From b3ec72ace939b0abd75d5d875e77cf0b777debb7 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 20 Feb 2017 19:46:10 +0200 Subject: ASoC: Intel: bdw-rt5677: Use devm_gpiod_get() Since index is always 0 replace devm_gpiod_get_index() by devm_gpiod_get() and apply proper flags. Signed-off-by: Andy Shevchenko Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5677.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc/intel/boards') 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", -- cgit v1.2.3 From 82cf89de2c9c2efcafde452ed76f85e7ef7f6ce0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 9 Mar 2017 18:18:59 -0600 Subject: ASoC: Intel: add machine driver for BYT/CHT + DA7213 Add new machine driver, tested with Ard-Audio-DA7212 [1] connected to MinnowBoardMAX Turbot. The MCLK is managed by the codec driver using the "mclk" handle to reuse existing code, but it could just as well be handled by this machine driver. [1] http://www.dialog-semiconductor.com/content/ard-audio-da7212 Signed-off-by: Pierre-Louis Bossart Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 12 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_da7213.c | 283 +++++++++++++++++++++++++++++++++ 3 files changed, 297 insertions(+) create mode 100644 sound/soc/intel/boards/bytcht_da7213.c (limited to 'sound/soc/intel/boards') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 526855ad479e..8d32fc336eb8 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -202,6 +202,18 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH platforms with MAX98090 audio codec it also can support TI jack chip as aux device. If unsure select "N". +config SND_SOC_INTEL_BYT_CHT_DA7213_MACH + tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail with DA7212/7213 codec" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_DA7213 + select SND_SST_ATOM_HIFI2_PLATFORM + select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI + help + This adds support for ASoC machine driver for Intel(R) Baytrail & CherryTrail + platforms with DA7212/7213 audio codec. + If unsure select "N". + config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 5639f10774e6..8a5d9edd16ee 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -10,6 +10,7 @@ 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-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 +27,7 @@ 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_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/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 + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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 "../../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-: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"); -- cgit v1.2.3 From 759db1c4660b558345573c7476a45c76a6aa07d2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 9 Mar 2017 18:19:01 -0600 Subject: ASoC: Intel: boards: add card for MinnowBoardMax/Up I2S access Add card with dummy codec and DAI to make I2S signals observable. Uses Mic and Speaker pins/widgets to control DAPM Signed-off-by: Pierre-Louis Bossart Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 12 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_nocodec.c | 208 ++++++++++++++++++++++++++++++++ 3 files changed, 222 insertions(+) create mode 100644 sound/soc/intel/boards/bytcht_nocodec.c (limited to 'sound/soc/intel/boards') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 8d32fc336eb8..67968ef3bbda 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -214,6 +214,18 @@ config SND_SOC_INTEL_BYT_CHT_DA7213_MACH platforms with DA7212/7213 audio codec. If unsure select "N". +config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH + tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SST_ATOM_HIFI2_PLATFORM + select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI + help + This adds support for ASoC machine driver for the MinnowBoard Max or + Up boards and provides access to I2S signals on the Low-Speed + connector + If unsure select "N". + config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 8a5d9edd16ee..56896e09445d 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -11,6 +11,7 @@ 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 @@ -28,6 +29,7 @@ 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/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 +#include +#include +#include +#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 "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bytcht_nocodec"); -- cgit v1.2.3 From d7fba9dcf6cd74cb8a8ab1c4c2629a3f7df00bc9 Mon Sep 17 00:00:00 2001 From: Harsha Priya Date: Wed, 15 Mar 2017 16:28:25 -0700 Subject: ASoC: Intel: Update bxt_da7219_max98357a to add a new This patch adds a platform clock widget to turn off the clock only when both headset capture and headset playback are not in use. This removes turning off the clock in hw_free so that the clock is on when either capture or playback of headset is in progress. Signed-off-by: Harsha Priya Signed-off-by: Sathyanarayana Nujella Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 97 +++++++++++++++------------ 1 file changed, 53 insertions(+), 44 deletions(-) (limited to 'sound/soc/intel/boards') 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, }, -- cgit v1.2.3 From e59ed0875b0681ebd1e5062b739742f98f24274c Mon Sep 17 00:00:00 2001 From: G Kranthi Date: Fri, 24 Mar 2017 23:10:27 +0530 Subject: ASoC: Intel: Skylake: Add 16-bit constraint to FE bxt_rt298 machine Add constraint to FE to restrict sample format to 16-bit for bxt_rt298 machine Signed-off-by: G Kranthi Signed-off-by: Jeeja KP Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_rt298.