March 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/5] ASoC: Intel: Skylake fixes
by Guneshwor Singh 01 Mar '18

01 Mar '18

This includes fixes for Skylake+ driver which are improving widget type check, fixes for simultaneous wake-on-voice (WoV) and capture use-case, fix to improve firmware download latency and lastly, fix to use uuid instead of module id for library modules. Guneshwor Singh (1): ASoC: Intel: Skylake: Refine skl widget type check Pawse, GuruprasadX (2): ASoC: Intel: Skylake: Add BE DAI for WoV usecase ASoC: Intel: board: Add BE DAI link for WoV and update DAPM machine map. Sanyog Kale (1): ASoC: Intel: Skylake: Disable clock and power gating during fw/lib download Sriram Periyasamy (1): ASoC: Intel: Skylake: Find module id from UUID for bind params sound/soc/intel/boards/bxt_rt298.c | 15 +++++ sound/soc/intel/skylake/skl-messages.c | 7 ++- sound/soc/intel/skylake/skl-pcm.c | 20 +++++- sound/soc/intel/skylake/skl-sst-ipc.h | 3 + sound/soc/intel/skylake/skl-topology.c | 107 +++++++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-topology.h | 13 +++- sound/soc/intel/skylake/skl.c | 21 +++++++ sound/soc/intel/skylake/skl.h | 4 ++ 8 files changed, 176 insertions(+), 14 deletions(-) -- 2.16.1

2 10

[alsa-devel] Applied "ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1310e737a68741c9b06cf63be0084bae6018b308 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 25 Feb 2018 11:46:55 +0100 Subject: [PATCH] ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection To determine if a plugged in jack is a headset (speakers + mic) or headphones (mic contact shorted to ground) we use the micbias1 OVer Current Detect (OVCD) functionality. For this to work we need to have a micbias current to actually cause an overcurrent condition when headphones are plugged in, so jack-type detection requires both the LDO and micbias1 supplies to be on. Before this commit there were 2 issues with the handling of this: 1) The LDO supply was force-enabled twice and never disabled again even though it only needs to be forced on when doing jack-type detection 2) micbias1 was not force-enabled, and thus may be off when doing jack-type detection This commit fixes both by force-enabling the LDO and micbias1 supplies before checking for an overcurrent condition and disabling them afterwards. Note that both supplies will still get turned on normally (and OVCD will protect against overcurrent) when the micbias1 is enabled normally because the user has activated a sound stream recording from the mic. Tested-by: Carlo Caione <carlo(a)endlessm.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5651.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index cbf9f56ecb14..be942886700d 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1561,6 +1561,28 @@ static int rt5651_set_bias_level(struct snd_soc_component *component, return 0; } +static void rt5651_enable_micbias1_for_ovcd(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "micbias1"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "micbias1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "LDO"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + static irqreturn_t rt5651_irq(int irq, void *data) { struct rt5651_priv *rt5651 = data; @@ -1574,7 +1596,6 @@ static irqreturn_t rt5651_irq(int irq, void *data) static int rt5651_set_jack(struct snd_soc_component *component, struct snd_soc_jack *hp_jack, void *data) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); int ret; @@ -1616,9 +1637,6 @@ static int rt5651_set_jack(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5651_PWR_ANLG2, RT5651_PWR_JD_M, RT5651_PWR_JD_M); - snd_soc_dapm_force_enable_pin(dapm, "LDO"); - snd_soc_dapm_sync(dapm); - snd_soc_component_update_bits(component, RT5651_MICBIAS, 0x38, 0x38); rt5651->hp_jack = hp_jack; @@ -1819,13 +1837,10 @@ static const struct dmi_system_id rt5651_quirk_table[] = { static int rt5651_jack_detect(struct snd_soc_component *component, int jack_insert) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); int jack_type; if (jack_insert) { - snd_soc_dapm_force_enable_pin(dapm, "LDO"); - snd_soc_dapm_sync(dapm); - + rt5651_enable_micbias1_for_ovcd(component); snd_soc_component_update_bits(component, RT5651_MICBIAS, RT5651_MIC1_OVCD_MASK | RT5651_MIC1_OVTH_MASK | @@ -1842,6 +1857,7 @@ static int rt5651_jack_detect(struct snd_soc_component *component, int jack_inse jack_type = SND_JACK_HEADSET; snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, RT5651_MB1_OC_CLR, 0); + rt5651_disable_micbias1_for_ovcd(component); } else { /* jack out */ jack_type = 0; -- 2.16.2

