December 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: hdmi-codec: re-introduce mutex locking again
by Jerome Brunet 09 Dec '19

09 Dec '19

The dai codec needs to ensure that on one dai is used at any time. This is currently protected by bit atomic operation. With this change, it done with a mutex instead. This change is not about functionality or efficiency. It is done with the hope that it help maintainability in the future. Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/codecs/hdmi-codec.c | 32 +++++++++++++++++++------------- 1 file changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index f8b5b960e597..543363102d03 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -274,7 +274,8 @@ struct hdmi_codec_priv { uint8_t eld[MAX_ELD_BYTES]; struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; - unsigned long busy; + struct mutex lock; + bool busy; struct snd_soc_jack *jack; unsigned int jack_status; }; @@ -390,9 +391,10 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); int ret = 0; - ret = test_and_set_bit(0, &hcp->busy); - if (ret) { + mutex_lock(&hcp->lock); + if (hcp->busy) { dev_err(dai->dev, "Only one simultaneous stream supported!\n"); + mutex_unlock(&hcp->lock); return -EINVAL; } @@ -405,21 +407,21 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, if (hcp->hcd.ops->get_eld) { ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); + if (ret) + goto err; + + ret = snd_pcm_hw_constraint_eld(substream->runtime, hcp->eld); + if (ret) + goto err; - if (!ret) { - ret = snd_pcm_hw_constraint_eld(substream->runtime, - hcp->eld); - if (ret) - goto err; - } /* Select chmap supported */ hdmi_codec_eld_chmap(hcp); } - return 0; + + hcp->busy = true; err: - /* Release the exclusive lock on error */ - clear_bit(0, &hcp->busy); + mutex_unlock(&hcp->lock); return ret; } @@ -431,7 +433,9 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data); - clear_bit(0, &hcp->busy); + mutex_lock(&hcp->lock); + hcp->busy = false; + mutex_unlock(&hcp->lock); } static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, @@ -811,6 +815,8 @@ static int hdmi_codec_probe(struct platform_device *pdev) return -ENOMEM; hcp->hcd = *hcd; + mutex_init(&hcp->lock); + daidrv = devm_kcalloc(dev, dai_count, sizeof(*daidrv), GFP_KERNEL); if (!daidrv) return -ENOMEM; -- 2.23.0

2 1

[alsa-devel] [PATCH v2 00/13] ASoC: SOF: initial cleanup for DT and multi-client support
by Pierre-Louis Bossart 09 Dec '19

09 Dec '19

This series does not add new functionality but introduces a better code partition to support machine drivers in Device Tree, as well as a 'componentization' of the PCM/topology parts needed for multi-client support. This cleanup was merged in the SOF tree and tested extensively with no regressions identified. Thanks Ranjani and Daniel for combining your proposals in a single change. Changes since v1: Rebased on top of Jaroslav's HDaudio changes (added in this series as discussed with Mark) Daniel Baluta (2): ASoC: SOF: Make creation of machine device from SOF core optional ASoC: SOF: Remove unused drv_name in sof_pdata Jaroslav Kysela (3): ASoC: intel/skl/hda - export number of digital microphones via control components ASoC: Intel - use control components to describe card config ASoC: Intel - do not describe I/O configuration in the long card name Ranjani Sridharan (8): ASoC: SOF: topology: remove snd_sof_init_topology() ASoC: SOF: core: modify the signature for snd_sof_create_page_table ASoC: SOF: core: move check for runtime callbacks to core ASoC: SOF: Introduce default_fw_filename member in sof_dev_desc ASoC: SOF: partition audio-related parts from SOF core ASoC: SOF: intel: hda: Modify signature for hda_codec_probe_bus() ASoC: SOF: remove nocodec_fw_filename ASoC: SOF: nocodec: Amend arguments for sof_nocodec_setup() include/sound/sof.h | 8 +- sound/soc/intel/boards/Kconfig | 13 + sound/soc/intel/boards/bytcht_es8316.c | 13 +- sound/soc/intel/boards/bytcr_rt5640.c | 10 + sound/soc/intel/boards/bytcr_rt5651.c | 22 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 8 + sound/soc/sof/Makefile | 2 +- sound/soc/sof/control.c | 55 +-- sound/soc/sof/core.c | 244 +--------- sound/soc/sof/intel/apl.c | 7 + sound/soc/sof/intel/bdw.c | 33 ++ sound/soc/sof/intel/byt.c | 45 ++ sound/soc/sof/intel/cnl.c | 7 + sound/soc/sof/intel/hda-codec.c | 16 +- sound/soc/sof/intel/hda-dai.c | 1 + sound/soc/sof/intel/hda-pcm.c | 4 +- sound/soc/sof/intel/hda-stream.c | 1 + sound/soc/sof/intel/hda.c | 228 ++++++---- sound/soc/sof/intel/hda.h | 8 +- sound/soc/sof/ipc.c | 16 +- sound/soc/sof/nocodec.c | 12 - sound/soc/sof/ops.h | 34 ++ sound/soc/sof/pcm.c | 131 +++--- sound/soc/sof/pm.c | 219 +-------- sound/soc/sof/sof-acpi-dev.c | 39 +- sound/soc/sof/sof-audio.c | 445 +++++++++++++++++++ sound/soc/sof/sof-audio.h | 211 +++++++++ sound/soc/sof/sof-of-dev.c | 24 +- sound/soc/sof/sof-pci-dev.c | 47 +- sound/soc/sof/sof-priv.h | 202 +-------- sound/soc/sof/topology.c | 374 ++++++++-------- sound/soc/sof/trace.c | 4 +- sound/soc/sof/utils.c | 60 +++ 33 files changed, 1380 insertions(+), 1163 deletions(-) create mode 100644 sound/soc/sof/sof-audio.c create mode 100644 sound/soc/sof/sof-audio.h -- 2.20.1

