diff options
author | Takashi Iwai <tiwai@suse.de> | 2020-01-27 17:45:44 +0100 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2020-01-27 17:45:44 +0100 |
commit | 90fb04f890bcb7384b4d4c216dc2640b0a870df3 (patch) | |
tree | f4c35d625351cd8034bdd37bb3e19926bad674db /sound/soc/intel | |
parent | 9b132f2764e5a04bfb2e6c6f51cf5a729617d6a4 (diff) | |
parent | a7196caf83ea9e9b56c5c8c95fbfe0d45acec46b (diff) | |
download | linux-90fb04f890bcb7384b4d4c216dc2640b0a870df3.tar.bz2 |
Merge tag 'asoc-v5.6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.6
A pretty big release this time around, a lot of new drivers and both
Morimoto-san and Takashi were doing subsystem wide updates as well:
- Further big refactorings from Morimoto-san simplifying the core
interfaces and moving things to the component level.
- Transition of drivers to managed buffer allocation and removal of
redundant PCM ioctls.
- New driver support for Ingenic JZ4770, Mediatek MT6660, Qualcomm
WCD934x and WSA881x, and Realtek RT700, RT711, RT715, RT1011, RT1015
and RT1308.
Diffstat (limited to 'sound/soc/intel')
38 files changed, 1268 insertions, 344 deletions
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 47e3d1943d7e..340bd2be39a7 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -387,27 +387,6 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, return ret_val; } -static int sst_media_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - int ret; - - ret = - snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(params)); - if (ret) - return ret; - memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - return 0; -} - -static int sst_media_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - return snd_pcm_lib_free_pages(substream); -} - static int sst_enable_ssp(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -473,8 +452,6 @@ static const struct snd_soc_dai_ops sst_media_dai_ops = { .startup = sst_media_open, .shutdown = sst_media_close, .prepare = sst_media_prepare, - .hw_params = sst_media_hw_params, - .hw_free = sst_media_hw_free, .mute_stream = sst_media_digital_mute, }; @@ -677,7 +654,7 @@ static int sst_soc_pcm_new(struct snd_soc_component *component, if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { - snd_pcm_lib_preallocate_pages_for_all(pcm, + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); @@ -705,7 +682,6 @@ static const struct snd_soc_component_driver sst_soc_platform_drv = { .probe = sst_soc_probe, .remove = sst_soc_remove, .open = sst_soc_open, - .ioctl = snd_soc_pcm_lib_ioctl, .trigger = sst_soc_trigger, .pointer = sst_soc_pointer, .compr_ops = &sst_platform_compr_ops, diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 1d780fcc448c..53383055c8dc 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -102,8 +102,6 @@ static int sst_byt_pcm_hw_params(struct snd_soc_component *component, return ret; } - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - ret = sst_byt_stream_buffer(byt, pcm_data->stream, substream->dma_buffer.addr, params_buffer_bytes(params)); @@ -121,17 +119,6 @@ static int sst_byt_pcm_hw_params(struct snd_soc_component *component, return 0; } -static int sst_byt_pcm_hw_free(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - dev_dbg(rtd->dev, "PCM: hw_free\n"); - snd_pcm_lib_free_pages(substream); - - return 0; -} - static int sst_byt_pcm_restore_stream_context(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -315,9 +302,8 @@ static int sst_byt_pcm_new(struct snd_soc_component *component, if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { size = sst_byt_pcm_hardware.buffer_bytes_max; - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - pdata->dma_dev, - size, size); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + pdata->dma_dev, size, size); } return 0; @@ -373,9 +359,7 @@ static const struct snd_soc_component_driver byt_dai_component = { .probe = sst_byt_pcm_probe, .open = sst_byt_pcm_open, .close = sst_byt_pcm_close, - .ioctl = snd_soc_pcm_lib_ioctl, .hw_params = sst_byt_pcm_hw_params, - .hw_free = sst_byt_pcm_hw_free, .trigger = sst_byt_pcm_trigger, .pointer = sst_byt_pcm_pointer, .mmap = sst_byt_pcm_mmap, diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index ef20316e83d1..9ca2567d0059 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -13,6 +13,19 @@ menuconfig SND_SOC_INTEL_MACH if SND_SOC_INTEL_MACH +config SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES + bool "Use more user friendly long card names" + help + Some drivers report the I/O configuration to userspace through the + soundcard's long card name in the control user space AP. An unfortunate + side effect is that this long name may also be used by the GUI, + confusing users with information they don't need. + This option prevents the long name from being modified, and the I/O + configuration will be provided through a different component interface. + Select Y if userspace like UCM (Use Case Manager) uses the component + interface. + If unsure select N. + if SND_SOC_INTEL_HASWELL config SND_SOC_INTEL_HASWELL_MACH @@ -31,12 +44,27 @@ endif ## SND_SOC_INTEL_HASWELL if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL +config SND_SOC_INTEL_BDW_RT5650_MACH + tristate "Broadwell with RT5650 codec" + depends on I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT5645 + help + This adds the ASoC machine driver for Intel Broadwell platforms with + the RT5650 codec. + Say Y if you have such a device. + If unsure select "N". + config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on I2C depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST depends on GPIOLIB || COMPILE_TEST depends on X86_INTEL_LPSS || COMPILE_TEST + depends on SPI_MASTER + select SPI_PXA2XX + select SND_SOC_RT5677_SPI select SND_SOC_RT5677 help This adds support for Intel Broadwell platform based boards with @@ -261,6 +289,7 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC + select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON @@ -387,6 +416,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH select SND_SOC_RT5682 select SND_SOC_MAX98357A select SND_SOC_DMIC + select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for Geminilake platforms @@ -400,6 +430,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" + select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI select SND_SOC_DMIC # SND_SOC_HDAC_HDA is already selected @@ -419,6 +450,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_RT5682 select SND_SOC_DMIC + select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for SOF platforms @@ -448,6 +480,7 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH select SND_SOC_RT5682 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI + select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platform with RT1011 + RT5682 I2S codec. @@ -456,4 +489,22 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK +if SND_SOC_SOF_JASPERLAKE + +config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH + tristate "SOF with DA7219 and MAX98373 in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_DA7219 + select SND_SOC_MAX98373 + select SND_SOC_DMIC + select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC + help + This adds support for ASoC machine driver for SOF platforms + with DA7219 + MAX98373 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + +endif ## SND_SOC_SOF_JASPERLAKE + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index ba1aa89db09d..b74ddd49bd39 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ 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-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o @@ -28,6 +29,7 @@ snd-soc-skl_rt286-objs := skl_rt286.o snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o hda_dsp_common.