From 9ca8dbcc65cfc63d6f5ef3312a33184e1d726e00 Mon Sep 17 00:00:00 2001 From: Yunhong Jiang Date: Tue, 4 Aug 2015 12:17:53 -0700 Subject: Add the rt linux 4.1.3-rt3 as base Import the rt linux 4.1.3-rt3 as OPNFV kvm base. It's from git://git.kernel.org/pub/scm/linux/kernel/git/rt/linux-rt-devel.git linux-4.1.y-rt and the base is: commit 0917f823c59692d751951bf5ea699a2d1e2f26a2 Author: Sebastian Andrzej Siewior Date: Sat Jul 25 12:13:34 2015 +0200 Prepare v4.1.3-rt3 Signed-off-by: Sebastian Andrzej Siewior We lose all the git history this way and it's not good. We should apply another opnfv project repo in future. Change-Id: I87543d81c9df70d99c5001fbdf646b202c19f423 Signed-off-by: Yunhong Jiang --- kernel/sound/soc/blackfin/bfin-eval-adau1373.c | 183 +++++++++++++++++++++++++ 1 file changed, 183 insertions(+) create mode 100644 kernel/sound/soc/blackfin/bfin-eval-adau1373.c (limited to 'kernel/sound/soc/blackfin/bfin-eval-adau1373.c') diff --git a/kernel/sound/soc/blackfin/bfin-eval-adau1373.c b/kernel/sound/soc/blackfin/bfin-eval-adau1373.c new file mode 100644 index 000000000..523baf582 --- /dev/null +++ b/kernel/sound/soc/blackfin/bfin-eval-adau1373.c @@ -0,0 +1,183 @@ +/* + * Machine driver for EVAL-ADAU1373 on Analog Devices bfin + * evaluation boards. + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2 or later. + */ + +#include +#include +#include +#include +#include +#include + +#include "../codecs/adau1373.h" + +static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line In1", NULL), + SND_SOC_DAPM_LINE("Line In2", NULL), + SND_SOC_DAPM_LINE("Line In3", NULL), + SND_SOC_DAPM_LINE("Line In4", NULL), + + SND_SOC_DAPM_LINE("Line Out1", NULL), + SND_SOC_DAPM_LINE("Line Out2", NULL), + SND_SOC_DAPM_LINE("Stereo Out", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = { + { "AIN1L", NULL, "Line In1" }, + { "AIN1R", NULL, "Line In1" }, + { "AIN2L", NULL, "Line In2" }, + { "AIN2R", NULL, "Line In2" }, + { "AIN3L", NULL, "Line In3" }, + { "AIN3R", NULL, "Line In3" }, + { "AIN4L", NULL, "Line In4" }, + { "AIN4R", NULL, "Line In4" }, + + /* MICBIAS can be connected via a jumper to the line-in jack, since w + don't know which one is going to be used, just power both. */ + { "Line In1", NULL, "MICBIAS1" }, + { "Line In2", NULL, "MICBIAS1" }, + { "Line In3", NULL, "MICBIAS1" }, + { "Line In4", NULL, "MICBIAS1" }, + { "Line In1", NULL, "MICBIAS2" }, + { "Line In2", NULL, "MICBIAS2" }, + { "Line In3", NULL, "MICBIAS2" }, + { "Line In4", NULL, "MICBIAS2" }, + + { "Line Out1", NULL, "LOUT1L" }, + { "Line Out1", NULL, "LOUT1R" }, + { "Line Out2", NULL, "LOUT2L" }, + { "Line Out2", NULL, "LOUT2R" }, + { "Headphone", NULL, "HPL" }, + { "Headphone", NULL, "HPR" }, + { "Earpiece", NULL, "EP" }, + { "Speaker", NULL, "SPKL" }, + { "Stereo Out", NULL, "SPKR" }, +}; + +static int bfin_eval_adau1373_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; + int ret; + int pll_rate; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, + ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, + SND_SOC_CLOCK_IN); + + return ret; +} + +static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int pll_rate = 48000 * 1024; + int ret; + + ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, + ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, + SND_SOC_CLOCK_IN); + + return ret; +} +static struct snd_soc_ops bfin_eval_adau1373_ops = { + .hw_params = bfin_eval_adau1373_hw_params, +}; + +static struct snd_soc_dai_link bfin_eval_adau1373_dai = { + .name = "adau1373", + .stream_name = "adau1373", + .cpu_dai_name = "bfin-i2s.0", + .codec_dai_name = "adau1373-aif1", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "adau1373.0-001a", + .ops = &bfin_eval_adau1373_ops, + .init = bfin_eval_adau1373_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, +}; + +static struct snd_soc_card bfin_eval_adau1373 = { + .name = "bfin-eval-adau1373", + .owner = THIS_MODULE, + .dai_link = &bfin_eval_adau1373_dai, + .num_links = 1, + + .dapm_widgets = bfin_eval_adau1373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets), + .dapm_routes = bfin_eval_adau1373_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1373_dapm_routes), +}; + +static int bfin_eval_adau1373_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &bfin_eval_adau1373; + + card->dev = &pdev->dev; + + return snd_soc_register_card(&bfin_eval_adau1373); +} + +static int bfin_eval_adau1373_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 bfin_eval_adau1373_driver = { + .driver = { + .name = "bfin-eval-adau1373", + .pm = &snd_soc_pm_ops, + }, + .probe = bfin_eval_adau1373_probe, + .remove = bfin_eval_adau1373_remove, +}; + +module_platform_driver(bfin_eval_adau1373_driver); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bfin-eval-adau1373"); -- cgit 1.2.3-korg