1 0

[alsa-devel] xhci: Not enough bandwidth for new device state
by Waldemar Brodkorb 01 Mar '18

01 Mar '18

Hi, I am getting these messages from a Debian 9.3 system using Linux 4.9.x: [ 547.352746] usb 3-5.2: 3:1: usb_set_interface failed (-28) [ 548.352865] usb 3-5.3: Not enough bandwidth for new device state. [ 548.352868] usb 3-5.3: Not enough bandwidth for altsetting 1 I have connected a 7-Port USB 3.0 Hub to my machine. I have 6 different Headsets and USB Soundcards connected to the HUB. As soon as I start a C++ application which opens all ALSA devices I get these errors and the USB stopps working correctly. I think the maximum count of USB sound devices, which behave stable is three or four. Is this some miscalculation of the Linux kernel regarding the USB bandwidth usage? How I can determine how much bandwidth the USB sound card driver preallocates? Or are five USB soundcards really too much for an USB 3.0 bus? Thanks in advance for any advise, best regards Waldemar P.s.: I sent this message to LKML, but maybe it is more appropriate to ask here.

2 3

[alsa-devel] Applied "ASoC: Intel: sst: Fix error-code check in sst_pause_stream()" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Fix error-code check in sst_pause_stream() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6f5062773747b3d3eb9f957030ac1be98e983ac2 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:38 +0100 Subject: [PATCH] ASoC: Intel: sst: Fix error-code check in sst_pause_stream() The value returned by sst_prepare_and_post_msg() is a negated SST_ERR_* value, so we must check for -SST_ERR_INVALID_STREAM_ID. Note that sst_pause_resume() already has the correct check. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index 7ee6aeb7e0af..b082b0922a7a 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -253,7 +253,7 @@ int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) if (retval == 0) { str_info->prev = str_info->status; str_info->status = STREAM_PAUSED; - } else if (retval == SST_ERR_INVALID_STREAM_ID) { + } else if (retval == -SST_ERR_INVALID_STREAM_ID) { retval = -EINVAL; mutex_lock(&sst_drv_ctx->sst_lock); sst_clean_stream(str_info); -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: Intel: sst: Remove 2 unused members from stream_info struct" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove 2 unused members from stream_info struct has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bd47469c9985b50831a55bdda5a36f4672452764 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:39 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove 2 unused members from stream_info struct Remove the unused ops and str_id members from the stream_info struct. While at it also remove some kernel-doc comments for members which have already been removed in the past. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 7 ------- sound/soc/intel/atom/sst/sst_pvt.c | 1 - 2 files changed, 8 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index e02e2b4cc08f..1f3107909044 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -181,21 +181,15 @@ struct sst_block { * * @status : stream current state * @prev : stream prev state - * @ops : stream operation pb/cp/drm... - * @bufs: stream buffer list * @lock : stream mutex for protecting state * @pcm_substream : PCM substream * @period_elapsed : PCM period elapsed callback * @sfreq : stream sampling freq - * @str_type : stream type * @cumm_bytes : cummulative bytes decoded - * @str_type : stream type - * @src : stream source */ struct stream_info { unsigned int status; unsigned int prev; - unsigned int ops; struct mutex lock; void *pcm_substream; @@ -212,7 +206,6 @@ struct stream_info { unsigned int num_ch; unsigned int pipe_id; - unsigned int str_id; unsigned int task_id; }; diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index b1e6b8f34a6a..d7d38b0f68db 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -365,7 +365,6 @@ void sst_init_stream(struct stream_info *stream, { stream->status = STREAM_INIT; stream->prev = STREAM_UN_INIT; - stream->ops = ops; } int sst_validate_strid( -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: Intel: sst: Remove unnecessary sst_init_stream() function" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove unnecessary sst_init_stream() function has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8cf732bbd8d6bc9a28ac48c7bac0ccc9f55cb151 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:40 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove unnecessary sst_init_stream() function sst_init_stream() has only one caller and all its function arguments are unused. Inline it on the one call site and remove it. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 2 -- sound/soc/intel/atom/sst/sst_pvt.c | 7 ------- sound/soc/intel/atom/sst/sst_stream.c | 5 ++--- 3 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 1f3107909044..164b0f674c20 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -494,8 +494,6 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, void sst_process_pending_msg(struct work_struct *work); int sst_assign_pvt_id(struct intel_sst_drv *sst_drv_ctx); -void sst_init_stream(struct stream_info *stream, - int codec, int sst_id, int ops, u8 slot); int sst_validate_strid(struct intel_sst_drv *sst_drv_ctx, int str_id); struct stream_info *get_stream_info(struct intel_sst_drv *sst_drv_ctx, int str_id); diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index d7d38b0f68db..af93244b4868 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -360,13 +360,6 @@ int sst_assign_pvt_id(struct intel_sst_drv *drv) return local; } -void sst_init_stream(struct stream_info *stream, - int codec, int sst_id, int ops, u8 slot) -{ - stream->status = STREAM_INIT; - stream->prev = STREAM_UN_INIT; -} - int sst_validate_strid( struct intel_sst_drv *sst_drv_ctx, int str_id) { diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index b082b0922a7a..7cf5dd1993bc 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -83,6 +83,8 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) pipe_id = str_params->device_type; task_id = str_params->task; + sst_drv_ctx->streams[str_id].status = STREAM_INIT; + sst_drv_ctx->streams[str_id].prev = STREAM_UN_INIT; sst_drv_ctx->streams[str_id].pipe_id = pipe_id; sst_drv_ctx->streams[str_id].task_id = task_id; sst_drv_ctx->streams[str_id].num_ch = num_ch; @@ -100,9 +102,6 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) pipe_id, task_id); /* allocate device type context */ - sst_init_stream(&sst_drv_ctx->streams[str_id], alloc_param.codec_type, - str_id, alloc_param.operation, 0); - dev_dbg(sst_drv_ctx->dev, "Alloc for str %d pipe %#x\n", str_id, pipe_id); ret = sst_prepare_and_post_msg(sst_drv_ctx, task_id, IPC_CMD, -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From de9833476de7eb1d4b8bdc599cbbd616fb0e1a28 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:41 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states STREAM_DECODE is completely unused, status == STREAM_RESET was checked for, but never set, remove both. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 4 +--- sound/soc/intel/atom/sst/sst_drv_interface.c | 19 ------------------- 2 files changed, 1 insertion(+), 22 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 164b0f674c20..f4fd442080b2 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -65,9 +65,7 @@ enum sst_stream_states { STREAM_UN_INIT = 0, /* Freed/Not used stream */ STREAM_RUNNING = 1, /* Running */ STREAM_PAUSED = 2, /* Paused stream */ - STREAM_DECODE = 3, /* stream is in decoding only state */ - STREAM_INIT = 4, /* stream init, waiting for data */ - STREAM_RESET = 5, /* force reset on recovery */ + STREAM_INIT = 3, /* stream init, waiting for data */ }; enum sst_ram_type { diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 71af5449be90..6a8b253c58d2 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -238,16 +238,7 @@ static int sst_cdev_close(struct device *dev, unsigned int str_id) return -EINVAL; } - if (stream->status == STREAM_RESET) { - dev_dbg(dev, "stream in reset state...\n"); - stream->status = STREAM_UN_INIT; - - retval = 0; - goto put; - } - retval = sst_free_stream(ctx, str_id); -put: stream->compr_cb_param = NULL; stream->compr_cb = NULL; @@ -256,7 +247,6 @@ static int sst_cdev_close(struct device *dev, unsigned int str_id) dev_dbg(dev, "End\n"); return retval; - } static int sst_cdev_ack(struct device *dev, unsigned int str_id, @@ -486,16 +476,7 @@ static int sst_close_pcm_stream(struct device *dev, unsigned int str_id) return -EINVAL; } - if (stream->status == STREAM_RESET) { - /* silently fail here as we have cleaned the stream earlier */ - dev_dbg(ctx->dev, "stream in reset state...\n"); - - retval = 0; - goto put; - } - retval = free_stream_context(ctx, str_id); -put: stream->pcm_substream = NULL; stream->status = STREAM_UN_INIT; stream->period_elapsed = NULL; -- 2.16.2

