diff options
Diffstat (limited to 'sound/soc/intel/boards')
| -rw-r--r-- | sound/soc/intel/boards/Makefile | 4 | ||||
| -rw-r--r-- | sound/soc/intel/boards/bdw-rt5677.c | 5 | ||||
| -rw-r--r-- | sound/soc/intel/boards/broadwell.c | 3 | ||||
| -rw-r--r-- | sound/soc/intel/boards/bxt_da7219_max98357a.c | 97 | ||||
| -rw-r--r-- | sound/soc/intel/boards/bxt_rt298.c | 3 | ||||
| -rw-r--r-- | sound/soc/intel/boards/bytcht_da7213.c | 283 | ||||
| -rw-r--r-- | sound/soc/intel/boards/bytcht_nocodec.c | 208 | ||||
| -rw-r--r-- | sound/soc/intel/boards/bytcr_rt5640.c | 109 |
8 files changed, 637 insertions, 75 deletions
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 5639f10774e6..56896e09445d 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -10,6 +10,8 @@ snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o +snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o @@ -26,6 +28,8 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o obj-$(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_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 53c6b4cbb1e1..14d9693c1641 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -193,13 +193,12 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) RT5677_CLK_SEL_I2S1_ASRC); /* Request rt5677 GPIO for headphone amp control */ - bdw_rt5677->gpio_hp_en = devm_gpiod_get_index(codec->dev, - "headphone-enable", 0, 0); + bdw_rt5677->gpio_hp_en = devm_gpiod_get(codec->dev, "headphone-enable", + GPIOD_OUT_LOW); if (IS_ERR(bdw_rt5677->gpio_hp_en)) { dev_err(codec->dev, "Can't find HP_AMP_SHDN_L gpio\n"); return PTR_ERR(bdw_rt5677->gpio_hp_en); } - gpiod_direction_output(bdw_rt5677->gpio_hp_en, 0); /* Create and initialize headphone jack */ if (!snd_soc_card_jack_new(rtd->card, "Headphone Jack", diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index faf865bb1765..6dcbbcefc25b 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -269,9 +269,6 @@ static struct snd_soc_card broadwell_rt286 = { static int broadwell_audio_probe(struct platform_device *pdev) { broadwell_rt286.dev = &pdev->dev; - - snd_soc_set_dmi_name(&broadwell_rt286, NULL); - return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286); } diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 2cda06cde4d1..3a8c4d954a91 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -55,6 +55,54 @@ enum { BXT_DPCM_AUDIO_HDMI3_PB, }; +static inline struct snd_soc_dai *bxt_get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + + if (!strncmp(rtd->codec_dai->name, BXT_DIALOG_CODEC_DAI, + strlen(BXT_DIALOG_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) +{ + int ret = 0; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = bxt_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if(SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, + DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); + if (ret) + dev_err(card->dev, "can't set codec sysclk configuration\n"); + + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -69,6 +117,8 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), }; static const struct snd_soc_dapm_route broxton_map[] = { @@ -109,6 +159,9 @@ static const struct snd_soc_dapm_route broxton_map[] = { /* DMIC */ {"dmic01_hifi", NULL, "DMIC01 Rx"}, {"DMIC01 Rx", NULL, "DMIC AIF"}, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, }; static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -243,49 +296,6 @@ static const struct snd_soc_ops broxton_da7219_fe_ops = { .startup = bxt_fe_startup, }; -static int broxton_da7219_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, - DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(codec_dai->dev, "can't set codec sysclk configuration\n"); - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - -static int broxton_da7219_hw_free(struct snd_pcm_substream *substream) -{ - 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_pll(codec_dai, 0, - DA7219_SYSCLK_MCLK, 0, 0); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - -static const struct snd_soc_ops broxton_da7219_ops = { - .hw_params = broxton_da7219_hw_params, - .hw_free = broxton_da7219_hw_free, -}; - static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -467,7 +477,6 @@ static struct snd_soc_dai_link broxton_dais[] = { SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, .be_hw_params_fixup = broxton_ssp_fixup, - .ops = &broxton_da7219_ops, .dpcm_playback = 1, .dpcm_capture = 