July 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.13 008/114] soundwire: bus: only use CLOCK_STOP_MODE0 and fix confusions
by Sasha Levin 10 Jul '21

10 Jul '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 345e9f5ca798600e44c0843646621f2804eb99f4 ] Existing devices and implementations only support the required CLOCK_STOP_MODE0. All the code related to CLOCK_STOP_MODE1 has not been tested and is highly questionable, with a clear confusion between CLOCK_STOP_MODE1 and the simple clock stop state machine. This patch removes all usages of CLOCK_STOP_MODE1 - which has no impact on any solution - and fixes the use of the simple clock stop state machine. The resulting code should be a lot more symmetrical and easier to maintain. Note that CLOCK_STOP_MODE1 is not supported in the SoundWire Device Class specification so it's rather unlikely that we need to re-add this mode later. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20210511030048.25622-2-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/bus.c | 100 ++++++++++++++-------------------- include/linux/soundwire/sdw.h | 2 - 2 files changed, 40 insertions(+), 62 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a9e0aa72654d..dc4033b6f2e9 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -821,26 +821,6 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, mutex_unlock(&bus->bus_lock); } -static enum sdw_clk_stop_mode sdw_get_clk_stop_mode(struct sdw_slave *slave) -{ - enum sdw_clk_stop_mode mode; - - /* - * Query for clock stop mode if Slave implements - * ops->get_clk_stop_mode, else read from property. - */ - if (slave->ops && slave->ops->get_clk_stop_mode) { - mode = slave->ops->get_clk_stop_mode(slave); - } else { - if (slave->prop.clk_stop_mode1) - mode = SDW_CLK_STOP_MODE1; - else - mode = SDW_CLK_STOP_MODE0; - } - - return mode; -} - static int sdw_slave_clk_stop_callback(struct sdw_slave *slave, enum sdw_clk_stop_mode mode, enum sdw_clk_stop_type type) @@ -933,7 +913,6 @@ static int sdw_bus_wait_for_clk_prep_deprep(struct sdw_bus *bus, u16 dev_num) */ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) { - enum sdw_clk_stop_mode slave_mode; bool simple_clk_stop = true; struct sdw_slave *slave; bool is_slave = false; @@ -955,10 +934,8 @@ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) /* Identify if Slave(s) are available on Bus */ is_slave = true; - slave_mode = sdw_get_clk_stop_mode(slave); - slave->curr_clk_stop_mode = slave_mode; - - ret = sdw_slave_clk_stop_callback(slave, slave_mode, + ret = sdw_slave_clk_stop_callback(slave, + SDW_CLK_STOP_MODE0, SDW_CLK_PRE_PREPARE); if (ret < 0) { dev_err(&slave->dev, @@ -966,22 +943,29 @@ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) return ret; } - ret = sdw_slave_clk_stop_prepare(slave, - slave_mode, true); - if (ret < 0) { - dev_err(&slave->dev, - "pre-prepare failed:%d", ret); - return ret; - } - - if (slave_mode == SDW_CLK_STOP_MODE1) + /* Only prepare a Slave device if needed */ + if (!slave->prop.simple_clk_stop_capable) { simple_clk_stop = false; + + ret = sdw_slave_clk_stop_prepare(slave, + SDW_CLK_STOP_MODE0, + true); + if (ret < 0) { + dev_err(&slave->dev, + "pre-prepare