From e56c72d5f201044b14191c5b83a25e17f2d68ccf Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 2 Apr 2015 15:37:02 +0800 Subject: ASoC: Intel: create boards folder and move sst boards files in Restructure the sound/soc/intel/ directory: create boards folder, and move sst boards files here. Signed-off-by: Jie Yang Acked-by: Vinod Koul Acked-by: Jarkko Nikula Tested-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 16 +- sound/soc/intel/boards/Makefile | 15 ++ sound/soc/intel/boards/broadwell.c | 292 ++++++++++++++++++++++ sound/soc/intel/boards/byt-max98090.c | 187 ++++++++++++++ sound/soc/intel/boards/byt-rt5640.c | 229 +++++++++++++++++ sound/soc/intel/boards/bytcr_rt5640.c | 227 +++++++++++++++++ sound/soc/intel/boards/cht_bsw_rt5645.c | 324 ++++++++++++++++++++++++ sound/soc/intel/boards/cht_bsw_rt5672.c | 366 +++++++++++++++++++++++++++ sound/soc/intel/boards/haswell.c | 209 ++++++++++++++++ sound/soc/intel/boards/mfld_machine.c | 430 ++++++++++++++++++++++++++++++++ sound/soc/intel/broadwell.c | 292 ---------------------- sound/soc/intel/byt-max98090.c | 187 -------------- sound/soc/intel/byt-rt5640.c | 229 ----------------- sound/soc/intel/bytcr_dpcm_rt5640.c | 227 ----------------- sound/soc/intel/cht_bsw_rt5645.c | 324 ------------------------ sound/soc/intel/cht_bsw_rt5672.c | 366 --------------------------- sound/soc/intel/haswell.c | 209 ---------------- sound/soc/intel/mfld_machine.c | 430 -------------------------------- 18 files changed, 2280 insertions(+), 2279 deletions(-) create mode 100644 sound/soc/intel/boards/Makefile create mode 100644 sound/soc/intel/boards/broadwell.c create mode 100644 sound/soc/intel/boards/byt-max98090.c create mode 100644 sound/soc/intel/boards/byt-rt5640.c create mode 100644 sound/soc/intel/boards/bytcr_rt5640.c create mode 100644 sound/soc/intel/boards/cht_bsw_rt5645.c create mode 100644 sound/soc/intel/boards/cht_bsw_rt5672.c create mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/mfld_machine.c delete mode 100644 sound/soc/intel/broadwell.c delete mode 100644 sound/soc/intel/byt-max98090.c delete mode 100644 sound/soc/intel/byt-rt5640.c delete mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c delete mode 100644 sound/soc/intel/cht_bsw_rt5645.c delete mode 100644 sound/soc/intel/cht_bsw_rt5672.c delete mode 100644 sound/soc/intel/haswell.c delete mode 100644 sound/soc/intel/mfld_machine.c (limited to 'sound/soc/intel') diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index eb3efce4ec24..ac0248f100ff 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -16,21 +16,7 @@ snd-soc-sst-baytrail-pcm-objs := \ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o # Machine support -snd-soc-sst-haswell-objs := haswell.o -snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o -snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o -snd-soc-sst-broadwell-objs := broadwell.o -snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o -snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o -snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o - -obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o -obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o -obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o -obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o -obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o -obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o -obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o +obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile new file mode 100644 index 000000000000..f8237f0044eb --- /dev/null +++ b/sound/soc/intel/boards/Makefile @@ -0,0 +1,15 @@ +snd-soc-sst-haswell-objs := haswell.o +snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o +snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o +snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o + +obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c new file mode 100644 index 000000000000..8bafaf6ceab1 --- /dev/null +++ b/sound/soc/intel/boards/broadwell.c @@ -0,0 +1,292 @@ +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" + +#include "../../codecs/rt286.h" + +static struct snd_soc_jack broadwell_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin broadwell_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broadwell_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), +}; + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret = 0; + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, + broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); + if (ret) + return ret; + + rt286_mic_detect(codec, &broadwell_headset); + return 0; +} + + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int broadwell_rt286_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; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback/Capture", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = broadwell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .init = broadwell_rt286_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +static int broadwell_suspend(struct snd_soc_card *card){ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); + rt286_mic_detect(codec, NULL); + break; + } + } + return 0; +} + +static int broadwell_resume(struct snd_soc_card *card){ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + dev_dbg(codec->dev, "enabling jack detect for resume.\n"); + rt286_mic_detect(codec, &broadwell_headset); + break; + } + } + return 0; +} + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .name = "broadwell-rt286", + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .controls = broadwell_controls, + .num_controls = ARRAY_SIZE(broadwell_controls), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, + .suspend_pre = broadwell_suspend, + .resume_post = broadwell_resume, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + broadwell_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&broadwell_rt286); + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c new file mode 100644 index 000000000000..7ab8cc9fbfd5 --- /dev/null +++ b/sound/soc/intel/boards/byt-max98090.c @@ -0,0 +1,187 @@ +/* + * Intel Baytrail SST MAX98090 machine driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/max98090.h" + +struct byt_max98090_private { + struct snd_soc_jack jack; +}; + +static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { + {"IN34", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS"}, + {"DMICL", NULL, "Int Mic"}, + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKL"}, + {"Ext Spk", NULL, "SPKR"}, +}; + +static const struct snd_kcontrol_new byt_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .name = "hp-gpio", + .idx = 0, + .report = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, + .debounce_time = 200, + }, + { + .name = "mic-gpio", + .idx = 1, + .invert = 1, + .report = SND_JACK_MICROPHONE, + .debounce_time = 200, + }, +}; + +static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); + struct snd_soc_jack *jack = &drv->jack; + + card->dapm.idle_bias_off = true; + + ret = snd_soc_dai_set_sysclk(runtime->codec_dai, + M98090_REG_SYSTEM_CLOCK, + 25000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + /* Enable jack detection */ + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack, + hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); + if (ret) + return ret; + + return snd_soc_jack_add_gpiods(card->dev->parent, jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); +} + +static struct snd_soc_dai_link byt_max98090_dais[] = { + { + .name = "Baytrail Audio", + .stream_name = "Audio", + .cpu_dai_name = "baytrail-pcm-audio", + .codec_dai_name = "HiFi", + .codec_name = "i2c-193C9890:00", + .platform_name = "baytrail-pcm-audio", + .init = byt_max98090_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card byt_max98090_card = { + .name = "byt-max98090", + .dai_link = byt_max98090_dais, + .num_links = ARRAY_SIZE(byt_max98090_dais), + .dapm_widgets = byt_max98090_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_max98090_widgets), + .dapm_routes = byt_max98090_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), + .controls = byt_max98090_controls, + .num_controls = ARRAY_SIZE(byt_max98090_controls), + .fully_routed = true, +}; + +static int byt_max98090_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct byt_max98090_private *priv; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); + if (!priv) { + dev_err(&pdev->dev, "allocation failed\n"); + return -ENOMEM; + } + + byt_max98090_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&byt_max98090_card, priv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_max98090_card); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + + return ret_val; +} + +static int byt_max98090_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct byt_max98090_private *priv = snd_soc_card_get_drvdata(card); + + snd_soc_jack_free_gpios(&priv->jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return 0; +} + +static struct platform_driver byt_max98090_driver = { + .probe = byt_max98090_probe, + .remove = byt_max98090_remove, + .driver = { + .name = "byt-max98090", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(byt_max98090_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); +MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:byt-max98090"); diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c new file mode 100644 index 000000000000..ae89b9b966d9 --- /dev/null +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -0,0 +1,229 @@ +/* + * Intel Baytrail SST RT5640 machine driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/rt5640.h" + +#include "../common/sst-dsp.h" + +static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { + {"Headset Mic", NULL, "MICBIAS1"}, + {"IN2P", NULL, "Headset Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Speaker", NULL, "SPOLP"}, + {"Speaker", NULL, "SPOLN"}, + {"Speaker", NULL, "SPORP"}, + {"Speaker", NULL, "SPORN"}, +}; + +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"DMIC1", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"DMIC2", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "MICBIAS1"}, + {"IN1P", NULL, "Internal Mic"}, +}; + +enum { + BYT_RT5640_DMIC1_MAP, + BYT_RT5640_DMIC2_MAP, + BYT_RT5640_IN1_MAP, +}; + +#define BYT_RT5640_MAP(quirk) ((quirk) & 0xff) +#define BYT_RT5640_DMIC_EN BIT(16) + +static unsigned long byt_rt5640_quirk = BYT_RT5640_DMIC1_MAP | + BYT_RT5640_DMIC_EN; + +static const struct snd_kcontrol_new byt_rt5640_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static int byt_rt5640_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; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set codec clock %d\n", ret); + return ret; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1, + params_rate(params) * 64, + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret); + return ret; + } + return 0; +} + +static int byt_rt5640_quirk_cb(const struct dmi_system_id *id) +{ + byt_rt5640_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id byt_rt5640_quirk_table[] = { + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), + }, + .