1 0

[alsa-devel] Applied "ASoC: Intel: sst: Add sst_realloc_stream() function" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Add sst_realloc_stream() function has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 473858cad1ea3d4e6c287bad23961cd5fb7382d9 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:42 +0100 Subject: [PATCH] ASoC: Intel: sst: Add sst_realloc_stream() function Move the struct snd_sst_alloc_mrfld alloc parameters from the stack into struct stream_info and add a new sst_realloc_stream() function which can re-alloc a stream with the same parameters as before. This is a preparation patch for fixing suspend/resume issues with some SST / DSP firmware versions. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 3 ++ sound/soc/intel/atom/sst/sst_stream.c | 88 +++++++++++++++++++++-------------- 2 files changed, 56 insertions(+), 35 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index f4fd442080b2..a357cd615b59 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -180,6 +180,7 @@ struct sst_block { * @status : stream current state * @prev : stream prev state * @lock : stream mutex for protecting state + * @alloc_param : parameters used for stream (re-)allocation * @pcm_substream : PCM substream * @period_elapsed : PCM period elapsed callback * @sfreq : stream sampling freq @@ -189,6 +190,7 @@ struct stream_info { unsigned int status; unsigned int prev; struct mutex lock; + struct snd_sst_alloc_mrfld alloc_param; void *pcm_substream; void (*period_elapsed)(void *pcm_substream); @@ -429,6 +431,7 @@ struct intel_sst_ops { void (*post_download)(struct intel_sst_drv *sst); }; +int sst_realloc_stream(struct intel_sst_drv *sst_drv_ctx, int str_id); int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int id); int sst_resume_stream(struct intel_sst_drv *sst_drv_ctx, int id); int sst_drop_stream(struct intel_sst_drv *sst_drv_ctx, int id); diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index 7cf5dd1993bc..fcedaa237505 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -35,29 +35,31 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) { - struct snd_sst_alloc_mrfld alloc_param; + struct snd_pcm_params *pcm_params; struct snd_sst_params *str_params; struct snd_sst_tstamp fw_tstamp; struct stream_info *str_info; - struct snd_sst_alloc_response *response; - unsigned int str_id, pipe_id, task_id; - int i, num_ch, ret = 0; - void *data = NULL; + int i, num_ch, str_id; dev_dbg(sst_drv_ctx->dev, "Enter\n"); str_params = (struct snd_sst_params *)params; - memset(&alloc_param, 0, sizeof(alloc_param)); - alloc_param.operation = str_params->ops; - alloc_param.codec_type = str_params->codec; - alloc_param.sg_count = str_params->aparams.sg_count; - alloc_param.ring_buf_info[0].addr = + str_id = str_params->stream_id; + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; + + memset(&str_info->alloc_param, 0, sizeof(str_info->alloc_param)); + str_info->alloc_param.operation = str_params->ops; + str_info->alloc_param.codec_type = str_params->codec; + str_info->alloc_param.sg_count = str_params->aparams.sg_count; + str_info->alloc_param.ring_buf_info[0].addr = str_params->aparams.ring_buf_info[0].addr; - alloc_param.ring_buf_info[0].size = + str_info->alloc_param.ring_buf_info[0].size = str_params->aparams.ring_buf_info[0].size; - alloc_param.frag_size = str_params->aparams.frag_size; + str_info->alloc_param.frag_size = str_params->aparams.frag_size; - memcpy(&alloc_param.codec_params, &str_params->sparams, + memcpy(&str_info->alloc_param.codec_params, &str_params->sparams, sizeof(struct snd_sst_stream_params)); /* @@ -67,46 +69,62 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) * Currently hardcoding as per FW reqm. */ num_ch = sst_get_num_channel(str_params); + pcm_params = &str_info->alloc_param.codec_params.uc.pcm_params; for (i = 0; i < 8; i++) { if (i < num_ch) - alloc_param.codec_params.uc.pcm_params.channel_map[i] = i; + pcm_params->channel_map[i] = i; else - alloc_param.codec_params.uc.pcm_params.channel_map[i] = 0xFF; + pcm_params->channel_map[i] = 0xff; } - str_id = str_params->stream_id; - str_info = get_stream_info(sst_drv_ctx, str_id); - if (str_info == NULL) { - dev_err(sst_drv_ctx->dev, "get stream info returned null\n"); - return -EINVAL; - } - - pipe_id = str_params->device_type; - task_id = str_params->task; sst_drv_ctx->streams[str_id].status = STREAM_INIT; sst_drv_ctx->streams[str_id].prev = STREAM_UN_INIT; - sst_drv_ctx->streams[str_id].pipe_id = pipe_id; - sst_drv_ctx->streams[str_id].task_id = task_id; + sst_drv_ctx->streams[str_id].pipe_id = str_params->device_type; + sst_drv_ctx->streams[str_id].task_id = str_params->task; sst_drv_ctx->streams[str_id].num_ch = num_ch; if (sst_drv_ctx->info.lpe_viewpt_rqd) - alloc_param.ts = sst_drv_ctx->info.mailbox_start + + str_info->alloc_param.ts = sst_drv_ctx->info.mailbox_start + sst_drv_ctx->tstamp + (str_id * sizeof(fw_tstamp)); else - alloc_param.ts = sst_drv_ctx->mailbox_add + + str_info->alloc_param.ts = sst_drv_ctx->mailbox_add + sst_drv_ctx->tstamp + (str_id * sizeof(fw_tstamp)); dev_dbg(sst_drv_ctx->dev, "alloc tstamp location = 0x%x\n", - alloc_param.ts); + str_info->alloc_param.ts); dev_dbg(sst_drv_ctx->dev, "assigned pipe id 0x%x to task %d\n", - pipe_id, task_id); + str_info->pipe_id, str_info->task_id); + + return sst_realloc_stream(sst_drv_ctx, str_id); +} + +/** + * sst_realloc_stream - Send msg for (re-)allocating a stream using the + * @sst_drv_ctx intel_sst_drv context pointer + * @str_id: stream ID + * + * Send a msg for (re-)allocating a stream using the parameters previously + * passed to sst_alloc_stream_mrfld() for the same stream ID. + * Return: 0 or negative errno value. + */ +int sst_realloc_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + struct snd_sst_alloc_response *response; + struct stream_info *str_info; + void *data = NULL; + int ret; + + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; - /* allocate device type context */ dev_dbg(sst_drv_ctx->dev, "Alloc for str %d pipe %#x\n", - str_id, pipe_id); - ret = sst_prepare_and_post_msg(sst_drv_ctx, task_id, IPC_CMD, - IPC_IA_ALLOC_STREAM_MRFLD, pipe_id, sizeof(alloc_param), - &alloc_param, &data, true, true, false, true); + str_id, str_info->pipe_id); + + ret = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, + IPC_IA_ALLOC_STREAM_MRFLD, str_info->pipe_id, + sizeof(str_info->alloc_param), &str_info->alloc_param, + &data, true, true, false, true); if (ret < 0) { dev_err(sst_drv_ctx->dev, "FW alloc failed ret %d\n", ret); -- 2.16.2