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-sof_da7219_max98373-objs := sof_da7219_max98373.o hda_dsp_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o @@ -37,6 +39,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da721 obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o @@ -58,3 +61,5 @@ 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-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH) += snd-soc-sof_da7219_max98373.o + diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c new file mode 100644 index 000000000000..1a302436d450 --- /dev/null +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -0,0 +1,327 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * ASoC machine driver for Intel Broadwell platforms with RT5650 codec + * + * Copyright 2019, The Chromium OS Authors. All rights reserved. + */ + +#include <linux/delay.h> +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> + +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" + +#include "../../codecs/rt5645.h" + +struct bdw_rt5650_priv { + struct gpio_desc *gpio_hp_en; + struct snd_soc_component *component; +}; + +static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("DMIC Pair1", NULL), + SND_SOC_DAPM_MIC("DMIC Pair2", NULL), +}; + +static const struct snd_soc_dapm_route bdw_rt5650_map[] = { + /* Speakers */ + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + + /* Headset jack connectors */ + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + + /* Digital MICs + * DMIC Pair1 are the two DMICs connected on the DMICN1 connector. + * DMIC Pair2 are the two DMICs connected on the DMICN2 connector. + * Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2 + * are on the right side. + */ + {"DMIC L1", NULL, "DMIC Pair1"}, + {"DMIC R1", NULL, "DMIC Pair1"}, + {"DMIC L2", NULL, "DMIC Pair2"}, + {"DMIC R2", NULL, "DMIC Pair2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static const struct snd_kcontrol_new bdw_rt5650_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("DMIC Pair1"), + SOC_DAPM_PIN_SWITCH("DMIC Pair2"), +}; + + +static struct snd_soc_jack headphone_jack; +static struct snd_soc_jack mic_jack; + +static struct snd_soc_jack_pin headphone_jack_pin = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +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, max 4-channels */ + rate->min = rate->max = 48000; + channels->min = 2; + channels->max = 4; + + /* set SSP0 to 24 bit */ + snd_mask_set_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int bdw_rt5650_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; + + /* Workaround: set codec PLL to 19.2MHz that PLL source is + * from MCLK(24MHz) to conform 2.4MHz DMIC clock. + */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + 24000000, 19200000); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* The actual MCLK freq is 24MHz. The codec is told that MCLK is + * 24.576MHz to satisfy the requirement of rl6231_get_clk_info. + * ASRC is enabled on AD and DA filters to ensure good audio quality. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000, + 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 bdw_rt5650_ops = { + .hw_params = bdw_rt5650_hw_params, +}; + +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) +static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct sst_pdata *pdata = dev_get_platdata(component->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings + * clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz + */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + return 0; +} +#endif + +static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd) +{ + struct bdw_rt5650_priv *bdw_rt5650 = + snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1. + * The ASRC clock source is clk_i2s1_asrc. + */ + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER | + RT5645_AD_MONO_L_FILTER | + RT5645_AD_MONO_R_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(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* Create and initialize headphone jack */ + if (snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &headphone_jack, + &headphone_jack_pin, 1)) { + dev_err(component->dev, "Can't create headphone jack\n"); + } + + /* Create and initialize mic jack */ + if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &mic_jack, &mic_jack_pin, 1)) { + dev_err(component->dev, "Can't create mic jack\n"); + } + + rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL); + + bdw_rt5650->component = component; + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +SND_SOC_DAILINK_DEF(dummy, + DAILINK_COMP_ARRAY(COMP_DUMMY())); + +SND_SOC_DAILINK_DEF(fe, + DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); + +SND_SOC_DAILINK_DEF(be, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) +SND_SOC_DAILINK_DEF(ssp0_port, + DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); +#else +SND_SOC_DAILINK_DEF(ssp0_port, + DAILINK_COMP_ARRAY(COMP_DUMMY())); +#endif + +static struct snd_soc_dai_link bdw_rt5650_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .dynamic = 1, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + .init = bdw_rt5650_rtd_init, +#endif + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(fe, dummy, platform), + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | 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 = &bdw_rt5650_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = bdw_rt5650_init, + SND_SOC_DAILINK_REG(ssp0_port, be, platform), + }, +}; + +/* ASoC machine driver for Broadwell DSP + RT5650 */ +static struct snd_soc_card bdw_rt5650_card = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + .dai_link = bdw_rt5650_dais, + .num_links = ARRAY_SIZE(bdw_rt5650_dais), + .dapm_widgets = bdw_rt5650_widgets, + .num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets), + .dapm_routes = bdw_rt5650_map, + .num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map), + .controls = bdw_rt5650_controls, + .num_controls = ARRAY_SIZE(bdw_rt5650_controls), + .fully_routed = true, +}; + +static int bdw_rt5650_probe(struct platform_device *pdev) +{ + struct bdw_rt5650_priv *bdw_rt5650; + struct snd_soc_acpi_mach *mach; + int ret; + + bdw_rt5650_card.dev = &pdev->dev; + + /* Allocate driver private struct */ + bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv), + GFP_KERNEL); + if (!bdw_rt5650) + return -ENOMEM; + + /* override plaform name, if required */ + mach = (&pdev->dev)->platform_data; + ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card, + mach->mach_params.platform); + + if (ret) + return ret; + + snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650); + + return devm_snd_soc_register_card(&pdev->dev, &bdw_rt5650_card); +} + +static struct platform_driver bdw_rt5650_audio = { + .probe = bdw_rt5650_probe, + .driver = { + .name = "bdw-rt5650", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(bdw_rt5650_audio) + +/* Module information */ +MODULE_AUTHOR("Ben Zhang <benzh@chromium.org>"); +MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bdw-rt5650"); diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 2af8e5a62da8..bb643c99069d 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -295,6 +295,14 @@ SND_SOC_DAILINK_DEF(platform, SND_SOC_DAILINK_DEF(be, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-RT5677CE:00", "rt5677-aif1"))); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) +SND_SOC_DAILINK_DEF(ssp0_port, + DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); +#else +SND_SOC_DAILINK_DEF(ssp0_port, + DAILINK_COMP_ARRAY(COMP_DUMMY())); +#endif + /* Wake on voice interface */ SND_SOC_DAILINK_DEFS(dsp, DAILINK_COMP_ARRAY(COMP_CPU("spi-RT5677AA:00")), @@ -342,7 +350,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, .init = bdw_rt5677_init, - SND_SOC_DAILINK_REG(dummy, be, dummy), + SND_SOC_DAILINK_REG(ssp0_port, be, platform), }, }; diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index db7e1e87156d..b9c12e24c70b 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -164,6 +164,14 @@ SND_SOC_DAILINK_DEF(platform, SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1"))); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) +SND_SOC_DAILINK_DEF(ssp0_port, + DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); +#else +SND_SOC_DAILINK_DEF(ssp0_port, + DAILINK_COMP_ARRAY(COMP_DUMMY())); +#endif + /* broadwell digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link broadwell_rt286_dais[] = { /* Front End DAI links */ @@ -218,7 +226,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .ops = &broadwell_rt286_ops, .dpcm_playback = 1, .dpcm_capture = 1, - SND_SOC_DAILINK_REG(dummy, codec, dummy), + SND_SOC_DAILINK_REG(ssp0_port, codec, platform), }, }; diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 5873abb46441..33b13f3ca152 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -161,13 +161,13 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* The ADSP will convert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = DUAL_CHANNEL; + chan->min = chan->max = DUAL_CHANNEL; /* set SSP to 24 bit */ snd_mask_none(fmt); @@ -313,12 +313,12 @@ static const struct snd_soc_ops broxton_da7219_fe_ops = { static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); if (params_channels(params) == 2) - channels->min = channels->max = 2; + chan->min = chan->max = 2; else - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index eabf9d8468ae..067a97e7e6a8 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -206,13 +206,13 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* The ADSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; /* set SSP5 to 24 bit */ snd_mask_none(fmt); @@ -255,9 +255,9 @@ static const struct snd_pcm_hw_constraint_list constraints_rates = { static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 54e97455d7f6..0adc5a5e134a 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -234,14 +234,6 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) return 0; } -static const struct snd_soc_pcm_stream byt_cht_es8316_dai_params = { - .formats = SNDRV_PCM_FMTBIT_S24_LE, - .rate_min = 48000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, -}; - static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -360,7 +352,10 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { /* SoC card */ static char codec_name[SND_ACPI_I2C_ID_LEN]; +#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) static char long_name[50]; /* = "bytcht-es8316-*-spk-*-mic" */ +#endif +static char components_string[32]; /* = "cfg-spk:* cfg-mic:* */ static int byt_cht_es8316_suspend(struct snd_soc_card *card) { @@ -573,11 +568,19 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) } } - /* register the soc card */ + snprintf(components_string, sizeof(components_string), + "cfg-spk:%s cfg-mic:%s", + (quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "1" : "2", + mic_name[BYT_CHT_ES8316_MAP(quirk)]); + byt_cht_es8316_card.components = components_string; +#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) snprintf(long_name, sizeof(long_name), "bytcht-es8316-%s-spk-%s-mic", (quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "mono" : "stereo", mic_name[BYT_CHT_ES8316_MAP(quirk)]); byt_cht_es8316_card.long_name = long_name; +#endif + + /* register the soc card */ snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); ret = devm_snd_soc_register_card(dev, &byt_cht_es8316_card); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 243f683bc02a..6bd9ae813be2 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -933,14 +933,6 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) return 0; } -static const struct snd_soc_pcm_stream byt_rt5640_dai_params = { - .formats = SNDRV_PCM_FMTBIT_S24_LE, - .rate_min = 48000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, -}; - static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -948,65 +940,43 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - int ret; + int ret, bits; /* The DSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; channels->min = channels->max = 2; if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { /* set SSP0 to 16-bit */ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); - - /* - * Default mode for SSP configuration is TDM 4 slot, override config - * with explicit setting to I2S 2ch 16-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, 16); - if (ret < 0) { - dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); - return ret; - } - + bits = 16; } else { - /* set SSP2 to 24-bit */ params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + bits = 24; + } - /* - * 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; - } + /* + * Default mode for SSP configuration is TDM 4 slot, override config + * with explicit setting to I2S 2ch. 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; - } + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits); + if (ret < 0) { + dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); + return ret; } + return 0; } @@ -1084,9 +1054,10 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { /* SoC card */ static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5640_codec_aif_name[12]; /* = "rt5640-aif[1|2]" */ -static char byt_rt5640_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ +#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) static char byt_rt5640_long_name[40]; /* = "bytcr-rt5640-*-spk-*-mic" */ +#endif +static char byt_rt5640_components[32]; /* = "cfg-spk:* cfg-mic:*" */ static int byt_rt5640_suspend(struct snd_soc_card *card) { @@ -1266,28 +1237,12 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) log_quirks(&pdev->dev); if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - - /* fixup codec aif name */ - snprintf(byt_rt5640_codec_aif_name, - sizeof(byt_rt5640_codec_aif_name), - "%s", "rt5640-aif2"); - - byt_rt5640_dais[dai_index].codecs->dai_name = - byt_rt5640_codec_aif_name; - } + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + byt_rt5640_dais[dai_index].codecs->dai_name = "rt5640-aif2"; if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - - /* fixup cpu dai name name */ - snprintf(byt_rt5640_cpu_dai_name, - sizeof(byt_rt5640_cpu_dai_name), - "%s", "ssp0-port"); - - byt_rt5640_dais[dai_index].cpus->dai_name = - byt_rt5640_cpu_dai_name; - } + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + byt_rt5640_dais[dai_index].cpus->dai_name = "ssp0-port"; if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); @@ -1309,12 +1264,19 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } } + snprintf(byt_rt5640_components, sizeof(byt_rt5640_components), + "cfg-spk:%s cfg-mic:%s", + (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? "1" : "2", + map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]); + byt_rt5640_card.components = byt_rt5640_components; +#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) snprintf(byt_rt5640_long_name, sizeof(byt_rt5640_long_name), "bytcr-rt5640-%s-spk-%s-mic", (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? "mono" : "stereo", map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]); byt_rt5640_card.long_name = byt_rt5640_long_name; +#endif /* override plaform name, if required */ platform_name = mach->mach_params.platform; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 