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP, + }, + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "DellInc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"), + }, + .driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP | + BYT_RT5640_DMIC_EN), + }, + {} +}; + +static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + const struct snd_soc_dapm_route *custom_map; + int num_routes; + + card->dapm.idle_bias_off = true; + + ret = snd_soc_add_card_controls(card, byt_rt5640_controls, + ARRAY_SIZE(byt_rt5640_controls)); + if (ret) { + dev_err(card->dev, "unable to add card controls\n"); + return ret; + } + + dmi_check_system(byt_rt5640_quirk_table); + switch (BYT_RT5640_MAP(byt_rt5640_quirk)) { + case BYT_RT5640_IN1_MAP: + custom_map = byt_rt5640_intmic_in1_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); + break; + case BYT_RT5640_DMIC2_MAP: + custom_map = byt_rt5640_intmic_dmic2_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map); + break; + default: + custom_map = byt_rt5640_intmic_dmic1_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); + } + + ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); + if (ret) + return ret; + + if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) { + ret = rt5640_dmic_enable(codec, 0, 0); + if (ret) + return ret; + } + + snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + + return ret; +} + +static struct snd_soc_ops byt_rt5640_ops = { + .hw_params = byt_rt5640_hw_params, +}; + +static struct snd_soc_dai_link byt_rt5640_dais[] = { + { + .name = "Baytrail Audio", + .stream_name = "Audio", + .cpu_dai_name = "baytrail-pcm-audio", + .codec_dai_name = "rt5640-aif1", + .codec_name = "i2c-10EC5640:00", + .platform_name = "baytrail-pcm-audio", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .init = byt_rt5640_init, + .ops = &byt_rt5640_ops, + }, +}; + +static struct snd_soc_card byt_rt5640_card = { + .name = "byt-rt5640", + .dai_link = byt_rt5640_dais, + .num_links = ARRAY_SIZE(byt_rt5640_dais), + .dapm_widgets = byt_rt5640_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets), + .dapm_routes = byt_rt5640_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map), + .fully_routed = true, +}; + +static int byt_rt5640_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &byt_rt5640_card; + + card->dev = &pdev->dev; + return devm_snd_soc_register_card(&pdev->dev, card); +} + +static struct platform_driver byt_rt5640_audio = { + .probe = byt_rt5640_probe, + .driver = { + .name = "byt-rt5640", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(byt_rt5640_audio) + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); +MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:byt-rt5640"); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c new file mode 100644 index 000000000000..5c2d8fabb5ed --- /dev/null +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -0,0 +1,227 @@ +/* + * byt_cr_dpcm_rt5640.c - ASoc Machine driver for Intel Byt CR platform + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/rt5640.h" +#include "../sst-atom-controls.h" + +static const struct snd_soc_dapm_widget byt_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route byt_audio_map[] = { + {"IN2P", NULL, "Headset Mic"}, + {"IN2N", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS1"}, + {"IN1P", NULL, "MICBIAS1"}, + {"LDO2", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new byt_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int byt_aif1_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; + + snd_soc_dai_set_bclk_ratio(codec_dai, 50); + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec clock %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1, + params_rate(params) * 50, + params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_pcm_stream byt_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int byt_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops byt_aif1_ops = { + .startup = byt_aif1_startup, +}; + +static struct snd_soc_ops byt_be_ssp2_ops = { + .hw_params = byt_aif1_hw_params, +}; + +static struct snd_soc_dai_link byt_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Baytrail Audio Port", + .stream_name = "Baytrail Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &byt_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Baytrail Compressed Port", + .stream_name = "Baytrail Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5640-aif1", + .codec_name = "i2c-10EC5640:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = byt_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &byt_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_byt = { + .name = "baytrailcraudio", + .dai_link = byt_dailink, + .num_links = ARRAY_SIZE(byt_dailink), + .dapm_widgets = byt_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_dapm_widgets), + .dapm_routes = byt_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_audio_map), + .controls = byt_mc_controls, + .num_controls = ARRAY_SIZE(byt_mc_controls), +}; + +static int snd_byt_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_byt.dev = &pdev->dev; + + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_byt); + if (ret_val) { + dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_byt); + return ret_val; +} + +static struct platform_driver snd_byt_mc_driver = { + .driver = { + .name = "bytt100_rt5640", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_byt_mc_probe, +}; + +module_platform_driver(snd_byt_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bytt100_rt5640"); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c new file mode 100644 index 000000000000..93bb6711ba3d --- /dev/null +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -0,0 +1,324 @@ +/* + * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5645 codec. + * + * Copyright (C) 2015 Intel Corp + * Author: Fang, Yang A + * N,Harshapriya + * This file is modified from cht_bsw_rt5672.c + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/rt5645.h" +#include "../sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "rt5645-aif1" + +struct cht_mc_private { + struct snd_soc_jack hp_jack; + struct snd_soc_jack mic_jack; +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + + if (!SND_SOC_DAPM_EVENT_OFF(event)) + return 0; + + /* Set codec sysclk source to its internal clock because codec PLL will + * be off when idle and MCLK will also be off by ACPI when codec is + * runtime suspended. Codec needs clock for jack detection and button + * press. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_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; + + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, + params_rate(params) * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + + /* Select clk_i2s1_asrc as ASRC clock source */ + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", + SND_JACK_HEADPHONE, &ctx->hp_jack, + NULL, 0); + if (ret) { + dev_err(runtime->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", + SND_JACK_MICROPHONE, &ctx->mic_jack, + NULL, 0); + if (ret) { + dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); + return ret; + } + + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* CODEC<->CODEC link */ + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5645:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "chtrt5645", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct cht_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-bsw-rt5645", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); +MODULE_AUTHOR("Fang, Yang A,N,Harshapriya"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5645"); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c new file mode 100644 index 000000000000..2cea002a31bb --- /dev/null +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -0,0 +1,366 @@ +/* + * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5672 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty + * Mengdong Lin + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/rt5670.h" +#include "../sst-atom-controls.h" + +/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */ +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "rt5670-aif1" + +static struct snd_soc_jack cht_bsw_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin cht_bsw_headset_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, 48000 * 512); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* set codec sysclk source to PLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + 48000 * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + } else { + /* Set codec sysclk source to its internal clock because codec + * PLL will be off when idle and MCLK will also be off by ACPI + * when codec is runtime suspended. Codec needs clock for jack + * detection and button press. + */ + snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, + 48000 * 512, SND_SOC_CLOCK_IN); + } + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_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; + + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* set codec sysclk source to PLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* Select codec ASRC clock source to track I2S1 clock, because codec + * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot + * be supported by RT5672. Otherwise, ASRC will be disabled and cause + * noise. + */ + rt5670_sel_asrc_clk_src(codec, + RT5670_DA_STEREO_FILTER + | RT5670_DA_MONO_L_FILTER + | RT5670_DA_MONO_R_FILTER + | RT5670_AD_STEREO_FILTER + | RT5670_AD_MONO_L_FILTER + | RT5670_AD_MONO_R_FILTER, + RT5670_CLK_SEL_I2S1_ASRC); + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2, &cht_bsw_headset, + cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins)); + if (ret) + return ret; + + rt5670_set_jack_detect(codec, &cht_bsw_headset); + return 0; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .nonatomic = true, + .codec_dai_name = "rt5670-aif1", + .codec_name = "i2c-10EC5670:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +static int cht_suspend_pre(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); + rt5670_jack_suspend(codec); + break; + } + } + return 0; +} + +static int cht_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + dev_dbg(codec->dev, "enabling jack detect for resume.