1, }, diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 176c080a9818..1a68d043c803 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -274,12 +274,15 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream) * on this platform for PCM device we support: * 48Khz * stereo + * 16-bit audio */ runtime->hw.channels_max = 2; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c new file mode 100644 index 000000000000..18873e23f404 --- /dev/null +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -0,0 +1,283 @@ +/* + * bytcht-da7213.c - ASoc Machine driver for Intel Baytrail and + * Cherrytrail-based platforms, with Dialog DA7213 codec + * + * Copyright (C) 2017 Intel Corporation + * Author: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include <linux/module.h> +#include <linux/acpi.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <asm/platform_sst_audio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/da7213.h" +#include "../atom/sst-atom-controls.h" +#include "../common/sst-acpi.h" + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Mic"), + SOC_DAPM_PIN_SWITCH("Aux In"), +}; + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Aux In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "HPL"}, + {"Headphone Jack", NULL, "HPR"}, + + {"AUXL", NULL, "Aux In"}, + {"AUXR", NULL, "Aux In"}, + + /* Assume Mic1 is linked to Headset and Mic2 to on-board mic */ + {"MIC1", NULL, "Headset Mic"}, + {"MIC2", NULL, "Mic"}, + + /* SOC-codec link */ + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + + {"Playback", NULL, "ssp2 Tx"}, + {"ssp2 Rx", NULL, "Capture"}, +}; + +static int codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + int ret; + 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 convert 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); + + /* + * Default mode for SSP configuration is TDM 4 slot, override config + * with explicit setting to I2S 2ch 24-bit. The word length is set with + * dai_set_tdm_slot() since there is no other API exposed + */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) { + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); + return ret; + } + + return 0; +} + +static int aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 48000); +} + +static int 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; + + ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK, + 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(codec_dai->dev, "can't set codec sysclk configuration\n"); + + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7213_SYSCLK_PLL_SRM, 0, DA7213_PLL_FREQ_OUT_98304000); + if (ret < 0) { + dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret); + return -EIO; + } + + return ret; +} + +static int aif1_hw_free(struct snd_pcm_substream *substream) +{ + 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_pll(codec_dai, 0, + DA7213_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret); + return -EIO; + } + + return ret; +} + +static const struct snd_soc_ops aif1_ops = { + .startup = aif1_startup, +}; + +static const struct snd_soc_ops ssp2_ops = { + .hw_params = aif1_hw_params, + .hw_free = aif1_hw_free, + +}; + +static struct snd_soc_dai_link 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", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &aif1_ops, + }, + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-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, + .ops = &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", + .id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "da7213-hifi", + .codec_name = "i2c-DLGS7213:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = codec_fixup, + .nonatomic = true, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card bytcht_da7213_card = { + .name = "bytcht-da7213", + .owner = THIS_MODULE, + .dai_link = dailink, + .num_links = ARRAY_SIZE(dailink), + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .dapm_widgets = dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), +}; + +static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ + +static int bytcht_da7213_probe(struct platform_device *pdev) +{ + int ret_val = 0; + int i; + struct snd_soc_card *card; + struct sst_acpi_mach *mach; + const char *i2c_name = NULL; + int dai_index = 0; + + mach = (&pdev->dev)->platform_data; + card = &bytcht_da7213_card; + card->dev = &pdev->dev; + + /* fix index of codec dai */ + dai_index = MERR_DPCM_COMPR + 1; + for (i = 0; i < ARRAY_SIZE(dailink); i++) { + if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) { + dai_index = i; + break; + } + } + + /* fixup codec name based on HID */ + i2c_name = sst_acpi_find_name_from_hid(mach->id); + if (i2c_name != NULL) { + snprintf(codec_name, sizeof(codec_name), + "%s%s", "i2c-", i2c_name); + dailink[dai_index].codec_name = codec_name; + } + + ret_val = devm_snd_soc_register_card(&pdev->dev, card); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, card); + return ret_val; +} + +static struct platform_driver bytcht_da7213_driver = { + .driver = { + .name = "bytcht_da7213", + }, + .probe = bytcht_da7213_probe, +}; +module_platform_driver(bytcht_da7213_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail+DA7213 Machine driver"); +MODULE_AUTHOR("Pierre-Louis Bossart"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bytcht_da7213"); diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c new file mode 100644 index 000000000000..89853eeaaf9d --- /dev/null +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -0,0 +1,208 @@ +/* + * bytcht_nocodec.c - ASoc Machine driver for MinnowBoard Max and Up + * to make I2S signals observable on the Low-Speed connector. Audio codec + * is not managed by ASoC/DAPM + * + * Copyright (C) 2015-2017 Intel Corp + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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 <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../atom/sst-atom-controls.h" + +static const struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + + {"ssp2 Rx", NULL, "Mic"}, + {"Speaker", NULL, "ssp2 Tx"}, +}; + +static int 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); + int ret; + + /* The DSP will convert 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); + + /* + * Default mode for SSP configuration is TDM 4 slot, override config + * with explicit setting to I2S 2ch 24-bit. The word length is set with + * dai_set_tdm_slot() since there is no other API exposed + */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + + if (ret < 0) { + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); + return ret; + } + + 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 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 aif1_ops = { + .startup = aif1_startup, +}; + +static struct snd_soc_dai_link dais[] = { + [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, + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &aif1_ops, + }, + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .ops = &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-LowSpeed Connector", + .id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = codec_fixup, + .ignore_suspend = 1, + .nonatomic = true, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* SoC card */ +static struct snd_soc_card bytcht_nocodec_card = { + .name = "bytcht-nocodec", + .owner = THIS_MODULE, + .dai_link = dais, + .num_links = ARRAY_SIZE(dais), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .fully_routed = true, +}; + +static int snd_bytcht_nocodec_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + bytcht_nocodec_card.dev = &pdev->dev; + + ret_val = devm_snd_soc_register_card(&pdev->dev, &bytcht_nocodec_card); + + if (ret_val) { + dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", + ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &bytcht_nocodec_card); + return ret_val; +} + +static struct platform_driver snd_bytcht_nocodec_mc_driver = { + .driver = { + .name = "bytcht_nocodec", + }, + .probe = snd_bytcht_nocodec_mc_probe, +}; +module_platform_driver(snd_bytcht_nocodec_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Nocodec Machine driver"); +MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bytcht_nocodec"); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 9e2a3404a836..4a76b099a508 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -19,6 +19,7 @@ #include <linux/init.h> #include <linux/module.h> +#include <linux/moduleparam.h> #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/device.h> @@ -56,35 +57,88 @@ enum { struct byt_rt5640_private { struct clk *mclk; }; +static bool is_bytcr; static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; +static unsigned int quirk_override; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) { - if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_DMIC1_MAP) - dev_info(dev, "quirk DMIC1_MAP