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-rw-r--r--kernel/sound/soc/pxa/Kconfig219
-rw-r--r--kernel/sound/soc/pxa/Makefile53
-rw-r--r--kernel/sound/soc/pxa/brownstone.c168
-rw-r--r--kernel/sound/soc/pxa/corgi.c328
-rw-r--r--kernel/sound/soc/pxa/e740_wm9705.c174
-rw-r--r--kernel/sound/soc/pxa/e750_wm9705.c156
-rw-r--r--kernel/sound/soc/pxa/e800_wm9712.c155
-rw-r--r--kernel/sound/soc/pxa/em-x270.c96
-rw-r--r--kernel/sound/soc/pxa/hx4700.c229
-rw-r--r--kernel/sound/soc/pxa/imote2.c104
-rw-r--r--kernel/sound/soc/pxa/magician.c534
-rw-r--r--kernel/sound/soc/pxa/mioa701_wm9713.c214
-rw-r--r--kernel/sound/soc/pxa/mmp-pcm.c257
-rw-r--r--kernel/sound/soc/pxa/mmp-sspa.c484
-rw-r--r--kernel/sound/soc/pxa/mmp-sspa.h92
-rw-r--r--kernel/sound/soc/pxa/palm27x.c170
-rw-r--r--kernel/sound/soc/pxa/poodle.c311
-rw-r--r--kernel/sound/soc/pxa/pxa-ssp.c841
-rw-r--r--kernel/sound/soc/pxa/pxa-ssp.h45
-rw-r--r--kernel/sound/soc/pxa/pxa2xx-ac97.c274
-rw-r--r--kernel/sound/soc/pxa/pxa2xx-ac97.h17
-rw-r--r--kernel/sound/soc/pxa/pxa2xx-i2s.c407
-rw-r--r--kernel/sound/soc/pxa/pxa2xx-i2s.h18
-rw-r--r--kernel/sound/soc/pxa/pxa2xx-pcm.c157
-rw-r--r--kernel/sound/soc/pxa/raumfeld.c320
-rw-r--r--kernel/sound/soc/pxa/spitz.c346
-rw-r--r--kernel/sound/soc/pxa/tosa.c280
-rw-r--r--kernel/sound/soc/pxa/ttc-dkb.c163
-rw-r--r--kernel/sound/soc/pxa/z2.c227
-rw-r--r--kernel/sound/soc/pxa/zylonite.c275
30 files changed, 7114 insertions, 0 deletions
diff --git a/kernel/sound/soc/pxa/Kconfig b/kernel/sound/soc/pxa/Kconfig
new file mode 100644
index 000000000..39cea8084
--- /dev/null
+++ b/kernel/sound/soc/pxa/Kconfig
@@ -0,0 +1,219 @@
+config SND_PXA2XX_SOC
+ tristate "SoC Audio for the Intel PXA2xx chip"
+ depends on ARCH_PXA
+ select SND_ARM
+ select SND_PXA2XX_LIB
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the PXA2xx AC97, I2S or SSP interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_MMP_SOC
+ bool "Soc Audio for Marvell MMP chips"
+ depends on ARCH_MMP
+ select MMP_SRAM
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ select SND_ARM
+ help
+ Say Y if you want to add support for codecs attached to
+ the MMP SSPA interface.
+
+config SND_PXA2XX_AC97
+ tristate
+ select SND_AC97_CODEC
+
+config SND_PXA2XX_SOC_AC97
+ tristate
+ select AC97_BUS
+ select SND_ARM
+ select SND_PXA2XX_LIB_AC97
+ select SND_SOC_AC97_BUS
+
+config SND_PXA2XX_SOC_I2S
+ tristate
+
+config SND_PXA_SOC_SSP
+ tristate
+ select PXA_SSP
+
+config SND_MMP_SOC_SSPA
+ tristate
+
+config SND_PXA2XX_SOC_CORGI
+ tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
+ depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-C7x0 models (Corgi, Shepherd, Husky).
+
+config SND_PXA2XX_SOC_SPITZ
+ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00"
+ depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8750
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita).
+
+config SND_PXA2XX_SOC_Z2
+ tristate "SoC Audio support for Zipit Z2"
+ depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8750
+ help
+ Say Y if you want to add support for SoC audio on Zipit Z2.
+
+config SND_PXA2XX_SOC_POODLE
+ tristate "SoC Audio support for Poodle"
+ depends on SND_PXA2XX_SOC && MACH_POODLE && I2C
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-5600 model (Poodle).
+
+config SND_PXA2XX_SOC_TOSA
+ tristate "SoC AC97 Audio support for Tosa"
+ depends on SND_PXA2XX_SOC && MACH_TOSA
+ depends on MFD_TC6393XB
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-C6000x models (Tosa).
+
+config SND_PXA2XX_SOC_E740
+ tristate "SoC AC97 Audio support for e740"
+ depends on SND_PXA2XX_SOC && MACH_E740
+ select SND_SOC_WM9705
+ select SND_PXA2XX_SOC_AC97
+ help
+ Say Y if you want to add support for SoC audio on the
+ toshiba e740 PDA
+
+config SND_PXA2XX_SOC_E750
+ tristate "SoC AC97 Audio support for e750"
+ depends on SND_PXA2XX_SOC && MACH_E750
+ select SND_SOC_WM9705
+ select SND_PXA2XX_SOC_AC97
+ help
+ Say Y if you want to add support for SoC audio on the
+ toshiba e750 PDA
+
+config SND_PXA2XX_SOC_E800
+ tristate "SoC AC97 Audio support for e800"
+ depends on SND_PXA2XX_SOC && MACH_E800
+ select SND_SOC_WM9712
+ select SND_PXA2XX_SOC_AC97
+ help
+ Say Y if you want to add support for SoC audio on the
+ Toshiba e800 PDA
+
+config SND_PXA2XX_SOC_EM_X270
+ tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
+ depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
+ MACH_CM_X300)
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for SoC audio on
+ CompuLab EM-x270, eXeda and CM-X300 machines.
+
+config SND_PXA2XX_SOC_PALM27X
+ bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive"
+ depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
+ MACH_PALMT5 || MACH_PALMTE2)
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for SoC audio on
+ Palm T|X, T5, E2 or LifeDrive handheld computer.
+
+config SND_PXA910_SOC
+ tristate "SoC Audio for Marvell PXA910 chip"
+ depends on ARCH_MMP && SND
+ select SND_PCM
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell PXA910 reference platform.
+
+config SND_SOC_TTC_DKB
+ tristate "SoC Audio support for TTC DKB"
+ depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y
+ select PXA_SSP
+ select SND_PXA_SOC_SSP
+ select SND_MMP_SOC
+ select MFD_88PM860X
+ select SND_SOC_88PM860X
+ help
+ Say Y if you want to add support for SoC audio on TTC DKB
+
+
+config SND_SOC_ZYLONITE
+ tristate "SoC Audio support for Marvell Zylonite"
+ depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+ select SND_PXA2XX_SOC_AC97
+ select SND_PXA_SOC_SSP
+ select SND_SOC_WM9713
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell Zylonite reference platform.
+
+config SND_SOC_RAUMFELD
+ tristate "SoC Audio support Raumfeld audio adapter"
+ depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR)
+ depends on I2C && SPI_MASTER
+ select SND_PXA_SOC_SSP
+ select SND_SOC_CS4270
+ select SND_SOC_AK4104
+ help
+ Say Y if you want to add support for SoC audio on Raumfeld devices
+
+config SND_PXA2XX_SOC_HX4700
+ tristate "SoC Audio support for HP iPAQ hx4700"
+ depends on SND_PXA2XX_SOC && MACH_H4700 && I2C
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_AK4641
+ help
+ Say Y if you want to add support for SoC audio on the
+ HP iPAQ hx4700.
+
+config SND_PXA2XX_SOC_MAGICIAN
+ tristate "SoC Audio support for HTC Magician"
+ depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C
+ select SND_PXA2XX_SOC_I2S
+ select SND_PXA_SOC_SSP
+ select SND_SOC_UDA1380
+ help
+ Say Y if you want to add support for SoC audio on the
+ HTC Magician.
+
+config SND_PXA2XX_SOC_MIOA701
+ tristate "SoC Audio support for MIO A701"
+ depends on SND_PXA2XX_SOC && MACH_MIOA701
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9713
+ help
+ Say Y if you want to add support for SoC audio on the
+ MIO A701.
+
+config SND_PXA2XX_SOC_IMOTE2
+ tristate "SoC Audio support for IMote 2"
+ depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8940
+ help
+ Say Y if you want to add support for SoC audio on the
+ IMote 2.
+
+config SND_MMP_SOC_BROWNSTONE
+ tristate "SoC Audio support for Marvell Brownstone"
+ depends on SND_MMP_SOC && MACH_BROWNSTONE
+ select SND_MMP_SOC_SSPA
+ select MFD_WM8994
+ select SND_SOC_WM8994
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell Brownstone reference platform.
diff --git a/kernel/sound/soc/pxa/Makefile b/kernel/sound/soc/pxa/Makefile
new file mode 100644
index 000000000..2cff67b61
--- /dev/null
+++ b/kernel/sound/soc/pxa/Makefile
@@ -0,0 +1,53 @@
+# PXA Platform Support
+snd-soc-pxa2xx-objs := pxa2xx-pcm.o
+snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
+snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+snd-soc-pxa-ssp-objs := pxa-ssp.o
+snd-soc-mmp-objs := mmp-pcm.o
+snd-soc-mmp-sspa-objs := mmp-sspa.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
+obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
+obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
+obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
+obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
+
+# PXA Machine Support
+snd-soc-corgi-objs := corgi.o
+snd-soc-poodle-objs := poodle.o
+snd-soc-tosa-objs := tosa.o
+snd-soc-e740-objs := e740_wm9705.o
+snd-soc-e750-objs := e750_wm9705.o
+snd-soc-e800-objs := e800_wm9712.o
+snd-soc-spitz-objs := spitz.o
+snd-soc-em-x270-objs := em-x270.o
+snd-soc-palm27x-objs := palm27x.o
+snd-soc-zylonite-objs := zylonite.o
+snd-soc-hx4700-objs := hx4700.o
+snd-soc-magician-objs := magician.o
+snd-soc-mioa701-objs := mioa701_wm9713.o
+snd-soc-z2-objs := z2.o
+snd-soc-imote2-objs := imote2.o
+snd-soc-raumfeld-objs := raumfeld.o
+snd-soc-brownstone-objs := brownstone.o
+snd-soc-ttc-dkb-objs := ttc-dkb.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
+obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
+obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
+obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
+obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
+obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
+obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
+obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
+obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
+obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
+obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
+obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o
diff --git a/kernel/sound/soc/pxa/brownstone.c b/kernel/sound/soc/pxa/brownstone.c
new file mode 100644
index 000000000..79936e3e8
--- /dev/null
+++ b/kernel/sound/soc/pxa/brownstone.c
@@ -0,0 +1,168 @@
+/*
+ * linux/sound/soc/pxa/brownstone.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8994.h"
+#include "mmp-sspa.h"
+
+static const struct snd_kcontrol_new brownstone_dapm_control[] = {
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Main Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route brownstone_audio_map[] = {
+ {"Ext Spk", NULL, "SPKOUTLP"},
+ {"Ext Spk", NULL, "SPKOUTLN"},
+ {"Ext Spk", NULL, "SPKOUTRP"},
+ {"Ext Spk", NULL, "SPKOUTRN"},
+
+ {"Headset Stereophone", NULL, "HPOUT1L"},
+ {"Headset Stereophone", NULL, "HPOUT1R"},
+
+ {"IN1RN", NULL, "Headset Mic"},
+
+ {"DMIC1DAT", NULL, "MICBIAS1"},
+ {"MICBIAS1", NULL, "Main Mic"},
+};
+
+static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ /* set endpoints to not connected */
+ snd_soc_dapm_nc_pin(dapm, "HPOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "HPOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "IN1LN");
+ snd_soc_dapm_nc_pin(dapm, "IN1LP");
+ snd_soc_dapm_nc_pin(dapm, "IN1RP");
+ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+ snd_soc_dapm_nc_pin(dapm, "IN2LN");
+
+ return 0;
+}
+
+static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int freq_out, sspa_mclk, sysclk;
+ int sspa_div;
+
+ if (params_rate(params) > 11025) {
+ freq_out = params_rate(params) * 512;
+ sysclk = params_rate(params) * 256;
+ sspa_mclk = params_rate(params) * 64;
+ } else {
+ freq_out = params_rate(params) * 1024;
+ sysclk = params_rate(params) * 512;
+ sspa_mclk = params_rate(params) * 64;
+ }
+ sspa_div = freq_out;
+ do_div(sspa_div, sspa_mclk);
+
+ snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0);
+ snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);
+ snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk);
+
+ /* set wm8994 sysclk */
+ snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0);
+
+ return 0;
+}
+
+/* machine stream operations */
+static struct snd_soc_ops brownstone_ops = {
+ .hw_params = brownstone_wm8994_hw_params,
+};
+
+static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
+{
+ .name = "WM8994",
+ .stream_name = "WM8994 HiFi",
+ .cpu_dai_name = "mmp-sspa-dai.0",
+ .codec_dai_name = "wm8994-aif1",
+ .platform_name = "mmp-pcm-audio",
+ .codec_name = "wm8994-codec",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &brownstone_ops,
+ .init = brownstone_wm8994_init,
+},
+};
+
+/* audio machine driver */
+static struct snd_soc_card brownstone = {
+ .name = "brownstone",
+ .owner = THIS_MODULE,
+ .dai_link = brownstone_wm8994_dai,
+ .num_links = ARRAY_SIZE(brownstone_wm8994_dai),
+
+ .controls = brownstone_dapm_control,
+ .num_controls = ARRAY_SIZE(brownstone_dapm_control),
+ .dapm_widgets = brownstone_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
+ .dapm_routes = brownstone_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
+};
+
+static int brownstone_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ brownstone.dev = &pdev->dev;
+ ret = snd_soc_register_card(&brownstone);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+}
+
+static int brownstone_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&brownstone);
+ return 0;
+}
+
+static struct platform_driver mmp_driver = {
+ .driver = {
+ .name = "brownstone-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = brownstone_probe,
+ .remove = brownstone_remove,
+};
+
+module_platform_driver(mmp_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC Brownstone");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/corgi.c b/kernel/sound/soc/pxa/corgi.c
new file mode 100644
index 000000000..3580d10c9
--- /dev/null
+++ b/kernel/sound/soc/pxa/corgi.c
@@ -0,0 +1,328 @@
+/*
+ * corgi.c -- SoC audio for Corgi
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/i2c.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/corgi.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm8731.h"
+#include "pxa2xx-i2s.h"
+
+#define CORGI_HP 0
+#define CORGI_MIC 1
+#define CORGI_LINE 2
+#define CORGI_HEADSET 3
+#define CORGI_HP_OFF 4
+#define CORGI_SPK_ON 0
+#define CORGI_SPK_OFF 1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define CORGI_AUDIO_CLOCK 12288000
+
+static int corgi_jack_func;
+static int corgi_spk_func;
+
+static void corgi_ext_control(struct snd_soc_dapm_context *dapm)
+{
+ snd_soc_dapm_mutex_lock(dapm);
+
+ /* set up jack connection */
+ switch (corgi_jack_func) {
+ case CORGI_HP:
+ /* set = unmute headphone */
+ gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ break;
+ case CORGI_MIC:
+ /* reset = mute headphone */
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 0);
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ break;
+ case CORGI_LINE:
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 0);
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ break;
+ case CORGI_HEADSET:
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
+ break;
+ }
+
+ if (corgi_spk_func == CORGI_SPK_ON)
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int corgi_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* check the jack status at stream startup */
+ corgi_ext_control(&rtd->card->dapm);
+
+ return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on corgi */
+static void corgi_shutdown(struct snd_pcm_substream *substream)
+{
+ /* set = unmute headphone */
+ gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+}
+
+static int corgi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops corgi_ops = {
+ .startup = corgi_startup,
+ .hw_params = corgi_hw_params,
+ .shutdown = corgi_shutdown,
+};
+
+static int corgi_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = corgi_jack_func;
+ return 0;
+}
+
+static int corgi_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (corgi_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ corgi_jack_func = ucontrol->value.integer.value[0];
+ corgi_ext_control(&card->dapm);
+ return 1;
+}
+
+static int corgi_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = corgi_spk_func;
+ return 0;
+}
+
+static int corgi_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (corgi_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ corgi_spk_func = ucontrol->value.integer.value[0];
+ corgi_ext_control(&card->dapm);
+ return 1;
+}
+
+static int corgi_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int corgi_mic_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event),
+SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event),
+SND_SOC_DAPM_LINE("Line Jack", NULL),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Corgi machine audio map (connections to the codec pins) */
+static const struct snd_soc_dapm_route corgi_audio_map[] = {
+
+ /* headset Jack - in = micin, out = LHPOUT*/
+ {"Headset Jack", NULL, "LHPOUT"},
+
+ /* headphone connected to LHPOUT1, RHPOUT1 */
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ /* speaker connected to LOUT, ROUT */
+ {"Ext Spk", NULL, "ROUT"},
+ {"Ext Spk", NULL, "LOUT"},
+
+ /* mic is connected to MICIN (via right channel of headphone jack) */
+ {"MICIN", NULL, "Mic Jack"},
+
+ /* Same as the above but no mic bias for line signals */
+ {"MICIN", NULL, "Line Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum corgi_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
+ SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack,
+ corgi_set_jack),
+ SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk,
+ corgi_set_spk),
+};
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731",
+ .cpu_dai_name = "pxa2xx-i2s",
+ .codec_dai_name = "wm8731-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm8731.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &corgi_ops,
+};
+
+/* corgi audio machine driver */
+static struct snd_soc_card corgi = {
+ .name = "Corgi",
+ .owner = THIS_MODULE,
+ .dai_link = &corgi_dai,
+ .num_links = 1,
+
+ .controls = wm8731_corgi_controls,
+ .num_controls = ARRAY_SIZE(wm8731_corgi_controls),
+ .dapm_widgets = wm8731_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets),
+ .dapm_routes = corgi_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(corgi_audio_map),
+ .fully_routed = true,
+};
+
+static int corgi_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &corgi;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+}
+
+static int corgi_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver corgi_driver = {
+ .driver = {
+ .name = "corgi-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = corgi_probe,
+ .remove = corgi_remove,
+};
+
+module_platform_driver(corgi_driver);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Corgi");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:corgi-audio");
diff --git a/kernel/sound/soc/pxa/e740_wm9705.c b/kernel/sound/soc/pxa/e740_wm9705.c
new file mode 100644
index 000000000..d72e124a3
--- /dev/null
+++ b/kernel/sound/soc/pxa/e740_wm9705.c
@@ -0,0 +1,174 @@
+/*
+ * e740-wm9705.c -- SoC audio for e740
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.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 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-ac97.h"
+
+
+#define E740_AUDIO_OUT 1
+#define E740_AUDIO_IN 2
+
+static int e740_audio_power;
+
+static void e740_sync_audio_power(int status)
+{
+ gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status);
+ gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0);
+ gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0);
+}
+
+static int e740_mic_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ e740_audio_power |= E740_AUDIO_IN;
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ e740_audio_power &= ~E740_AUDIO_IN;
+
+ e740_sync_audio_power(e740_audio_power);
+
+ return 0;
+}
+
+static int e740_output_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ e740_audio_power |= E740_AUDIO_OUT;
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ e740_audio_power &= ~E740_AUDIO_OUT;
+
+ e740_sync_audio_power(e740_audio_power);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e740_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+ SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e740_output_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Output Amp", NULL, "LOUT"},
+ {"Output Amp", NULL, "ROUT"},
+ {"Output Amp", NULL, "MONOOUT"},
+
+ {"Speaker", NULL, "Output Amp"},
+ {"Headphone Jack", NULL, "Output Amp"},
+
+ {"MIC1", NULL, "Mic Amp"},
+ {"Mic Amp", NULL, "Mic (Internal)"},
+};
+
+static struct snd_soc_dai_link e740_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai_name = "pxa2xx-ac97",
+ .codec_dai_name = "wm9705-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9705-codec",
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
+ .codec_dai_name = "wm9705-aux",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9705-codec",
+ },
+};
+
+static struct snd_soc_card e740 = {
+ .name = "Toshiba e740",
+ .owner = THIS_MODULE,
+ .dai_link = e740_dai,
+ .num_links = ARRAY_SIZE(e740_dai),
+
+ .dapm_widgets = e740_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
+};
+
+static struct gpio e740_audio_gpios[] = {
+ { GPIO_E740_MIC_ON, GPIOF_OUT_INIT_LOW, "Mic amp" },
+ { GPIO_E740_AMP_ON, GPIOF_OUT_INIT_LOW, "Output amp" },
+ { GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, "Audio power" },
+};
+
+static int e740_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &e740;
+ int ret;
+
+ ret = gpio_request_array(e740_audio_gpios,
+ ARRAY_SIZE(e740_audio_gpios));
+ if (ret)
+ return ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios));
+ }
+ return ret;
+}
+
+static int e740_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios));
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver e740_driver = {
+ .driver = {
+ .name = "e740-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = e740_probe,
+ .remove = e740_remove,
+};
+
+module_platform_driver(e740_driver);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e740");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:e740-audio");
diff --git a/kernel/sound/soc/pxa/e750_wm9705.c b/kernel/sound/soc/pxa/e750_wm9705.c
new file mode 100644
index 000000000..48f2d7c2e
--- /dev/null
+++ b/kernel/sound/soc/pxa/e750_wm9705.c
@@ -0,0 +1,156 @@
+/*
+ * e750-wm9705.c -- SoC audio for e750
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.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 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-ac97.h"
+
+static int e750_spk_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1);
+
+ return 0;
+}
+
+static int e750_hp_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E750_HP_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E750_HP_AMP_OFF, 1);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e750_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+ SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Amp", NULL, "HPOUTL"},
+ {"Headphone Amp", NULL, "HPOUTR"},
+ {"Headphone Jack", NULL, "Headphone Amp"},
+
+ {"Speaker Amp", NULL, "MONOOUT"},
+ {"Speaker", NULL, "Speaker Amp"},
+
+ {"MIC1", NULL, "Mic (Internal)"},
+};
+
+static struct snd_soc_dai_link e750_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai_name = "pxa2xx-ac97",
+ .codec_dai_name = "wm9705-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9705-codec",
+ /* use ops to check startup state */
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
+ .codec_dai_name ="wm9705-aux",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9705-codec",
+ },
+};
+
+static struct snd_soc_card e750 = {
+ .name = "Toshiba e750",
+ .owner = THIS_MODULE,
+ .dai_link = e750_dai,
+ .num_links = ARRAY_SIZE(e750_dai),
+
+ .dapm_widgets = e750_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
+};
+
+static struct gpio e750_audio_gpios[] = {
+ { GPIO_E750_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Headphone amp" },
+ { GPIO_E750_SPK_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" },
+};
+
+static int e750_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &e750;
+ int ret;
+
+ ret = gpio_request_array(e750_audio_gpios,
+ ARRAY_SIZE(e750_audio_gpios));
+ if (ret)
+ return ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios));
+ }
+ return ret;
+}
+
+static int e750_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios));
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver e750_driver = {
+ .driver = {
+ .name = "e750-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = e750_probe,
+ .remove = e750_remove,
+};
+
+module_platform_driver(e750_driver);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e750");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:e750-audio");
diff --git a/kernel/sound/soc/pxa/e800_wm9712.c b/kernel/sound/soc/pxa/e800_wm9712.c
new file mode 100644
index 000000000..45d4bd46f
--- /dev/null
+++ b/kernel/sound/soc/pxa/e800_wm9712.c
@@ -0,0 +1,155 @@
+/*
+ * e800-wm9712.c -- SoC audio for e800
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.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 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-ac97.h"
+
+static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E800_SPK_AMP_ON, 1);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E800_SPK_AMP_ON, 0);
+
+ return 0;
+}
+
+static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E800_HP_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E800_HP_AMP_OFF, 1);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e800_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal1)", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal2)", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Jack", NULL, "HPOUTL"},
+ {"Headphone Jack", NULL, "HPOUTR"},
+ {"Headphone Jack", NULL, "Headphone Amp"},
+
+ {"Speaker Amp", NULL, "MONOOUT"},
+ {"Speaker", NULL, "Speaker Amp"},
+
+ {"MIC1", NULL, "Mic (Internal1)"},
+ {"MIC2", NULL, "Mic (Internal2)"},
+};
+
+static struct snd_soc_dai_link e800_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai_name = "pxa2xx-ac97",
+ .codec_dai_name = "wm9712-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9712-codec",
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
+ .codec_dai_name ="wm9712-aux",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9712-codec",
+ },
+};
+
+static struct snd_soc_card e800 = {
+ .name = "Toshiba e800",
+ .owner = THIS_MODULE,
+ .dai_link = e800_dai,
+ .num_links = ARRAY_SIZE(e800_dai),
+
+ .dapm_widgets = e800_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e800_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static struct gpio e800_audio_gpios[] = {
+ { GPIO_E800_SPK_AMP_ON, GPIOF_OUT_INIT_HIGH, "Headphone amp" },
+ { GPIO_E800_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" },
+};
+
+static int e800_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &e800;
+ int ret;
+
+ ret = gpio_request_array(e800_audio_gpios,
+ ARRAY_SIZE(e800_audio_gpios));
+ if (ret)
+ return ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios));
+ }
+ return ret;
+}
+
+static int e800_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios));
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver e800_driver = {
+ .driver = {
+ .name = "e800-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = e800_probe,
+ .remove = e800_remove,
+};
+
+module_platform_driver(e800_driver);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e800");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:e800-audio");
diff --git a/kernel/sound/soc/pxa/em-x270.c b/kernel/sound/soc/pxa/em-x270.c
new file mode 100644
index 000000000..64743a05a
--- /dev/null
+++ b/kernel/sound/soc/pxa/em-x270.c
@@ -0,0 +1,96 @@
+/*
+ * SoC audio driver for EM-X270, eXeda and CM-X300
+ *
+ * Copyright 2007, 2009 CompuLab, Ltd.
+ *
+ * Author: Mike Rapoport <mike@compulab.co.il>
+ *
+ * Copied from tosa.c:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_dai_link em_x270_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai_name = "pxa2xx-ac97",
+ .codec_dai_name = "wm9712-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9712-codec",
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
+ .codec_dai_name ="wm9712-aux",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9712-codec",
+ },
+};
+
+static struct snd_soc_card em_x270 = {
+ .name = "EM-X270",
+ .owner = THIS_MODULE,
+ .dai_link = em_x270_dai,
+ .num_links = ARRAY_SIZE(em_x270_dai),
+};
+
+static struct platform_device *em_x270_snd_device;
+
+static int __init em_x270_init(void)
+{
+ int ret;
+
+ if (!(machine_is_em_x270() || machine_is_exeda()
+ || machine_is_cm_x300()))
+ return -ENODEV;
+
+ em_x270_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!em_x270_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(em_x270_snd_device, &em_x270);
+ ret = platform_device_add(em_x270_snd_device);
+
+ if (ret)
+ platform_device_put(em_x270_snd_device);
+
+ return ret;
+}
+
+static void __exit em_x270_exit(void)
+{
+ platform_device_unregister(em_x270_snd_device);
+}
+
+module_init(em_x270_init);
+module_exit(em_x270_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mike Rapoport");
+MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/hx4700.c b/kernel/sound/soc/pxa/hx4700.c
new file mode 100644
index 000000000..9f8be7cd5
--- /dev/null
+++ b/kernel/sound/soc/pxa/hx4700.c
@@ -0,0 +1,229 @@
+/*
+ * SoC audio for HP iPAQ hx4700
+ *
+ * Copyright (c) 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 as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/hx4700.h>
+#include <asm/mach-types.h>
+#include "pxa2xx-i2s.h"
+
+#include "../codecs/ak4641.h"
+
+static struct snd_soc_jack hs_jack;
+
+/* Headphones jack detection DAPM pin */
+static struct snd_soc_jack_pin hs_jack_pin[] = {
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Speaker",
+ /* disable speaker when hp jack is inserted */
+ .mask = SND_JACK_HEADPHONE,
+ .invert = 1,
+ },
+};
+
+/* Headphones jack detection GPIO */
+static struct snd_soc_jack_gpio hs_jack_gpio = {
+ .gpio = GPIO75_HX4700_EARPHONE_nDET,
+ .invert = true,
+ .name = "hp-gpio",
+ .report = SND_JACK_HEADPHONE,
+ .debounce_time = 200,
+};
+
+/*
+ * iPAQ hx4700 uses I2S for capture and playback.
+ */
+static int hx4700_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+
+ /* set the I2S system clock as output */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ /* inform codec driver about clock freq *
+ * (PXA I2S always uses divider 256) */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params),
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops hx4700_ops = {
+ .hw_params = hx4700_hw_params,
+};
+
+static int hx4700_spk_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(GPIO107_HX4700_SPK_nSD, !!SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int hx4700_hp_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(GPIO92_HX4700_HP_DRIVER, !!SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* hx4700 machine dapm widgets */
+static const struct snd_soc_dapm_widget hx4700_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", hx4700_hp_power),
+ SND_SOC_DAPM_SPK("Speaker", hx4700_spk_power),
+ SND_SOC_DAPM_MIC("Built-in Microphone", NULL),
+};
+
+/* hx4700 machine audio_map */
+static const struct snd_soc_dapm_route hx4700_audio_map[] = {
+
+ /* Headphone connected to LOUT, ROUT */
+ {"Headphone Jack", NULL, "LOUT"},
+ {"Headphone Jack", NULL, "ROUT"},
+
+ /* Speaker connected to MOUT2 */
+ {"Speaker", NULL, "MOUT2"},
+
+ /* Microphone connected to MICIN */
+ {"MICIN", NULL, "Built-in Microphone"},
+ {"AIN", NULL, "MICOUT"},
+};
+
+/*
+ * Logic for a ak4641 as connected on a HP iPAQ hx4700
+ */
+static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int err;
+
+ /* Jack detection API stuff */
+ err = snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+ SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin,
+ ARRAY_SIZE(hs_jack_pin));
+ if (err)
+ return err;
+
+ err = snd_soc_jack_add_gpios(&hs_jack, 1, &hs_jack_gpio);
+
+ return err;
+}
+
+static int hx4700_card_remove(struct snd_soc_card *card)
+{
+ snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio);
+
+ return 0;
+}
+
+/* hx4700 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link hx4700_dai = {
+ .name = "ak4641",
+ .stream_name = "AK4641",
+ .cpu_dai_name = "pxa2xx-i2s",
+ .codec_dai_name = "ak4641-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "ak4641.0-0012",
+ .init = hx4700_ak4641_init,
+ .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &hx4700_ops,
+};
+
+/* hx4700 audio machine driver */
+static struct snd_soc_card snd_soc_card_hx4700 = {
+ .name = "iPAQ hx4700",
+ .owner = THIS_MODULE,
+ .remove = hx4700_card_remove,
+ .dai_link = &hx4700_dai,
+ .num_links = 1,
+ .dapm_widgets = hx4700_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(hx4700_dapm_widgets),
+ .dapm_routes = hx4700_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(hx4700_audio_map),
+ .fully_routed = true,
+};
+
+static struct gpio hx4700_audio_gpios[] = {
+ { GPIO107_HX4700_SPK_nSD, GPIOF_OUT_INIT_HIGH, "SPK_POWER" },
+ { GPIO92_HX4700_HP_DRIVER, GPIOF_OUT_INIT_LOW, "EP_POWER" },
+};
+
+static int hx4700_audio_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ if (!machine_is_h4700())
+ return -ENODEV;
+
+ ret = gpio_request_array(hx4700_audio_gpios,
+ ARRAY_SIZE(hx4700_audio_gpios));
+ if (ret)
+ return ret;
+
+ snd_soc_card_hx4700.dev = &pdev->dev;
+ ret = snd_soc_register_card(&snd_soc_card_hx4700);
+ if (ret)
+ gpio_free_array(hx4700_audio_gpios,
+ ARRAY_SIZE(hx4700_audio_gpios));
+
+ return ret;
+}
+
+static int hx4700_audio_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&snd_soc_card_hx4700);
+
+ gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
+ gpio_set_value(GPIO107_HX4700_SPK_nSD, 0);
+
+ gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios));
+ return 0;
+}
+
+static struct platform_driver hx4700_audio_driver = {
+ .driver = {
+ .name = "hx4700-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = hx4700_audio_probe,
+ .remove = hx4700_audio_remove,
+};
+
+module_platform_driver(hx4700_audio_driver);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:hx4700-audio");
diff --git a/kernel/sound/soc/pxa/imote2.c b/kernel/sound/soc/pxa/imote2.c
new file mode 100644
index 000000000..29fabbfd2
--- /dev/null
+++ b/kernel/sound/soc/pxa/imote2.c
@@ -0,0 +1,104 @@
+
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm8940.h"
+#include "pxa2xx-i2s.h"
+
+static int imote2_asoc_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int clk = 0;
+ int ret;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk,
+ SND_SOC_CLOCK_OUT);
+
+ return ret;
+}
+
+static struct snd_soc_ops imote2_asoc_ops = {
+ .hw_params = imote2_asoc_hw_params,
+};
+
+static struct snd_soc_dai_link imote2_dai = {
+ .name = "WM8940",
+ .stream_name = "WM8940",
+ .cpu_dai_name = "pxa2xx-i2s",
+ .codec_dai_name = "wm8940-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm8940-codec.0-0034",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &imote2_asoc_ops,
+};
+
+static struct snd_soc_card imote2 = {
+ .name = "Imote2",
+ .owner = THIS_MODULE,
+ .dai_link = &imote2_dai,
+ .num_links = 1,
+};
+
+static int imote2_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &imote2;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+}
+
+static int imote2_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver imote2_driver = {
+ .driver = {
+ .name = "imote2-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = imote2_probe,
+ .remove = imote2_remove,
+};
+
+module_platform_driver(imote2_driver);
+
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_DESCRIPTION("ALSA SoC Imote 2");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imote2-audio");
diff --git a/kernel/sound/soc/pxa/magician.c b/kernel/sound/soc/pxa/magician.c
new file mode 100644
index 000000000..241d0be42
--- /dev/null
+++ b/kernel/sound/soc/pxa/magician.c
@@ -0,0 +1,534 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/uda1380.h>
+
+#include <mach/magician.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-i2s.h"
+#include "pxa-ssp.h"
+
+#define MAGICIAN_MIC 0
+#define MAGICIAN_MIC_EXT 1
+
+static int magician_hp_switch;
+static int magician_spk_switch = 1;
+static int magician_in_sel = MAGICIAN_MIC;
+
+static void magician_ext_control(struct snd_soc_dapm_context *dapm)
+{
+
+ snd_soc_dapm_mutex_lock(dapm);
+
+ if (magician_spk_switch)
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
+ else
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+ if (magician_hp_switch)
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic");
+ break;
+ case MAGICIAN_MIC_EXT:
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic");
+ break;
+ }
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* check the jack status at stream startup */
+ magician_ext_control(&rtd->card->dapm);
+
+ return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int acps, acds, width;
+ unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+ int ret = 0;
+
+ width = snd_pcm_format_physical_width(params_format(params));
+
+ /*
+ * rate = SSPSCLK / (2 * width(16 or 32))
+ * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+ */
+ switch (params_rate(params)) {
+ case 8000:
+ /* off by a factor of 2: bug in the PXA27x audio clock? */
+ acps = 32842000;
+ switch (width) {
+ case 16:
+ /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_16;
+ break;
+ default: /* 32 */
+ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_8;
+ }
+ break;
+ case 11025:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_4;
+ break;
+ default: /* 32 */
+ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ }
+ break;
+ case 22050:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ default: /* 32 */
+ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 44100:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ default: /* 32 */
+ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 48000:
+ acps = 12235000;
+ switch (width) {
+ case 16:
+ /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ default: /* 32 */
+ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 96000:
+ default:
+ acps = 12235000;
+ switch (width) {
+ case 16:
+ /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ break;
+ default: /* 32 */
+ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ div4 = PXA_SSP_CLK_SCDB_1;
+ break;
+ }
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
+ if (ret < 0)
+ return ret;
+
+ /* set audio clock as clock source */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock ACDS divider */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ PXA_SSP_AUDIO_DIV_ACDS, acds);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock SCDB divider4 */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ PXA_SSP_AUDIO_DIV_SCDB, div4);
+ if (ret < 0)
+ return ret;
+
+ /* set SSP audio pll clock */
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as output */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_capture_hw_params,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_playback_hw_params,
+};
+
+static int magician_get_hp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_hp_switch;
+ return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (magician_hp_switch == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_hp_switch = ucontrol->value.integer.value[0];
+ magician_ext_control(&card->dapm);
+ return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_spk_switch;
+ return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (magician_spk_switch == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_spk_switch = ucontrol->value.integer.value[0];
+ magician_ext_control(&card->dapm);
+ return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_in_sel;
+ return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (magician_in_sel == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_in_sel = ucontrol->value.integer.value[0];
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
+ break;
+ case MAGICIAN_MIC_EXT:
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
+ }
+
+ return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+ SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+ SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+ SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* Headphone connected to VOUTL, VOUTR */
+ {"Headphone Jack", NULL, "VOUTL"},
+ {"Headphone Jack", NULL, "VOUTR"},
+
+ /* Speaker connected to VOUTL, VOUTR */
+ {"Speaker", NULL, "VOUTL"},
+ {"Speaker", NULL, "VOUTR"},
+
+ /* Mics are connected to VINM */
+ {"VINM", NULL, "Headset Mic"},
+ {"VINM", NULL, "Call Mic"},
+};
+
+static const char *input_select[] = {"Call Mic", "Headset Mic"};
+static const struct soc_enum magician_in_sel_enum =
+ SOC_ENUM_SINGLE_EXT(2, input_select);
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+ SOC_SINGLE_BOOL_EXT("Headphone Switch",
+ (unsigned long)&magician_hp_switch,
+ magician_get_hp, magician_set_hp),
+ SOC_SINGLE_BOOL_EXT("Speaker Switch",
+ (unsigned long)&magician_spk_switch,
+ magician_get_spk, magician_set_spk),
+ SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
+ magician_get_input, magician_set_input),
+};
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+ .name = "uda1380",
+ .stream_name = "UDA1380 Playback",
+ .cpu_dai_name = "pxa-ssp-dai.0",
+ .codec_dai_name = "uda1380-hifi-playback",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "uda1380-codec.0-0018",
+ .ops = &magician_playback_ops,
+},
+{
+ .name = "uda1380",
+ .stream_name = "UDA1380 Capture",
+ .cpu_dai_name = "pxa2xx-i2s",
+ .codec_dai_name = "uda1380-hifi-capture",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "uda1380-codec.0-0018",
+ .ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_card snd_soc_card_magician = {
+ .name = "Magician",
+ .owner = THIS_MODULE,
+ .dai_link = magician_dai,
+ .num_links = ARRAY_SIZE(magician_dai),
+
+ .controls = uda1380_magician_controls,
+ .num_controls = ARRAY_SIZE(uda1380_magician_controls),
+ .dapm_widgets = uda1380_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
+};
+
+static struct platform_device *magician_snd_device;
+
+/*
+ * FIXME: move into magician board file once merged into the pxa tree
+ */
+static struct uda1380_platform_data uda1380_info = {
+ .gpio_power = EGPIO_MAGICIAN_CODEC_POWER,
+ .gpio_reset = EGPIO_MAGICIAN_CODEC_RESET,
+ .dac_clk = UDA1380_DAC_CLK_WSPLL,
+};
+
+static struct i2c_board_info i2c_board_info[] = {
+ {
+ I2C_BOARD_INFO("uda1380", 0x18),
+ .platform_data = &uda1380_info,
+ },
+};
+
+static int __init magician_init(void)
+{
+ int ret;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+
+ if (!machine_is_magician())
+ return -ENODEV;
+
+ adapter = i2c_get_adapter(0);
+ if (!adapter)
+ return -ENODEV;
+ client = i2c_new_device(adapter, i2c_board_info);
+ i2c_put_adapter(adapter);
+ if (!client)
+ return -ENODEV;
+
+ ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
+ if (ret)
+ goto err_request_spk;
+ ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
+ if (ret)
+ goto err_request_ep;
+ ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
+ if (ret)
+ goto err_request_mic;
+ ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
+ if (ret)
+ goto err_request_in_sel0;
+ ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
+ if (ret)
+ goto err_request_in_sel1;
+
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
+
+ magician_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!magician_snd_device) {
+ ret = -ENOMEM;
+ goto err_pdev;
+ }
+
+ platform_set_drvdata(magician_snd_device, &snd_soc_card_magician);
+ ret = platform_device_add(magician_snd_device);
+ if (ret) {
+ platform_device_put(magician_snd_device);
+ goto err_pdev;
+ }
+
+ return 0;
+
+err_pdev:
+ gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+err_request_in_sel1:
+ gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+err_request_in_sel0:
+ gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+err_request_mic:
+ gpio_free(EGPIO_MAGICIAN_EP_POWER);
+err_request_ep:
+ gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+err_request_spk:
+ return ret;
+}
+
+static void __exit magician_exit(void)
+{
+ platform_device_unregister(magician_snd_device);
+
+ gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
+
+ gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+ gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+ gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+ gpio_free(EGPIO_MAGICIAN_EP_POWER);
+ gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/mioa701_wm9713.c b/kernel/sound/soc/pxa/mioa701_wm9713.c
new file mode 100644
index 000000000..a9615a574
--- /dev/null
+++ b/kernel/sound/soc/pxa/mioa701_wm9713.c
@@ -0,0 +1,214 @@
+/*
+ * Handles the Mitac mioa701 SoC system
+ *
+ * Copyright (C) 2008 Robert Jarzmik
+ *
+ * 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 in 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
+ *
+ * This is a little schema of the sound interconnections :
+ *
+ * Sagem X200 Wolfson WM9713
+ * +--------+ +-------------------+ Rear Speaker
+ * | | | | /-+
+ * | +--->----->---+MONOIN SPKL+--->----+-+ |
+ * | GSM | | | | | |
+ * | +--->----->---+PCBEEP SPKR+--->----+-+ |
+ * | CHIP | | | \-+
+ * | +---<-----<---+MONO |
+ * | | | | Front Speaker
+ * +--------+ | | /-+
+ * | HPL+--->----+-+ |
+ * | | | | |
+ * | OUT3+--->----+-+ |
+ * | | \-+
+ * | |
+ * | | Front Micro
+ * | | +
+ * | MIC1+-----<--+o+
+ * | | +
+ * +-------------------+ ---
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+
+#include "pxa2xx-ac97.h"
+#include "../codecs/wm9713.h"
+
+#define AC97_GPIO_PULL 0x58
+
+/* Use GPIO8 for rear speaker amplifier */
+static int rear_amp_power(struct snd_soc_codec *codec, int power)
+{
+ unsigned short reg;
+
+ if (power) {
+ reg = snd_soc_read(codec, AC97_GPIO_CFG);
+ snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100);
+ reg = snd_soc_read(codec, AC97_GPIO_PULL);
+ snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15));
+ } else {
+ reg = snd_soc_read(codec, AC97_GPIO_CFG);
+ snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100);
+ reg = snd_soc_read(codec, AC97_GPIO_PULL);
+ snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15));
+ }
+
+ return 0;
+}
+
+static int rear_amp_event(struct snd_soc_dapm_widget *widget,
+ struct snd_kcontrol *kctl, int event)
+{
+ struct snd_soc_codec *codec = widget->dapm->card->rtd[0].codec;
+
+ return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event));
+}
+
+/* mioa701 machine dapm widgets */
+static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Front Speaker", NULL),
+ SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event),
+ SND_SOC_DAPM_MIC("Headset", NULL),
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Front Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Call Mic */
+ {"Mic Bias", NULL, "Front Mic"},
+ {"MIC1", NULL, "Mic Bias"},
+
+ /* Headset Mic */
+ {"LINEL", NULL, "Headset Mic"},
+ {"LINER", NULL, "Headset Mic"},
+
+ /* GSM Module */
+ {"MONOIN", NULL, "GSM Line Out"},
+ {"PCBEEP", NULL, "GSM Line Out"},
+ {"GSM Line In", NULL, "MONO"},
+
+ /* headphone connected to HPL, HPR */
+ {"Headset", NULL, "HPL"},
+ {"Headset", NULL, "HPR"},
+
+ /* front speaker connected to HPL, OUT3 */
+ {"Front Speaker", NULL, "HPL"},
+ {"Front Speaker", NULL, "OUT3"},
+
+ /* rear speaker connected to SPKL, SPKR */
+ {"Rear Speaker", NULL, "SPKL"},
+ {"Rear Speaker", NULL, "SPKR"},
+};
+
+static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+
+ /* Prepare GPIO8 for rear speaker amplifier */
+ snd_soc_update_bits(codec, AC97_GPIO_CFG, 0x100, 0x100);
+
+ /* Prepare MIC input */
+ snd_soc_update_bits(codec, AC97_3D_CONTROL, 0xc000, 0xc000);
+
+ return 0;
+}
+
+static struct snd_soc_ops mioa701_ops;
+
+static struct snd_soc_dai_link mioa701_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai_name = "pxa2xx-ac97",
+ .codec_dai_name = "wm9713-hifi",
+ .codec_name = "wm9713-codec",
+ .init = mioa701_wm9713_init,
+ .platform_name = "pxa-pcm-audio",
+ .ops = &mioa701_ops,
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
+ .codec_dai_name ="wm9713-aux",
+ .codec_name = "wm9713-codec",
+ .platform_name = "pxa-pcm-audio",
+ .ops = &mioa701_ops,
+ },
+};
+
+static struct snd_soc_card mioa701 = {
+ .name = "MioA701",
+ .owner = THIS_MODULE,
+ .dai_link = mioa701_dai,
+ .num_links = ARRAY_SIZE(mioa701_dai),
+
+ .dapm_widgets = mioa701_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mioa701_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static int mioa701_wm9713_probe(struct platform_device *pdev)
+{
+ int rc;
+
+ if (!machine_is_mioa701())
+ return -ENODEV;
+
+ mioa701.dev = &pdev->dev;
+ rc = snd_soc_register_card(&mioa701);
+ if (!rc)
+ dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+ "lead to overheating and possible destruction of your device."
+ " Do not use without a good knowledge of mio's board design!\n");
+ return rc;
+}
+
+static int mioa701_wm9713_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver mioa701_wm9713_driver = {
+ .probe = mioa701_wm9713_probe,
+ .remove = mioa701_wm9713_remove,
+ .driver = {
+ .name = "mioa701-wm9713",
+ .pm = &snd_soc_pm_ops,
+ },
+};
+
+module_platform_driver(mioa701_wm9713_driver);
+
+/* Module information */
+MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
+MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/mmp-pcm.c b/kernel/sound/soc/pxa/mmp-pcm.c
new file mode 100644
index 000000000..1eb45dcfb
--- /dev/null
+++ b/kernel/sound/soc/pxa/mmp-pcm.c
@@ -0,0 +1,257 @@
+/*
+ * linux/sound/soc/pxa/mmp-pcm.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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.
+ *
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/platform_data/dma-mmp_tdma.h>
+#include <linux/platform_data/mmp_audio.h>
+
+#include <sound/pxa2xx-lib.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+struct mmp_dma_data {
+ int ssp_id;
+ struct resource *dma_res;
+};
+
+#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \
+ SNDRV_PCM_INFO_MMAP_VALID | \
+ SNDRV_PCM_INFO_INTERLEAVED | \
+ SNDRV_PCM_INFO_PAUSE | \
+ SNDRV_PCM_INFO_RESUME | \
+ SNDRV_PCM_INFO_NO_PERIOD_WAKEUP)
+
+static struct snd_pcm_hardware mmp_pcm_hardware[] = {
+ {
+ .info = MMP_PCM_INFO,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 2048,
+ .periods_min = 2,
+ .periods_max = 32,
+ .buffer_bytes_max = 4096,
+ .fifo_size = 32,
+ },
+ {
+ .info = MMP_PCM_INFO,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 2048,
+ .periods_min = 2,
+ .periods_max = 32,
+ .buffer_bytes_max = 4096,
+ .fifo_size = 32,
+ },
+};
+
+static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+ struct dma_slave_config slave_config;
+ int ret;
+
+ ret =
+ snd_dmaengine_pcm_prepare_slave_config(substream, params,
+ &slave_config);
+ if (ret)
+ return ret;
+
+ ret = dmaengine_slave_config(chan, &slave_config);
+ if (ret)
+ return ret;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ struct mmp_dma_data *dma_data = param;
+ bool found = false;
+ char *devname;
+
+ devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name,
+ dma_data->ssp_id);
+ if ((strcmp(dev_name(chan->device->dev), devname) == 0) &&
+ (chan->chan_id == dma_data->dma_res->start)) {
+ found = true;
+ }
+
+ kfree(devname);
+ return found;
+}
+
+static int mmp_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct mmp_dma_data dma_data;
+ struct resource *r;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
+ if (!r)
+ return -EBUSY;
+
+ snd_soc_set_runtime_hwparams(substream,
+ &mmp_pcm_hardware[substream->stream]);
+
+ dma_data.dma_res = r;
+ dma_data.ssp_id = cpu_dai->id;
+
+ return snd_dmaengine_pcm_open_request_chan(substream, filter,
+ &dma_data);
+}
+
+static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long off = vma->vm_pgoff;
+
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ return remap_pfn_range(vma, vma->vm_start,
+ __phys_to_pfn(runtime->dma_addr) + off,
+ vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+static struct snd_pcm_ops mmp_pcm_ops = {
+ .open = mmp_pcm_open,
+ .close = snd_dmaengine_pcm_close_release_chan,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = mmp_pcm_hw_params,
+ .trigger = snd_dmaengine_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer,
+ .mmap = mmp_pcm_mmap,
+};
+
+static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+ struct gen_pool *gpool;
+
+ gpool = sram_get_gpool("asram");
+ if (!gpool)
+ return;
+
+ for (stream = 0; stream < 2; stream++) {
+ size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ gen_pool_free(gpool, (unsigned long)buf->area, size);
+ buf->area = NULL;
+ }
+
+ return;
+}
+
+static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
+ int stream)
+{
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+ struct gen_pool *gpool;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = substream->pcm->card->dev;
+ buf->private_data = NULL;
+
+ gpool = sram_get_gpool("asram");
+ if (!gpool)
+ return -ENOMEM;
+
+ buf->area = gen_pool_dma_alloc(gpool, size, &buf->addr);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+ return 0;
+}
+
+static int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0, stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+
+ ret = mmp_pcm_preallocate_dma_buffer(substream, stream);
+ if (ret)
+ goto err;
+ }
+
+ return 0;
+
+err:
+ mmp_pcm_free_dma_buffers(pcm);
+ return ret;
+}
+
+static struct snd_soc_platform_driver mmp_soc_platform = {
+ .ops = &mmp_pcm_ops,
+ .pcm_new = mmp_pcm_new,
+ .pcm_free = mmp_pcm_free_dma_buffers,
+};
+
+static int mmp_pcm_probe(struct platform_device *pdev)
+{
+ struct mmp_audio_platdata *pdata = pdev->dev.platform_data;
+
+ if (pdata) {
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max =
+ pdata->buffer_max_playback;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max =
+ pdata->period_max_playback;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max =
+ pdata->buffer_max_capture;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max =
+ pdata->period_max_capture;
+ }
+ return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform);
+}
+
+static int mmp_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver mmp_pcm_driver = {
+ .driver = {
+ .name = "mmp-pcm-audio",
+ },
+
+ .probe = mmp_pcm_probe,
+ .remove = mmp_pcm_remove,
+};
+
+module_platform_driver(mmp_pcm_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP Soc Audio DMA module");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/mmp-sspa.c b/kernel/sound/soc/pxa/mmp-sspa.c
new file mode 100644
index 000000000..eca60c297
--- /dev/null
+++ b/kernel/sound/soc/pxa/mmp-sspa.c
@@ -0,0 +1,484 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.c
+ * Base on pxa2xx-ssp.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/slab.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/io.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+#include "mmp-sspa.h"
+
+/*
+ * SSPA audio private data
+ */
+struct sspa_priv {
+ struct ssp_device *sspa;
+ struct snd_dmaengine_dai_dma_data *dma_params;
+ struct clk *audio_clk;
+ struct clk *sysclk;
+ int dai_fmt;
+ int running_cnt;
+};
+
+static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val)
+{
+ __raw_writel(val, sspa->mmio_base + reg);
+}
+
+static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg)
+{
+ return __raw_readl(sspa->mmio_base + reg);
+}
+
+static void mmp_sspa_tx_enable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+ sspa_sp |= SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_tx_disable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+ sspa_sp &= ~SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_enable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+ sspa_sp |= SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_disable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+ sspa_sp &= ~SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static int mmp_sspa_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_enable(priv->sysclk);
+ clk_enable(priv->sspa->clk);
+
+ return 0;
+}
+
+static void mmp_sspa_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable(priv->sspa->clk);
+ clk_disable(priv->sysclk);
+
+ return;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (clk_id) {
+ case MMP_SSPA_CLK_AUDIO:
+ ret = clk_set_rate(priv->audio_clk, freq);
+ if (ret)
+ return ret;
+ break;
+ case MMP_SSPA_CLK_PLL:
+ case MMP_SSPA_CLK_VCXO:
+ /* not support yet */
+ return -EINVAL;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+ int source, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (pll_id) {
+ case MMP_SYSCLK:
+ ret = clk_set_rate(priv->sysclk, freq_out);
+ if (ret)
+ return ret;
+ break;
+ case MMP_SSPA_CLK:
+ ret = clk_set_rate(priv->sspa->clk, freq_out);
+ if (ret)
+ return ret;
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+/*
+ * Set up the sspa dai format. The sspa port must be inactive
+ * before calling this function as the physical
+ * interface format is changed.