4606f6f582d6..5074bb53f98e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -656,14 +656,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) return 0; } -static const struct snd_soc_pcm_stream byt_rt5651_dai_params = { - .formats = SNDRV_PCM_FMTBIT_S24_LE, - .rate_min = 48000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, -}; - static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -795,9 +787,10 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { /* SoC card */ static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ -static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ +#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */ +#endif +static char byt_rt5651_components[50]; /* = "cfg-spk:* cfg-mic:*" */ static int byt_rt5651_suspend(struct snd_soc_card *card) { @@ -876,7 +869,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) const char *platform_name; struct acpi_device *adev; struct device *codec_dev; - const char *hp_swapped; bool is_bytcr = false; int ret_val = 0; int dai_index = 0; @@ -996,10 +988,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) if (byt_rt5651_gpios) { devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); - priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( - &pdev->dev, "ext-amp-enable", 0, - codec_dev->fwnode, - GPIOD_OUT_LOW, "speaker-amp"); + priv->ext_amp_gpio = devm_fwnode_gpiod_get(&pdev->dev, + codec_dev->fwnode, + "ext-amp-enable", + GPIOD_OUT_LOW, + "speaker-amp"); if (IS_ERR(priv->ext_amp_gpio)) { ret_val = PTR_ERR(priv->ext_amp_gpio); switch (ret_val) { @@ -1015,10 +1008,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) return ret_val; } } - priv->hp_detect = devm_fwnode_get_index_gpiod_from_child( - &pdev->dev, "hp-detect", 0, - codec_dev->fwnode, - GPIOD_IN, "hp-detect"); + priv->hp_detect = devm_fwnode_gpiod_get(&pdev->dev, + codec_dev->fwnode, + "hp-detect", + GPIOD_IN, + "hp-detect"); if (IS_ERR(priv->hp_detect)) { ret_val = PTR_ERR(priv->hp_detect); switch (ret_val) { @@ -1041,26 +1035,12 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) log_quirks(&pdev->dev); if ((byt_rt5651_quirk & BYT_RT5651_SSP2_AIF2) || - (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)) { - /* fixup codec aif name */ - snprintf(byt_rt5651_codec_aif_name, - sizeof(byt_rt5651_codec_aif_name), - "%s", "rt5651-aif2"); - - byt_rt5651_dais[dai_index].codecs->dai_name = - byt_rt5651_codec_aif_name; - } + (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)) + byt_rt5651_dais[dai_index].codecs->dai_name = "rt5651-aif2"; if ((byt_rt5651_quirk & BYT_RT5651_SSP0_AIF1) || - (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)) { - /* fixup cpu dai name name */ - snprintf(byt_rt5651_cpu_dai_name, - sizeof(byt_rt5651_cpu_dai_name), - "%s", "ssp0-port"); - - byt_rt5651_dais[dai_index].cpus->dai_name = - byt_rt5651_cpu_dai_name; - } + (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)) + byt_rt5651_dais[dai_index].cpus->dai_name = "ssp0-port"; if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); @@ -1080,17 +1060,23 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } } - if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) - hp_swapped = "-hp-swapped"; - else - hp_swapped = ""; - + snprintf(byt_rt5651_components, sizeof(byt_rt5651_components), + "cfg-spk:%s cfg-mic:%s%s", + (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? "1" : "2", + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], + (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) ? + " cfg-hp:lrswap" : ""); + byt_rt5651_card.components = byt_rt5651_components; +#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), "bytcr-rt5651-%s-spk-%s-mic%s", (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? "mono" : "stereo", - mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped); + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], + (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) ? + "-hp-swapped" : ""); byt_rt5651_card.long_name = byt_rt5651_long_name; +#endif /* override plaform name, if required */ platform_name = mach->mach_params.platform; diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index c68a5b85a4a0..b5b016d493f1 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -515,8 +515,6 @@ static struct cht_acpi_card snd_soc_cards[] = { }; static char cht_rt5645_codec_name[SND_ACPI_I2C_ID_LEN]; -static char cht_rt5645_codec_aif_name[12]; /* = "rt5645-aif[1|2]" */ -static char cht_rt5645_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ @@ -641,28 +639,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) log_quirks(&pdev->dev); if ((cht_rt5645_quirk & CHT_RT5645_SSP2_AIF2) || - (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)) { - - /* fixup codec aif name */ - snprintf(cht_rt5645_codec_aif_name, - sizeof(cht_rt5645_codec_aif_name), - "%s", "rt5645-aif2"); - - cht_dailink[dai_index].codecs->dai_name = - cht_rt5645_codec_aif_name; - } + (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)) + cht_dailink[dai_index].codecs->dai_name = "rt5645-aif2"; if ((cht_rt5645_quirk & CHT_RT5645_SSP0_AIF1) || - (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)) { - - /* fixup cpu dai name name */ - snprintf(cht_rt5645_cpu_dai_name, - sizeof(cht_rt5645_cpu_dai_name), - "%s", "ssp0-port"); - - cht_dailink[dai_index].cpus->dai_name = - cht_rt5645_cpu_dai_name; - } + (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)) + cht_dailink[dai_index].cpus->dai_name = "ssp0-port"; /* override plaform name, if required */ platform_name = mach->mach_params.platform; diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index 5f1bf6d3800c..d6efc554898c 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -405,19 +405,19 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { static struct snd_soc_codec_conf rt1011_conf[] = { { - .dev_name = "i2c-10EC1011:00", + .dlc = COMP_CODEC_CONF("i2c-10EC1011:00"), .name_prefix = "WL", }, { - .dev_name = "i2c-10EC1011:01", + .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), .name_prefix = "WR", }, { - .dev_name = "i2c-10EC1011:02", + .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), .name_prefix = "TL", }, { - .dev_name = "i2c-10EC1011:03", + .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), .name_prefix = "TR", }, }; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index b36264d1d1cd..4a6d117ea7af 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -118,13 +118,13 @@ static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* The ADSP will convert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = DUAL_CHANNEL; + chan->min = chan->max = DUAL_CHANNEL; /* set SSP to 24 bit */ snd_mask_none(fmt); @@ -249,16 +249,6 @@ static const struct snd_pcm_hw_constraint_list constraints_rates = { .mask = 0, }; -static const unsigned int channels[] = { - DUAL_CHANNEL, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - static unsigned int channels_quad[] = { QUAD_CHANNEL, }; @@ -272,13 +262,13 @@ static struct snd_pcm_hw_constraint_list constraints_channels_quad = { static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); /* * set BE channel constraint as user FE channels */ - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c index ed36b68d6705..9179f07f9ee4 100644 --- a/sound/soc/intel/boards/hda_dsp_common.c +++ b/sound/soc/intel/boards/hda_dsp_common.c @@ -63,7 +63,7 @@ int hda_dsp_hdmi_build_controls(struct snd_soc_card *card, "%s: mapping HDMI converter %d to PCM %d (%p)\n", __func__, i, hpcm->device, spcm); } else { - hpcm->pcm = 0; + hpcm->pcm = NULL; hpcm->device = SNDRV_PCM_INVALID_DEVICE; dev_warn(card->dev, "%s: no PCM in topology for HDMI converter %d\n\n", diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb6..bc7f9a9ce9af 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -141,13 +141,13 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* The ADSP will convert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = DUAL_CHANNEL; + chan->min = chan->max = DUAL_CHANNEL; /* set SSP to 24 bit */ snd_mask_none(fmt); @@ -305,7 +305,7 @@ static const struct snd_soc_ops kabylake_da7219_fe_ops = { static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); /* @@ -313,9 +313,9 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, */ if (params_channels(params) == 2) - channels->min = channels->max = 2; + chan->min = chan->max = 2; else - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } @@ -336,19 +336,6 @@ static struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 829f95fc4179..7a13e9b35187 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -176,7 +176,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *runtime = substream->private_data; - int ret = 0, j; + int ret, j; for (j = 0; j < runtime->num_codecs; j++) { struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; @@ -279,7 +279,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_soc_dpcm *dpcm = container_of( @@ -298,7 +298,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, !strcmp(rtd->card->name, "kblmax98373")) { /* The ADSP will convert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = DUAL_CHANNEL; + chan->min = chan->max = DUAL_CHANNEL; /* set SSP to 24 bit */ snd_mask_none(fmt); @@ -313,7 +313,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } @@ -341,6 +341,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) kabylake_ssp1_map, ARRAY_SIZE(kabylake_ssp1_map)); + if (ret) + return ret; + /* * Headset buttons map to the google Reference headset. * These can be configured by userspace. @@ -491,7 +494,7 @@ static const struct snd_soc_ops kabylake_da7219_fe_ops = { static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); /* @@ -499,9 +502,9 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, */ if (params_channels(params) == 2) - channels->min = channels->max = 2; + chan->min = chan->max = 2; else - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } @@ -571,12 +574,12 @@ static struct snd_soc_ops skylake_refcap_ops = { static struct snd_soc_codec_conf max98927_codec_conf[] = { { - .dev_name = MAX98927_DEV0_NAME, + .dlc = COMP_CODEC_CONF(MAX98927_DEV0_NAME), .name_prefix = "Right", }, { - .dev_name = MAX98927_DEV1_NAME, + .dlc = COMP_CODEC_CONF(MAX98927_DEV1_NAME), .name_prefix = "Left", }, }; @@ -584,12 +587,12 @@ static struct snd_soc_codec_conf max98927_codec_conf[] = { static struct snd_soc_codec_conf max98373_codec_conf[] = { { - .dev_name = MAX98373_DEV0_NAME, + .dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME), .name_prefix = "Right", }, { - .dev_name = MAX98373_DEV1_NAME, + .dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME), .name_prefix = "Left", }, }; @@ -1092,7 +1095,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) struct kbl_codec_private *ctx; struct snd_soc_dai_link *kbl_dai_link; struct snd_soc_dai_link_component **codecs; - int i = 0; + int i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 74fe1f3a5479..e23dea9ab79a 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -138,13 +138,13 @@ static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* The ADSP will convert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = DUAL_CHANNEL; + chan->min = chan->max = DUAL_CHANNEL; /* set SSP0 to 24 bit */ snd_mask_none(fmt); diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 7cefda341fbf..d8f2ff7139a9 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -229,11 +229,11 @@ static const struct snd_soc_dapm_route kabylake_5663_map[] = { static struct snd_soc_codec_conf max98927_codec_conf[] = { { - .dev_name = MAXIM_DEV0_NAME, + .dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME), .name_prefix = "Right", }, { - .dev_name = MAXIM_DEV1_NAME, + .dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME), .name_prefix = "Left", }, }; @@ -398,7 +398,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_soc_dpcm *dpcm = container_of( @@ -413,7 +413,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } @@ -454,13 +454,13 @@ static struct snd_soc_ops kabylake_rt5663_ops = { static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - channels->min = channels->max = 2; + chan->min = chan->max = 2; else - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 3e5f6bead229..96c814f36458 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -193,11 +193,11 @@ static const struct snd_soc_dapm_route kabylake_map[] = { static struct snd_soc_codec_conf max98927_codec_conf[] = { { - .dev_name = MAXIM_DEV0_NAME, + .dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME), .name_prefix = "Right", }, { - .dev_name = MAXIM_DEV1_NAME, + .dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME), .name_prefix = "Left", }, }; @@ -333,7 +333,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_soc_dpcm *dpcm = container_of( @@ -348,15 +348,15 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - channels->min = channels->max = 2; + chan->min = chan->max = 2; else - channels->min = channels->max = 4; + chan->min = chan->max = 4; } /* * The speaker on the SSP0 supports S16_LE and not S24_LE. @@ -761,7 +761,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; - int ret = 0; + int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index eb419e1ec42b..78ff5f24c40e 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -41,16 +41,19 @@ int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) return 0; } -SND_SOC_DAILINK_DEFS(idisp1, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin")), +SND_SOC_DAILINK_DEF(idisp1_cpu, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); +SND_SOC_DAILINK_DEF(idisp1_codec, DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); -SND_SOC_DAILINK_DEFS(idisp2, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin")), +SND_SOC_DAILINK_DEF(idisp2_cpu, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); +SND_SOC_DAILINK_DEF(idisp2_codec, DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); -SND_SOC_DAILINK_DEFS(idisp3, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin")), +SND_SOC_DAILINK_DEF(idisp3_cpu, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); +SND_SOC_DAILINK_DEF(idisp3_codec, DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); SND_SOC_DAILINK_DEF(analog_cpu, @@ -83,21 +86,21 @@ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { .id = 1, .dpcm_playback = 1, .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1), + SND_SOC_DAILINK_REG(idisp1_cpu, idisp1_codec, platform), }, { .name = "iDisp2", .id = 2, .dpcm_playback = 1, .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2), + SND_SOC_DAILINK_REG(idisp2_cpu, idisp2_codec, platform), }, { .name = "iDisp3", .id = 3, .dpcm_playback = 1, .