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc/intel/boards') 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); -- cgit v1.2.3 From 861886d338f7dd802be972e770a695bba62f397a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 24 Apr 2017 08:54:42 +0200 Subject: ASoC: Call snd_soc_set_dmi_name() unconditionally Since recently UCM can pick up a configuration specific to the board via card longname field, and we introduced a helper function snd_soc_set_dmi_name() for that. So far, it was used only in one place (sound/soc/intel/boards/broadwell.c), but it should be more widely applied. This patch puts a big hammer for that: it lets snd_soc_register_card() calling snd_soc_set_dmi_name() unconditionally, so that all x86 devices get the better longname string. This would have no impact for other systems without DMI support, as snd_soc_set_dmi_name() is no-op on them. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/broadwell.c | 3 --- sound/soc/soc-core.c | 3 +++ 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/intel/boards') 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/soc-core.c b/sound/soc/soc-core.c index 52f6d9c28df9..05c4d9564b0b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2192,6 +2192,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, card->num_of_dapm_routes); + /* try to set some sane longname if DMI is available */ + snd_soc_set_dmi_name(card, NULL); + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), -- cgit v1.2.3 From 9f2cf73ed65b598514e4858ca3d602710718ab93 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 23 Apr 2017 09:22:56 +0200 Subject: ASoC: bytcr_rt5640: Allow quirk set via module option The bytcr-rt5640 driver has a few quirk setups depending on the board, where the quirk value is set by DMI matching. When you have a new device to add the support, you often experience to try the different quirk by trial-and-error. Or, you may have a development model that still has no proper DMI string. In either case, you'd need to compile the driver at each time. This patch introduces a module option to override the quirk value on the fly. User can boot like snd-soc-sst-bytcr-rt5640.quirk=0x4004 to override the default value without recompilation. It's a raw value, so user needs to check the source code for the meaning of each bit. Signed-off-by: Takashi Iwai Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound/soc/intel/boards') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5c7219fb3aa8..0ac32788f216 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -19,6 +19,7 @@ #include #include +#include #include #include #include @@ -58,6 +59,9 @@ struct byt_rt5640_private { }; static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; +static unsigned int quirk_override; +module_param_named(quirk, quirk_override, int, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) { @@ -806,6 +810,11 @@ 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 => 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) || -- cgit v1.2.3 From 0b2c9f88b94cd40f6a27641f0bac02a7ba185e39 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 24 Apr 2017 23:34:30 +0200 Subject: ASoC: bytcr_rt5640: Fix a typo and quirk parameter type The previous patch for adding the quirk module option had a typo in its info print, which results in a weird output. Also, the parameter type should be rather unsigned int instead of signed int. Fixes: 9f2cf73ed65b ("ASoC: bytcr_rt5640: Allow quirk set via module option") Reported-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/intel/boards') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 0ac32788f216..5ca09cadf39f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -60,7 +60,7 @@ struct byt_rt5640_private { static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; static unsigned int quirk_override; -module_param_named(quirk, quirk_override, int, 0444); +module_param_named(quirk, quirk_override, uint, 0444); MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) @@ -811,7 +811,7 @@ 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 => 0x%x\n", + dev_info(&pdev->dev, "Overriding quirk 0x%x => 0x%x\n", (unsigned int)byt_rt5640_quirk, quirk_override); byt_rt5640_quirk = quirk_override; } -- cgit v1.2.3 From cb67d7651676e8c8f2e40587ef591da057806c57 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 24 Apr 2017 16:34:33 -0500 Subject: ASoC: bytcr_rt5640: log quirk configuration errors Now that quirks can be overridden with a module parameter, log errors so that non-sensical quirks introduced by mistake are identified. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 100 +++++++++++++++++++++++++--------- 1 file changed, 75 insertions(+), 25 deletions(-) (limited to 'sound/soc/intel/boards') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5ca09cadf39f..f063368edef9 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -57,6 +57,7 @@ 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; @@ -65,30 +66,79 @@ 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"); + } } @@ -132,7 +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; } } @@ -714,8 +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; -- cgit v1.2.3