+ */
+static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ u32 sspa_sp, sspa_ctrl;
+
+ /* check if we need to change anything at all */
+ if (sspa_priv->dai_fmt == fmt)
+ return 0;
+
+ /* we can only change the settings if the port is not in use */
+ if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) ||
+ (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) {
+ dev_err(&sspa->pdev->dev,
+ "can't change hardware dai format: stream is in use\n");
+ return -EINVAL;
+ }
+
+ /* reset port settings */
+ sspa_sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH;
+ sspa_ctrl = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ sspa_sp |= SSPA_SP_MSL;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sspa_sp |= SSPA_SP_FSP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ sspa_sp |= SSPA_TXSP_FPER(63);
+ sspa_sp |= SSPA_SP_FWID(31);
+ sspa_ctrl |= SSPA_CTL_XDATDLY(1);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+ sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH);
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+ /*
+ * FIXME: hw issue, for the tx serial port,
+ * can not config the master/slave mode;
+ * so must clean this bit.
+ * The master/slave mode has been set in the
+ * rx port.
+ */
+ sspa_sp &= ~SSPA_SP_MSL;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+
+ mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+
+ /* Since we are configuring the timings for the format by hand
+ * we have to defer some things until hw_params() where we
+ * know parameters like the sample size.
+ */
+ sspa_priv->dai_fmt = fmt;
+ return 0;
+}
+
+/*
+ * Set the SSPA audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int mmp_sspa_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ struct snd_dmaengine_dai_dma_data *dma_params;
+ u32 sspa_ctrl;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL);
+ else
+ sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL);
+
+ sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK;
+ sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1);
+ sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK;
+ sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS);
+ sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1);
+ } else {
+ mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0);
+ }
+
+ dma_params = &sspa_priv->dma_params[substream->stream];
+ dma_params->addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ (sspa->phys_base + SSPA_TXD) :
+ (sspa->phys_base + SSPA_RXD);
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
+ return 0;
+}
+
+static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /*
+ * whatever playback or capture, must enable rx.
+ * this is a hw issue, so need check if rx has been
+ * enabled or not; if has been enabled by another
+ * stream, do not enable again.
+ */
+ if (!sspa_priv->running_cnt)
+ mmp_sspa_rx_enable(sspa);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mmp_sspa_tx_enable(sspa);
+
+ sspa_priv->running_cnt++;
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sspa_priv->running_cnt--;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mmp_sspa_tx_disable(sspa);
+
+ /* have no capture stream, disable rx port */
+ if (!sspa_priv->running_cnt)
+ mmp_sspa_rx_disable(sspa);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static int mmp_sspa_probe(struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = dev_get_drvdata(dai->dev);
+
+ snd_soc_dai_set_drvdata(dai, priv);
+ return 0;
+
+}
+
+#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000
+#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
+ .startup = mmp_sspa_startup,
+ .shutdown = mmp_sspa_shutdown,
+ .trigger = mmp_sspa_trigger,
+ .hw_params = mmp_sspa_hw_params,
+ .set_sysclk = mmp_sspa_set_dai_sysclk,
+ .set_pll = mmp_sspa_set_dai_pll,
+ .set_fmt = mmp_sspa_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver mmp_sspa_dai = {
+ .probe = mmp_sspa_probe,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 128,
+ .rates = MMP_SSPA_RATES,
+ .formats = MMP_SSPA_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MMP_SSPA_RATES,
+ .formats = MMP_SSPA_FORMATS,
+ },
+ .ops = &mmp_sspa_dai_ops,
+};
+
+static const struct snd_soc_component_driver mmp_sspa_component = {
+ .name = "mmp-sspa",
+};
+
+static int asoc_mmp_sspa_probe(struct platform_device *pdev)
+{
+ struct sspa_priv *priv;
+ struct resource *res;
+
+ priv = devm_kzalloc(&pdev->dev,
+ sizeof(struct sspa_priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->sspa = devm_kzalloc(&pdev->dev,
+ sizeof(struct ssp_device), GFP_KERNEL);
+ if (priv->sspa == NULL)
+ return -ENOMEM;
+
+ priv->dma_params = devm_kzalloc(&pdev->dev,
+ 2 * sizeof(struct snd_dmaengine_dai_dma_data),
+ GFP_KERNEL);
+ if (priv->dma_params == NULL)
+ return -ENOMEM;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(priv->sspa->mmio_base))
+ return PTR_ERR(priv->sspa->mmio_base);
+
+ priv->sspa->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(priv->sspa->clk))
+ return PTR_ERR(priv->sspa->clk);
+
+ priv->audio_clk = clk_get(NULL, "mmp-audio");
+ if (IS_ERR(priv->audio_clk))
+ return PTR_ERR(priv->audio_clk);
+
+ priv->sysclk = clk_get(NULL, "mmp-sysclk");
+ if (IS_ERR(priv->sysclk)) {
+ clk_put(priv->audio_clk);
+ return PTR_ERR(priv->sysclk);
+ }
+ clk_enable(priv->audio_clk);
+ priv->dai_fmt = (unsigned int) -1;
+ platform_set_drvdata(pdev, priv);
+
+ return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component,
+ &mmp_sspa_dai, 1);
+}
+
+static int asoc_mmp_sspa_remove(struct platform_device *pdev)
+{
+ struct sspa_priv *priv = platform_get_drvdata(pdev);
+
+ clk_disable(priv->audio_clk);
+ clk_put(priv->audio_clk);
+ clk_put(priv->sysclk);
+ return 0;
+}
+
+static struct platform_driver asoc_mmp_sspa_driver = {
+ .driver = {
+ .name = "mmp-sspa-dai",
+ },
+ .probe = asoc_mmp_sspa_probe,
+ .remove = asoc_mmp_sspa_remove,
+};
+
+module_platform_driver(asoc_mmp_sspa_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP SSPA SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/mmp-sspa.h b/kernel/sound/soc/pxa/mmp-sspa.h
new file mode 100644
index 000000000..ea365cb9e
--- /dev/null
+++ b/kernel/sound/soc/pxa/mmp-sspa.h
@@ -0,0 +1,92 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.h
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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 _MMP_SSPA_H
+#define _MMP_SSPA_H
+
+/*
+ * SSPA Registers
+ */
+#define SSPA_RXD (0x00)
+#define SSPA_RXID (0x04)
+#define SSPA_RXCTL (0x08)
+#define SSPA_RXSP (0x0c)
+#define SSPA_RXFIFO_UL (0x10)
+#define SSPA_RXINT_MASK (0x14)
+#define SSPA_RXC (0x18)
+#define SSPA_RXFIFO_NOFS (0x1c)
+#define SSPA_RXFIFO_SIZE (0x20)
+
+#define SSPA_TXD (0x80)
+#define SSPA_TXID (0x84)
+#define SSPA_TXCTL (0x88)
+#define SSPA_TXSP (0x8c)
+#define SSPA_TXFIFO_LL (0x90)
+#define SSPA_TXINT_MASK (0x94)
+#define SSPA_TXC (0x98)
+#define SSPA_TXFIFO_NOFS (0x9c)
+#define SSPA_TXFIFO_SIZE (0xa0)
+
+/* SSPA Control Register */
+#define SSPA_CTL_XPH (1 << 31) /* Read Phase */
+#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */
+#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */
+#define SSPA_CTL_XFRLEN2_MASK (7 << 24)
+#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */
+#define SSPA_CTL_XWDLEN2_MASK (7 << 21)
+#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */
+#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Tansmit Data Delay */
+#define SSPA_CTL_XSSZ2_MASK (7 << 16)
+#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */
+#define SSPA_CTL_XFRLEN1_MASK (7 << 8)
+#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */
+#define SSPA_CTL_XWDLEN1_MASK (7 << 5)
+#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */
+#define SSPA_CTL_XSSZ1_MASK (7 << 0)
+#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */
+
+#define SSPA_CTL_8_BITS (0x0) /* Sample Size */
+#define SSPA_CTL_12_BITS (0x1)
+#define SSPA_CTL_16_BITS (0x2)
+#define SSPA_CTL_20_BITS (0x3)
+#define SSPA_CTL_24_BITS (0x4)
+#define SSPA_CTL_32_BITS (0x5)
+
+/* SSPA Serial Port Register */
+#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */
+#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */
+#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */
+#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */
+#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */
+#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */
+#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */
+#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */
+#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */
+
+/* sspa clock sources */
+#define MMP_SSPA_CLK_PLL 0
+#define MMP_SSPA_CLK_VCXO 1
+#define MMP_SSPA_CLK_AUDIO 3
+
+/* sspa pll id */
+#define MMP_SYSCLK 0
+#define MMP_SSPA_CLK 1
+
+#endif /* _MMP_SSPA_H */
diff --git a/kernel/sound/soc/pxa/palm27x.c b/kernel/sound/soc/pxa/palm27x.c
new file mode 100644
index 000000000..c20bbc042
--- /dev/null
+++ b/kernel/sound/soc/pxa/palm27x.c
@@ -0,0 +1,170 @@
+/*
+ * linux/sound/soc/pxa/palm27x.c
+ *
+ * SoC Audio driver for Palm T|X, T5 and LifeDrive
+ *
+ * based on tosa.c
+ *
+ * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.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/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <linux/platform_data/asoc-palm27x.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_jack hs_jack;
+
+/* Headphones jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+/* Headphones jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+ [0] = {
+ /* gpio is set on per-platform basis */
+ .name = "hp-gpio",
+ .report = SND_JACK_HEADPHONE,
+ .debounce_time = 200,
+ },
+};
+
+/* Palm27x machine dapm widgets */
+static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext. Speaker", NULL),
+ SND_SOC_DAPM_MIC("Ext. Microphone", NULL),
+};
+
+/* PalmTX audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* headphone connected to HPOUTL, HPOUTR */
+ {"Headphone Jack", NULL, "HPOUTL"},
+ {"Headphone Jack", NULL, "HPOUTR"},
+
+ /* ext speaker connected to ROUT2, LOUT2 */
+ {"Ext. Speaker", NULL, "LOUT2"},
+ {"Ext. Speaker", NULL, "ROUT2"},
+
+ /* mic connected to MIC1 */
+ {"MIC1", NULL, "Ext. Microphone"},
+};
+
+static struct snd_soc_card palm27x_asoc;
+
+static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int err;
+
+ /* Jack detection API stuff */
+ err = snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+ SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins,
+ ARRAY_SIZE(hs_jack_pins));
+ if (err)
+ return err;
+
+ err = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
+
+ return err;
+}
+
+static struct snd_soc_dai_link palm27x_dai[] = {
+{
+ .name = "AC97 HiFi",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai_name = "pxa2xx-ac97",
+ .codec_dai_name = "wm9712-hifi",
+ .codec_name = "wm9712-codec",
+ .platform_name = "pxa-pcm-audio",
+ .init = palm27x_ac97_init,
+},
+{
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
+ .codec_dai_name = "wm9712-aux",
+ .codec_name = "wm9712-codec",
+ .platform_name = "pxa-pcm-audio",
+},
+};
+
+static struct snd_soc_card palm27x_asoc = {
+ .name = "Palm/PXA27x",
+ .owner = THIS_MODULE,
+ .dai_link = palm27x_dai,
+ .num_links = ARRAY_SIZE(palm27x_dai),
+ .dapm_widgets = palm27x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(palm27x_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
+};
+
+static int palm27x_asoc_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ if (!(machine_is_palmtx() || machine_is_palmt5() ||
+ machine_is_palmld() || machine_is_palmte2()))
+ return -ENODEV;
+
+ if (!pdev->dev.platform_data) {
+ dev_err(&pdev->dev, "please supply platform_data\n");
+ return -ENODEV;
+ }
+
+ hs_jack_gpios[0].gpio = ((struct palm27x_asoc_info *)
+ (pdev->dev.platform_data))->jack_gpio;
+
+ palm27x_asoc.dev = &pdev->dev;
+
+ ret = snd_soc_register_card(&palm27x_asoc);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+}
+
+static int palm27x_asoc_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&palm27x_asoc);
+ return 0;
+}
+
+static struct platform_driver palm27x_wm9712_driver = {
+ .probe = palm27x_asoc_probe,
+ .remove = palm27x_asoc_remove,
+ .driver = {
+ .name = "palm27x-asoc",
+ .pm = &snd_soc_pm_ops,
+ },
+};
+
+module_platform_driver(palm27x_wm9712_driver);
+
+/* Module information */
+MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/poodle.c b/kernel/sound/soc/pxa/poodle.c
new file mode 100644
index 000000000..0fce8c420
--- /dev/null
+++ b/kernel/sound/soc/pxa/poodle.c
@@ -0,0 +1,311 @@
+/*
+ * poodle.c -- SoC audio for Poodle
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/i2c.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/locomo.h>
+#include <mach/poodle.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm8731.h"
+#include "pxa2xx-i2s.h"
+
+#define POODLE_HP 1
+#define POODLE_HP_OFF 0
+#define POODLE_SPK_ON 1
+#define POODLE_SPK_OFF 0
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define POODLE_AUDIO_CLOCK 12288000
+
+static int poodle_jack_func;
+static int poodle_spk_func;
+
+static void poodle_ext_control(struct snd_soc_dapm_context *dapm)
+{
+ /* set up jack connection */
+ if (poodle_jack_func == POODLE_HP) {
+ /* set = unmute headphone */
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 1);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 1);
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ } else {
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 0);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 0);
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ }
+
+ /* set the enpoints to their new connetion states */
+ if (poodle_spk_func == POODLE_SPK_ON)
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync(dapm);
+}
+
+static int poodle_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* check the jack status at stream startup */
+ poodle_ext_control(&rtd->card->dapm);
+
+ return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on poodle */
+static void poodle_shutdown(struct snd_pcm_substream *substream)
+{
+ /* set = unmute headphone */
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 1);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 1);
+}
+
+static int poodle_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops poodle_ops = {
+ .startup = poodle_startup,
+ .hw_params = poodle_hw_params,
+ .shutdown = poodle_shutdown,
+};
+
+static int poodle_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = poodle_jack_func;
+ return 0;
+}
+
+static int poodle_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (poodle_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ poodle_jack_func = ucontrol->value.integer.value[0];
+ poodle_ext_control(&card->dapm);
+ return 1;
+}
+
+static int poodle_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = poodle_spk_func;
+ return 0;
+}
+
+static int poodle_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (poodle_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ poodle_spk_func = ucontrol->value.integer.value[0];
+ poodle_ext_control(&card->dapm);
+ return 1;
+}
+
+static int poodle_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 0);
+ else
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 1);
+
+ return 0;
+}
+
+/* poodle machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
+};
+
+/* Corgi machine connections to the codec pins */
+static const struct snd_soc_dapm_route poodle_audio_map[] = {
+
+ /* headphone connected to LHPOUT1, RHPOUT1 */
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ /* speaker connected to LOUT, ROUT */
+ {"Ext Spk", NULL, "ROUT"},
+ {"Ext Spk", NULL, "LOUT"},
+};
+
+static const char *jack_function[] = {"Off", "Headphone"};
+static const char *spk_function[] = {"Off", "On"};
+static const struct soc_enum poodle_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
+ SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
+ poodle_set_jack),
+ SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
+ poodle_set_spk),
+};
+
+/*
+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
+ */
+static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_nc_pin(dapm, "LLINEIN");
+ snd_soc_dapm_nc_pin(dapm, "RLINEIN");
+
+ return 0;
+}
+
+/* poodle digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link poodle_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731",
+ .cpu_dai_name = "pxa2xx-i2s",
+ .codec_dai_name = "wm8731-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm8731.0-001b",
+ .init = poodle_wm8731_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &poodle_ops,
+};
+
+/* poodle audio machine driver */
+static struct snd_soc_card poodle = {
+ .name = "Poodle",
+ .dai_link = &poodle_dai,
+ .num_links = 1,
+ .owner = THIS_MODULE,
+
+ .controls = wm8731_poodle_controls,
+ .num_controls = ARRAY_SIZE(wm8731_poodle_controls),
+ .dapm_widgets = wm8731_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets),
+ .dapm_routes = poodle_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(poodle_audio_map),
+};
+
+static int poodle_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &poodle;
+ int ret;
+
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 0);
+ /* should we mute HP at startup - burning power ?*/
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 0);
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 0);
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+}
+
+static int poodle_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver poodle_driver = {
+ .driver = {
+ .name = "poodle-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = poodle_probe,
+ .remove = poodle_remove,
+};
+
+module_platform_driver(poodle_driver);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Poodle");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:poodle-audio");
diff --git a/kernel/sound/soc/pxa/pxa-ssp.c b/kernel/sound/soc/pxa/pxa-ssp.c
new file mode 100644
index 000000000..fbe2e93d6
--- /dev/null
+++ b/kernel/sound/soc/pxa/pxa-ssp.c
@@ -0,0 +1,841 @@
+/*
+ * pxa-ssp.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2005,2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * 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.