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3), + SND_SOC_DAILINK_REG(idisp3_cpu, idisp3_codec, platform), }, { .name = "Analog Playback and Capture", diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 4e45901e3a2f..11eaee9ae41f 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -100,6 +100,8 @@ static struct snd_soc_card hda_soc_card = { .late_probe = skl_hda_card_late_probe, }; +static char hda_soc_components[30]; + #define IDISP_DAI_COUNT 3 #define HDAC_DAI_COUNT 2 #define DMIC_DAI_COUNT 2 @@ -183,6 +185,12 @@ static int skl_hda_audio_probe(struct platform_device *pdev) hda_soc_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&hda_soc_card, ctx); + if (mach->mach_params.dmic_num > 0) { + snprintf(hda_soc_components, sizeof(hda_soc_components), + "cfg-dmics:%d", mach->mach_params.dmic_num); + hda_soc_card.components = hda_soc_components; + } + return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card); } diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 3ce8efbeed12..e6de3b28d840 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -139,13 +139,13 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* The ADSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; /* set SSP0 to 24 bit */ snd_mask_none(fmt); @@ -315,13 +315,13 @@ static const struct snd_soc_ops skylake_nau8825_ops = { static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - channels->min = channels->max = 2; + chan->min = chan->max = 2; else - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 1a7ac8bdf543..c99c8b23e509 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -147,11 +147,11 @@ static const struct snd_soc_dapm_route skylake_map[] = { static struct snd_soc_codec_conf ssm4567_codec_conf[] = { { - .dev_name = "i2c-INT343B:00", + .dlc = COMP_CODEC_CONF("i2c-INT343B:00"), .name_prefix = "Left", }, { - .dev_name = "i2c-INT343B:01", + .dlc = COMP_CODEC_CONF("i2c-INT343B:01"), .name_prefix = "Right", }, }; @@ -317,13 +317,13 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* The ADSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; /* set SSP0 to 24 bit */ snd_mask_none(fmt); @@ -334,12 +334,12 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - channels->min = channels->max = 2; + chan->min = chan->max = 2; else - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 231349a47cc9..a9aec66a2351 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -211,13 +211,13 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); /* The output is 48KHz, stereo, 16bits */ rate->min = rate->max = 48000; - channels->min = channels->max = 2; + chan->min = chan->max = 2; /* set SSP0 to 24 bit */ snd_mask_none(fmt); @@ -247,12 +247,12 @@ static const struct snd_soc_ops skylake_rt286_ops = { static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct snd_interval *channels = hw_param_interval(params, + struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); if (params_channels(params) == 2) - channels->min = channels->max = 2; + chan->min = chan->max = 2; else - channels->min = channels->max = 4; + chan->min = chan->max = 4; return 0; } diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c new file mode 100644 index 000000000000..8f44f13d2848 --- /dev/null +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -0,0 +1,371 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2019 Intel Corporation. + +/* + * Intel SOF Machine driver for DA7219 + MAX98373 codec + */ + +#include <linux/input.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../codecs/da7219.h" +#include "../../codecs/da7219-aad.h" +#include "hda_dsp_common.h" + +#define DIALOG_CODEC_DAI "da7219-hifi" +#define MAX98373_CODEC_DAI "max98373-aif1" +#define MAXIM_DEV0_NAME "i2c-MX98373:00" +#define MAXIM_DEV1_NAME "i2c-MX98373:01" + +struct hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct card_private { + struct snd_soc_jack headset; + struct list_head hdmi_pcm_list; + struct snd_soc_jack hdmi[3]; +}; + +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 = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, DIALOG_CODEC_DAI); + 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_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 controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", 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 audio_map[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, + + { "MIC", NULL, "Headset Mic" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static struct snd_soc_jack headset; + +static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24000000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, jack); + + return ret; +} + +static int ssp1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = substream->private_data; + int ret, j; + + for (j = 0; j < runtime->num_codecs; j++) { + struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + + if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { + /* vmon_slot_no = 0 imon_slot_no = 1 for TX slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 3, 4, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { + /* vmon_slot_no = 2 imon_slot_no = 3 for TX slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC, 3, 4, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + + return 0; +} + +static struct snd_soc_ops ssp1_ops = { + .hw_params = ssp1_hw_params, +}; + +static struct snd_soc_codec_conf max98373_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static int hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int card_late_probe(struct snd_soc_card *card) +{ + struct card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_acpi_mach *mach = (card->dev)->platform_data; + struct hdmi_pcm *pcm; + + if (mach->mach_params.common_hdmi_codec_drv) { + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm, + head); + return hda_dsp_hdmi_build_controls(card, + pcm->codec_dai->component); + } + + return -EINVAL; +} + +SND_SOC_DAILINK_DEF(ssp0_pin, + DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); +SND_SOC_DAILINK_DEF(ssp0_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", DIALOG_CODEC_DAI))); + +SND_SOC_DAILINK_DEF(ssp1_pin, + DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); +SND_SOC_DAILINK_DEF(ssp1_amps, + DAILINK_COMP_ARRAY( + /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI), + /* Right */ COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI))); + +SND_SOC_DAILINK_DEF(dmic_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); +SND_SOC_DAILINK_DEF(dmic_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); + +SND_SOC_DAILINK_DEF(idisp1_pin, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); +SND_SOC_DAILINK_DEF(idisp1_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); + +SND_SOC_DAILINK_DEF(idisp2_pin, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); +SND_SOC_DAILINK_DEF(idisp2_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); + +SND_SOC_DAILINK_DEF(idisp3_pin, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); +SND_SOC_DAILINK_DEF(idisp3_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); + +SND_SOC_DAILINK_DEF(platform, /* subject to be overridden during probe */ + DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); + +static struct snd_soc_dai_link dais[] = { + /* Back End DAI links */ + { + .name = "SSP1-Codec", + .id = 0, + .ignore_pmdown_time = 1, + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, /* IV feedback */ + .ops = &ssp1_ops, + SND_SOC_DAILINK_REG(ssp1_pin, ssp1_amps, platform), + }, + { + .name = "SSP0-Codec", + .id = 1, + .no_pcm = 1, + .init = da7219_codec_init, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), + }, + { + .name = "dmic01", + .id = 2, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), + }, + { + .name = "iDisp1", + .id = 3, + .init = hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), + }, + { + .name = "iDisp2", + .id = 4, + .init = hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), + }, + { + .name = "iDisp3", + .id = 5, + .init = hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), + }, +}; + +static struct snd_soc_card card_da7219_m98373 = { + .name = "da7219max", + .owner = THIS_MODULE, + .dai_link = dais, + .num_links = ARRAY_SIZE(dais), + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), + .codec_conf = max98373_codec_conf, + .num_configs = ARRAY_SIZE(max98373_codec_conf), + .fully_routed = true, + .late_probe = card_late_probe, +}; + +static int