failed:%d", ret); + return ret; + } + } } /* Skip remaining clock stop preparation if no Slave is attached */ if (!is_slave) return ret; + /* + * Don't wait for all Slaves to be ready if they follow the simple + * state machine + */ if (!simple_clk_stop) { ret = sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); @@ -998,17 +982,13 @@ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) slave->status != SDW_SLAVE_ALERT) continue; - slave_mode = slave->curr_clk_stop_mode; - - if (slave_mode == SDW_CLK_STOP_MODE1) { - ret = sdw_slave_clk_stop_callback(slave, - slave_mode, - SDW_CLK_POST_PREPARE); + ret = sdw_slave_clk_stop_callback(slave, + SDW_CLK_STOP_MODE0, + SDW_CLK_POST_PREPARE); - if (ret < 0) { - dev_err(&slave->dev, - "post-prepare failed:%d", ret); - } + if (ret < 0) { + dev_err(&slave->dev, + "post-prepare failed:%d", ret); } } @@ -1059,7 +1039,6 @@ EXPORT_SYMBOL(sdw_bus_clk_stop); */ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) { - enum sdw_clk_stop_mode mode; bool simple_clk_stop = true; struct sdw_slave *slave; bool is_slave = false; @@ -1081,31 +1060,33 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) /* Identify if Slave(s) are available on Bus */ is_slave = true; - mode = slave->curr_clk_stop_mode; - - if (mode == SDW_CLK_STOP_MODE1) { - simple_clk_stop = false; - continue; - } - - ret = sdw_slave_clk_stop_callback(slave, mode, + ret = sdw_slave_clk_stop_callback(slave, SDW_CLK_STOP_MODE0, SDW_CLK_PRE_DEPREPARE); if (ret < 0) dev_warn(&slave->dev, "clk stop deprep failed:%d", ret); - ret = sdw_slave_clk_stop_prepare(slave, mode, - false); + /* Only de-prepare a Slave device if needed */ + if (!slave->prop.simple_clk_stop_capable) { + simple_clk_stop = false; - if (ret < 0) - dev_warn(&slave->dev, - "clk stop deprep failed:%d", ret); + ret = sdw_slave_clk_stop_prepare(slave, SDW_CLK_STOP_MODE0, + false); + + if (ret < 0) + dev_warn(&slave->dev, + "clk stop deprep failed:%d", ret); + } } /* Skip remaining clock stop de-preparation if no Slave is attached */ if (!is_slave) return 0; + /* + * Don't wait for all Slaves to be ready if they follow the simple + * state machine + */ if (!simple_clk_stop) sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); @@ -1117,8 +1098,7 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) slave->status != SDW_SLAVE_ALERT) continue; - mode = slave->curr_clk_stop_mode; - sdw_slave_clk_stop_callback(slave, mode, + sdw_slave_clk_stop_callback(slave, SDW_CLK_STOP_MODE0, SDW_CLK_POST_DEPREPARE); } diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ced07f8fde87..5d93d9949653 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -624,7 +624,6 @@ struct sdw_slave_ops { int (*port_prep)(struct sdw_slave *slave, struct sdw_prepare_ch *prepare_ch, enum sdw_port_prep_ops pre_ops); - int (*get_clk_stop_mode)(struct sdw_slave *slave); int (*clk_stop)(struct sdw_slave *slave, enum sdw_clk_stop_mode mode, enum sdw_clk_stop_type type); @@ -675,7 +674,6 @@ struct sdw_slave { struct list_head node; struct completion port_ready[SDW_MAX_PORTS]; unsigned int m_port_map[SDW_MAX_PORTS]; - enum sdw_clk_stop_mode curr_clk_stop_mode; u16 dev_num; u16 dev_num_sticky; bool probed; -- 2.30.2