+ *
+ * TODO:
+ * o Test network mode for > 16bit sample size
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/of.h>
+#include <linux/dmaengine.h>
+
+#include <asm/irq.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "../../arm/pxa2xx-pcm.h"
+#include "pxa-ssp.h"
+
+/*
+ * SSP audio private data
+ */
+struct ssp_priv {
+ struct ssp_device *ssp;
+ unsigned int sysclk;
+ int dai_fmt;
+#ifdef CONFIG_PM
+ uint32_t cr0;
+ uint32_t cr1;
+ uint32_t to;
+ uint32_t psp;
+#endif
+};
+
+static void dump_registers(struct ssp_device *ssp)
+{
+ dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
+ pxa_ssp_read_reg(ssp, SSCR0), pxa_ssp_read_reg(ssp, SSCR1),
+ pxa_ssp_read_reg(ssp, SSTO));
+
+ dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n",
+ pxa_ssp_read_reg(ssp, SSPSP), pxa_ssp_read_reg(ssp, SSSR),
+ pxa_ssp_read_reg(ssp, SSACD));
+}
+
+static void pxa_ssp_enable(struct ssp_device *ssp)
+{
+ uint32_t sscr0;
+
+ sscr0 = __raw_readl(ssp->mmio_base + SSCR0) | SSCR0_SSE;
+ __raw_writel(sscr0, ssp->mmio_base + SSCR0);
+}
+
+static void pxa_ssp_disable(struct ssp_device *ssp)
+{
+ uint32_t sscr0;
+
+ sscr0 = __raw_readl(ssp->mmio_base + SSCR0) & ~SSCR0_SSE;
+ __raw_writel(sscr0, ssp->mmio_base + SSCR0);
+}
+
+static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4,
+ int out, struct snd_dmaengine_dai_dma_data *dma)
+{
+ dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES :
+ DMA_SLAVE_BUSWIDTH_2_BYTES;
+ dma->maxburst = 16;
+ dma->addr = ssp->phys_base + SSDR;
+}
+
+static int pxa_ssp_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ struct snd_dmaengine_dai_dma_data *dma;
+ int ret = 0;
+
+ if (!cpu_dai->active) {
+ clk_prepare_enable(ssp->clk);
+ pxa_ssp_disable(ssp);
+ }
+
+ dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
+ if (!dma)
+ return -ENOMEM;
+
+ dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ &ssp->drcmr_tx : &ssp->drcmr_rx;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma);
+
+ return ret;
+}
+
+static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+
+ if (!cpu_dai->active) {
+ pxa_ssp_disable(ssp);
+ clk_disable_unprepare(ssp->clk);
+ }
+
+ kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+ snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+}
+
+#ifdef CONFIG_PM
+
+static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+
+ if (!cpu_dai->active)
+ clk_prepare_enable(ssp->clk);
+
+ priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0);
+ priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1);
+ priv->to = __raw_readl(ssp->mmio_base + SSTO);
+ priv->psp = __raw_readl(ssp->mmio_base + SSPSP);
+
+ pxa_ssp_disable(ssp);
+ clk_disable_unprepare(ssp->clk);
+ return 0;
+}
+
+static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE;
+
+ clk_prepare_enable(ssp->clk);
+
+ __raw_writel(sssr, ssp->mmio_base + SSSR);
+ __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0);
+ __raw_writel(priv->cr1, ssp->mmio_base + SSCR1);
+ __raw_writel(priv->to, ssp->mmio_base + SSTO);
+ __raw_writel(priv->psp, ssp->mmio_base + SSPSP);
+
+ if (cpu_dai->active)
+ pxa_ssp_enable(ssp);
+ else
+ clk_disable_unprepare(ssp->clk);
+
+ return 0;
+}
+
+#else
+#define pxa_ssp_suspend NULL
+#define pxa_ssp_resume NULL
+#endif
+
+/**
+ * ssp_set_clkdiv - set SSP clock divider
+ * @div: serial clock rate divider
+ */
+static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
+{
+ u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+
+ if (ssp->type == PXA25x_SSP) {
+ sscr0 &= ~0x0000ff00;
+ sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
+ } else {
+ sscr0 &= ~0x000fff00;
+ sscr0 |= (div - 1) << 8; /* 1..4096 */
+ }
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+}
+
+/**
+ * pxa_ssp_get_clkdiv - get SSP clock divider
+ */
+static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
+{
+ u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+ u32 div;
+
+ if (ssp->type == PXA25x_SSP)
+ div = ((sscr0 >> 8) & 0xff) * 2 + 2;
+ else
+ div = ((sscr0 >> 8) & 0xfff) + 1;
+ return div;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ int val;
+
+ u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
+ ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
+
+ dev_dbg(&ssp->pdev->dev,
+ "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n",
+ cpu_dai->id, clk_id, freq);
+
+ switch (clk_id) {
+ case PXA_SSP_CLK_NET_PLL:
+ sscr0 |= SSCR0_MOD;
+ break;
+ case PXA_SSP_CLK_PLL:
+ /* Internal PLL is fixed */
+ if (ssp->type == PXA25x_SSP)
+ priv->sysclk = 1843200;
+ else
+ priv->sysclk = 13000000;
+ break;
+ case PXA_SSP_CLK_EXT:
+ priv->sysclk = freq;
+ sscr0 |= SSCR0_ECS;
+ break;
+ case PXA_SSP_CLK_NET:
+ priv->sysclk = freq;
+ sscr0 |= SSCR0_NCS | SSCR0_MOD;
+ break;
+ case PXA_SSP_CLK_AUDIO:
+ priv->sysclk = 0;
+ pxa_ssp_set_scr(ssp, 1);
+ sscr0 |= SSCR0_ACS;
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ /* The SSP clock must be disabled when changing SSP clock mode
+ * on PXA2xx. On PXA3xx it must be enabled when doing so. */
+ if (ssp->type != PXA3xx_SSP)
+ clk_disable_unprepare(ssp->clk);
+ val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0;
+ pxa_ssp_write_reg(ssp, SSCR0, val);
+ if (ssp->type != PXA3xx_SSP)
+ clk_prepare_enable(ssp->clk);
+
+ return 0;
+}
+
+/*
+ * Set the SSP clock dividers.
+ */
+static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ int val;
+
+ switch (div_id) {
+ case PXA_SSP_AUDIO_DIV_ACDS:
+ val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
+ pxa_ssp_write_reg(ssp, SSACD, val);
+ break;
+ case PXA_SSP_AUDIO_DIV_SCDB:
+ val = pxa_ssp_read_reg(ssp, SSACD);
+ val &= ~SSACD_SCDB;
+ if (ssp->type == PXA3xx_SSP)
+ val &= ~SSACD_SCDX8;
+ switch (div) {
+ case PXA_SSP_CLK_SCDB_1:
+ val |= SSACD_SCDB;
+ break;
+ case PXA_SSP_CLK_SCDB_4:
+ break;
+ case PXA_SSP_CLK_SCDB_8:
+ if (ssp->type == PXA3xx_SSP)
+ val |= SSACD_SCDX8;
+ else
+ return -EINVAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ pxa_ssp_write_reg(ssp, SSACD, val);
+ break;
+ case PXA_SSP_DIV_SCR:
+ pxa_ssp_set_scr(ssp, div);
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+/*
+ * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
+ */
+static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
+
+ if (ssp->type == PXA3xx_SSP)
+ pxa_ssp_write_reg(ssp, SSACDD, 0);
+
+ switch (freq_out) {
+ case 5622000:
+ break;
+ case 11345000:
+ ssacd |= (0x1 << 4);
+ break;
+ case 12235000:
+ ssacd |= (0x2 << 4);
+ break;
+ case 14857000:
+ ssacd |= (0x3 << 4);
+ break;
+ case 32842000:
+ ssacd |= (0x4 << 4);
+ break;
+ case 48000000:
+ ssacd |= (0x5 << 4);
+ break;
+ case 0:
+ /* Disable */
+ break;
+
+ default:
+ /* PXA3xx has a clock ditherer which can be used to generate
+ * a wider range of frequencies - calculate a value for it.
+ */
+ if (ssp->type == PXA3xx_SSP) {
+ u32 val;
+ u64 tmp = 19968;
+ tmp *= 1000000;
+ do_div(tmp, freq_out);
+ val = tmp;
+
+ val = (val << 16) | 64;
+ pxa_ssp_write_reg(ssp, SSACDD, val);
+
+ ssacd |= (0x6 << 4);
+
+ dev_dbg(&ssp->pdev->dev,
+ "Using SSACDD %x to supply %uHz\n",
+ val, freq_out);
+ break;
+ }
+
+ return -EINVAL;
+ }
+
+ pxa_ssp_write_reg(ssp, SSACD, ssacd);
+
+ return 0;
+}
+
+/*
+ * Set the active slots in TDM/Network mode
+ */
+static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ u32 sscr0;
+
+ sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+ sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS);
+
+ /* set slot width */
+ if (slot_width > 16)
+ sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16);
+ else
+ sscr0 |= SSCR0_DataSize(slot_width);
+
+ if (slots > 1) {
+ /* enable network mode */
+ sscr0 |= SSCR0_MOD;
+
+ /* set number of active slots */
+ sscr0 |= SSCR0_SlotsPerFrm(slots);
+
+ /* set active slot mask */
+ pxa_ssp_write_reg(ssp, SSTSA, tx_mask);
+ pxa_ssp_write_reg(ssp, SSRSA, rx_mask);
+ }
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+
+ return 0;
+}
+
+/*
+ * Tristate the SSP DAI lines
+ */
+static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
+ int tristate)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ u32 sscr1;
+
+ sscr1 = pxa_ssp_read_reg(ssp, SSCR1);
+ if (tristate)
+ sscr1 &= ~SSCR1_TTE;
+ else
+ sscr1 |= SSCR1_TTE;
+ pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+
+ return 0;
+}
+
+/*
+ * Set up the SSP DAI format.
+ * The SSP Port must be inactive before calling this function as the
+ * physical interface format is changed.
+ */
+static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ u32 sscr0, sscr1, sspsp, scfr;
+
+ /* check if we need to change anything at all */
+ if (priv->dai_fmt == fmt)
+ return 0;
+
+ /* we can only change the settings if the port is not in use */
+ if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) {
+ dev_err(&ssp->pdev->dev,
+ "can't change hardware dai format: stream is in use");
+ return -EINVAL;
+ }
+
+ /* reset port settings */
+ sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
+ ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
+ sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
+ sspsp = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ sscr1 |= SSCR1_SCLKDIR | SSCR1_SCFR;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sspsp |= SSPSP_SFRMP;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ sspsp |= SSPSP_SCMODE(2);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ sscr0 |= SSCR0_PSP;
+ sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
+ /* See hw_params() */
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A:
+ sspsp |= SSPSP_FSRT;
+ case SND_SOC_DAIFMT_DSP_B:
+ sscr0 |= SSCR0_MOD | SSCR0_PSP;
+ sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+ pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+ pxa_ssp_write_reg(ssp, SSPSP, sspsp);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR;
+ pxa_ssp_write_reg(ssp, SSCR1, scfr);
+
+ while (pxa_ssp_read_reg(ssp, SSSR) & SSSR_BSY)
+ cpu_relax();
+ break;
+ }
+
+ dump_registers(ssp);
+
+ /* Since we are configuring the timings for the format by hand
+ * we have to defer some things until hw_params() where we
+ * know parameters like the sample size.
+ */
+ priv->dai_fmt = fmt;
+
+ return 0;
+}
+
+/*
+ * Set the SSP audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ int chn = params_channels(params);
+ u32 sscr0;
+ u32 sspsp;
+ int width = snd_pcm_format_physical_width(params_format(params));
+ int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
+ struct snd_dmaengine_dai_dma_data *dma_data;
+
+ dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
+
+ /* Network mode with one active slot (ttsa == 1) can be used
+ * to force 16-bit frame width on the wire (for S16_LE), even
+ * with two channels. Use 16-bit DMA transfers for this case.
+ */
+ pxa_ssp_set_dma_params(ssp,
+ ((chn == 2) && (ttsa != 1)) || (width == 32),
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK, dma_data);
+
+ /* we can only change the settings if the port is not in use */
+ if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
+ return 0;
+
+ /* clear selected SSP bits */
+ sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ if (ssp->type == PXA3xx_SSP)
+ sscr0 |= SSCR0_FPCKE;
+ sscr0 |= SSCR0_DataSize(16);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
+ break;
+ }
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+
+ switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ sspsp = pxa_ssp_read_reg(ssp, SSPSP);
+
+ if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) {
+ /* This is a special case where the bitclk is 64fs
+ * and we're not dealing with 2*32 bits of audio
+ * samples.
+ *
+ * The SSP values used for that are all found out by
+ * trying and failing a lot; some of the registers
+ * needed for that mode are only available on PXA3xx.
+ */
+ if (ssp->type != PXA3xx_SSP)
+ return -EINVAL;
+
+ sspsp |= SSPSP_SFRMWDTH(width * 2);
+ sspsp |= SSPSP_SFRMDLY(width * 4);
+ sspsp |= SSPSP_EDMYSTOP(3);
+ sspsp |= SSPSP_DMYSTOP(3);
+ sspsp |= SSPSP_DMYSTRT(1);
+ } else {
+ /* The frame width is the width the LRCLK is
+ * asserted for; the delay is expressed in
+ * half cycle units. We need the extra cycle
+ * because the data starts clocking out one BCLK
+ * after LRCLK changes polarity.
+ */
+ sspsp |= SSPSP_SFRMWDTH(width + 1);
+ sspsp |= SSPSP_SFRMDLY((width + 1) * 2);
+ sspsp |= SSPSP_DMYSTRT(1);
+ }
+
+ pxa_ssp_write_reg(ssp, SSPSP, sspsp);
+ break;
+ default:
+ break;
+ }
+
+ /* When we use a network mode, we always require TDM slots
+ * - complain loudly and fail if they've not been set up yet.
+ */
+ if ((sscr0 & SSCR0_MOD) && !ttsa) {
+ dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
+ return -EINVAL;
+ }
+
+ dump_registers(ssp);
+
+ return 0;
+}
+
+static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream,
+ struct ssp_device *ssp, int value)
+{
+ uint32_t sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+ uint32_t sscr1 = pxa_ssp_read_reg(ssp, SSCR1);
+ uint32_t sspsp = pxa_ssp_read_reg(ssp, SSPSP);
+ uint32_t sssr = pxa_ssp_read_reg(ssp, SSSR);
+
+ if (value && (sscr0 & SSCR0_SSE))
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (value)
+ sscr1 |= SSCR1_TSRE;
+ else
+ sscr1 &= ~SSCR1_TSRE;
+ } else {
+ if (value)
+ sscr1 |= SSCR1_RSRE;
+ else
+ sscr1 &= ~SSCR1_RSRE;
+ }
+
+ pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+
+ if (value) {
+ pxa_ssp_write_reg(ssp, SSSR, sssr);
+ pxa_ssp_write_reg(ssp, SSPSP, sspsp);
+ pxa_ssp_write_reg(ssp, SSCR0, sscr0 | SSCR0_SSE);
+ }
+}
+
+static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *cpu_dai)
+{
+ int ret = 0;
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *ssp = priv->ssp;
+ int val;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_RESUME:
+ pxa_ssp_enable(ssp);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pxa_ssp_set_running_bit(substream, ssp, 1);
+ val = pxa_ssp_read_reg(ssp, SSSR);
+ pxa_ssp_write_reg(ssp, SSSR, val);
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ pxa_ssp_set_running_bit(substream, ssp, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pxa_ssp_set_running_bit(substream, ssp, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ pxa_ssp_disable(ssp);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pxa_ssp_set_running_bit(substream, ssp, 0);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ dump_registers(ssp);
+
+ return ret;
+}
+
+static int pxa_ssp_probe(struct snd_soc_dai *dai)
+{
+ struct device *dev = dai->dev;
+ struct ssp_priv *priv;
+ int ret;
+
+ priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ if (dev->of_node) {
+ struct device_node *ssp_handle;
+
+ ssp_handle = of_parse_phandle(dev->of_node, "port", 0);
+ if (!ssp_handle) {
+ dev_err(dev, "unable to get 'port' phandle\n");
+ ret = -ENODEV;
+ goto err_priv;
+ }
+
+ priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio");
+ if (priv->ssp == NULL) {
+ ret = -ENODEV;
+ goto err_priv;
+ }
+ } else {
+ priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
+ if (priv->ssp == NULL) {
+ ret = -ENODEV;
+ goto err_priv;
+ }
+ }
+
+ priv->dai_fmt = (unsigned int) -1;
+ snd_soc_dai_set_drvdata(dai, priv);
+
+ return 0;
+
+err_priv:
+ kfree(priv);
+ return ret;
+}
+
+static int pxa_ssp_remove(struct snd_soc_dai *dai)
+{
+ struct ssp_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ pxa_ssp_free(priv->ssp);
+ kfree(priv);
+ return 0;
+}
+
+#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
+ .startup = pxa_ssp_startup,
+ .shutdown = pxa_ssp_shutdown,
+ .trigger = pxa_ssp_trigger,
+ .hw_params = pxa_ssp_hw_params,
+ .set_sysclk = pxa_ssp_set_dai_sysclk,
+ .set_clkdiv = pxa_ssp_set_dai_clkdiv,
+ .set_pll = pxa_ssp_set_dai_pll,
+ .set_fmt = pxa_ssp_set_dai_fmt,
+ .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+ .set_tristate = pxa_ssp_set_dai_tristate,
+};
+
+static struct snd_soc_dai_driver pxa_ssp_dai = {
+ .probe = pxa_ssp_probe,
+ .remove = pxa_ssp_remove,
+ .suspend = pxa_ssp_suspend,
+ .resume = pxa_ssp_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = PXA_SSP_RATES,
+ .formats = PXA_SSP_FORMATS,
+ },
+ .ops = &pxa_ssp_dai_ops,
+};
+
+static const struct snd_soc_component_driver pxa_ssp_component = {
+ .name = "pxa-ssp",
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pxa_ssp_of_ids[] = {
+ { .compatible = "mrvl,pxa-ssp-dai" },
+ {}
+};
+#endif
+
+static int asoc_ssp_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_component(&pdev->dev, &pxa_ssp_component,
+ &pxa_ssp_dai, 1);
+}
+
+static int asoc_ssp_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_component(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver asoc_ssp_driver = {
+ .driver = {
+ .name = "pxa-ssp-dai",
+ .of_match_table = of_match_ptr(pxa_ssp_of_ids),
+ },
+
+ .probe = asoc_ssp_probe,
+ .remove = asoc_ssp_remove,
+};
+
+module_platform_driver(asoc_ssp_driver);
+
+/* Module information */
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/pxa-ssp.h b/kernel/sound/soc/pxa/pxa-ssp.h
new file mode 100644
index 000000000..bc79da221
--- /dev/null
+++ b/kernel/sound/soc/pxa/pxa-ssp.h
@@ -0,0 +1,45 @@
+/*
+ * ASoC PXA SSP port support
+ *
+ * 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 _PXA_SSP_H
+#define _PXA_SSP_H
+
+/* pxa DAI SSP IDs */
+#define PXA_DAI_SSP1 0
+#define PXA_DAI_SSP2 1
+#define PXA_DAI_SSP3 2
+#define PXA_DAI_SSP4 3
+
+/* SSP clock sources */
+#define PXA_SSP_CLK_PLL 0
+#define PXA_SSP_CLK_EXT 1
+#define PXA_SSP_CLK_NET 2
+#define PXA_SSP_CLK_AUDIO 3
+#define PXA_SSP_CLK_NET_PLL 4
+
+/* SSP audio dividers */
+#define PXA_SSP_AUDIO_DIV_ACDS 0
+#define PXA_SSP_AUDIO_DIV_SCDB 1
+#define PXA_SSP_DIV_SCR 2
+
+/* SSP ACDS audio dividers values */
+#define PXA_SSP_CLK_AUDIO_DIV_1 0
+#define PXA_SSP_CLK_AUDIO_DIV_2 1
+#define PXA_SSP_CLK_AUDIO_DIV_4 2
+#define PXA_SSP_CLK_AUDIO_DIV_8 3
+#define PXA_SSP_CLK_AUDIO_DIV_16 4
+#define PXA_SSP_CLK_AUDIO_DIV_32 5
+
+/* SSP divider bypass */
+#define PXA_SSP_CLK_SCDB_4 0
+#define PXA_SSP_CLK_SCDB_1 1
+#define PXA_SSP_CLK_SCDB_8 2
+
+#define PXA_SSP_PLL_OUT 0
+
+#endif
diff --git a/kernel/sound/soc/pxa/pxa2xx-ac97.c b/kernel/sound/soc/pxa/pxa2xx-ac97.c
new file mode 100644
index 000000000..1f6054650
--- /dev/null
+++ b/kernel/sound/soc/pxa/pxa2xx-ac97.c
@@ -0,0 +1,274 @@
+/*
+ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip.