\n"); + rt5670_jack_resume(codec); + break; + } + } + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailcraudio", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), + .suspend_pre = cht_suspend_pre, + .resume_post = cht_resume_post, +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-bsw-rt5672", + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5672"); diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c new file mode 100644 index 000000000000..22558572cb9c --- /dev/null +++ b/sound/soc/intel/boards/haswell.c @@ -0,0 +1,209 @@ +/* + * Intel Haswell Lynxpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include +#include +#include +#include +#include +#include + +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" + +#include "../../codecs/rt5640.h" + +/* Haswell ULT platforms have a Headphone and Mic jack */ +static const struct snd_soc_dapm_widget haswell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route haswell_rt5640_map[] = { + + {"Headphones", NULL, "HPOR"}, + {"Headphones", NULL, "HPOL"}, + {"IN2P", NULL, "Mic"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int haswell_rt5640_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; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + /* set correct codec filter for DAI format and clock config */ + snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000); + + return ret; +} + +static struct snd_soc_ops haswell_rt5640_ops = { + .hw_params = haswell_rt5640_hw_params, +}; + +static int haswell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *haswell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(haswell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "failed to set device config\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_dai_link haswell_rt5640_dais[] = { + /* Front End DAI links */ + { + .name = "System", + .stream_name = "System Playback/Capture", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = haswell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT33CA:00", + .codec_dai_name = "rt5640-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = haswell_ssp0_fixup, + .ops = &haswell_rt5640_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ +static struct snd_soc_card haswell_rt5640 = { + .name = "haswell-rt5640", + .owner = THIS_MODULE, + .dai_link = haswell_rt5640_dais, + .num_links = ARRAY_SIZE(haswell_rt5640_dais), + .dapm_widgets = haswell_widgets, + .num_dapm_widgets = ARRAY_SIZE(haswell_widgets), + .dapm_routes = haswell_rt5640_map, + .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map), + .fully_routed = true, +}; + +static int haswell_audio_probe(struct platform_device *pdev) +{ + haswell_rt5640.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &haswell_rt5640); +} + +static struct platform_driver haswell_audio = { + .probe = haswell_audio_probe, + .driver = { + .name = "haswell-audio", + }, +}; + +module_platform_driver(haswell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:haswell-audio"); diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c new file mode 100644 index 000000000000..49c09a0add79 --- /dev/null +++ b/sound/soc/intel/boards/mfld_machine.c @@ -0,0 +1,430 @@ +/* + * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../codecs/sn95031.h" + +#define MID_MONO 1 +#define MID_STEREO 2 +#define MID_MAX_CAP 5 +#define MFLD_JACK_INSERT 0x04 + +enum soc_mic_bias_zones { + MFLD_MV_START = 0, + /* mic bias volutage range for Headphones*/ + MFLD_MV_HP = 400, + /* mic bias volutage range for American Headset*/ + MFLD_MV_AM_HS = 650, + /* mic bias volutage range for Headset*/ + MFLD_MV_HS = 2000, + MFLD_MV_UNDEFINED, +}; + +static unsigned int hs_switch; +static unsigned int lo_dac; +static struct snd_soc_codec *mfld_codec; + +struct mfld_mc_private { + void __iomem *int_base; + u8 interrupt_status; +}; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; + +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, + {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, +}; + +/* sound card controls */ +static const char *headset_switch_text[] = {"Earpiece", "Headset"}; + +static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; + +static const struct soc_enum headset_enum = + SOC_ENUM_SINGLE_EXT(2, headset_switch_text); + +static const struct soc_enum lo_enum = + SOC_ENUM_SINGLE_EXT(4, lo_text); + +static int headset_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = hs_switch; + return 0; +} + +static int headset_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = &card->dapm; + + if (ucontrol->value.integer.value[0] == hs_switch) + return 0; + + snd_soc_dapm_mutex_lock(dapm); + + if (ucontrol->value.integer.value[0]) { + pr_debug("hs_set HS path\n"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); + } else { + pr_debug("hs_set EP path\n"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT"); + } + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + hs_switch = ucontrol->value.integer.value[0]; + + return 0; +} + +static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) +{ + snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL"); + snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR"); + snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL"); + snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR"); + snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT"); + snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT"); + if (hs_switch) { + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); + } else { + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT"); + } +} + +static int lo_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lo_dac; + return 0; +} + +static int lo_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = &card->dapm; + + if (ucontrol->value.integer.value[0] == lo_dac) + return 0; + + snd_soc_dapm_mutex_lock(dapm); + + /* we dont want to work with last state of lineout so just enable all + * pins and then disable pins not required + */ + lo_enable_out_pins(dapm); + + switch (ucontrol->value.integer.value[0]) { + case 0: + pr_debug("set vibra path\n"); + snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); + snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT"); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0); + break; + + case 1: + pr_debug("set hs path\n"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22); + break; + + case 2: + pr_debug("set spkr path\n"); + snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL"); + snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR"); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44); + break; + + case 3: + pr_debug("set null path\n"); + snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR"); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66); + break; + } + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + lo_dac = ucontrol->value.integer.value[0]; + return 0; +} + +static const struct snd_kcontrol_new mfld_snd_controls[] = { + SOC_ENUM_EXT("Playback Switch", headset_enum, + headset_get_switch, headset_set_switch), + SOC_ENUM_EXT("Lineout Mux", lo_enum, + lo_get_switch, lo_set_switch), +}; + +static const struct snd_soc_dapm_widget mfld_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route mfld_map[] = { + {"Headphones", NULL, "HPOUTR"}, + {"Headphones", NULL, "HPOUTL"}, + {"Mic", NULL, "AMIC1"}, +}; + +static void mfld_jack_check(unsigned int intr_status) +{ + struct mfld_jack_data jack_data; + + if (!mfld_codec) + return; + + jack_data.mfld_jack = &mfld_jack; + jack_data.intr_id = intr_status; + + sn95031_jack_detection(mfld_codec, &jack_data); + /* TODO: add american headset detection post gpiolib support */ +} + +static int mfld_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_dapm_context *dapm = &runtime->card->dapm; + int ret_val; + + /* default is earpiece pin, userspace sets it explcitly */ + snd_soc_dapm_disable_pin(dapm, "Headphones"); + /* default is lineout NC, userspace sets it explcitly */ + snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); + lo_dac = 3; + hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); + + /* Headset and button jack detection */ + ret_val = snd_soc_card_jack_new(runtime->card, + "Intel(R) MID Audio Jack", SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack, + mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins)); + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } + + mfld_codec = runtime->codec; + + /* we want to check if anything is inserted at boot, + * so send a fake event to codec and it will read adc + * to find if anything is there or not */ + mfld_jack_check(MFLD_JACK_INSERT); + return ret_val; +} + +static struct snd_soc_dai_link mfld_msic_dailink[] = { + { + .name = "Medfield Headset", + .stream_name = "Headset", + .cpu_dai_name = "Headset-cpu-dai", + .codec_dai_name = "SN95031 Headset", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = mfld_init, + }, + { + .name = "Medfield Speaker", + .stream_name = "Speaker", + .cpu_dai_name = "Speaker-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Vibra", + .stream_name = "Vibra1", + .cpu_dai_name = "Vibra1-cpu-dai", + .codec_dai_name = "SN95031 Vibra1", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Haptics", + .stream_name = "Vibra2", + .cpu_dai_name = "Vibra2-cpu-dai", + .codec_dai_name = "SN95031 Vibra2", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Compress", + .stream_name = "Speaker", + .cpu_dai_name = "Compress-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mfld = { + .name = "medfield_audio", + .owner = THIS_MODULE, + .dai_link = mfld_msic_dailink, + .num_links = ARRAY_SIZE(mfld_msic_dailink), + + .controls = mfld_snd_controls, + .num_controls = ARRAY_SIZE(mfld_snd_controls), + .dapm_widgets = mfld_widgets, + .num_dapm_widgets = ARRAY_SIZE(mfld_widgets), + .dapm_routes = mfld_map, + .num_dapm_routes = ARRAY_SIZE(mfld_map), +}; + +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + memcpy_fromio(&mc_private->interrupt_status, + ((void *)(mc_private->int_base)), + sizeof(u8)); + return IRQ_WAKE_THREAD; +} + +static irqreturn_t snd_mfld_jack_detection(int irq, void *data) +{ + struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; + + mfld_jack_check(mc_drv_ctx->interrupt_status); + + return IRQ_HANDLED; +} + +static int snd_mfld_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0, irq; + struct mfld_mc_private *mc_drv_ctx; + struct resource *irq_mem; + + pr_debug("snd_mfld_mc_probe called\n"); + + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); + if (!mc_drv_ctx) { + pr_err("allocation failed\n"); + return -ENOMEM; + } + + irq_mem = platform_get_resource_byname( + pdev, IORESOURCE_MEM, "IRQ_BASE"); + if (!irq_mem) { + pr_err("no mem resource given\n"); + return -ENODEV; + } + mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, + resource_size(irq_mem)); + if (!mc_drv_ctx->int_base) { + pr_err("Mapping of cache failed\n"); + return -ENOMEM; + } + /* register for interrupt */ + ret_val = devm_request_threaded_irq(&pdev->dev, irq, + snd_mfld_jack_intr_handler, + snd_mfld_jack_detection, + IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); + if (ret_val) { + pr_err("cannot register IRQ\n"); + return ret_val; + } + /* register the soc card */ + snd_soc_card_mfld.dev = &pdev->dev; + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); + if (ret_val) { + pr_debug("snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, mc_drv_ctx); + pr_debug("successfully exited probe\n"); + return 0; +} + +static struct platform_driver snd_mfld_mc_driver = { + .driver = { + .name = "msic_audio", + }, + .probe = snd_mfld_mc_probe, +}; + +module_platform_driver(snd_mfld_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:msic-audio"); diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c deleted file mode 100644 index 6c75b6bd0049..000000000000 --- a/sound/soc/intel/broadwell.c +++ /dev/null @@ -1,292 +0,0 @@ -/* - * Intel Broadwell Wildcatpoint SST Audio - * - * Copyright (C) 2013, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - */ - -#include -#include -#include -#include -#include -#include -#include - -#include "sst-dsp.h" -#include "sst-haswell-ipc.h" - -#include "../codecs/rt286.h" - -static struct snd_soc_jack broadwell_headset; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin broadwell_headset_pins[] = { - { - .pin = "Mic Jack", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, -}; - -static const struct snd_kcontrol_new broadwell_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), - SOC_DAPM_PIN_SWITCH("Headphone Jack"), -}; - -static const struct snd_soc_dapm_widget broadwell_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_MIC("DMIC1", NULL), - SND_SOC_DAPM_MIC("DMIC2", NULL), - SND_SOC_DAPM_LINE("Line Jack", NULL), -}; - -static const struct snd_soc_dapm_route broadwell_rt286_map[] = { - - /* speaker */ - {"Speaker", NULL, "SPOR"}, - {"Speaker", NULL, "SPOL"}, - - /* HP jack connectors - unknown if we have jack deteck */ - {"Headphone Jack", NULL, "HPO Pin"}, - - /* other jacks */ - {"MIC1", NULL, "Mic Jack"}, - {"LINE1", NULL, "Line Jack"}, - - /* digital mics */ - {"DMIC1 Pin", NULL, "DMIC1"}, - {"DMIC2 Pin", NULL, "DMIC2"}, - - /* CODEC BE connections */ - {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, - {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, -}; - -static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - int ret = 0; - ret = snd_soc_card_jack_new(rtd->card, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, - broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); - if (ret) - return ret; - - rt286_mic_detect(codec, &broadwell_headset); - return 0; -} - - -static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* The ADSP will covert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - /* set SSP0 to 16 bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); - return 0; -} - -static int broadwell_rt286_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; - - ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, - SND_SOC_CLOCK_IN); - - if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - return ret; -} - -static struct snd_soc_ops broadwell_rt286_ops = { - .hw_params = broadwell_rt286_hw_params, -}; - -static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) -{ - struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); - struct sst_hsw *broadwell = pdata->dsp; - int ret; - - /* Set ADSP SSP port settings */ - ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, - SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, - SST_HSW_DEVICE_CLOCK_MASTER, 9); - if (ret < 0) { - dev_err(rtd->dev, "error: failed to set device config\n"); - return ret; - } - - return 0; -} - -/* broadwell digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link broadwell_rt286_dais[] = { - /* Front End DAI links */ - { - .name = "System PCM", - .stream_name = "System Playback/Capture", - .cpu_dai_name = "System Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .init = broadwell_rtd_init, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .dpcm_capture = 1, - }, - { - .name = "Offload0", - .stream_name = "Offload0 Playback", - .cpu_dai_name = "Offload0 Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - }, - { - .name = "Offload1", - .stream_name = "Offload1 Playback", - .cpu_dai_name = "Offload1 Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - }, - { - .name = "Loopback PCM", - .stream_name = "Loopback", - .cpu_dai_name = "Loopback Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 0, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "Codec", - .be_id = 0, - .cpu_dai_name = "snd-soc-dummy-dai", - .platform_name = "snd-soc-dummy", - .no_pcm = 1, - .codec_name = "i2c-INT343A:00", - .codec_dai_name = "rt286-aif1", - .init = broadwell_rt286_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ignore_suspend = 1, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = broadwell_ssp0_fixup, - .ops = &broadwell_rt286_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - }, -}; - -static int broadwell_suspend(struct snd_soc_card *card){ - struct snd_soc_codec *codec; - - list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-INT343A:00")) { - dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); - rt286_mic_detect(codec, NULL); - break; - } - } - return 0; -} - -static int broadwell_resume(struct snd_soc_card *card){ - struct snd_soc_codec *codec; - - list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-INT343A:00")) { - dev_dbg(codec->dev, "enabling jack detect for resume.\n"); - rt286_mic_detect(codec, &broadwell_headset); - break; - } - } - return 0; -} - -/* broadwell audio machine driver for WPT + RT286S */ -static struct snd_soc_card broadwell_rt286 = { - .name = "broadwell-rt286", - .owner = THIS_MODULE, - .dai_link = broadwell_rt286_dais, - .num_links = ARRAY_SIZE(broadwell_rt286_dais), - .controls = broadwell_controls, - .num_controls = ARRAY_SIZE(broadwell_controls), - .dapm_widgets = broadwell_widgets, - .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), - .dapm_routes = broadwell_rt286_map, - .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), - .fully_routed = true, - .suspend_pre = broadwell_suspend, - .resume_post = broadwell_resume, -}; - -static int broadwell_audio_probe(struct platform_device *pdev) -{ - broadwell_rt286.dev = &pdev->dev; - - return snd_soc_register_card(&broadwell_rt286); -} - -static int broadwell_audio_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&broadwell_rt286); - return 0; -} - -static struct platform_driver broadwell_audio = { - .probe = broadwell_audio_probe, - .remove = broadwell_audio_remove, - .driver = { - .name = "broadwell-audio", - }, -}; - -module_platform_driver(broadwell_audio) - -/* Module information */ -MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:broadwell-audio"); diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c deleted file mode 100644 index d8b1f038da1c..000000000000 --- a/sound/soc/intel/byt-max98090.c +++ /dev/null @@ -1,187 +0,0 @@ -/* - * Intel Baytrail SST MAX98090 machine driver - * Copyright (c) 2014, Intel Corporation. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms and conditions of the GNU General Public License, - * version 2, as published by the Free Software Foundation. - * - * This program is distributed in the hope it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/max98090.h" - -struct byt_max98090_private { - struct snd_soc_jack jack; -}; - -static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { - {"IN34", NULL, "Headset Mic"}, - {"Headset Mic", NULL, "MICBIAS"}, - {"DMICL", NULL, "Int Mic"}, - {"Headphone", NULL, "HPL"}, - {"Headphone", NULL, "HPR"}, - {"Ext Spk", NULL, "SPKL"}, - {"Ext Spk", NULL, "SPKR"}, -}; - -static const struct snd_kcontrol_new byt_max98090_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Int Mic"), - SOC_DAPM_PIN_SWITCH("Ext Spk"), -}; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static struct snd_soc_jack_gpio hs_jack_gpios[] = { - { - .name = "hp-gpio", - .idx = 0, - .report = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, - .debounce_time = 200, - }, - { - .name = "mic-gpio", - .idx = 1, - .invert = 1, - .report = SND_JACK_MICROPHONE, - .debounce_time = 200, - }, -}; - -static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) -{ - int ret; - struct snd_soc_card *card = runtime->card; - struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); - struct snd_soc_jack *jack = &drv->jack; - - card->dapm.idle_bias_off = true; - - ret = snd_soc_dai_set_sysclk(runtime->codec_dai, - M98090_REG_SYSTEM_CLOCK, - 25000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "Can't set codec clock %d\n", ret); - return ret; - } - - /* Enable jack detection */ - ret = snd_soc_card_jack_new(runtime->card, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET, jack, - hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); - if (ret) - return ret; - - return snd_soc_jack_add_gpiods(card->dev->parent, jack, - ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); -} - -static struct snd_soc_dai_link byt_max98090_dais[] = { - { - .name = "Baytrail Audio", - .stream_name = "Audio", - .cpu_dai_name = "baytrail-pcm-audio", - .codec_dai_name = "HiFi", - .codec_name = "i2c-193C9890:00", - .platform_name = "baytrail-pcm-audio", - .init = byt_max98090_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - }, -}; - -static struct snd_soc_card byt_max98090_card = { - .name = "byt-max98090", - .dai_link = byt_max98090_dais, - .num_links = ARRAY_SIZE(byt_max98090_dais), - .dapm_widgets = byt_max98090_widgets, - .num_dapm_widgets = ARRAY_SIZE(byt_max98090_widgets), - .dapm_routes = byt_max98090_audio_map, - .