audio_probe(struct platform_device *pdev) +{ + static struct snd_soc_card *card; + struct snd_soc_acpi_mach *mach; + struct card_private *ctx; + int ret; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + card = (struct snd_soc_card *)pdev->id_entry->driver_data; + card->dev = &pdev->dev; + + mach = (&pdev->dev)->platform_data; + ret = snd_soc_fixup_dai_links_platform_name(card, + mach->mach_params.platform); + if (ret) + return ret; + + snd_soc_card_set_drvdata(card, ctx); + + return devm_snd_soc_register_card(&pdev->dev, card); +} + +static const struct platform_device_id board_ids[] = { + { + .name = "sof_da7219_max98373", + .driver_data = (kernel_ulong_t)&card_da7219_m98373, + }, + { } +}; + +static struct platform_driver audio = { + .probe = audio_probe, + .driver = { + .name = "sof_da7219_max98373", + .pm = &snd_soc_pm_ops, + }, + .id_table = board_ids, +}; +module_platform_driver(audio) + +/* Module information */ +MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver"); +MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_da7219_max98373"); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 751b8ea6ae1f..8a13231dee15 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -35,6 +35,10 @@ #define SOF_RT5682_SSP_AMP(quirk) \ (((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK) #define SOF_RT5682_MCLK_BYTCHT_EN BIT(9) +#define SOF_RT5682_NUM_HDMIDEV_SHIFT 10 +#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) +#define SOF_RT5682_NUM_HDMIDEV(quirk) \ + ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -594,6 +598,19 @@ static int sof_audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; + + dmi_check_system(sof_rt5682_quirk_table); + + mach = (&pdev->dev)->platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -604,11 +621,13 @@ static int sof_audio_probe(struct platform_device *pdev) SOF_RT5682_SSP_CODEC(2); } else { dmic_be_num = 2; - hdmi_num = 3; + hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >> + SOF_RT5682_NUM_HDMIDEV_SHIFT; + /* default number of HDMI DAI's */ + if (!hdmi_num) + hdmi_num = 3; } - dmi_check_system(sof_rt5682_quirk_table); - /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); @@ -652,7 +671,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_rt5682.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682, @@ -683,6 +701,21 @@ static int sof_rt5682_remove(struct platform_device *pdev) return 0; } +static const struct platform_device_id board_ids[] = { + { + .name = "sof_rt5682", + }, + { + .name = "tgl_max98357a_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { } +}; + static struct platform_driver sof_audio = { .probe = sof_audio_probe, .remove = sof_rt5682_remove, @@ -690,6 +723,7 @@ static struct platform_driver sof_audio = { .name = "sof_rt5682", .pm = &snd_soc_pm_ops, }, + .id_table = board_ids, }; module_platform_driver(sof_audio) @@ -699,3 +733,4 @@ MODULE_AUTHOR("Bard Liao <bard.liao@intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r@intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); +MODULE_ALIAS("platform:tgl_max98357a_rt5682"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cfl-match.c b/sound/soc/intel/common/soc-acpi-intel-cfl-match.c index d6fd2026d0b8..ff9d6938b9f6 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cfl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cfl-match.c @@ -14,5 +14,10 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cfl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cfl_machines); +struct snd_soc_acpi_mach snd_soc_acpi_intel_cfl_sdw_machines[] = { + {} +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cfl_sdw_machines); + MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index fb9ba8819706..f55634c4c2e8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -59,5 +59,94 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); +static const u64 rt711_0_adr[] = { + 0x000010025D071100 +}; + +static const u64 rt1308_1_adr[] = { + 0x000110025D130800 +}; + +static const u64 rt1308_2_adr[] = { + 0x000210025D130800 +}; + +static const u64 rt715_3_adr[] = { + 0x000310025D071500 +}; + +static const struct snd_soc_acpi_link_adr cml_3_in_1_default[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr = rt711_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1308_1_adr), + .adr = rt1308_1_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1308_2_adr), + .adr = rt1308_2_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt715_3_adr), + .adr = rt715_3_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr cml_3_in_1_mono_amp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr = rt711_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1308_1_adr), + .adr = rt1308_1_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt715_3_adr), + .adr = rt715_3_adr, + }, + {} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[] = { + { + .link_mask = 0xF, /* 4 active links required */ + .links = cml_3_in_1_default, + .drv_name = "sdw_rt711_rt1308_rt715", + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-rt711-rt1308-rt715.tplg", + }, + { + /* + * link_mask should be 0xB, but all links are enabled by BIOS. + * This entry will be selected if there is no rt1308 exposed + * on link2 since it will fail to match the above entry. + */ + .link_mask = 0xF, + .links = cml_3_in_1_mono_amp, + .drv_name = "sdw_rt711_rt1308_rt715", + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-rt711-rt1308-mono-rt715.tplg", + }, + { + .link_mask = 0x2, /* RT700 connected on Link1 */ + .drv_name = "sdw_rt700", + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-rt700.tplg", + }, + {} +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_sdw_machines); + MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index 27588841c8b0..828980d5630d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -27,5 +27,10 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines); +struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[] = { + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_sdw_machines); + MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 34eb0baaa951..35958553652e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -30,6 +30,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .sof_tplg_filename = "sof-bdw-rt286.tplg", }, { + .id = "10EC5650", + .drv_name = "bdw-rt5650", + .fw_filename = "intel/IntcSST2.bin", + .sof_fw_filename = "sof-bdw.ri", + .sof_tplg_filename = "sof-bdw-rt5650.tplg", + }, + { .id = "RT5677CE", .drv_name = "bdw-rt5677", .fw_filename = "intel/IntcSST2.bin", diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index 38977669b576..67e9da4635f2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -33,5 +33,103 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines); +static const u64 rt700_0_adr[] = { + 0x000010025D070000 +}; + +static const struct snd_soc_acpi_link_adr icl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt700_0_adr), + .adr = rt700_0_adr, + }, + {} +}; + +static const u64 rt711_0_adr[] = { + 0x000010025D071100 +}; + +static const u64 rt1308_1_adr[] = { + 0x000110025D130800 +}; + +static const u64 rt1308_2_adr[] = { + 0x000210025D130800 +}; + +static const u64 rt715_3_adr[] = { + 0x000310025D715000 +}; + +static const struct snd_soc_acpi_link_adr icl_3_in_1_default[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr = rt711_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1308_1_adr), + .adr = rt1308_1_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1308_2_adr), + .adr = rt1308_2_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt715_3_adr), + .adr = rt715_3_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr icl_3_in_1_mono_amp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr = rt711_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1308_1_adr), + .adr = rt1308_1_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt715_3_adr), + .adr = rt715_3_adr, + }, + {} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[] = { + { + .link_mask = 0xF, /* 4 active links required */ + .links = icl_3_in_1_default, + .drv_name = "sdw_rt711_rt1308_rt715", + .sof_fw_filename = "sof-icl.ri", + .sof_tplg_filename = "sof-icl-rt711-rt1308-rt715.tplg", + }, + { + .link_mask = 0xB, /* 3 active links required */ + .links = icl_3_in_1_mono_amp, + .drv_name = "sdw_rt711_rt1308_rt715", + .sof_fw_filename = "sof-icl.ri", + .sof_tplg_filename = "sof-icl-rt711-rt1308-rt715-mono.tplg", + }, + { + .link_mask = 0x1, /* rt700 connected on link0 */ + .links = icl_rvp, + .drv_name = "sdw_rt700", + .sof_fw_filename = "sof-icl.ri", + .sof_tplg_filename = "sof-icl-rt700.