2 26

[alsa-devel] [PATCH] ALSA: hda/ca0132 - add missed snd_hda_gen_parse_auto_config
by Chuhong Yuan 09 Dec '19

09 Dec '19

It is found that all usages of snd_hda_parse_pin_defcfg() are followed with snd_hda_gen_parse_auto_config() except here. This should be a miss and needs to add the missed call. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index b7a1abb3e231..ea1187a13241 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -24,6 +24,7 @@ #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" +#include "hda_generic.h" #include "ca0132_regs.h" @@ -8844,6 +8845,10 @@ static int patch_ca0132(struct hda_codec *codec) if (err < 0) goto error; + err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg); + if (err < 0) + goto error; + return 0; error: -- 2.24.0

2 1

[alsa-devel] [PATCH] ASoC: tas2552: add missed regulator_bulk_disable in remove
by Chuhong Yuan 09 Dec '19

09 Dec '19

The driver forgets to call regulator_bulk_disable() in remove like that in probe failure. Add the missed call to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/tas2552.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 56671f21cfe5..0e19ec76aae0 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -616,6 +616,9 @@ static void tas2552_component_remove(struct snd_soc_component *component) pm_runtime_put(component->dev); gpiod_set_value(tas2552->enable_gpio, 0); + + regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); }; #ifdef CONFIG_PM -- 2.24.0

2 1

[alsa-devel] [PATCH] ASoC: cs35l32: add missed regulator_bulk_disable in remove
by Chuhong Yuan 09 Dec '19

09 Dec '19

The driver forgets to call regulator_bulk_disable() in remove like that in probe failure. Add the missed call to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/cs35l32.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 3a644a35c464..be8159bd8ec8 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -501,6 +501,8 @@ static int cs35l32_i2c_remove(struct i2c_client *i2c_client) /* Hold down reset */ gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); return 0; } -- 2.24.0

2 1

[alsa-devel] [PATCH] firewire-motu: fix double unlocked 'motu->mutex'
by Takashi Sakamoto 09 Dec '19

09 Dec '19

Mutex is doubly unlocked in some error path of pcm.open. This commit fixes ALSA firewire-motu driver in Linux kernel v5.5. Reported-by: kbuild test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 3fd80b200388 ("ALSA: firewire-motu: use the same size of period for PCM substream in AMDTP streams") Fixes: 0f5482e7875b ("ALSA: firewire-motu: share PCM buffer size for both direction") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-pcm.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 349b4d09e84f..005970931030 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -177,18 +177,14 @@ static int pcm_open(struct snd_pcm_substream *substream) err = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, frames_per_period, frames_per_period); - if (err < 0) { - mutex_unlock(&motu->mutex); + if (err < 0) goto err_locked; - } err = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, frames_per_buffer, frames_per_buffer); - if (err < 0) { - mutex_unlock(&motu->mutex); + if (err < 0) goto err_locked; - } } } -- 2.20.1