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), - .controls = byt_max98090_controls, - .num_controls = ARRAY_SIZE(byt_max98090_controls), - .fully_routed = true, -}; - -static int byt_max98090_probe(struct platform_device *pdev) -{ - int ret_val = 0; - struct byt_max98090_private *priv; - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); - if (!priv) { - dev_err(&pdev->dev, "allocation failed\n"); - return -ENOMEM; - } - - byt_max98090_card.dev = &pdev->dev; - snd_soc_card_set_drvdata(&byt_max98090_card, priv); - ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_max98090_card); - if (ret_val) { - dev_err(&pdev->dev, - "snd_soc_register_card failed %d\n", ret_val); - return ret_val; - } - - return ret_val; -} - -static int byt_max98090_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct byt_max98090_private *priv = snd_soc_card_get_drvdata(card); - - snd_soc_jack_free_gpios(&priv->jack, ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - return 0; -} - -static struct platform_driver byt_max98090_driver = { - .probe = byt_max98090_probe, - .remove = byt_max98090_remove, - .driver = { - .name = "byt-max98090", - .pm = &snd_soc_pm_ops, - }, -}; -module_platform_driver(byt_max98090_driver) - -MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); -MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:byt-max98090"); diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c deleted file mode 100644 index 354eaad886e1..000000000000 --- a/sound/soc/intel/byt-rt5640.c +++ /dev/null @@ -1,229 +0,0 @@ -/* - * Intel Baytrail SST RT5640 machine driver - * Copyright (c) 2014, Intel Corporation. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms and conditions of the GNU General Public License, - * version 2, as published by the Free Software Foundation. - * - * This program is distributed in the hope it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/rt5640.h" - -#include "sst-dsp.h" - -static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Internal Mic", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - -static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { - {"Headset Mic", NULL, "MICBIAS1"}, - {"IN2P", NULL, "Headset Mic"}, - {"Headphone", NULL, "HPOL"}, - {"Headphone", NULL, "HPOR"}, - {"Speaker", NULL, "SPOLP"}, - {"Speaker", NULL, "SPOLN"}, - {"Speaker", NULL, "SPORP"}, - {"Speaker", NULL, "SPORN"}, -}; - -static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { - {"DMIC1", NULL, "Internal Mic"}, -}; - -static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { - {"DMIC2", NULL, "Internal Mic"}, -}; - -static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { - {"Internal Mic", NULL, "MICBIAS1"}, - {"IN1P", NULL, "Internal Mic"}, -}; - -enum { - BYT_RT5640_DMIC1_MAP, - BYT_RT5640_DMIC2_MAP, - BYT_RT5640_IN1_MAP, -}; - -#define BYT_RT5640_MAP(quirk) ((quirk) & 0xff) -#define BYT_RT5640_DMIC_EN BIT(16) - -static unsigned long byt_rt5640_quirk = BYT_RT5640_DMIC1_MAP | - BYT_RT5640_DMIC_EN; - -static const struct snd_kcontrol_new byt_rt5640_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Internal Mic"), - SOC_DAPM_PIN_SWITCH("Speaker"), -}; - -static int byt_rt5640_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; - - ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, - params_rate(params) * 256, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(codec_dai->dev, "can't set codec clock %d\n", ret); - return ret; - } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1, - params_rate(params) * 64, - params_rate(params) * 256); - if (ret < 0) { - dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret); - return ret; - } - return 0; -} - -static int byt_rt5640_quirk_cb(const struct dmi_system_id *id) -{ - byt_rt5640_quirk = (unsigned long)id->driver_data; - return 1; -} - -static const struct dmi_system_id byt_rt5640_quirk_table[] = { - { - .callback = byt_rt5640_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), - }, - .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP, - }, - { - .callback = byt_rt5640_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "DellInc."), - DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"), - }, - .driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP | - BYT_RT5640_DMIC_EN), - }, - {} -}; - -static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) -{ - int ret; - struct snd_soc_codec *codec = runtime->codec; - struct snd_soc_card *card = runtime->card; - const struct snd_soc_dapm_route *custom_map; - int num_routes; - - card->dapm.idle_bias_off = true; - - ret = snd_soc_add_card_controls(card, byt_rt5640_controls, - ARRAY_SIZE(byt_rt5640_controls)); - if (ret) { - dev_err(card->dev, "unable to add card controls\n"); - return ret; - } - - dmi_check_system(byt_rt5640_quirk_table); - switch (BYT_RT5640_MAP(byt_rt5640_quirk)) { - case BYT_RT5640_IN1_MAP: - custom_map = byt_rt5640_intmic_in1_map; - num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); - break; - case BYT_RT5640_DMIC2_MAP: - custom_map = byt_rt5640_intmic_dmic2_map; - num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map); - break; - default: - custom_map = byt_rt5640_intmic_dmic1_map; - num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); - } - - ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); - if (ret) - return ret; - - if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) { - ret = rt5640_dmic_enable(codec, 0, 0); - if (ret) - return ret; - } - - snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); - - return ret; -} - -static struct snd_soc_ops byt_rt5640_ops = { - .hw_params = byt_rt5640_hw_params, -}; - -static struct snd_soc_dai_link byt_rt5640_dais[] = { - { - .name = "Baytrail Audio", - .stream_name = "Audio", - .cpu_dai_name = "baytrail-pcm-audio", - .codec_dai_name = "rt5640-aif1", - .codec_name = "i2c-10EC5640:00", - .platform_name = "baytrail-pcm-audio", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .init = byt_rt5640_init, - .ops = &byt_rt5640_ops, - }, -}; - -static struct snd_soc_card byt_rt5640_card = { - .name = "byt-rt5640", - .dai_link = byt_rt5640_dais, - .num_links = ARRAY_SIZE(byt_rt5640_dais), - .dapm_widgets = byt_rt5640_widgets, - .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets), - .dapm_routes = byt_rt5640_audio_map, - .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map), - .fully_routed = true, -}; - -static int byt_rt5640_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &byt_rt5640_card; - - card->dev = &pdev->dev; - return devm_snd_soc_register_card(&pdev->dev, card); -} - -static struct platform_driver byt_rt5640_audio = { - .probe = byt_rt5640_probe, - .driver = { - .name = "byt-rt5640", - .pm = &snd_soc_pm_ops, - }, -}; -module_platform_driver(byt_rt5640_audio) - -MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); -MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:byt-rt5640"); diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c deleted file mode 100644 index 3b262d01c1b3..000000000000 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ /dev/null @@ -1,227 +0,0 @@ -/* - * byt_cr_dpcm_rt5640.c - ASoc Machine driver for Intel Byt CR platform - * - * Copyright (C) 2014 Intel Corp - * Author: Subhransu S. Prusty - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/rt5640.h" -#include "sst-atom-controls.h" - -static const struct snd_soc_dapm_widget byt_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route byt_audio_map[] = { - {"IN2P", NULL, "Headset Mic"}, - {"IN2N", NULL, "Headset Mic"}, - {"Headset Mic", NULL, "MICBIAS1"}, - {"IN1P", NULL, "MICBIAS1"}, - {"LDO2", NULL, "Int Mic"}, - {"Headphone", NULL, "HPOL"}, - {"Headphone", NULL, "HPOR"}, - {"Ext Spk", NULL, "SPOLP"}, - {"Ext Spk", NULL, "SPOLN"}, - {"Ext Spk", NULL, "SPORP"}, - {"Ext Spk", NULL, "SPORN"}, - - {"AIF1 Playback", NULL, "ssp2 Tx"}, - {"ssp2 Tx", NULL, "codec_out0"}, - {"ssp2 Tx", NULL, "codec_out1"}, - {"codec_in0", NULL, "ssp2 Rx"}, - {"codec_in1", NULL, "ssp2 Rx"}, - {"ssp2 Rx", NULL, "AIF1 Capture"}, -}; - -static const struct snd_kcontrol_new byt_mc_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Int Mic"), - SOC_DAPM_PIN_SWITCH("Ext Spk"), -}; - -static int byt_aif1_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; - - snd_soc_dai_set_bclk_ratio(codec_dai, 50); - - ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, - params_rate(params) * 512, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec clock %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1, - params_rate(params) * 50, - params_rate(params) * 512); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec pll: %d\n", ret); - return ret; - } - - return 0; -} - -static const struct snd_soc_pcm_stream byt_dai_params = { - .formats = SNDRV_PCM_FMTBIT_S24_LE, - .rate_min = 48000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, -}; - -static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* The DSP will covert the FE rate to 48k, stereo, 24bits */ - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - /* set SSP2 to 24-bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); - return 0; -} - -static unsigned int rates_48000[] = { - 48000, -}; - -static struct snd_pcm_hw_constraint_list constraints_48000 = { - .count = ARRAY_SIZE(rates_48000), - .list = rates_48000, -}; - -static int byt_aif1_startup(struct snd_pcm_substream *substream) -{ - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_48000); -} - -static struct snd_soc_ops byt_aif1_ops = { - .startup = byt_aif1_startup, -}; - -static struct snd_soc_ops byt_be_ssp2_ops = { - .hw_params = byt_aif1_hw_params, -}; - -static struct snd_soc_dai_link byt_dailink[] = { - [MERR_DPCM_AUDIO] = { - .name = "Baytrail Audio Port", - .stream_name = "Baytrail Audio", - .cpu_dai_name = "media-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ops = &byt_aif1_ops, - }, - [MERR_DPCM_COMPR] = { - .name = "Baytrail Compressed Port", - .stream_name = "Baytrail Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, - /* back ends */ - { - .name = "SSP2-Codec", - .be_id = 1, - .cpu_dai_name = "ssp2-port", - .platform_name = "sst-mfld-platform", - .no_pcm = 1, - .codec_dai_name = "rt5640-aif1", - .codec_name = "i2c-10EC5640:00", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, - .be_hw_params_fixup = byt_codec_fixup, - .ignore_suspend = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ops = &byt_be_ssp2_ops, - }, -}; - -/* SoC card */ -static struct snd_soc_card snd_soc_card_byt = { - .name = "baytrailcraudio", - .dai_link = byt_dailink, - .num_links = ARRAY_SIZE(byt_dailink), - .dapm_widgets = byt_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(byt_dapm_widgets), - .dapm_routes = byt_audio_map, - .num_dapm_routes = ARRAY_SIZE(byt_audio_map), - .controls = byt_mc_controls, - .num_controls = ARRAY_SIZE(byt_mc_controls), -}; - -static int snd_byt_mc_probe(struct platform_device *pdev) -{ - int ret_val = 0; - - /* register the soc card */ - snd_soc_card_byt.dev = &pdev->dev; - - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_byt); - if (ret_val) { - dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val); - return ret_val; - } - platform_set_drvdata(pdev, &snd_soc_card_byt); - return ret_val; -} - -static struct platform_driver snd_byt_mc_driver = { - .driver = { - .name = "bytt100_rt5640", - .pm = &snd_soc_pm_ops, - }, - .probe = snd_byt_mc_probe, -}; - -module_platform_driver(snd_byt_mc_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); -MODULE_AUTHOR("Subhransu S. Prusty "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:bytt100_rt5640"); diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c deleted file mode 100644 index 012227997ed9..000000000000 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ /dev/null @@ -1,324 +0,0 @@ -/* - * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms - * Cherrytrail and Braswell, with RT5645 codec. - * - * Copyright (C) 2015 Intel Corp - * Author: Fang, Yang A - * N,Harshapriya - * This file is modified from cht_bsw_rt5672.c - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/rt5645.h" -#include "sst-atom-controls.h" - -#define CHT_PLAT_CLK_3_HZ 19200000 -#define CHT_CODEC_DAI "rt5645-aif1" - -struct cht_mc_private { - struct snd_soc_jack hp_jack; - struct snd_soc_jack mic_jack; -}; - -static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) -{ - int i; - - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd; - - rtd = card->rtd + i; - if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, - strlen(CHT_CODEC_DAI))) - return rtd->codec_dai; - } - return NULL; -} - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret; - - codec_dai = cht_get_codec_dai(card); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); - return -EIO; - } - - if (!SND_SOC_DAPM_EVENT_OFF(event)) - return 0; - - /* Set codec sysclk source to its internal clock because codec PLL will - * be off when idle and MCLK will also be off by ACPI when codec is - * runtime suspended. Codec needs clock for jack detection and button - * press. - */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "can't set codec sysclk: %d\n", ret); - return ret; - } - - return 0; -} - -static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_POST_PMD), -}; - -static const struct snd_soc_dapm_route cht_audio_map[] = { - {"IN1P", NULL, "Headset Mic"}, - {"IN1N", NULL, "Headset Mic"}, - {"DMIC L1", NULL, "Int Mic"}, - {"DMIC R1", NULL, "Int Mic"}, - {"Headphone", NULL, "HPOL"}, - {"Headphone", NULL, "HPOR"}, - {"Ext Spk", NULL, "SPOL"}, - {"Ext Spk", NULL, "SPOR"}, - {"AIF1 Playback", NULL, "ssp2 Tx"}, - {"ssp2 Tx", NULL, "codec_out0"}, - {"ssp2 Tx", NULL, "codec_out1"}, - {"codec_in0", NULL, "ssp2 Rx" }, - {"codec_in1", NULL, "ssp2 Rx" }, - {"ssp2 Rx", NULL, "AIF1 Capture"}, - {"Headphone", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "Platform Clock"}, - {"Int Mic", NULL, "Platform Clock"}, - {"Ext Spk", NULL, "Platform Clock"}, -}; - -static const struct snd_kcontrol_new cht_mc_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Int Mic"), - SOC_DAPM_PIN_SWITCH("Ext Spk"), -}; - -static int cht_aif1_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; - - /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, - CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec pll: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, - params_rate(params) * 512, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); - return ret; - } - - return 0; -} - -static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) -{ - int ret; - struct snd_soc_codec *codec = runtime->codec; - struct snd_soc_dai *codec_dai = runtime->codec_dai; - struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); - - /* Select clk_i2s1_asrc as ASRC clock source */ - rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_DA_MONO_L_FILTER | - RT5645_DA_MONO_R_FILTER | - RT5645_AD_STEREO_FILTER, - RT5645_CLK_SEL_I2S1_ASRC); - - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); - if (ret < 0) { - dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); - return ret; - } - - ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", - SND_JACK_HEADPHONE, &ctx->hp_jack, - NULL, 0); - if (ret) { - dev_err(runtime->dev, "HP jack creation failed %d\n", ret); - return ret; - } - - ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", - SND_JACK_MICROPHONE, &ctx->mic_jack, - NULL, 0); - if (ret) { - dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); - return ret; - } - - rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); - - return ret; -} - -static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* The DSP will covert the FE rate to 48k, stereo, 24bits */ - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - /* set SSP2 to 24-bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); - return 0; -} - -static unsigned int rates_48000[] = { - 48000, -}; - -static struct snd_pcm_hw_constraint_list constraints_48000 = { - .count = ARRAY_SIZE(rates_48000), - .list = rates_48000, -}; - -static int cht_aif1_startup(struct snd_pcm_substream *substream) -{ - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_48000); -} - -static struct snd_soc_ops cht_aif1_ops = { - .startup = cht_aif1_startup, -}; - -static struct snd_soc_ops cht_be_ssp2_ops = { - .hw_params = cht_aif1_hw_params, -}; - -static struct snd_soc_dai_link cht_dailink[] = { - [MERR_DPCM_AUDIO] = { - .name = "Audio Port", - .stream_name = "Audio", - .cpu_dai_name = "media-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ops = &cht_aif1_ops, - }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, - /* CODEC<->CODEC link */ - /* back ends */ - { - .name = "SSP2-Codec", - .be_id = 1, - .cpu_dai_name = "ssp2-port", - .platform_name = "sst-mfld-platform", - .no_pcm = 1, - .codec_dai_name = "rt5645-aif1", - .codec_name = "i2c-10EC5645:00", - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS, - .init = cht_codec_init, - .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ops = &cht_be_ssp2_ops, - }, -}; - -/* SoC card */ -static struct snd_soc_card snd_soc_card_cht = { - .name = "chtrt5645", - .dai_link = cht_dailink, - .num_links = ARRAY_SIZE(cht_dailink), - .dapm_widgets = cht_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), - .dapm_routes = cht_audio_map, - .num_dapm_routes = ARRAY_SIZE(cht_audio_map), - .controls = cht_mc_controls, - .num_controls = ARRAY_SIZE(cht_mc_controls), -}; - -static int snd_cht_mc_probe(struct platform_device *pdev) -{ - int ret_val = 0; - struct cht_mc_private *drv; - - drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); - if (!drv) - return -ENOMEM; - - snd_soc_card_cht.dev = &pdev->dev; - snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); - if (ret_val) { - dev_err(&pdev->dev, - "snd_soc_register_card failed %d\n", ret_val); - return ret_val; - } - platform_set_drvdata(pdev, &snd_soc_card_cht); - return ret_val; -} - -static struct platform_driver snd_cht_mc_driver = { - .driver = { - .name = "cht-bsw-rt5645", - .pm = &snd_soc_pm_ops, - }, - .probe = snd_cht_mc_probe, -}; - -module_platform_driver(snd_cht_mc_driver) - -MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); -MODULE_AUTHOR("Fang, Yang A,N,Harshapriya"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:cht-bsw-rt5645"); diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c deleted file mode 100644 index 4204fc4f1bad..000000000000 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ /dev/null @@ -1,366 +0,0 @@ -/* - * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms - * Cherrytrail and Braswell, with RT5672 codec. - * - * Copyright (C) 2014 Intel Corp - * Author: Subhransu S. Prusty - * Mengdong Lin - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - */ - -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/rt5670.h" -#include "sst-atom-controls.h" - -/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */ -#define CHT_PLAT_CLK_3_HZ 19200000 -#define CHT_CODEC_DAI "rt5670-aif1" - -static struct snd_soc_jack cht_bsw_headset; - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin cht_bsw_headset_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, -}; - -static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) -{ - int i; - - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd; - - rtd = card->rtd + i; - if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, - strlen(CHT_CODEC_DAI))) - return rtd->codec_dai; - } - return NULL; -} - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret; - - codec_dai = cht_get_codec_dai(card); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, - CHT_PLAT_CLK_3_HZ, 48000 * 512); - if (ret < 0) { - dev_err(card->dev, "can't set codec pll: %d\n", ret); - return ret; - } - - /* set codec sysclk source to PLL */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, - 48000 * 512, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "can't set codec sysclk: %d\n", ret); - return ret; - } - } else { - /* Set codec sysclk source to its internal clock because codec - * PLL will be off when idle and MCLK will also be off by ACPI - * when codec is runtime suspended. Codec needs clock for jack - * detection and button press. - */ - snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, - 48000 * 512, SND_SOC_CLOCK_IN); - } - return 0; -} - -static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static const struct snd_soc_dapm_route cht_audio_map[] = { - {"IN1P", NULL, "Headset Mic"}, - {"IN1N", NULL, "Headset Mic"}, - {"DMIC L1", NULL, "Int Mic"}, - {"DMIC R1", NULL, "Int Mic"}, - {"Headphone", NULL, "HPOL"}, - {"Headphone", NULL, "HPOR"}, - {"Ext Spk", NULL, "SPOLP"}, - {"Ext Spk", NULL, "SPOLN"}, - {"Ext Spk", NULL, "SPORP"}, - {"Ext Spk", NULL, "SPORN"}, - {"AIF1 Playback", NULL, "ssp2 Tx"}, - {"ssp2 Tx", NULL, "codec_out0"}, - {"ssp2 Tx", NULL, "codec_out1"}, - {"codec_in0", NULL, "ssp2 Rx"}, - {"codec_in1", NULL, "ssp2 Rx"}, - {"ssp2 Rx", NULL, "AIF1 Capture"}, - {"Headphone", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "Platform Clock"}, - {"Int Mic", NULL, "Platform Clock"}, - {"Ext Spk", NULL, "Platform Clock"}, -}; - -static const struct snd_kcontrol_new cht_mc_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Int Mic"), - SOC_DAPM_PIN_SWITCH("Ext Spk"), -}; - -static int cht_aif1_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; - - /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, - CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec pll: %d\n", ret); - return ret; - } - - /* set codec sysclk source to PLL */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, - params_rate(params) * 512, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); - return ret; - } - return 0; -} - -static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) -{ - int ret; - struct snd_soc_dai *codec_dai = runtime->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); - if (ret < 0) { - dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); - return ret; - } - - /* Select codec ASRC clock source to track I2S1 clock, because codec - * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot - * be supported by RT5672. Otherwise, ASRC will be disabled and cause - * noise. - */ - rt5670_sel_asrc_clk_src(codec, - RT5670_DA_STEREO_FILTER - | RT5670_DA_MONO_L_FILTER - | RT5670_DA_MONO_R_FILTER - | RT5670_AD_STEREO_FILTER - | RT5670_AD_MONO_L_FILTER - | RT5670_AD_MONO_R_FILTER, - RT5670_CLK_SEL_I2S1_ASRC); - - ret = snd_soc_card_jack_new(runtime->card, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2, &cht_bsw_headset, - cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins)); - if (ret) - return ret; - - rt5670_set_jack_detect(codec, &cht_bsw_headset); - return 0; -} - -static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* The DSP will covert the FE rate to 48k, stereo, 24bits */ - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - /* set SSP2 to 24-bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); - return 0; -} - -static unsigned int rates_48000[] = { - 48000, -}; - -static struct snd_pcm_hw_constraint_list constraints_48000 = { - .count = ARRAY_SIZE(rates_48000), - .list = rates_48000, -}; - -static int cht_aif1_startup(struct snd_pcm_substream *substream) -{ - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_48000); -} - -static struct snd_soc_ops cht_aif1_ops = { - .startup = cht_aif1_startup, -}; - -static struct snd_soc_ops cht_be_ssp2_ops = { - .hw_params = cht_aif1_hw_params, -}; - -static struct snd_soc_dai_link cht_dailink[] = { - /* Front End DAI links */ - [MERR_DPCM_AUDIO] = { - .name = "Audio Port", - .stream_name = "Audio", - .cpu_dai_name = "media-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - .nonatomic = true, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ops = &cht_aif1_ops, - }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, - - /* Back End DAI links */ - { - /* SSP2 - Codec */ - .name = "SSP2-Codec", - .be_id = 1, - .cpu_dai_name = "ssp2-port", - .platform_name = "sst-mfld-platform", - .no_pcm = 1, - .nonatomic = true, - .codec_dai_name = "rt5670-aif1", - .codec_name = "i2c-10EC5670:00", - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS, - .init = cht_codec_init, - .be_hw_params_fixup = cht_codec_fixup, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ops = &cht_be_ssp2_ops, - }, -}; - -static int cht_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_codec *codec; - - list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { - dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); - rt5670_jack_suspend(codec); - break; - } - } - return 0; -} - -static int cht_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_codec *codec; - - list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { - dev_dbg(codec->dev, "enabling jack detect for resume.\n"); - rt5670_jack_resume(codec); - break; - } - } - - return 0; -} - -/* SoC card */ -static struct snd_soc_card snd_soc_card_cht = { - .name = "cherrytrailcraudio", - .dai_link = cht_dailink, - .num_links = ARRAY_SIZE(cht_dailink), - .dapm_widgets = cht_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), - .dapm_routes = cht_audio_map, - .num_dapm_routes = ARRAY_SIZE(cht_audio_map), - .controls = cht_mc_controls, - .num_controls = ARRAY_SIZE(cht_mc_controls), - .suspend_pre = cht_suspend_pre, - .resume_post = cht_resume_post, -}; - -static int snd_cht_mc_probe(struct platform_device *pdev) -{ - int ret_val = 0; - - /* register the soc card */ - snd_soc_card_cht.dev = &pdev->dev; - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); - if (ret_val) { - dev_err(&pdev->dev, - "snd_soc_register_card failed %d\n", ret_val); - return ret_val; - } - platform_set_drvdata(pdev, &snd_soc_card_cht); - return ret_val; -} - -static struct platform_driver snd_cht_mc_driver = { - .driver = { - .name = "cht-bsw-rt5672", - }, - .probe = snd_cht_mc_probe, -}; - -module_platform_driver(snd_cht_mc_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); -MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:cht-bsw-rt5672"); diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c deleted file mode 100644 index 00fddd3f5dfb..000000000000 --- a/sound/soc/intel/haswell.c +++ /dev/null @@ -1,209 +0,0 @@ -/* - * Intel Haswell Lynxpoint SST Audio - * - * Copyright (C) 2013, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - */ - -#include -#include -#include -#include -#include -#include - -#include "sst-dsp.h" -#include "sst-haswell-ipc.h" - -#include "../codecs/rt5640.h" - -/* Haswell ULT platforms have a Headphone and Mic jack */ -static const struct snd_soc_dapm_widget haswell_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route haswell_rt5640_map[] = { - - {"Headphones", NULL, "HPOR"}, - {"Headphones", NULL, "HPOL"}, - {"IN2P", NULL, "Mic"}, - - /* CODEC BE connections */ - {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, - {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, -}; - -static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* The ADSP will covert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - /* set SSP0 to 16 bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); - return 0; -} - -static int haswell_rt5640_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; - - ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, - SND_SOC_CLOCK_IN); - - if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - /* set correct codec filter for DAI format and clock config */ - snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000); - - return ret; -} - -static struct snd_soc_ops haswell_rt5640_ops = { - .hw_params = haswell_rt5640_hw_params, -}; - -static int haswell_rtd_init(struct snd_soc_pcm_runtime *rtd) -{ - struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); - struct sst_hsw *haswell = pdata->dsp; - int ret; - - /* Set ADSP SSP port settings */ - ret = sst_hsw_device_set_config(haswell, SST_HSW_DEVICE_SSP_0, - SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, - SST_HSW_DEVICE_CLOCK_MASTER, 9); - if (ret < 0) { - dev_err(rtd->dev, "failed to set device config\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_dai_link haswell_rt5640_dais[] = { - /* Front End DAI links */ - { - .name = "System", - .stream_name = "System Playback/Capture", - .cpu_dai_name = "System Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .init = haswell_rtd_init, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .dpcm_capture = 1, - }, - { - .name = "Offload0", - .stream_name = "Offload0 Playback", - .cpu_dai_name = "Offload0 Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - }, - { - .name = "Offload1", - .stream_name = "Offload1 Playback", - .cpu_dai_name = "Offload1 Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - }, - { - .name = "Loopback", - .stream_name = "Loopback", - .cpu_dai_name = "Loopback Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 0, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "Codec", - .be_id = 0, - .cpu_dai_name = "snd-soc-dummy-dai", - .platform_name = "snd-soc-dummy", - .no_pcm = 1, - .codec_name = "i2c-INT33CA:00", - .codec_dai_name = "rt5640-aif1", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ignore_suspend = 1, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = haswell_ssp0_fixup, - .ops = &haswell_rt5640_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - }, -}; - -/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ -static struct snd_soc_card haswell_rt5640 = { - .name = "haswell-rt5640", - .owner = THIS_MODULE, - .dai_link = haswell_rt5640_dais, - .num_links = ARRAY_SIZE(haswell_rt5640_dais), - .dapm_widgets = haswell_widgets, - .num_dapm_widgets = ARRAY_SIZE(haswell_widgets), - .dapm_routes = haswell_rt5640_map, - .