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[PATCH AUTOSEL 5.13 003/114] ASoC: Intel: sof_sdw: add SOF_RT715_DAI_ID_FIX for AlderLake
by Sasha Levin 10 Jul '21

10 Jul '21

From: Libin Yang <libin.yang(a)intel.com> [ Upstream commit 81cd42e5174ba7918edd3d006406ce21ebaa8507 ] AlderLake needs the flag SOF_RT715_DAI_ID_FIX if it is using the rt715 DMIC. Reviewed-by: Bard Liao <bard.liao(a)intel.com> Signed-off-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210505163705.305616-11-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 85a2797c2550..5827a16773c9 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -196,6 +196,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 | SOF_SDW_TGL_HDMI | + SOF_RT715_DAI_ID_FIX | SOF_SDW_PCH_DMIC), }, {} -- 2.30.2

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[PATCH AUTOSEL 5.13 002/114] ASoC: Intel: sof_sdw: add mutual exclusion between PCH DMIC and RT715
by Sasha Levin 10 Jul '21

10 Jul '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 35564e2bf94611c3eb51d35362addb3cb394ad54 ] When external RT714/715 devices are used for capture, we don't want the PCH DMICs to be used. Any information provided by the SOF platform driver or DMI quirks will be overridden. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Libin Yang <libin.yang(a)intel.com> Link: https://lore.kernel.org/r/20210505163705.305616-5-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 19 +++++++++++++++++-- sound/soc/intel/boards/sof_sdw_common.h | 1 + 2 files changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index ecd3f90f4bbe..85a2797c2550 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -353,6 +353,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .part_id = 0x714, .version_id = 3, .direction = {false, true}, + .ignore_pch_dmic = true, .dai_name = "rt715-aif2", .init = sof_sdw_rt715_sdca_init, }, @@ -360,6 +361,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .part_id = 0x715, .version_id = 3, .direction = {false, true}, + .ignore_pch_dmic = true, .dai_name = "rt715-aif2", .init = sof_sdw_rt715_sdca_init, }, @@ -367,6 +369,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .part_id = 0x714, .version_id = 2, .direction = {false, true}, + .ignore_pch_dmic = true, .dai_name = "rt715-aif2", .init = sof_sdw_rt715_init, }, @@ -374,6 +377,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .part_id = 0x715, .version_id = 2, .direction = {false, true}, + .ignore_pch_dmic = true, .dai_name = "rt715-aif2", .init = sof_sdw_rt715_init, }, @@ -729,7 +733,8 @@ static int create_sdw_dailink(struct device *dev, int *be_index, int *cpu_id, bool *group_generated, struct snd_soc_codec_conf *codec_conf, int codec_count, - int *codec_conf_index) + int *codec_conf_index, + bool *ignore_pch_dmic) { const struct snd_soc_acpi_link_adr *link_next; struct snd_soc_dai_link_component *codecs; @@ -782,6 +787,9 @@ static int create_sdw_dailink(struct device *dev, int *be_index, if (codec_index < 0) return codec_index; + if (codec_info_list[codec_index].ignore_pch_dmic) + *ignore_pch_dmic = true; + cpu_dai_index = *cpu_id; for_each_pcm_streams(stream) { char *name, *cpu_name; @@ -913,6 +921,7 @@ static int sof_card_dai_links_create(struct device *dev, const struct snd_soc_acpi_link_adr *adr_link; struct snd_soc_dai_link_component *cpus; struct snd_soc_codec_conf *codec_conf; + bool ignore_pch_dmic = false; int codec_conf_count; int codec_conf_index = 0; bool group_generated[SDW_MAX_GROUPS]; @@ -1019,7 +1028,8 @@ static int sof_card_dai_links_create(struct device *dev, sdw_cpu_dai_num, cpus, adr_link, &cpu_id, group_generated, codec_conf, codec_conf_count, - &codec_conf_index); + &codec_conf_index, + &ignore_pch_dmic); if (ret < 0) { dev_err(dev, "failed to create dai link %d", be_id); return -ENOMEM; @@ -1087,6 +1097,10 @@ static int sof_card_dai_links_create(struct device *dev, DMIC: /* dmic */ if (dmic_num > 0) { + if (ignore_pch_dmic) { + dev_warn(dev, "Ignoring PCH DMIC\n"); + goto HDMI; + } cpus[cpu_id].dai_name = "DMIC01 Pin"; init_dai_link(dev, links + link_id, be_id, "dmic01", 0, 1, // DMIC only supports capture @@ -1105,6 +1119,7 @@ static int sof_card_dai_links_create(struct device *dev, INC_ID(be_id, cpu_id, link_id); } +HDMI: /* HDMI */ if (hdmi_num > 0) { idisp_components = devm_kcalloc(dev, hdmi_num, diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index f3cb6796363e..ea60e8ed215c 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -56,6 +56,7 @@ struct sof_sdw_codec_info { int amp_num; const u8 acpi_id[ACPI_ID_LEN]; const bool direction[2]; // playback & capture support + const bool ignore_pch_dmic; const char *dai_name; const struct snd_soc_ops *ops; -- 2.30.2

