HW_AD1847 0x0801 /* AD1847 chip */
+#define WSS_HW_AD1848 0x0802 /* AD1848 chip */
+#define WSS_HW_CS4248 0x0803 /* CS4248 chip */
+#define WSS_HW_CMI8330 0x0804 /* CMI8330 chip */
+#define WSS_HW_THINKPAD 0x0805 /* Thinkpad 360/750/755 */
+/* compatible, but clones */
+#define WSS_HW_INTERWAVE 0x1000 /* InterWave chip */
+#define WSS_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */
+#define WSS_HW_OPTI93X 0x1102 /* Opti 930/931/933 */
+
+/* defines for codec.hwshare */
+#define WSS_HWSHARE_IRQ (1<<0)
+#define WSS_HWSHARE_DMA1 (1<<1)
+#define WSS_HWSHARE_DMA2 (1<<2)
+
+/* IBM Thinkpad specific stuff */
+#define AD1848_THINKPAD_CTL_PORT1 0x15e8
+#define AD1848_THINKPAD_CTL_PORT2 0x15e9
+#define AD1848_THINKPAD_CS4248_ENABLE_BIT 0x02
+
+struct snd_wss {
+ unsigned long port; /* base i/o port */
+ struct resource *res_port;
+ unsigned long cport; /* control base i/o port (CS4236) */
+ struct resource *res_cport;
+ int irq; /* IRQ line */
+ int dma1; /* playback DMA */
+ int dma2; /* record DMA */
+ unsigned short version; /* version of CODEC chip */
+ unsigned short mode; /* see to WSS_MODE_XXXX */
+ unsigned short hardware; /* see to WSS_HW_XXXX */
+ unsigned short hwshare; /* shared resources */
+ unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit */
+ /* daughter board) or dma1 == dma2 */
+ ebus_flag:1, /* SPARC: EBUS present */
+ thinkpad_flag:1; /* Thinkpad CS4248 needs extra help */
+
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+ struct snd_timer *timer;
+
+ unsigned char image[32]; /* registers image */
+ unsigned char eimage[32]; /* extended registers image */
+ unsigned char cimage[16]; /* control registers image */
+ int mce_bit;
+ int calibrate_mute;
+ int sw_3d_bit;
+ unsigned int p_dma_size;
+ unsigned int c_dma_size;
+
+ spinlock_t reg_lock;
+ struct mutex mce_mutex;
+ struct mutex open_mutex;
+
+ int (*rate_constraint) (struct snd_pcm_runtime *runtime);
+ void (*set_playback_format) (struct snd_wss *chip,
+ struct snd_pcm_hw_params *hw_params,
+ unsigned char pdfr);
+ void (*set_capture_format) (struct snd_wss *chip,
+ struct snd_pcm_hw_params *hw_params,
+ unsigned char cdfr);
+ void (*trigger) (struct snd_wss *chip, unsigned int what, int start);
+#ifdef CONFIG_PM
+ void (*suspend) (struct snd_wss *chip);
+ void (*resume) (struct snd_wss *chip);
+#endif
+ void *dma_private_data;
+ int (*claim_dma) (struct snd_wss *chip,
+ void *dma_private_data, int dma);
+ int (*release_dma) (struct snd_wss *chip,
+ void *dma_private_data, int dma);
+};
+
+/* exported functions */
+
+void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char val);
+unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg);
+void snd_cs4236_ext_out(struct snd_wss *chip,
+ unsigned char reg, unsigned char val);
+unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg);
+void snd_wss_mce_up(struct snd_wss *chip);
+void snd_wss_mce_down(struct snd_wss *chip);
+
+void snd_wss_overrange(struct snd_wss *chip);
+
+irqreturn_t snd_wss_interrupt(int irq, void *dev_id);
+
+const char *snd_wss_chip_id(struct snd_wss *chip);
+
+int snd_wss_create(struct snd_card *card,
+ unsigned long port,
+ unsigned long cport,
+ int irq, int dma1, int dma2,
+ unsigned short hardware,
+ unsigned short hwshare,
+ struct snd_wss **rchip);
+int snd_wss_pcm(struct snd_wss *chip, int device);
+int snd_wss_timer(struct snd_wss *chip, int device);
+int snd_wss_mixer(struct snd_wss *chip);
+
+const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction);
+
+int snd_cs4236_create(struct snd_card *card,
+ unsigned long port,
+ unsigned long cport,
+ int irq, int dma1, int dma2,
+ unsigned short hardware,
+ unsigned short hwshare,
+ struct snd_wss **rchip);
+int snd_cs4236_pcm(struct snd_wss *chip, int device);
+int snd_cs4236_mixer(struct snd_wss *chip);
+
+/*
+ * mixer library
+ */
+
+#define WSS_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_wss_info_single, \
+ .get = snd_wss_get_single, \
+ .put = snd_wss_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+int snd_wss_info_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_wss_get_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_wss_put_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
+#define WSS_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_wss_info_double, \
+ .get = snd_wss_get_double, \
+ .put = snd_wss_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \
+ (shift_right << 19) | (mask << 24) | (invert << 22) }
+
+#define WSS_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_wss_info_single, \
+ .get = snd_wss_get_single, \
+ .put = snd_wss_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \
+ .tlv = { .p = (xtlv) } }
+
+#define WSS_DOUBLE_TLV(xname, xindex, left_reg, right_reg, \
+ shift_left, shift_right, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_wss_info_double, \
+ .get = snd_wss_get_double, \
+ .put = snd_wss_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \
+ (shift_right << 19) | (mask << 24) | (invert << 22), \
+ .tlv = { .p = (xtlv) } }
+
+
+int snd_wss_info_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_wss_get_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_wss_put_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
+#endif /* __SOUND_WSS_H */