2 2

[alsa-devel] [PATCH v2 1/2] ALSA: core: add snd_pcm_format_by_name()
by Daniel Mack 08 Dec '19

08 Dec '19

Add a new function to look up PCM format codes by name. This can be used by ASoC drivers to look up formats through device-tree properties, for instance. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- include/sound/pcm.h | 1 + sound/core/pcm.c | 23 +++++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8a89fa6fdd5e..1bfde6f2f180 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1339,6 +1339,7 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) (IEC958_AES3_CON_FS_48000<<24)) const char *snd_pcm_format_name(snd_pcm_format_t format); +int snd_pcm_format_by_name(const char *name, snd_pcm_format_t *format); /** * snd_pcm_stream_str - Get a string naming the direction of a stream diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9a72d641743d..c02d8df4f92b 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -225,6 +225,29 @@ const char *snd_pcm_format_name(snd_pcm_format_t format) } EXPORT_SYMBOL_GPL(snd_pcm_format_name); +/** + * snd_pcm_format_by_name - Return the PCM format code for the given name + * @name: PCM format name ('S16_LE', 'S24_3LE', ...) + * @format: Pointer to returned PCM format code + * + * The string comparison is done in a case-insensitive way. + * + * Return: 0 on success, or -ENOENT if the given format is not valid. + */ +int snd_pcm_format_by_name(const char *name, snd_pcm_format_t *format) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(snd_pcm_format_names); i++) + if (strcasecmp(name, snd_pcm_format_names[i]) == 0) { + *format = i; + return 0; + } + + return -ENOENT; +} +EXPORT_SYMBOL_GPL(snd_pcm_format_by_name); + #ifdef CONFIG_SND_VERBOSE_PROCFS #define STATE(v) [SNDRV_PCM_STATE_##v] = #v -- 2.23.0

2 3

[alsa-devel] [PATCH] ALSA: echoaudio: simplify get_audio_levels
by Olof Johansson 08 Dec '19

08 Dec '19

The loop optimizer seems to go astray here, and produces some warnings that don't seem valid. Still, the code can be simplified -- just clear the whole array at the beginning, and fill in whatever values are valid on the platform. Warnings before this change (GCC 8.2.0 ARM allmodconfig): In file included from ../sound/pci/echoaudio/gina24.c:115: ../sound/pci/echoaudio/echoaudio.c: In function 'snd_echo_vumeters_get': ../sound/pci/echoaudio/echoaudio_dsp.c:647:9: warning: iteration 1073741824 invokes undefined behavior [-Waggressive-loop-optimizations] In file included from ../sound/pci/echoaudio/layla24.c:112: ../sound/pci/echoaudio/echoaudio.c: In function 'snd_echo_vumeters_get': ../sound/pci/echoaudio/echoaudio_dsp.c:658:9: warning: iteration 1073741824 invokes undefined behavior [-Waggressive-loop-optimizations] ../sound/pci/echoaudio/echoaudio_dsp.c:647:9: warning: iteration 1073741824 invokes undefined behavior [-Waggressive-loop-optimizations] Signed-off-by: Olof Johansson <olof(a)lixom.net> --- sound/pci/echoaudio/echoaudio_dsp.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 50d4a87a6bb34..f02f5b1568dee 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -635,36 +635,30 @@ This function assumes there are no more than 16 in/out busses or pipes Meters is an array [3][16][2] of long. */ static void get_audio_meters(struct echoaudio *chip, long *meters) { - int i, m, n; + unsigned int i, m, n; - m = 0; - n = 0; - for (i = 0; i < num_busses_out(chip); i++, m++) { + for (i = 0 ; i < 96; i++) + meters[i] = 0; + + for (m = 0, n = 0, i = 0; i < num_busses_out(chip); i++, m++) { meters[n++] = chip->comm_page->vu_meter[m]; meters[n++] = chip->comm_page->peak_meter[m]; } - for (; n < 32; n++) - meters[n] = 0; #ifdef ECHOCARD_ECHO3G m = E3G_MAX_OUTPUTS; /* Skip unused meters */ #endif - for (i = 0; i < num_busses_in(chip); i++, m++) { + for (n = 32, i = 0; i < num_busses_in(chip); i++, m++) { meters[n++] = chip->comm_page->vu_meter[m]; meters[n++] = chip->comm_page->peak_meter[m]; } - for (; n < 64; n++) - meters[n] = 0; - #ifdef ECHOCARD_HAS_VMIXER - for (i = 0; i < num_pipes_out(chip); i++, m++) { + for (n = 64, i = 0; i < num_pipes_out(chip); i++, m++) { meters[n++] = chip->comm_page->vu_meter[m]; meters[n++] = chip->comm_page->peak_meter[m]; } #endif - for (; n < 96; n++) - meters[n] = 0; } -- 2.11.0