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Re: Audio out on DisplayPort but not HDMI, on AMD Ryzen APU/Vega
by Takashi Iwai 09 Jul '21

09 Jul '21

On Thu, 08 Jul 2021 14:26:54 +0200, Damjan Georgievski wrote: > > On Thu, 8 Jul 2021 at 13:37, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > On Fri, 02 Jul 2021 15:27:30 +0200, > > Damjan Georgievski wrote: > > > > > > I have a desktop computer with an AMD Ryzen 5 4650G PRO APU (CPU + on-die GPU). > > > The motherboard (MSI Mortar B550m wifi / bios 1.6). has one > > > displayport and one HDMI outputs. > > > The displayport is connected to my Lenovo P27 monitor, and the HDMI to > > > my Samsung TV. > > > > > > When I choose the GPU as an audio out, I only get audio on the 3.5" > > > headphones jack of the Lenovo monitor, while I would like to get the > > > audio on the Samsung TV over HDMI. > > > I can't find a way to switch the different outputs (neither over > > > pipewire or just using native alsa). > > > > > > pavucontrol configuration https://i.imgur.com/9nbVtNz.png > > > alsamixer -c0 only has only one option to mute s/pdif, which mutes the > > > output on the displayport -> Lenovo -> 3.5" jack. > > > > > > > > > $ aplay -l > > > **** List of PLAYBACK Hardware Devices **** > > > card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] > > > Subdevices: 1/1 > > > Subdevice #0: subdevice #0 > > > card 2: Generic_1 [HD-Audio Generic], device 0: ALCS1200A Analog > > > [ALCS1200A Analog] > > > Subdevices: 1/1 > > > Subdevice #0: subdevice #0 > > > card 2: Generic_1 [HD-Audio Generic], device 1: ALCS1200A Digital > > > [ALCS1200A Digital] > > > Subdevices: 1/1 > > > Subdevice #0: subdevice #0 > > > > > > Am I correct that card 1 here ^ should have at least two devices? > > > > It could have more entries, depending on the configuration. > > I suppose you enabled CONFIG_SND_HDA_CODEC_HDMI properly? > > yes - CONFIG_SND_HDA_CODEC_HDMI=m is enabled. > I've been using the Arch distro kernel, which has that enabled; but > also with self-compiled 5.13.0 from > stable, with the same distro config; and with 5.14-git (77d34a468) > with the same config. > All have the same behaviour. > > Should this hardware be supported with the > sound/soc/amd/renoir/rn-pci-acp3x.c driver (instead of snd_hda_intel)? > … since it is a Renoir desktop APU. > > > Please give alsa-info.sh output for more detailed analysis. > > http://alsa-project.org/db/?f=00b31f3f0e6c58b56c73f98aec7c6b5ccaff700e The codec proc output there shows that only one pin is enabled for HDMI/DP output, and that's the reason why the driver created only one stream. It's basically a BIOS setup. The patch below should enable all pins forcibly and this should give more streams. Give it a try. Takashi --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1940,6 +1940,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static const struct snd_pci_quirk force_connect_list[] = { SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1), SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1), {} };

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[PATCH] ALSA: hda/proc - dump the coeff with the capital letters
by Hui Wang 09 Jul '21