+ *
+ * Author: Nicolas Pitre
+ * Created: Dec 02, 2004
+ * Copyright: MontaVista Software 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.
+ */
+
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/ac97_codec.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+
+#include <mach/hardware.h>
+#include <mach/regs-ac97.h>
+#include <mach/audio.h>
+
+#include "pxa2xx-ac97.h"
+
+static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ pxa2xx_ac97_try_warm_reset(ac97);
+
+ pxa2xx_ac97_finish_reset(ac97);
+}
+
+static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ pxa2xx_ac97_try_cold_reset(ac97);
+
+ pxa2xx_ac97_finish_reset(ac97);
+}
+
+static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
+ .read = pxa2xx_ac97_read,
+ .write = pxa2xx_ac97_write,
+ .warm_reset = pxa2xx_ac97_warm_reset,
+ .reset = pxa2xx_ac97_cold_reset,
+};
+
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
+ .addr = __PREG(PCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_ac97_pcm_stereo_in_req,
+};
+
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
+ .addr = __PREG(PCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_ac97_pcm_stereo_out_req,
+};
+
+static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
+ .addr = __PREG(MODR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req,
+};
+
+static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
+ .addr = __PREG(MODR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req,
+};
+
+static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
+ .addr = __PREG(MCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req,
+};
+
+static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_dmaengine_dai_dma_data *dma_data;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dma_data = &pxa2xx_ac97_pcm_stereo_out;
+ else
+ dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
+ return 0;
+}
+
+static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_dmaengine_dai_dma_data *dma_data;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+ else
+ dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
+ return 0;
+}
+
+static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+ else
+ snd_soc_dai_set_dma_data(cpu_dai, substream,
+ &pxa2xx_ac97_pcm_mic_mono_in);
+
+ return 0;
+}
+
+#define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
+ .hw_params = pxa2xx_ac97_hw_params,
+};
+
+static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
+ .hw_params = pxa2xx_ac97_hw_aux_params,
+};
+
+static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
+ .hw_params = pxa2xx_ac97_hw_mic_params,
+};
+
+/*
+ * There is only 1 physical AC97 interface for pxa2xx, but it
+ * has extra fifo's that can be used for aux DACs and ADCs.
+ */
+static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = {
+{
+ .name = "pxa2xx-ac97",
+ .bus_control = true,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &pxa_ac97_hifi_dai_ops,
+},
+{
+ .name = "pxa2xx-ac97-aux",
+ .bus_control = true,
+ .playback = {
+ .stream_name = "AC97 Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Aux Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &pxa_ac97_aux_dai_ops,
+},
+{
+ .name = "pxa2xx-ac97-mic",
+ .bus_control = true,
+ .capture = {
+ .stream_name = "AC97 Mic Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &pxa_ac97_mic_dai_ops,
+},
+};
+
+static const struct snd_soc_component_driver pxa_ac97_component = {
+ .name = "pxa-ac97",
+};
+
+static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ if (pdev->id != -1) {
+ dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
+ return -ENXIO;
+ }
+
+ ret = pxa2xx_ac97_hw_probe(pdev);
+ if (ret) {
+ dev_err(&pdev->dev, "PXA2xx AC97 hw probe error (%d)\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
+ if (ret != 0)
+ return ret;
+
+ /* Punt most of the init to the SoC probe; we may need the machine
+ * driver to do interesting things with the clocking to get us up
+ * and running.
+ */
+ return snd_soc_register_component(&pdev->dev, &pxa_ac97_component,
+ pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver));
+}
+
+static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_component(&pdev->dev);
+ snd_soc_set_ac97_ops(NULL);
+ pxa2xx_ac97_hw_remove(pdev);
+ return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int pxa2xx_ac97_dev_suspend(struct device *dev)
+{
+ return pxa2xx_ac97_hw_suspend();
+}
+
+static int pxa2xx_ac97_dev_resume(struct device *dev)
+{
+ return pxa2xx_ac97_hw_resume();
+}
+
+static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops,
+ pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume);
+#endif
+
+static struct platform_driver pxa2xx_ac97_driver = {
+ .probe = pxa2xx_ac97_dev_probe,
+ .remove = pxa2xx_ac97_dev_remove,
+ .driver = {
+ .name = "pxa2xx-ac97",
+#ifdef CONFIG_PM_SLEEP
+ .pm = &pxa2xx_ac97_pm_ops,
+#endif
+ },
+};
+
+module_platform_driver(pxa2xx_ac97_driver);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/pxa2xx-ac97.h b/kernel/sound/soc/pxa/pxa2xx-ac97.h
new file mode 100644
index 000000000..a49c21ba3
--- /dev/null
+++ b/kernel/sound/soc/pxa/pxa2xx-ac97.h
@@ -0,0 +1,17 @@
+/*
+ * linux/sound/soc/pxa/pxa2xx-ac97.h
+ *
+ * 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 _PXA2XX_AC97_H
+#define _PXA2XX_AC97_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_AC97_HIFI 0
+#define PXA2XX_DAI_AC97_AUX 1
+#define PXA2XX_DAI_AC97_MIC 2
+
+#endif
diff --git a/kernel/sound/soc/pxa/pxa2xx-i2s.c b/kernel/sound/soc/pxa/pxa2xx-i2s.c
new file mode 100644
index 000000000..e68290c15
--- /dev/null
+++ b/kernel/sound/soc/pxa/pxa2xx-i2s.c
@@ -0,0 +1,407 @@
+/*
+ * pxa2xx-i2s.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * lrg@slimlogic.co.uk
+ *
+ * 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.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+
+#include <mach/hardware.h>
+#include <mach/audio.h>
+
+#include "pxa2xx-i2s.h"
+
+/*
+ * I2S Controller Register and Bit Definitions
+ */
+#define SACR0 __REG(0x40400000) /* Global Control Register */
+#define SACR1 __REG(0x40400004) /* Serial Audio I 2 S/MSB-Justified Control Register */
+#define SASR0 __REG(0x4040000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */
+#define SAIMR __REG(0x40400014) /* Serial Audio Interrupt Mask Register */
+#define SAICR __REG(0x40400018) /* Serial Audio Interrupt Clear Register */
+#define SADIV __REG(0x40400060) /* Audio Clock Divider Register. */
+#define SADR __REG(0x40400080) /* Serial Audio Data Register (TX and RX FIFO access Register). */
+
+#define SACR0_RFTH(x) ((x) << 12) /* Rx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_TFTH(x) ((x) << 8) /* Tx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */
+#define SACR0_EFWR (1 << 4) /* Enable EFWR Function */
+#define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */
+#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */
+#define SACR0_ENB (1 << 0) /* Enable I2S Link */
+#define SACR1_ENLBF (1 << 5) /* Enable Loopback */
+#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */
+#define SACR1_DREC (1 << 3) /* Disable Recording Function */
+#define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */
+
+#define SASR0_I2SOFF (1 << 7) /* Controller Status */
+#define SASR0_ROR (1 << 6) /* Rx FIFO Overrun */
+#define SASR0_TUR (1 << 5) /* Tx FIFO Underrun */
+#define SASR0_RFS (1 << 4) /* Rx FIFO Service Request */
+#define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */
+#define SASR0_BSY (1 << 2) /* I2S Busy */
+#define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */
+#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */
+
+#define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */
+#define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */
+
+#define SAIMR_ROR (1 << 6) /* Enable Rx FIFO Overrun Condition Interrupt */
+#define SAIMR_TUR (1 << 5) /* Enable Tx FIFO Underrun Condition Interrupt */
+#define SAIMR_RFS (1 << 4) /* Enable Rx FIFO Service Interrupt */
+#define SAIMR_TFS (1 << 3) /* Enable Tx FIFO Service Interrupt */
+
+struct pxa_i2s_port {
+ u32 sadiv;
+ u32 sacr0;
+ u32 sacr1;
+ u32 saimr;
+ int master;
+ u32 fmt;
+};
+static struct pxa_i2s_port pxa_i2s;
+static struct clk *clk_i2s;
+static int clk_ena = 0;
+
+static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = {
+ .addr = __PREG(SADR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_i2s_pcm_stereo_out_req,
+};
+
+static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = {
+ .addr = __PREG(SADR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_i2s_pcm_stereo_in_req,
+};
+
+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ if (IS_ERR(clk_i2s))
+ return PTR_ERR(clk_i2s);
+
+ if (!cpu_dai->active)
+ SACR0 = 0;
+
+ return 0;
+}
+
+/* wait for I2S controller to be ready */
+static int pxa_i2s_wait(void)
+{
+ int i;
+
+ /* flush the Rx FIFO */
+ for(i = 0; i < 16; i++)
+ SADR;
+ return 0;
+}
+
+static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ pxa_i2s.fmt = 0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ pxa_i2s.fmt = SACR1_AMSL;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ pxa_i2s.master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ pxa_i2s.master = 0;
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ if (clk_id != PXA2XX_I2S_SYSCLK)
+ return -ENODEV;
+
+ return 0;
+}
+
+static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_dmaengine_dai_dma_data *dma_data;
+
+ if (WARN_ON(IS_ERR(clk_i2s)))
+ return -EINVAL;
+ clk_prepare_enable(clk_i2s);
+ clk_ena = 1;
+ pxa_i2s_wait();
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dma_data = &pxa2xx_i2s_pcm_stereo_out;
+ else
+ dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+ snd_soc_dai_set_dma_data(dai, substream, dma_data);
+
+ /* is port used by another stream */
+ if (!(SACR0 & SACR0_ENB)) {
+ SACR0 = 0;
+ if (pxa_i2s.master)
+ SACR0 |= SACR0_BCKD;
+
+ SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1);
+ SACR1 |= pxa_i2s.fmt;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ SAIMR |= SAIMR_TFS;
+ else
+ SAIMR |= SAIMR_RFS;
+
+ switch (params_rate(params)) {
+ case 8000:
+ SADIV = 0x48;
+ break;
+ case 11025:
+ SADIV = 0x34;
+ break;
+ case 16000:
+ SADIV = 0x24;
+ break;
+ case 22050:
+ SADIV = 0x1a;
+ break;
+ case 44100:
+ SADIV = 0xd;
+ break;
+ case 48000:
+ SADIV = 0xc;
+ break;
+ case 96000: /* not in manual and possibly slightly inaccurate */
+ SADIV = 0x6;
+ break;
+ }
+
+ return 0;
+}
+
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ SACR1 &= ~SACR1_DRPL;
+ else
+ SACR1 &= ~SACR1_DREC;
+ SACR0 |= SACR0_ENB;
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ SACR1 |= SACR1_DRPL;
+ SAIMR &= ~SAIMR_TFS;
+ } else {
+ SACR1 |= SACR1_DREC;
+ SAIMR &= ~SAIMR_RFS;
+ }
+
+ if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
+ SACR0 &= ~SACR0_ENB;
+ pxa_i2s_wait();
+ if (clk_ena) {
+ clk_disable_unprepare(clk_i2s);
+ clk_ena = 0;
+ }
+ }
+}
+
+#ifdef CONFIG_PM
+static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
+{
+ /* store registers */
+ pxa_i2s.sacr0 = SACR0;
+ pxa_i2s.sacr1 = SACR1;
+ pxa_i2s.saimr = SAIMR;
+ pxa_i2s.sadiv = SADIV;
+
+ /* deactivate link */
+ SACR0 &= ~SACR0_ENB;
+ pxa_i2s_wait();
+ return 0;
+}
+
+static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
+{
+ pxa_i2s_wait();
+
+ SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB;
+ SACR1 = pxa_i2s.sacr1;
+ SAIMR = pxa_i2s.saimr;
+ SADIV = pxa_i2s.sadiv;
+
+ SACR0 = pxa_i2s.sacr0;
+
+ return 0;
+}
+
+#else
+#define pxa2xx_i2s_suspend NULL
+#define pxa2xx_i2s_resume NULL
+#endif
+
+static int pxa2xx_i2s_probe(struct snd_soc_dai *dai)
+{
+ clk_i2s = clk_get(dai->dev, "I2SCLK");
+ if (IS_ERR(clk_i2s))
+ return PTR_ERR(clk_i2s);
+
+ /*
+ * PXA Developer's Manual:
+ * If SACR0[ENB] is toggled in the middle of a normal operation,
+ * the SACR0[RST] bit must also be set and cleared to reset all
+ * I2S controller registers.
+ */
+ SACR0 = SACR0_RST;
+ SACR0 = 0;
+ /* Make sure RPL and REC are disabled */
+ SACR1 = SACR1_DRPL | SACR1_DREC;
+ /* Along with FIFO servicing */
+ SAIMR &= ~(SAIMR_RFS | SAIMR_TFS);
+
+ return 0;
+}
+
+static int pxa2xx_i2s_remove(struct snd_soc_dai *dai)
+{
+ clk_put(clk_i2s);
+ clk_i2s = ERR_PTR(-ENOENT);
+ return 0;
+}
+
+#define PXA2XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+static const struct snd_soc_dai_ops pxa_i2s_dai_ops = {
+ .startup = pxa2xx_i2s_startup,
+ .shutdown = pxa2xx_i2s_shutdown,
+ .trigger = pxa2xx_i2s_trigger,
+ .hw_params = pxa2xx_i2s_hw_params,
+ .set_fmt = pxa2xx_i2s_set_dai_fmt,
+ .set_sysclk = pxa2xx_i2s_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver pxa_i2s_dai = {
+ .probe = pxa2xx_i2s_probe,
+ .remove = pxa2xx_i2s_remove,
+ .suspend = pxa2xx_i2s_suspend,
+ .resume = pxa2xx_i2s_resume,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &pxa_i2s_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static const struct snd_soc_component_driver pxa_i2s_component = {
+ .name = "pxa-i2s",
+};
+
+static int pxa2xx_i2s_drv_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_component(&pdev->dev, &pxa_i2s_component,
+ &pxa_i2s_dai, 1);
+}
+
+static int pxa2xx_i2s_drv_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_component(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver pxa2xx_i2s_driver = {
+ .probe = pxa2xx_i2s_drv_probe,
+ .remove = pxa2xx_i2s_drv_remove,
+
+ .driver = {
+ .name = "pxa2xx-i2s",
+ },
+};
+
+static int __init pxa2xx_i2s_init(void)
+{
+ clk_i2s = ERR_PTR(-ENOENT);
+ return platform_driver_register(&pxa2xx_i2s_driver);
+}
+
+static void __exit pxa2xx_i2s_exit(void)
+{
+ platform_driver_unregister(&pxa2xx_i2s_driver);
+}
+
+module_init(pxa2xx_i2s_init);
+module_exit(pxa2xx_i2s_exit);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
+MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa2xx-i2s");
diff --git a/kernel/sound/soc/pxa/pxa2xx-i2s.h b/kernel/sound/soc/pxa/pxa2xx-i2s.h
new file mode 100644
index 000000000..070f3c605
--- /dev/null
+++ b/kernel/sound/soc/pxa/pxa2xx-i2s.h
@@ -0,0 +1,18 @@
+/*
+ * linux/sound/soc/pxa/pxa2xx-i2s.h
+ *
+ * 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 _PXA2XX_I2S_H
+#define _PXA2XX_I2S_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_I2S 0
+
+/* I2S clock */
+#define PXA2XX_I2S_SYSCLK 0
+
+#endif
diff --git a/kernel/sound/soc/pxa/pxa2xx-pcm.c b/kernel/sound/soc/pxa/pxa2xx-pcm.c
new file mode 100644
index 000000000..a51c9da66
--- /dev/null
+++ b/kernel/sound/soc/pxa/pxa2xx-pcm.c
@@ -0,0 +1,157 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: (C) 2004 MontaVista Software, 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.