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_sdw_machines); + MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 1c68a04f0c6e..ed2b125f6a11 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -10,6 +10,13 @@ #include <sound/soc-acpi-intel-match.h> struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { + { + .id = "DLGS7219", + .drv_name = "sof_da7219_max98373", + .machine_quirk = snd_soc_acpi_codec_list, + .sof_fw_filename = "sof-jsl.ri", + .sof_tplg_filename = "sof-jsl-da7219.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 57a6298d6dca..5984dd151f3e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -9,16 +9,76 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> +static struct snd_soc_acpi_codecs tgl_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +static const u64 rt711_0_adr[] = { + 0x000010025D071100 +}; + +static const u64 rt1308_1_adr[] = { + 0x000120025D130800, + 0x000122025D130800 +}; + +static const struct snd_soc_acpi_link_adr tgl_i2s_rt1308[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr = rt711_0_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr tgl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr = rt711_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1308_1_adr), + .adr = rt1308_1_adr, + }, + {} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", - .drv_name = "tgl_rt1308", + .drv_name = "rt711_rt1308", + .link_mask = 0x1, /* RT711 on SoundWire link0 */ + .links = tgl_i2s_rt1308, + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg", + }, + { + .id = "10EC5682", + .drv_name = "tgl_max98357a_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &tgl_codecs, .sof_fw_filename = "sof-tgl.ri", - .sof_tplg_filename = "sof-tgl-rt1308.tplg", + .sof_tplg_filename = "sof-tgl-max98357a-rt5682.tplg", }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); +/* this table is used when there is no I2S codec present */ +struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { + { + .link_mask = 0x3, /* rt711 on link 0 and 2 rt1308s on link 1 */ + .links = tgl_rvp, + .drv_name = "sdw_rt711_rt1308_rt715", + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_sdw_machines); + MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index a3a5bba2fbd9..033d7c05d7fb 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -592,13 +592,6 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, return ret; } - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) { - dev_err(rtd->dev, "error: could not allocate %d bytes for PCM %d\n", - params_buffer_bytes(params), ret); - return ret; - } - dmab = snd_pcm_get_dma_buf(substream); ret = create_adsp_page_table(substream, pdata, rtd, runtime->dma_area, @@ -656,13 +649,6 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, return 0; } -static int hsw_pcm_hw_free(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - return 0; -} - static int hsw_pcm_trigger(struct snd_soc_component *component, struct snd_pcm_substream *substream, int cmd) { @@ -930,7 +916,7 @@ static int hsw_pcm_new(struct snd_soc_component *component, if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - snd_pcm_lib_preallocate_pages_for_all(pcm, + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV_SG, dev, hsw_pcm_hardware.buffer_bytes_max, @@ -1114,10 +1100,8 @@ static const struct snd_soc_component_driver hsw_dai_component = { .open = hsw_pcm_open, .close = hsw_pcm_close, .hw_params = hsw_pcm_hw_params, - .hw_free = hsw_pcm_hw_free, .trigger = hsw_pcm_trigger, .pointer = hsw_pcm_pointer, - .ioctl = snd_soc_pcm_lib_ioctl, .pcm_construct = hsw_pcm_new, .controls = hsw_volume_controls, .num_controls = ARRAY_SIZE(hsw_volume_controls), diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 900746c30bff..b99509675d29 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -77,13 +77,7 @@ static int skl_substream_alloc_pages(struct hdac_bus *bus, hdac_stream(stream)->period_bytes = 0; hdac_stream(stream)->format_val = 0; - return snd_pcm_lib_malloc_pages(substream, size); -} - -static int skl_substream_free_pages(struct hdac_bus *bus, - struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); + return 0; } static void skl_set_pcm_constrains(struct hdac_bus *bus, @@ -385,7 +379,6 @@ static void skl_pcm_close(struct snd_pcm_substream *substream, static int skl_pcm_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct skl_dev *skl = get_skl_ctx(dai->dev); struct skl_module_cfg *mconfig; @@ -405,7 +398,7 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream, snd_hdac_stream_cleanup(hdac_stream(stream)); hdac_stream(stream)->prepared = 0; - return skl_substream_free_pages(bus, substream); + return 0; } static int skl_be_hw_params(struct snd_pcm_substream *substream, @@ -1289,12 +1282,6 @@ static int skl_platform_soc_get_time_info( return 0; } -static void skl_platform_soc_free(struct snd_soc_component *component, - struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - #define MAX_PREALLOC_SIZE (32 * 1024 * 1024) static int skl_platform_soc_new(struct snd_soc_component *component, @@ -1312,10 +1299,10 @@ static int skl_platform_soc_new(struct snd_soc_component *component, size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024; if (size > MAX_PREALLOC_SIZE) size = MAX_PREALLOC_SIZE; - snd_pcm_lib_preallocate_pages_for_all(pcm, - SNDRV_DMA_TYPE_DEV_SG, - &skl->pci->dev, - size, MAX_PREALLOC_SIZE); + snd_pcm_set_managed_buffer_all(pcm, + SNDRV_DMA_TYPE_DEV_SG, + &skl->pci->dev, + size, MAX_PREALLOC_SIZE); } return 0; @@ -1477,13 +1464,11 @@ static const struct snd_soc_component_driver skl_component = { .probe = skl_platform_soc_probe, .remove = skl_platform_soc_remove, .open = skl_platform_soc_open, - .ioctl = snd_soc_pcm_lib_ioctl, .trigger = skl_platform_soc_trigger, .pointer = skl_platform_soc_pointer, .get_time_info = skl_platform_soc_get_time_info, .mmap = skl_platform_soc_mmap, .pcm_construct = skl_platform_soc_new, - .pcm_destruct = skl_platform_soc_free, .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ }; diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 58ba3e9469ba..f755ca2484cf 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -773,11 +773,6 @@ static void skl_codec_create(struct hdac_bus *bus) } } -static const struct hdac_bus_ops bus_core_ops = { - .command = snd_hdac_bus_send_cmd, - .get_response = snd_hdac_bus_get_response, -}; - static int skl_i915_init(struct hdac_bus *bus) { int err; @@ -888,7 +883,7 @@ static int skl_create(struct pci_dev *pci, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) ext_ops = snd_soc_hdac_hda_get_ops(); #endif - snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, ext_ops); + snd_hdac_ext_bus_init(bus, &pci->dev, NULL, ext_ops); bus->use_posbuf = 1; skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); |