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map), - .fully_routed = true, -}; - -static int haswell_audio_probe(struct platform_device *pdev) -{ - haswell_rt5640.dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &haswell_rt5640); -} - -static struct platform_driver haswell_audio = { - .probe = haswell_audio_probe, - .driver = { - .name = "haswell-audio", - }, -}; - -module_platform_driver(haswell_audio) - -/* Module information */ -MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:haswell-audio"); diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c deleted file mode 100644 index 49c09a0add79..000000000000 --- a/sound/soc/intel/mfld_machine.c +++ /dev/null @@ -1,430 +0,0 @@ -/* - * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/sn95031.h" - -#define MID_MONO 1 -#define MID_STEREO 2 -#define MID_MAX_CAP 5 -#define MFLD_JACK_INSERT 0x04 - -enum soc_mic_bias_zones { - MFLD_MV_START = 0, - /* mic bias volutage range for Headphones*/ - MFLD_MV_HP = 400, - /* mic bias volutage range for American Headset*/ - MFLD_MV_AM_HS = 650, - /* mic bias volutage range for Headset*/ - MFLD_MV_HS = 2000, - MFLD_MV_UNDEFINED, -}; - -static unsigned int hs_switch; -static unsigned int lo_dac; -static struct snd_soc_codec *mfld_codec; - -struct mfld_mc_private { - void __iomem *int_base; - u8 interrupt_status; -}; - -struct snd_soc_jack mfld_jack; - -/*Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin mfld_jack_pins[] = { - { - .pin = "Headphones", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "AMIC1", - .mask = SND_JACK_MICROPHONE, - }, -}; - -/* jack detection voltage zones */ -static struct snd_soc_jack_zone mfld_zones[] = { - {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, - {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, -}; - -/* sound card controls */ -static const char *headset_switch_text[] = {"Earpiece", "Headset"}; - -static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; - -static const struct soc_enum headset_enum = - SOC_ENUM_SINGLE_EXT(2, headset_switch_text); - -static const struct soc_enum lo_enum = - SOC_ENUM_SINGLE_EXT(4, lo_text); - -static int headset_get_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = hs_switch; - return 0; -} - -static int headset_set_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_context *dapm = &card->dapm; - - if (ucontrol->value.integer.value[0] == hs_switch) - return 0; - - snd_soc_dapm_mutex_lock(dapm); - - if (ucontrol->value.integer.value[0]) { - pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); - } else { - pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT"); - } - - snd_soc_dapm_sync_unlocked(dapm); - - snd_soc_dapm_mutex_unlock(dapm); - - hs_switch = ucontrol->value.integer.value[0]; - - return 0; -} - -static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) -{ - snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL"); - snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR"); - snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL"); - snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR"); - snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT"); - snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT"); - if (hs_switch) { - snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); - } else { - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT"); - } -} - -static int lo_get_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = lo_dac; - return 0; -} - -static int lo_set_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_context *dapm = &card->dapm; - - if (ucontrol->value.integer.value[0] == lo_dac) - return 0; - - snd_soc_dapm_mutex_lock(dapm); - - /* we dont want to work with last state of lineout so just enable all - * pins and then disable pins not required - */ - lo_enable_out_pins(dapm); - - switch (ucontrol->value.integer.value[0]) { - case 0: - pr_debug("set vibra path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); - snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT"); - snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0); - break; - - case 1: - pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); - snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22); - break; - - case 2: - pr_debug("set spkr path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL"); - snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR"); - snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44); - break; - - case 3: - pr_debug("set null path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR"); - snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66); - break; - } - - snd_soc_dapm_sync_unlocked(dapm); - - snd_soc_dapm_mutex_unlock(dapm); - - lo_dac = ucontrol->value.integer.value[0]; - return 0; -} - -static const struct snd_kcontrol_new mfld_snd_controls[] = { - SOC_ENUM_EXT("Playback Switch", headset_enum, - headset_get_switch, headset_set_switch), - SOC_ENUM_EXT("Lineout Mux", lo_enum, - lo_get_switch, lo_set_switch), -}; - -static const struct snd_soc_dapm_widget mfld_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route mfld_map[] = { - {"Headphones", NULL, "HPOUTR"}, - {"Headphones", NULL, "HPOUTL"}, - {"Mic", NULL, "AMIC1"}, -}; - -static void mfld_jack_check(unsigned int intr_status) -{ - struct mfld_jack_data jack_data; - - if (!mfld_codec) - return; - - jack_data.mfld_jack = &mfld_jack; - jack_data.intr_id = intr_status; - - sn95031_jack_detection(mfld_codec, &jack_data); - /* TODO: add american headset detection post gpiolib support */ -} - -static int mfld_init(struct snd_soc_pcm_runtime *runtime) -{ - struct snd_soc_dapm_context *dapm = &runtime->card->dapm; - int ret_val; - - /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "Headphones"); - /* default is lineout NC, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); - lo_dac = 3; - hs_switch = 0; - /* we dont use linein in this so set to NC */ - snd_soc_dapm_disable_pin(dapm, "LINEINL"); - snd_soc_dapm_disable_pin(dapm, "LINEINR"); - - /* Headset and button jack detection */ - ret_val = snd_soc_card_jack_new(runtime->card, - "Intel(R) MID Audio Jack", SND_JACK_HEADSET | - SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack, - mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins)); - if (ret_val) { - pr_err("jack creation failed\n"); - return ret_val; - } - - ret_val = snd_soc_jack_add_zones(&mfld_jack, - ARRAY_SIZE(mfld_zones), mfld_zones); - if (ret_val) { - pr_err("adding jack zones failed\n"); - return ret_val; - } - - mfld_codec = runtime->codec; - - /* we want to check if anything is inserted at boot, - * so send a fake event to codec and it will read adc - * to find if anything is there or not */ - mfld_jack_check(MFLD_JACK_INSERT); - return ret_val; -} - -static struct snd_soc_dai_link mfld_msic_dailink[] = { - { - .name = "Medfield Headset", - .stream_name = "Headset", - .cpu_dai_name = "Headset-cpu-dai", - .codec_dai_name = "SN95031 Headset", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = mfld_init, - }, - { - .name = "Medfield Speaker", - .stream_name = "Speaker", - .cpu_dai_name = "Speaker-cpu-dai", - .codec_dai_name = "SN95031 Speaker", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Vibra", - .stream_name = "Vibra1", - .cpu_dai_name = "Vibra1-cpu-dai", - .codec_dai_name = "SN95031 Vibra1", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Haptics", - .stream_name = "Vibra2", - .cpu_dai_name = "Vibra2-cpu-dai", - .codec_dai_name = "SN95031 Vibra2", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Compress", - .stream_name = "Speaker", - .cpu_dai_name = "Compress-cpu-dai", - .codec_dai_name = "SN95031 Speaker", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, -}; - -/* SoC card */ -static struct snd_soc_card snd_soc_card_mfld = { - .name = "medfield_audio", - .owner = THIS_MODULE, - .dai_link = mfld_msic_dailink, - .num_links = ARRAY_SIZE(mfld_msic_dailink), - - .controls = mfld_snd_controls, - .num_controls = ARRAY_SIZE(mfld_snd_controls), - .dapm_widgets = mfld_widgets, - .num_dapm_widgets = ARRAY_SIZE(mfld_widgets), - .dapm_routes = mfld_map, - .num_dapm_routes = ARRAY_SIZE(mfld_map), -}; - -static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) -{ - struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; - - memcpy_fromio(&mc_private->interrupt_status, - ((void *)(mc_private->int_base)), - sizeof(u8)); - return IRQ_WAKE_THREAD; -} - -static irqreturn_t snd_mfld_jack_detection(int irq, void *data) -{ - struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - - mfld_jack_check(mc_drv_ctx->interrupt_status); - - return IRQ_HANDLED; -} - -static int snd_mfld_mc_probe(struct platform_device *pdev) -{ - int ret_val = 0, irq; - struct mfld_mc_private *mc_drv_ctx; - struct resource *irq_mem; - - pr_debug("snd_mfld_mc_probe called\n"); - - /* retrive the irq number */ - irq = platform_get_irq(pdev, 0); - - /* audio interrupt base of SRAM location where - * interrupts are stored by System FW */ - mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); - if (!mc_drv_ctx) { - pr_err("allocation failed\n"); - return -ENOMEM; - } - - irq_mem = platform_get_resource_byname( - pdev, IORESOURCE_MEM, "IRQ_BASE"); - if (!irq_mem) { - pr_err("no mem resource given\n"); - return -ENODEV; - } - mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, - resource_size(irq_mem)); - if (!mc_drv_ctx->int_base) { - pr_err("Mapping of cache failed\n"); - return -ENOMEM; - } - /* register for interrupt */ - ret_val = devm_request_threaded_irq(&pdev->dev, irq, - snd_mfld_jack_intr_handler, - snd_mfld_jack_detection, - IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); - if (ret_val) { - pr_err("cannot register IRQ\n"); - return ret_val; - } - /* register the soc card */ - snd_soc_card_mfld.dev = &pdev->dev; - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); - if (ret_val) { - pr_debug("snd_soc_register_card failed %d\n", ret_val); - return ret_val; - } - platform_set_drvdata(pdev, mc_drv_ctx); - pr_debug("successfully exited probe\n"); - return 0; -} - -static struct platform_driver snd_mfld_mc_driver = { - .driver = { - .name = "msic_audio", - }, - .probe = snd_mfld_mc_probe, -}; - -module_platform_driver(snd_mfld_mc_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); -MODULE_AUTHOR("Vinod Koul "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:msic-audio"); -- cgit v1.2.3-59-g8ed1b