+ */
+
+#include <linux/dma-mapping.h>
+#include <linux/module.h>
+#include <linux/dmaengine.h>
+#include <linux/of.h>
+
+#include <mach/dma.h>
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "../../arm/pxa2xx-pcm.h"
+
+static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_dmaengine_dai_dma_data *dma;
+ int ret;
+
+ dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
+ if (!dma)
+ return 0;
+
+ /* this may get called several times by oss emulation
+ * with different params */
+ if (prtd->params == NULL) {
+ prtd->params = dma;
+ ret = pxa_request_dma("name", DMA_PRIO_LOW,
+ pxa2xx_pcm_dma_irq, substream);
+ if (ret < 0)
+ return ret;
+ prtd->dma_ch = ret;
+ } else if (prtd->params != dma) {
+ pxa_free_dma(prtd->dma_ch);
+ prtd->params = dma;
+ ret = pxa_request_dma("name", DMA_PRIO_LOW,
+ pxa2xx_pcm_dma_irq, substream);
+ if (ret < 0)
+ return ret;
+ prtd->dma_ch = ret;
+ }
+
+ return __pxa2xx_pcm_hw_params(substream, params);
+}
+
+static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+
+ __pxa2xx_pcm_hw_free(substream);
+
+ if (prtd->dma_ch >= 0) {
+ pxa_free_dma(prtd->dma_ch);
+ prtd->dma_ch = -1;
+ prtd->params = NULL;
+ }
+
+ return 0;
+}
+
+static struct snd_pcm_ops pxa2xx_pcm_ops = {
+ .open = __pxa2xx_pcm_open,
+ .close = __pxa2xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pxa2xx_pcm_hw_params,
+ .hw_free = pxa2xx_pcm_hw_free,
+ .prepare = __pxa2xx_pcm_prepare,
+ .trigger = pxa2xx_pcm_trigger,
+ .pointer = pxa2xx_pcm_pointer,
+ .mmap = pxa2xx_pcm_mmap,
+};
+
+static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret;
+
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+static struct snd_soc_platform_driver pxa2xx_soc_platform = {
+ .ops = &pxa2xx_pcm_ops,
+ .pcm_new = pxa2xx_soc_pcm_new,
+ .pcm_free = pxa2xx_pcm_free_dma_buffers,
+};
+
+static int pxa2xx_soc_platform_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_platform(&pdev->dev, &pxa2xx_soc_platform);
+}
+
+static int pxa2xx_soc_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id snd_soc_pxa_audio_match[] = {
+ { .compatible = "mrvl,pxa-pcm-audio" },
+ { }
+};
+#endif
+
+static struct platform_driver pxa_pcm_driver = {
+ .driver = {
+ .name = "pxa-pcm-audio",
+ .of_match_table = of_match_ptr(snd_soc_pxa_audio_match),
+ },
+
+ .probe = pxa2xx_soc_platform_probe,
+ .remove = pxa2xx_soc_platform_remove,
+};
+
+module_platform_driver(pxa_pcm_driver);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/raumfeld.c b/kernel/sound/soc/pxa/raumfeld.c
new file mode 100644
index 000000000..552b76300
--- /dev/null
+++ b/kernel/sound/soc/pxa/raumfeld.c
@@ -0,0 +1,320 @@
+/*
+ * raumfeld_audio.c -- SoC audio for Raumfeld audio devices
+ *
+ * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de>
+ *
+ * based on code from:
+ *
+ * Wolfson Microelectronics PLC.
+ * Openedhand Ltd.
+ * Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.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.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+
+#include "pxa-ssp.h"
+
+#define GPIO_SPDIF_RESET (38)
+#define GPIO_MCLK_RESET (111)
+#define GPIO_CODEC_RESET (120)
+
+static struct i2c_client *max9486_client;
+static struct i2c_board_info max9486_hwmon_info = {
+ I2C_BOARD_INFO("max9485", 0x63),
+};
+
+#define MAX9485_MCLK_FREQ_112896 0x22
+#define MAX9485_MCLK_FREQ_122880 0x23
+#define MAX9485_MCLK_FREQ_225792 0x32
+#define MAX9485_MCLK_FREQ_245760 0x33
+
+static void set_max9485_clk(char clk)
+{
+ i2c_master_send(max9486_client, &clk, 1);
+}
+
+static void raumfeld_enable_audio(bool en)
+{
+ if (en) {
+ gpio_set_value(GPIO_MCLK_RESET, 1);
+
+ /* wait some time to let the clocks become stable */
+ msleep(100);
+
+ gpio_set_value(GPIO_SPDIF_RESET, 1);
+ gpio_set_value(GPIO_CODEC_RESET, 1);
+ } else {
+ gpio_set_value(GPIO_MCLK_RESET, 0);
+ gpio_set_value(GPIO_SPDIF_RESET, 0);
+ gpio_set_value(GPIO_CODEC_RESET, 0);
+ }
+}
+
+/* CS4270 */
+static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ /* set freq to 0 to enable all possible codec sample rates */
+ return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+}
+
+static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ /* set freq to 0 to enable all possible codec sample rates */
+ snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+}
+
+static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 44100:
+ set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+ clk = 11289600;
+ break;
+ case 48000:
+ set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+ clk = 12288000;
+ break;
+ case 88200:
+ set_max9485_clk(MAX9485_MCLK_FREQ_225792);
+ clk = 22579200;
+ break;
+ case 96000:
+ set_max9485_clk(MAX9485_MCLK_FREQ_245760);
+ clk = 24576000;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0);
+ if (ret < 0)
+ return ret;
+
+ /* setup the CPU DAI */
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops raumfeld_cs4270_ops = {
+ .startup = raumfeld_cs4270_startup,
+ .shutdown = raumfeld_cs4270_shutdown,
+ .hw_params = raumfeld_cs4270_hw_params,
+};
+
+static int raumfeld_analog_suspend(struct snd_soc_card *card)
+{
+ raumfeld_enable_audio(false);
+ return 0;
+}
+
+static int raumfeld_analog_resume(struct snd_soc_card *card)
+{
+ raumfeld_enable_audio(true);
+ return 0;
+}
+
+/* AK4104 */
+
+static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0, clk = 0;
+
+ switch (params_rate(params)) {
+ case 44100:
+ set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+ clk = 11289600;
+ break;
+ case 48000:
+ set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+ clk = 12288000;
+ break;
+ case 88200:
+ set_max9485_clk(MAX9485_MCLK_FREQ_225792);
+ clk = 22579200;
+ break;
+ case 96000:
+ set_max9485_clk(MAX9485_MCLK_FREQ_245760);
+ clk = 24576000;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* setup the CPU DAI */
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops raumfeld_ak4104_ops = {
+ .hw_params = raumfeld_ak4104_hw_params,
+};
+
+#define DAI_LINK_CS4270 \
+{ \
+ .name = "CS4270", \
+ .stream_name = "CS4270", \
+ .cpu_dai_name = "pxa-ssp-dai.0", \
+ .platform_name = "pxa-pcm-audio", \
+ .codec_dai_name = "cs4270-hifi", \
+ .codec_name = "cs4270.0-0048", \
+ .dai_fmt = SND_SOC_DAIFMT_I2S | \
+ SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBS_CFS, \
+ .ops = &raumfeld_cs4270_ops, \
+}
+
+#define DAI_LINK_AK4104 \
+{ \
+ .name = "ak4104", \
+ .stream_name = "Playback", \
+ .cpu_dai_name = "pxa-ssp-dai.1", \
+ .codec_dai_name = "ak4104-hifi", \
+ .platform_name = "pxa-pcm-audio", \
+ .dai_fmt = SND_SOC_DAIFMT_I2S | \
+ SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBS_CFS, \
+ .ops = &raumfeld_ak4104_ops, \
+ .codec_name = "spi0.0", \
+}
+
+static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] =
+{
+ DAI_LINK_CS4270,
+ DAI_LINK_AK4104,
+};
+
+static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] =
+{
+ DAI_LINK_CS4270,
+};
+
+static struct snd_soc_card snd_soc_raumfeld_connector = {
+ .name = "Raumfeld Connector",
+ .owner = THIS_MODULE,
+ .dai_link = snd_soc_raumfeld_connector_dai,
+ .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai),
+ .suspend_post = raumfeld_analog_suspend,
+ .resume_pre = raumfeld_analog_resume,
+};
+
+static struct snd_soc_card snd_soc_raumfeld_speaker = {
+ .name = "Raumfeld Speaker",
+ .owner = THIS_MODULE,
+ .dai_link = snd_soc_raumfeld_speaker_dai,
+ .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai),
+ .suspend_post = raumfeld_analog_suspend,
+ .resume_pre = raumfeld_analog_resume,
+};
+
+static struct platform_device *raumfeld_audio_device;
+
+static int __init raumfeld_audio_init(void)
+{
+ int ret;
+
+ if (!machine_is_raumfeld_speaker() &&
+ !machine_is_raumfeld_connector())
+ return 0;
+
+ max9486_client = i2c_new_device(i2c_get_adapter(0),
+ &max9486_hwmon_info);
+
+ if (!max9486_client)
+ return -ENOMEM;
+
+ set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+
+ /* Register analog device */
+ raumfeld_audio_device = platform_device_alloc("soc-audio", 0);
+ if (!raumfeld_audio_device)
+ return -ENOMEM;
+
+ if (machine_is_raumfeld_speaker())
+ platform_set_drvdata(raumfeld_audio_device,
+ &snd_soc_raumfeld_speaker);
+
+ if (machine_is_raumfeld_connector())
+ platform_set_drvdata(raumfeld_audio_device,
+ &snd_soc_raumfeld_connector);
+
+ ret = platform_device_add(raumfeld_audio_device);
+ if (ret < 0) {
+ platform_device_put(raumfeld_audio_device);
+ return ret;
+ }
+
+ raumfeld_enable_audio(true);
+ return 0;
+}
+
+static void __exit raumfeld_audio_exit(void)
+{
+ raumfeld_enable_audio(false);
+
+ platform_device_unregister(raumfeld_audio_device);
+
+ i2c_unregister_device(max9486_client);
+
+ gpio_free(GPIO_MCLK_RESET);
+ gpio_free(GPIO_CODEC_RESET);
+ gpio_free(GPIO_SPDIF_RESET);
+}
+
+module_init(raumfeld_audio_init);
+module_exit(raumfeld_audio_exit);
+
+/* Module information */
+MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
+MODULE_DESCRIPTION("Raumfeld audio SoC");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/spitz.c b/kernel/sound/soc/pxa/spitz.c
new file mode 100644
index 000000000..461123ad5
--- /dev/null
+++ b/kernel/sound/soc/pxa/spitz.c
@@ -0,0 +1,346 @@
+/*
+ * spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/spitz.h>
+#include "../codecs/wm8750.h"
+#include "pxa2xx-i2s.h"
+
+#define SPITZ_HP 0
+#define SPITZ_MIC 1
+#define SPITZ_LINE 2
+#define SPITZ_HEADSET 3
+#define SPITZ_HP_OFF 4
+#define SPITZ_SPK_ON 0
+#define SPITZ_SPK_OFF 1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define SPITZ_AUDIO_CLOCK 12288000
+
+static int spitz_jack_func;
+static int spitz_spk_func;
+static int spitz_mic_gpio;
+
+static void spitz_ext_control(struct snd_soc_dapm_context *dapm)
+{
+ snd_soc_dapm_mutex_lock(dapm);
+
+ if (spitz_spk_func == SPITZ_SPK_ON)
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
+
+ /* set up jack connection */
+ switch (spitz_jack_func) {
+ case SPITZ_HP:
+ /* enable and unmute hp jack, disable mic bias */
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
+ break;
+ case SPITZ_MIC:
+ /* enable mic jack and bias, mute hp */
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+ break;
+ case SPITZ_LINE:
+ /* enable line jack, disable mic bias and mute hp */
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+ break;
+ case SPITZ_HEADSET:
+ /* enable and unmute headset jack enable mic bias, mute L hp */
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
+ break;
+ case SPITZ_HP_OFF:
+
+ /* jack removed, everything off */
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+ break;
+ }
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int spitz_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* check the jack status at stream startup */
+ spitz_ext_control(&rtd->card->dapm);
+
+ return 0;
+}
+
+static int spitz_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops spitz_ops = {
+ .startup = spitz_startup,
+ .hw_params = spitz_hw_params,
+};
+
+static int spitz_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spitz_jack_func;
+ return 0;
+}
+
+static int spitz_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (spitz_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ spitz_jack_func = ucontrol->value.integer.value[0];
+ spitz_ext_control(&card->dapm);
+ return 1;
+}
+
+static int spitz_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spitz_spk_func;
+ return 0;
+}
+
+static int spitz_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (spitz_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ spitz_spk_func = ucontrol->value.integer.value[0];
+ spitz_ext_control(&card->dapm);
+ return 1;
+}
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value_cansleep(spitz_mic_gpio, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* spitz machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_LINE("Line Jack", NULL),
+
+ /* headset is a mic and mono headphone */
+ SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Spitz machine audio_map */
+static const struct snd_soc_dapm_route spitz_audio_map[] = {
+
+ /* headphone connected to LOUT1, ROUT1 */
+ {"Headphone Jack", NULL, "LOUT1"},
+ {"Headphone Jack", NULL, "ROUT1"},
+
+ /* headset connected to ROUT1 and LINPUT1 with bias (def below) */
+ {"Headset Jack", NULL, "ROUT1"},
+
+ /* ext speaker connected to LOUT2, ROUT2 */
+ {"Ext Spk", NULL , "ROUT2"},
+ {"Ext Spk", NULL , "LOUT2"},
+
+ /* mic is connected to input 1 - with bias */
+ {"LINPUT1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Mic Jack"},
+
+ /* line is connected to input 1 - no bias */
+ {"LINPUT1", NULL, "Line Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum spitz_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
+ SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack,
+ spitz_set_jack),
+ SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk,
+ spitz_set_spk),
+};
+
+/* spitz digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link spitz_dai = {
+ .name = "wm8750",
+ .stream_name = "WM8750",
+ .cpu_dai_name = "pxa2xx-i2s",
+ .codec_dai_name = "wm8750-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm8750.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &spitz_ops,
+};
+
+/* spitz audio machine driver */
+static struct snd_soc_card snd_soc_spitz = {
+ .name = "Spitz",
+ .owner = THIS_MODULE,
+ .dai_link = &spitz_dai,
+ .num_links = 1,
+
+ .controls = wm8750_spitz_controls,
+ .num_controls = ARRAY_SIZE(wm8750_spitz_controls),
+ .dapm_widgets = wm8750_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
+ .dapm_routes = spitz_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(spitz_audio_map),
+ .fully_routed = true,
+};
+
+static int spitz_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &snd_soc_spitz;
+ int ret;
+
+ if (machine_is_akita())
+ spitz_mic_gpio = AKITA_GPIO_MIC_BIAS;
+ else
+ spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS;
+
+ ret = gpio_request(spitz_mic_gpio, "MIC GPIO");
+ if (ret)
+ goto err1;
+
+ ret = gpio_direction_output(spitz_mic_gpio, 0);
+ if (ret)
+ goto err2;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ goto err2;
+ }
+
+ return 0;
+
+err2:
+ gpio_free(spitz_mic_gpio);
+err1:
+ return ret;
+}
+
+static int spitz_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ gpio_free(spitz_mic_gpio);
+ return 0;
+}
+
+static struct platform_driver spitz_driver = {
+ .driver = {
+ .name = "spitz-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = spitz_probe,
+ .remove = spitz_remove,
+};
+
+module_platform_driver(spitz_driver);
+
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Spitz");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spitz-audio");
diff --git a/kernel/sound/soc/pxa/tosa.c b/kernel/sound/soc/pxa/tosa.c
new file mode 100644
index 000000000..cb49284e8
--- /dev/null
+++ b/kernel/sound/soc/pxa/tosa.c
@@ -0,0 +1,280 @@
+/*
+ * tosa.c -- SoC audio for Tosa
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.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.