09 Jul '21

Sometimes we need to dump the coeff to debug the audio issues, and need to compare it with the log of RTHDDump.exe under Windows, this is the coeff under Windows and Linux: Vendor widget coefficient value under Windows Index 0x00 0x0002 Index 0x01 0xAAAA Index 0x02 0x8AAA Processing caps: benign=0, ncoeff=91 under Linux Coeff 0x00: 0x0002 Coeff 0x01: 0xaaaa Coeff 0x02: 0x8aaa Windows prints the hex number with capital letter while Linux prints the hex number with lower case letter, it adds the trouble when comparing them. Let us change it to use capital letter. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 00c2eeb2c472..9037c87665b6 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -590,7 +590,7 @@ static void print_proc_caps(struct snd_info_buffer *buffer, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, i); val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0); - snd_iprintf(buffer, " Coeff 0x%02x: 0x%04x\n", i, val); + snd_iprintf(buffer, " Coeff 0x%02X: 0x%04X\n", i, val); } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, oldindex); } -- 2.25.1

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[Patch v2] ALSA: compress: allow to leave draining state when pausing in draining
by Robert Lee 09 Jul '21

09 Jul '21

When compress offload pauses in draining state, not all platforms need to keep in draining state. Some platforms may call drain or partial drain again when resume from pause in draining, so it needs to wake up from snd_compress_wait_for_drain() in this case. Call API snd_compr_leave_draining_in_pause(), if the platform doesn't need to keep in draining state when pause in draining state. Signed-off-by: Robert Lee <lerobert(a)google.com> --- include/sound/compress_driver.h | 14 ++++++++++++++ sound/core/compress_offload.c | 8 +++++++- 2 files changed, 21 insertions(+), 1 deletion(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 277087f635f3..e16524a93a14 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -145,6 +145,7 @@ struct snd_compr_ops { * @lock: device lock * @device: device id * @use_pause_in_draining: allow pause in draining, true when set + * @leave_draining_in_pause: leave draining state when pausing in draining */ struct snd_compr { const char *name; @@ -156,6 +157,7 @@ struct snd_compr { struct mutex lock; int device; bool use_pause_in_draining; + bool leave_draining_in_pause; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -182,6 +184,18 @@ static inline void snd_compr_use_pause_in_draining(struct snd_compr_stream *subs substream->device->use_pause_in_draining = true; } +/** + * snd_compr_leave_draining_in_pause - Leave draining state when pause in draining + * @substream: compress substream to set + * + * In some platform, we need to leave draining state when we use pause in draining. + * Add API to allow leave draining state. + */ +static inline void snd_compr_leave_draining_in_pause(struct snd_compr_stream *substream) +{ + substream->device->leave_draining_in_pause = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 21ce4c056a92..c6e5c8f072d7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -719,8 +719,14 @@ static int snd_compr_pause(struct snd_compr_stream *stream) if (!stream->device->use_pause_in_draining) return -EPERM; retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) + if (!retval) { + if (stream->device->leave_draining_in_pause) { + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + wake_up(&stream->runtime->sleep); + break; + } stream->pause_in_draining = true; + } break; default: return -EPERM; -- 2.32.0.93.g670b81a890-goog

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[PATCH v4] ASoC: soc-pcm: add a flag to reverse the stop sequence
by Vijendar Mukunda 09 Jul '21