2 1

Re: [alsa-devel] [PATCH v3 06/15] clk: tegra: Remove tegra_pmc_clk_init along with clk ids
by Dmitry Osipenko 08 Dec '19

08 Dec '19

07.12.2019 22:35, Sowjanya Komatineni пишет: > > On 12/7/19 7:04 AM, Dmitry Osipenko wrote: >> 07.12.2019 17:43, Dmitry Osipenko пишет: >>> 07.12.2019 17:33, Dmitry Osipenko пишет: >>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>> Current Tegra clock driver registers PMC clocks clk_out_1, clk_out_2, >>>>> clk_out_3 and blink output in tegra_pmc_init() which does direct Tegra >>>>> PMC access during clk_ops and these PMC register read and write access >>>>> will not happen when PMC is in secure mode. >>>>> >>>>> Any direct PMC register access from non-secure world will not go >>>>> through and all the PMC clocks and blink control are done in Tegra PMC >>>>> driver with PMC as clock provider. >>>>> >>>>> This patch removes tegra_pmc_clk_init along with corresponding clk ids >>>>> from Tegra clock driver. >>>>> >>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>> --- >>>> [snip] >>>> >>>>> @@ -1230,9 +1222,6 @@ static struct tegra_clk_init_table >>>>> init_table[] __initdata = { >>>>> { TEGRA30_CLK_PLL_A, TEGRA30_CLK_CLK_MAX, 564480000, 1 }, >>>>> { TEGRA30_CLK_PLL_A_OUT0, TEGRA30_CLK_CLK_MAX, 11289600, 1 }, >>>>> { TEGRA30_CLK_EXTERN1, TEGRA30_CLK_PLL_A_OUT0, 0, 1 }, >>>> Perhaps these clocks do not need to be always-enabled? >>>> >>>> [snip] >>>> >>> Also, EXTERN1 parent configuration should be moved to the audio >>> driver/device-tree as well. >> Ah, I missed that it's done in the patch #10. > Yes its done in Patch#10 >> >>> Maybe it even makes sense to move the whole configuration, including >>> PLLA. I don't see why clk driver need to do something for the audio >>> driver. > > Current ASoC driver already takes care of PLLA rate and enables. > > So PLLA init can be removed from clock driver too. I didn't went through > complete audio driver to be confident to remove this. > > But PLLA is needed for i2s clock also and currently I2S driver takes > care of only I2S clock rate using PLLA as parent set by clock driver and > clock driver enables PLLA earlier to have it ready by the time both I2S > driver and ASoC driver . I2S could use assigned-clocks, but probably it's not really necessary and predefined configuration in the clk driver is good enough. At least PLLA doesn't need to be always-enabled since audio drivers enable it when necessary.

1 0

[alsa-devel] [PATCH v3] ASoC: intel: Add Broadwell rt5650 machine driver
by Curtis Malainey 06 Dec '19

06 Dec '19

From: Ben Zhang <benzh(a)chromium.org> Add machine driver for Broadwell + rt5650. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Jon Flatley <jflat(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- v2 (Curtis): * Addressed Pierre's comments * Added SOF dependencies * Added platform override * Fixed Kconfig * Moved to devm register v3: * Removed THIS_MODULE * Added PM Ops sound/soc/intel/boards/Kconfig | 12 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 319 ++++++++++++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 7 + 4 files changed, 340 insertions(+) create mode 100644 sound/soc/intel/boards/bdw-rt5650.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6c9fd9ad566ed..de613ae5052bf 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -31,6 +31,18 @@ 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 diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index ba1aa89db09d5..4efb3689503f8 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 @@ -37,6 +38,7 @@ 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 diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c new file mode 100644 index 0000000000000..274b21680d624 --- /dev/null +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -0,0 +1,319 @@ +// 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(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + 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"))); + +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(dummy, be, dummy), + }, +}; + +/* 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(a)chromium.org>"); +MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bdw-rt5650"); 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 34eb0baaa9517..35958553652ec 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 @@ -29,6 +29,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .sof_fw_filename = "sof-bdw.ri", .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", -- 2.24.0.393.g34dc348eaf-goog

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