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[alsa-devel] Applied "ASoC: Intel: sst: Free streams on suspend, re-alloc on resume" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Free streams on suspend, re-alloc on resume has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bf642bf57398572327072d306e26e38aebd94fa7 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:43 +0100 Subject: [PATCH] ASoC: Intel: sst: Free streams on suspend, re-alloc on resume The Bay Trail SST-DSP firmware version looses track of all streams over a suspend/resume, failing any attempts to resume and/or free streams, with a SST_ERR_INVALID_STREAM_ID error. This commit adds support for free-ing the streams on suspend and re-allocating them on resume, fixing suspend/resume issues on devices using this firmware version. This new behavior gets triggered by a new flag in sst_platform_info which only gets set on Bay Trail platforms. This has been tested on the following devices: -Asus T100TA, Bay Trail + ALC5642 codec -Ployer MOMO7W, Bay Trail CR + ALC5652 codec Tested-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/x86/include/asm/platform_sst_audio.h | 1 + sound/soc/intel/atom/sst/sst.c | 24 +++++++++++++++++++++++- sound/soc/intel/atom/sst/sst.h | 4 ++++ sound/soc/intel/atom/sst/sst_acpi.c | 3 ++- sound/soc/intel/atom/sst/sst_stream.c | 24 +++++++++++++++++++++++- 5 files changed, 53 insertions(+), 3 deletions(-) diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h index 5973a2f3db3d..059823bb8af7 100644 --- a/arch/x86/include/asm/platform_sst_audio.h +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -135,6 +135,7 @@ struct sst_platform_info { const struct sst_res_info *res_info; const struct sst_lib_dnld_info *lib_info; const char *platform; + bool streams_lost_on_suspend; }; int add_sst_platform_device(void); #endif diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 8afdff457579..0962bc9adc62 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -449,6 +449,13 @@ static int intel_sst_suspend(struct device *dev) dev_err(dev, "stream %d is running, can't suspend, abort\n", i); return -EBUSY; } + + if (ctx->pdata->streams_lost_on_suspend) { + stream->resume_status = stream->status; + stream->resume_prev = stream->prev; + if (stream->status != STREAM_UN_INIT) + sst_free_stream(ctx, i); + } } synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); @@ -509,8 +516,8 @@ static int intel_sst_resume(struct device *dev) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); struct sst_fw_save *fw_save = ctx->fw_save; - int ret = 0; struct sst_block *block; + int i, ret = 0; if (!fw_save) return 0; @@ -550,6 +557,21 @@ static int intel_sst_resume(struct device *dev) sst_set_fw_state_locked(ctx, SST_FW_RUNNING); } + if (ctx->pdata->streams_lost_on_suspend) { + for (i = 1; i <= ctx->info.max_streams; i++) { + struct stream_info *stream = &ctx->streams[i]; + + if (stream->resume_status != STREAM_UN_INIT) { + dev_dbg(ctx->dev, "Re-allocing stream %d status %d prev %d\n", + i, stream->resume_status, + stream->resume_prev); + sst_realloc_stream(ctx, i); + stream->status = stream->resume_status; + stream->prev = stream->resume_prev; + } + } + } + sst_free_block(ctx, block); return ret; } diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index a357cd615b59..b2a705dc9304 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -179,6 +179,8 @@ struct sst_block { * * @status : stream current state * @prev : stream prev state + * @resume_status : stream current state to restore on resume + * @resume_prev : stream prev state to restore on resume * @lock : stream mutex for protecting state * @alloc_param : parameters used for stream (re-)allocation * @pcm_substream : PCM substream @@ -189,6 +191,8 @@ struct sst_block { struct stream_info { unsigned int status; unsigned int prev; + unsigned int resume_status; + unsigned int resume_prev; struct mutex lock; struct snd_sst_alloc_mrfld alloc_param; diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 6cd481bec275..c90b04cc071d 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -143,10 +143,11 @@ static struct sst_platform_info byt_rvp_platform_data = { .lib_info = &byt_lib_dnld_info, .res_info = &byt_rvp_res_info, .platform = "sst-mfld-platform", + .streams_lost_on_suspend = true, }; /* Cherryview (Cherrytrail and Braswell) uses same mrfld dpcm fw as Baytrail, - * so pdata is same as Baytrail. + * so pdata is same as Baytrail, minus the streams_lost_on_suspend quirk. */ static struct sst_platform_info chv_platform_data = { .probe_data = &byt_fwparse_info, diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index fcedaa237505..107271f7dd63 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -302,7 +302,29 @@ int sst_resume_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) return -EINVAL; if (str_info->status == STREAM_RUNNING) return 0; - if (str_info->status == STREAM_PAUSED) { + + if (str_info->resume_status == STREAM_PAUSED && + str_info->resume_prev == STREAM_RUNNING) { + /* + * Stream was running before suspend and re-created on resume, + * start it to get back to running state. + */ + dev_dbg(sst_drv_ctx->dev, "restart recreated stream after resume\n"); + str_info->status = STREAM_RUNNING; + str_info->prev = STREAM_PAUSED; + retval = sst_start_stream(sst_drv_ctx, str_id); + str_info->resume_status = STREAM_UN_INIT; + } else if (str_info->resume_status == STREAM_PAUSED && + str_info->resume_prev == STREAM_INIT) { + /* + * Stream was idle before suspend and re-created on resume, + * keep it as is. + */ + dev_dbg(sst_drv_ctx->dev, "leaving recreated stream idle after resume\n"); + str_info->status = STREAM_INIT; + str_info->prev = STREAM_PAUSED; + str_info->resume_status = STREAM_UN_INIT; + } else if (str_info->status == STREAM_PAUSED) { retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, IPC_IA_RESUME_STREAM_MRFLD, str_info->pipe_id, 0, NULL, NULL, -- 2.16.2

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[alsa-devel] [PATCH] ASoC: Intel: kbl: Enable mclk and ssp sclk early for rt5663
by Naveen Manohar 01 Mar '18

01 Mar '18

rt5663 requires mclk/sclk early to synchronize its internal clocks. Enable these clocks early. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Kevin Cheng <kevin.cheng(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3c9585eb6860..0e6b7e79441c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -204,13 +204,18 @@ static const struct snd_soc_dapm_widget kabylake_5663_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("DP", NULL), SND_SOC_DAPM_SPK("HDMI", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_5663_map[] = { + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, -- 2.7.4

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