09 Jul '21

From: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> On stream stop, currently CPU DAI stop sequence invoked first followed by DMA. For Few platforms, it is required to stop the DMA first before stopping CPU DAI. For Stoneyridge platform, it is required to invoke DMA driver stop first rather than invoking DWC I2S controller stop. Introduced new flag in dai_link structure for reordering stop sequence. Based on flag check, ASoC core will re-order the stop sequence. Fixes: 4378f1fbe92405 ("ASoC: soc-pcm: Use different sequence for start/stop trigger") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v3 -> v4: Updated Fixes tag in commit message v2 -> v3: moved "stop_dma_first" flag from card structure to dai_link structure and modified code to use. "stop_dma_first" flag in acp machine driver and soc pcm trigger api v1 -> v2: renamed flag as "stop_dma_fist" fixed build error by removing extra + symbol sound/soc/soc-pcm.c:1019:3: error: expected expression before 'struct' 1019 | + struct snd_soc_card *card = rtd->card; include/sound/soc.h | 6 ++++++ sound/soc/amd/acp-da7219-max98357a.c | 5 +++++ sound/soc/soc-pcm.c | 22 ++++++++++++++++------ 3 files changed, 27 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 675849d07284..8e6dd8a257c5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -712,6 +712,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* This flag will reorder stop sequence. By enabling this flag + * DMA controller stop sequence will be invoked first followed by + * CPU DAI driver stop sequence + */ + unsigned int stop_dma_first:1; + #ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; /* For topology */ #endif diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 84e3906abd4f..9449fb40a956 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_rt5682_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_play_ops, SND_SOC_DAILINK_REG(designware1, rt5682, platform), }, @@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, SND_SOC_DAILINK_REG(designware2, rt5682, platform), }, @@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, SND_SOC_DAILINK_REG(designware2, adau, platform), }, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 46513bb97904..d1c570ca21ea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1015,6 +1015,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int ret = -EINVAL, _ret = 0; int rollback = 0; @@ -1055,14 +1056,23 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); - if (ret < 0) - break; + if (rtd->dai_link->stop_dma_first) { + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; - ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); - if (ret < 0) - break; + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } else { + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } ret = snd_soc_link_trigger(substream, cmd, rollback); break; } -- 2.17.1

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[PATCH v4] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Randy Dunlap 08 Jul '21

08 Jul '21

On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Fixes: eb17726b00b3 ("m32r: add simple dma") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Acked-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- v2: rebase & resend; drop Cc: to Bo Shen <voice.shen(a)atmel.com> # bounced v3: drop "depends on HAS_DMA" (as suggested by Codrin; Thanks) v4: update Fixes: tag as suggested by Codrin (Thanks) --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) --- linux-next-20210707.orig/sound/soc/atmel/Kconfig +++ linux-next-20210707/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool

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Can't get snd-aloop in sync with another device
by Mark Hills 08 Jul '21

08 Jul '21

I use dmix+snd-aloop for all local applications; then forwarded to the dmix+Layla3G using a custom C program. Despite asking for clock sync, they slowly drift. Am I doing something wrong? $ echo hw:Layla3G,0,0 > /proc/asound/Loopback/timer_source The forwarder drains its buffer and eventually underruns. Loopback is not providing enough audio to feed Layla3G -- drifts by about 9 samples/second at 48kHz. This _feels_ like a regression; I'm pretty sure this was perfectly in time before. I'd need to bisect kernel and alsa-lib changes. Below shows the status during playback/capture. The timer is definitely connected in some way. It's listed below, and aloop ticking corresponds to whether Layla3G is open by any application (not just the forwarder) -- Mark $ cat /proc/asound/Loopback/cable#0 Cable 0 substream 0: inactive Cable 0 substream 1: inactive Cable 0 substream 2: inactive Cable 0 substream 3: valid: 3 running: 3 pause: 0 Playback buffer_size: 65536 buffer_pos: 12800 silent_size: 0 period_size: 512 bytes_per_sec: 384000 sample_align: 8 rate_shift: 0 sound timer: hw:0,0,0 timer open: playback Capture buffer_size: 65536 buffer_pos: 53760 silent_size: 0 period_size: 512 bytes_per_sec: 384000 sample_align: 8 rate_shift: 0 sound timer: hw:0,0,0 timer open: playback Cable 0 substream 4: inactive Cable 0 substream 5: inactive Cable 0 substream 6: inactive Cable 0 substream 7: inactive

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trigger timestamp
by Muni Sekhar 08 Jul '21

08 Jul '21

Hi all, >From the user space test application , Is it possible to get the trigger timestamp corresponding to SNDRV_PCM_TRIGGER_START & SNDRV_PCM_TRIGGER_STOP events for audio playback capture? -- Thanks, Sekhar

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