enabled"); - if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_DMIC2_MAP) - dev_info(dev, "quirk DMIC2_MAP enabled"); - if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_IN1_MAP) - dev_info(dev, "quirk IN1_MAP enabled"); - if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_IN3_MAP) - dev_info(dev, "quirk IN3_MAP enabled"); - if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) - dev_info(dev, "quirk DMIC enabled"); + int map; + bool has_dmic = false; + bool has_mclk = false; + bool has_ssp0 = false; + bool has_ssp0_aif1 = false; + bool has_ssp0_aif2 = false; + bool has_ssp2_aif2 = false; + + map = BYT_RT5640_MAP(byt_rt5640_quirk); + switch (map) { + case BYT_RT5640_DMIC1_MAP: + dev_info(dev, "quirk DMIC1_MAP enabled\n"); + has_dmic = true; + break; + case BYT_RT5640_DMIC2_MAP: + dev_info(dev, "quirk DMIC2_MAP enabled\n"); + has_dmic = true; + break; + case BYT_RT5640_IN1_MAP: + dev_info(dev, "quirk IN1_MAP enabled\n"); + break; + case BYT_RT5640_IN3_MAP: + dev_info(dev, "quirk IN3_MAP enabled\n"); + break; + default: + dev_err(dev, "quirk map 0x%x is not supported, microphone input will not work\n", map); + break; + } + if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) { + if (has_dmic) + dev_info(dev, "quirk DMIC enabled\n"); + else + dev_err(dev, "quirk DMIC enabled but no DMIC input set, will be ignored\n"); + } if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) - dev_info(dev, "quirk MONO_SPEAKER enabled"); - if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC) - dev_info(dev, "quirk DIFF_MIC enabled"); - if (byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) - dev_info(dev, "quirk SSP2_AIF2 enabled"); - if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) - dev_info(dev, "quirk SSP0_AIF1 enabled"); - if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2) - dev_info(dev, "quirk SSP0_AIF2 enabled"); - if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) - dev_info(dev, "quirk MCLK_EN enabled"); - if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) - dev_info(dev, "quirk MCLK_25MHZ enabled"); + dev_info(dev, "quirk MONO_SPEAKER enabled\n"); + if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC) { + if (!has_dmic) + dev_info(dev, "quirk DIFF_MIC enabled\n"); + else + dev_info(dev, "quirk DIFF_MIC enabled but DMIC input selected, will be ignored\n"); + } + if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) { + dev_info(dev, "quirk SSP0_AIF1 enabled\n"); + has_ssp0 = true; + has_ssp0_aif1 = true; + } + if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2) { + dev_info(dev, "quirk SSP0_AIF2 enabled\n"); + has_ssp0 = true; + has_ssp0_aif2 = true; + } + if (byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) { + dev_info(dev, "quirk SSP2_AIF2 enabled\n"); + has_ssp2_aif2 = true; + } + if (is_bytcr && !has_ssp0) + dev_err(dev, "Invalid routing, bytcr detected but no SSP0-based quirk, audio cannot work with SSP2 on bytcr\n"); + if (has_ssp0_aif1 && has_ssp0_aif2) + dev_err(dev, "Invalid routing, SSP0 cannot be connected to both AIF1 and AIF2\n"); + if (has_ssp0 && has_ssp2_aif2) + dev_err(dev, "Invalid routing, cannot have both SSP0 and SSP2 connected to codec\n"); + + if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) { + dev_info(dev, "quirk MCLK_EN enabled\n"); + has_mclk = true; + } + if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) { + if (has_mclk) + dev_info(dev, "quirk MCLK_25MHZ enabled\n"); + else + dev_err(dev, "quirk MCLK_25MHZ enabled but quirk MCLK not selected, will be ignored\n"); + } } @@ -128,7 +182,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, ret = clk_prepare_enable(priv->mclk); if (ret < 0) { dev_err(card->dev, - "could not configure MCLK state"); + "could not configure MCLK state\n"); return ret; } } @@ -710,8 +764,8 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) int i; int dai_index; struct byt_rt5640_private *priv; - bool is_bytcr = false; + is_bytcr = false; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); if (!priv) return -ENOMEM; @@ -806,6 +860,11 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* check quirks before creating card */ dmi_check_system(byt_rt5640_quirk_table); + if (quirk_override) { + dev_info(&pdev->dev, "Overriding quirk 0x%x => 0x%x\n", + (unsigned int)byt_rt5640_quirk, quirk_override); + byt_rt5640_quirk = quirk_override; + } log_quirks(&pdev->dev); if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) || |