+ *
+ * GPIO's
+ * 1 - Jack Insertion
+ * 5 - Hookswitch (headset answer/hang up switch)
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/tosa.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-ac97.h"
+
+#define TOSA_HP 0
+#define TOSA_MIC_INT 1
+#define TOSA_HEADSET 2
+#define TOSA_HP_OFF 3
+#define TOSA_SPK_ON 0
+#define TOSA_SPK_OFF 1
+
+static int tosa_jack_func;
+static int tosa_spk_func;
+
+static void tosa_ext_control(struct snd_soc_dapm_context *dapm)
+{
+
+ snd_soc_dapm_mutex_lock(dapm);
+
+ /* set up jack connection */
+ switch (tosa_jack_func) {
+ case TOSA_HP:
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ break;
+ case TOSA_MIC_INT:
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ break;
+ case TOSA_HEADSET:
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
+ break;
+ }
+
+ if (tosa_spk_func == TOSA_SPK_ON)
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
+ else
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int tosa_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* check the jack status at stream startup */
+ tosa_ext_control(&rtd->card->dapm);
+
+ return 0;
+}
+
+static struct snd_soc_ops tosa_ops = {
+ .startup = tosa_startup,
+};
+
+static int tosa_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = tosa_jack_func;
+ return 0;
+}
+
+static int tosa_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (tosa_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ tosa_jack_func = ucontrol->value.integer.value[0];
+ tosa_ext_control(&card->dapm);
+ return 1;
+}
+
+static int tosa_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = tosa_spk_func;
+ return 0;
+}
+
+static int tosa_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+ if (tosa_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ tosa_spk_func = ucontrol->value.integer.value[0];
+ tosa_ext_control(&card->dapm);
+ return 1;
+}
+
+/* tosa dapm event handlers */
+static int tosa_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 :0);
+ return 0;
+}
+
+/* tosa machine dapm widgets */
+static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* tosa audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* headphone connected to HPOUTL, HPOUTR */
+ {"Headphone Jack", NULL, "HPOUTL"},
+ {"Headphone Jack", NULL, "HPOUTR"},
+
+ /* ext speaker connected to LOUT2, ROUT2 */
+ {"Speaker", NULL, "LOUT2"},
+ {"Speaker", NULL, "ROUT2"},
+
+ /* internal mic is connected to mic1, mic2 differential - with bias */
+ {"MIC1", NULL, "Mic Bias"},
+ {"MIC2", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Mic (Internal)"},
+
+ /* headset is connected to HPOUTR, and LINEINR with bias */
+ {"Headset Jack", NULL, "HPOUTR"},
+ {"LINEINR", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum tosa_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new tosa_controls[] = {
+ SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
+ tosa_set_jack),
+ SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
+ tosa_set_spk),
+};
+
+static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_nc_pin(dapm, "OUT3");
+ snd_soc_dapm_nc_pin(dapm, "MONOOUT");
+
+ return 0;
+}
+
+static struct snd_soc_dai_link tosa_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai_name = "pxa2xx-ac97",
+ .codec_dai_name = "wm9712-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9712-codec",
+ .init = tosa_ac97_init,
+ .ops = &tosa_ops,
+},
+{
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
+ .codec_dai_name = "wm9712-aux",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm9712-codec",
+ .ops = &tosa_ops,
+},
+};
+
+static struct snd_soc_card tosa = {
+ .name = "Tosa",
+ .owner = THIS_MODULE,
+ .dai_link = tosa_dai,
+ .num_links = ARRAY_SIZE(tosa_dai),
+
+ .controls = tosa_controls,
+ .num_controls = ARRAY_SIZE(tosa_controls),
+ .dapm_widgets = tosa_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static int tosa_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &tosa;
+ int ret;
+
+ ret = gpio_request_one(TOSA_GPIO_L_MUTE, GPIOF_OUT_INIT_LOW,
+ "Headphone Jack");
+ if (ret)
+ return ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ gpio_free(TOSA_GPIO_L_MUTE);
+ }
+ return ret;
+}
+
+static int tosa_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ gpio_free(TOSA_GPIO_L_MUTE);
+ snd_soc_unregister_card(card);
+ return 0;
+}
+
+static struct platform_driver tosa_driver = {
+ .driver = {
+ .name = "tosa-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = tosa_probe,
+ .remove = tosa_remove,
+};
+
+module_platform_driver(tosa_driver);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Tosa");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:tosa-audio");
diff --git a/kernel/sound/soc/pxa/ttc-dkb.c b/kernel/sound/soc/pxa/ttc-dkb.c
new file mode 100644
index 000000000..1753c7d9e
--- /dev/null
+++ b/kernel/sound/soc/pxa/ttc-dkb.c
@@ -0,0 +1,163 @@
+/*
+ * linux/sound/soc/pxa/ttc_dkb.c
+ *
+ * Copyright (C) 2012 Marvell International Ltd.
+ *
+ * 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/module.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <sound/pcm_params.h>
+#include "../codecs/88pm860x-codec.h"
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+ { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
+};
+
+/* ttc machine dapm widgets */
+static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+ SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+ SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+ SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+ SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* ttc machine audio map */
+static const struct snd_soc_dapm_route ttc_audio_map[] = {
+ {"Headset Stereophone", NULL, "HS1"},
+ {"Headset Stereophone", NULL, "HS2"},
+
+ {"Ext Speaker", NULL, "LSP"},
+ {"Ext Speaker", NULL, "LSN"},
+
+ {"Lineout Out 1", NULL, "LINEOUT1"},
+ {"Lineout Out 2", NULL, "LINEOUT2"},
+
+ {"MIC1P", NULL, "Mic1 Bias"},
+ {"MIC1N", NULL, "Mic1 Bias"},
+ {"Mic1 Bias", NULL, "Ext Mic 1"},
+
+ {"MIC2P", NULL, "Mic1 Bias"},
+ {"MIC2N", NULL, "Mic1 Bias"},
+ {"Mic1 Bias", NULL, "Headset Mic 2"},
+
+ {"MIC3P", NULL, "Mic3 Bias"},
+ {"MIC3N", NULL, "Mic3 Bias"},
+ {"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+
+ /* Headset jack detection */
+ snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+ &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins));
+ snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE,
+ &mic_jack, mic_jack_pins,
+ ARRAY_SIZE(mic_jack_pins));
+
+ /* headphone, microphone detection & headset short detection */
+ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+ SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+ pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+
+ return 0;
+}
+
+/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
+{
+ .name = "88pm860x i2s",
+ .stream_name = "audio playback",
+ .codec_name = "88pm860x-codec",
+ .platform_name = "mmp-pcm-audio",
+ .cpu_dai_name = "pxa-ssp-dai.1",
+ .codec_dai_name = "88pm860x-i2s",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .init = ttc_pm860x_init,
+},
+};
+
+/* ttc/td audio machine driver */
+static struct snd_soc_card ttc_dkb_card = {
+ .name = "ttc-dkb-hifi",
+ .owner = THIS_MODULE,
+ .dai_link = ttc_pm860x_hifi_dai,
+ .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
+
+ .dapm_widgets = ttc_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
+ .dapm_routes = ttc_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
+};
+
+static int ttc_dkb_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &ttc_dkb_card;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+
+ return ret;
+}
+
+static int ttc_dkb_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver ttc_dkb_driver = {
+ .driver = {
+ .name = "ttc-dkb-audio",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = ttc_dkb_probe,
+ .remove = ttc_dkb_remove,
+};
+
+module_platform_driver(ttc_dkb_driver);
+
+/* Module information */
+MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC TTC DKB");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ttc-dkb-audio");
diff --git a/kernel/sound/soc/pxa/z2.c b/kernel/sound/soc/pxa/z2.c
new file mode 100644
index 000000000..bcbfbe830
--- /dev/null
+++ b/kernel/sound/soc/pxa/z2.c
@@ -0,0 +1,227 @@
+/*
+ * linux/sound/soc/pxa/z2.c
+ *
+ * SoC Audio driver for Aeronix Zipit Z2
+ *
+ * Copyright (C) 2009 Ken McGuire <kenm@desertweyr.com>
+ * Copyright (C) 2010 Marek Vasut <marek.vasut@gmail.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/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
+#include <mach/z2.h>
+
+#include "../codecs/wm8750.h"
+#include "pxa2xx-i2s.h"
+
+static struct snd_soc_card snd_soc_z2;
+
+static int z2_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_jack hs_jack;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ {
+ .pin = "Mic Jack",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Ext Spk",
+ .mask = SND_JACK_HEADPHONE,
+ .invert = 1
+ },
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+ {
+ .gpio = GPIO37_ZIPITZ2_HEADSET_DETECT,
+ .name = "hsdet-gpio",
+ .report = SND_JACK_HEADSET,
+ .debounce_time = 200,
+ .invert = 1,
+ },
+};
+
+/* z2 machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+
+ /* headset is a mic and mono headphone */
+ SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Z2 machine audio_map */
+static const struct snd_soc_dapm_route z2_audio_map[] = {
+
+ /* headphone connected to LOUT1, ROUT1 */
+ {"Headphone Jack", NULL, "LOUT1"},
+ {"Headphone Jack", NULL, "ROUT1"},
+
+ /* ext speaker connected to LOUT2, ROUT2 */
+ {"Ext Spk", NULL , "ROUT2"},
+ {"Ext Spk", NULL , "LOUT2"},
+
+ /* mic is connected to R input 2 - with bias */
+ {"RINPUT2", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Mic Jack"},
+};
+
+/*
+ * Logic for a wm8750 as connected on a Z2 Device
+ */
+static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret;
+
+ /* NC codec pins */
+ snd_soc_dapm_disable_pin(dapm, "LINPUT3");
+ snd_soc_dapm_disable_pin(dapm, "RINPUT3");
+ snd_soc_dapm_disable_pin(dapm, "OUT3");
+ snd_soc_dapm_disable_pin(dapm, "MONO1");
+
+ /* Jack detection API stuff */
+ ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET,
+ &hs_jack, hs_jack_pins,
+ ARRAY_SIZE(hs_jack_pins));
+ if (ret)
+ goto err;
+
+ ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
+ if (ret)
+ goto err;
+
+ return 0;
+
+err:
+ return ret;
+}
+
+static struct snd_soc_ops z2_ops = {
+ .hw_params = z2_hw_params,
+};
+
+/* z2 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link z2_dai = {
+ .name = "wm8750",
+ .stream_name = "WM8750",
+ .cpu_dai_name = "pxa2xx-i2s",
+ .codec_dai_name = "wm8750-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm8750.0-001b",
+ .init = z2_wm8750_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &z2_ops,
+};
+
+/* z2 audio machine driver */
+static struct snd_soc_card snd_soc_z2 = {
+ .name = "Z2",
+ .owner = THIS_MODULE,
+ .dai_link = &z2_dai,
+ .num_links = 1,
+
+ .dapm_widgets = wm8750_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
+ .dapm_routes = z2_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(z2_audio_map),
+};
+
+static struct platform_device *z2_snd_device;
+
+static int __init z2_init(void)
+{
+ int ret;
+
+ if (!machine_is_zipit2())
+ return -ENODEV;
+
+ z2_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!z2_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(z2_snd_device, &snd_soc_z2);
+ ret = platform_device_add(z2_snd_device);
+
+ if (ret)
+ platform_device_put(z2_snd_device);
+
+ return ret;
+}
+
+static void __exit z2_exit(void)
+{
+ platform_device_unregister(z2_snd_device);
+}
+
+module_init(z2_init);
+module_exit(z2_exit);
+
+MODULE_AUTHOR("Ken McGuire <kenm@desertweyr.com>, "
+ "Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC ZipitZ2");
+MODULE_LICENSE("GPL");
diff --git a/kernel/sound/soc/pxa/zylonite.c b/kernel/sound/soc/pxa/zylonite.c
new file mode 100644
index 000000000..8f301c72e
--- /dev/null
+++ b/kernel/sound/soc/pxa/zylonite.c
@@ -0,0 +1,275 @@
+/*
+ * zylonite.c -- SoC audio for Zylonite
+ *
+ * Copyright 2008 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "../codecs/wm9713.h"
+#include "pxa2xx-ac97.h"
+#include "pxa-ssp.h"
+
+/*
+ * There is a physical switch SW15 on the board which changes the MCLK
+ * for the WM9713 between the standard AC97 master clock and the
+ * output of the CLK_POUT signal from the PXA.
+ */
+static int clk_pout;
+module_param(clk_pout, int, 0);
+MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board).");
+
+static struct clk *pout;
+
+static struct snd_soc_card zylonite;
+
+static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Microphone", NULL),
+ SND_SOC_DAPM_MIC("Handset Microphone", NULL),
+ SND_SOC_DAPM_SPK("Multiactor", NULL),
+ SND_SOC_DAPM_SPK("Headset Earpiece", NULL),
+};
+
+/* Currently supported audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* Headphone output connected to HPL/HPR */
+ { "Headphone", NULL, "HPL" },
+ { "Headphone", NULL, "HPR" },
+
+ /* On-board earpiece */
+ { "Headset Earpiece", NULL, "OUT3" },
+
+ /* Headphone mic */
+ { "MIC2A", NULL, "Mic Bias" },
+ { "Mic Bias", NULL, "Headset Microphone" },
+
+ /* On-board mic */
+ { "MIC1", NULL, "Mic Bias" },
+ { "Mic Bias", NULL, "Handset Microphone" },
+
+ /* Multiactor differentially connected over SPKL/SPKR */
+ { "Multiactor", NULL, "SPKL" },
+ { "Multiactor", NULL, "SPKR" },
+};
+
+static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
+{
+ if (clk_pout)
+ snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
+ clk_get_rate(pout), 0);
+
+ return 0;
+}
+
+static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int pll_out = 0;
+ unsigned int wm9713_div = 0;
+ int ret = 0;
+ int rate = params_rate(params);
+ int width = snd_pcm_format_physical_width(params_format(params));
+
+ /* Only support ratios that we can generate neatly from the AC97
+ * based master clock - in particular, this excludes 44.1kHz.
+ * In most applications the voice DAC will be used for telephony
+ * data so multiples of 8kHz will be the common case.
+ */
+ switch (rate) {
+ case 8000:
+ wm9713_div = 12;
+ break;
+ case 16000:
+ wm9713_div = 6;
+ break;
+ case 48000:
+ wm9713_div = 2;
+ break;
+ default:
+ /* Don't support OSS emulation */
+ return -EINVAL;
+ }
+
+ /* Add 1 to the width for the leading clock cycle */
+ pll_out = rate * (width + 1) * 8;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
+ if (ret < 0)
+ return ret;
+
+ if (clk_pout)
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
+ WM9713_PCMDIV(wm9713_div));
+ else
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+ WM9713_PCMDIV(wm9713_div));
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops zylonite_voice_ops = {
+ .hw_params = zylonite_voice_hw_params,
+};
+
+static struct snd_soc_dai_link zylonite_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .codec_name = "wm9713-codec",
+ .platform_name = "pxa-pcm-audio",
+ .cpu_dai_name = "pxa2xx-ac97",
+ .codec_dai_name = "wm9713-hifi",
+ .init = zylonite_wm9713_init,
+},
+{
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .codec_name = "wm9713-codec",
+ .platform_name = "pxa-pcm-audio",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
+ .codec_dai_name = "wm9713-aux",
+},
+{
+ .name = "WM9713 Voice",
+ .stream_name = "WM9713 Voice",
+ .codec_name = "wm9713-codec",
+ .platform_name = "pxa-pcm-audio",
+ .cpu_dai_name = "pxa-ssp-dai.2",
+ .codec_dai_name = "wm9713-voice",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &zylonite_voice_ops,
+},
+};
+
+static int zylonite_probe(struct snd_soc_card *card)
+{
+ int ret;
+
+ if (clk_pout) {
+ pout = clk_get(NULL, "CLK_POUT");
+ if (IS_ERR(pout)) {
+ dev_err(card->dev, "Unable to obtain CLK_POUT: %ld\n",
+ PTR_ERR(pout));
+ return PTR_ERR(pout);
+ }
+
+ ret = clk_enable(pout);
+ if (ret != 0) {
+ dev_err(card->dev, "Unable to enable CLK_POUT: %d\n",
+ ret);
+ clk_put(pout);
+ return ret;
+ }
+
+ dev_dbg(card->dev, "MCLK enabled at %luHz\n",
+ clk_get_rate(pout));
+ }
+
+ return 0;
+}
+
+static int zylonite_remove(struct snd_soc_card *card)
+{
+ if (clk_pout) {
+ clk_disable(pout);
+ clk_put(pout);
+ }
+
+ return 0;
+}
+
+static int zylonite_suspend_post(struct snd_soc_card *card)
+{
+ if (clk_pout)
+ clk_disable(pout);
+
+ return 0;
+}
+
+static int zylonite_resume_pre(struct snd_soc_card *card)
+{
+ int ret = 0;
+
+ if (clk_pout) {
+ ret = clk_enable(pout);
+ if (ret != 0)
+ dev_err(card->dev, "Unable to enable CLK_POUT: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+
+static struct snd_soc_card zylonite = {
+ .name = "Zylonite",
+ .owner = THIS_MODULE,
+ .probe = &zylonite_probe,
+ .remove = &zylonite_remove,
+ .suspend_post = &zylonite_suspend_post,
+ .resume_pre = &zylonite_resume_pre,
+ .dai_link = zylonite_dai,
+ .num_links = ARRAY_SIZE(zylonite_dai),
+
+ .dapm_widgets = zylonite_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(zylonite_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static struct platform_device *zylonite_snd_ac97_device;
+
+static int __init zylonite_init(void)
+{
+ int ret;
+
+ zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+ if (!zylonite_snd_ac97_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(zylonite_snd_ac97_device, &zylonite);
+
+ ret = platform_device_add(zylonite_snd_ac97_device);
+ if (ret != 0)
+ platform_device_put(zylonite_snd_ac97_device);
+
+ return ret;
+}
+
+static void __exit zylonite_exit(void)
+{
+ platform_device_unregister(zylonite_snd_ac97_device);
+}
+
+module_init(zylonite_init);
+module_exit(zylonite_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite");
+MODULE_LICENSE("GPL");