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

[alsa-devel] [PATCH AUTOSEL 5.4 292/350] ASoC: wm8904: fix regcache handling
by Sasha Levin 10 Dec '19

10 Dec '19

From: Michael Walle <michael(a)walle.cc> [ Upstream commit e9149b8c00d25dbaef1aa174fc604bed207e576d ] The current code assumes that the power is turned off in SND_SOC_BIAS_OFF. If there are no actual regulator the codec isn't turned off and the registers are not reset to their default values but the regcache is still marked as dirty. Thus a value might not be written to the hardware if it is set to the default value. Do a software reset before turning off the power to make sure the registers are always reset to their default states. Signed-off-by: Michael Walle <michael(a)walle.cc> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20191112223629.21867-1-michael@walle.cc Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index bcb3c9d5abf0c..9e8c564f6e9c4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1917,6 +1917,7 @@ static int wm8904_set_bias_level(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); + snd_soc_component_write(component, WM8904_SW_RESET_AND_ID, 0); regcache_cache_only(wm8904->regmap, true); regcache_mark_dirty(wm8904->regmap); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 269/350] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Sasha Levin 10 Dec '19

10 Dec '19

From: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> [ Upstream commit e2db787bdcb4f2722ecf410168f0583764634e45 ] On KBL platform, the microphone is attached to external codec(rt5514) instead of PCH. However, TDM slot between PCH and codec is 16 bits only. In order to avoid setting wrong format, we should add a constraint to force to use 16 bits format forever. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20190923162940.199580-1-yuhsuan@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a0..67b276a65a8d2 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -400,6 +400,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 265/350] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile
by Sasha Levin 10 Dec '19

10 Dec '19

From: Ben Zhang <benzh(a)chromium.org> [ Upstream commit eabf424f7b60246c76dcb0ea6f1e83ef9abbeaa6 ] The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1 while it's streaming audio over SPI. The DSP firmware turns on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap does not believe that register can change by itself. When BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't read the register first before write, so PLL2 power bit is cleared by accident. Marking MX-64h as volatile in regmap solved the issue. Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 315a3d39bc091..8bc9450da79cf 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -298,6 +298,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_I2C_MASTER_CTRL7: case RT5677_I2C_MASTER_CTRL8: case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_ANLG2: /* Modified by DSP firmware */ case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: case RT5677_ASRC_22: -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 253/350] soundwire: intel: fix PDI/stream mapping for Bulk
by Sasha Levin 10 Dec '19

10 Dec '19

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit c134f914e9f55b7817e2bae625ec0e5f1379f7cd ] The previous formula is incorrect for PDI0/1, the mapping is not linear but has a discontinuity between PDI1 and PDI2. This change has no effect on PCM PDIs (same mapping). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191022232948.17156-1-pierre-louis.bossart@linux… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 13c54eac0cc3e..d1839707128a8 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -479,7 +479,10 @@ intel_pdi_shim_configure(struct sdw_intel *sdw, struct sdw_cdns_pdi *pdi) unsigned int link_id = sdw->instance; int pdi_conf = 0; - pdi->intel_alh_id = (link_id * 16) + pdi->num + 5; + /* the Bulk and PCM streams are not contiguous */ + pdi->intel_alh_id = (link_id * 16) + pdi->num + 3; + if (pdi->num >= 2) + pdi->intel_alh_id += 2; /* * Program stream parameters to stream SHIM register @@ -508,7 +511,10 @@ intel_pdi_alh_configure(struct sdw_intel *sdw, struct sdw_cdns_pdi *pdi) unsigned int link_id = sdw->instance; unsigned int conf; - pdi->intel_alh_id = (link_id * 16) + pdi->num + 5; + /* the Bulk and PCM streams are not contiguous */ + pdi->intel_alh_id = (link_id * 16) + pdi->num + 3; + if (pdi->num >= 2) + pdi->intel_alh_id += 2; /* Program Stream config ALH register */ conf = intel_readl(alh, SDW_ALH_STRMZCFG(pdi->intel_alh_id)); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 214/350] ALSA: timer: Limit max amount of slave instances
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit fdea53fe5de532969a332d6e5e727f2ad8bf084d ] The fuzzer tries to open the timer instances as much as possible, and this may cause a system hiccup easily. We've already introduced the cap for the max number of available instances for the h/w timers, and we should put such a limit also to the slave timers, too. This patch introduces the limit to the multiple opened slave timers. The upper limit is hard-coded to 1000 for now, which should suffice for any practical usages up to now. Link: https://lore.kernel.org/r/20191106154257.5853-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/core/timer.c b/sound/core/timer.c index 59ae21b0bb936..013f0e69ff0f7 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -74,6 +74,9 @@ static LIST_HEAD(snd_timer_slave_list); /* lock for slave active lists */ static DEFINE_SPINLOCK(slave_active_lock); +#define MAX_SLAVE_INSTANCES 1000 +static int num_slaves; + static DEFINE_MUTEX(register_mutex); static int snd_timer_free(struct snd_timer *timer); @@ -252,6 +255,10 @@ int snd_timer_open(struct snd_timer_instance **ti, err = -EINVAL; goto unlock; } + if (num_slaves >= MAX_SLAVE_INSTANCES) { + err = -EBUSY; + goto unlock; + } timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { err = -ENOMEM; @@ -261,6 +268,7 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->slave_id = tid->device; timeri->flags |= SNDRV_TIMER_IFLG_SLAVE; list_add_tail(&timeri->open_list, &snd_timer_slave_list); + num_slaves++; err = snd_timer_check_slave(timeri); if (err < 0) { snd_timer_close_locked(timeri, &card_dev_to_put); @@ -356,6 +364,8 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri, } list_del(&timeri->open_list); + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + num_slaves--; /* force to stop the timer */ snd_timer_stop(timeri); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 208/350] ASoC: SOF: topology: set trigger order for FE DAI link
by Sasha Levin 10 Dec '19

10 Dec '19

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> [ Upstream commit 5eee2b3f60065a2530d13f28e771be48b989eb4c ] Set trigger order for FE DAI links to SND_SOC_DPCM_TRIGGER_POST to trigger the BE DAI's before the FE DAI's. This prevents the xruns seen on playback pipelines using the link DMA. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191104224812.3393-3-ranjani.sridharan@linux.int… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/topology.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 4452594c2e17a..fa299e0781561 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2828,6 +2828,10 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, if (!link->no_pcm) { link->nonatomic = true; + /* set trigger order */ + link->trigger[0] = SND_SOC_DPCM_TRIGGER_POST; + link->trigger[1] = SND_SOC_DPCM_TRIGGER_POST; + /* nothing more to do for FE dai links */ return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 195/350] ALSA: pcm: Fix missing check of the new non-cached buffer type
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 6111fd2370eecae9f11bfdc08ba097e0b51fcfd3 ] The check for the mmap support via hw_support_mmap() function misses the case where the device is with SNDRV_DMA_TYPE_DEV_UC, which should have been treated equally as SNDRV_DMA_TYPE_DEV. Let's fix it. Note that this bug doesn't hit any practical problem, because SNDRV_DMA_TYPE_DEV_UC is used only for x86-specific drivers (snd-hda-intel and snd-intel8x0) for the specific platforms that need the non-cached buffers. And, on such platforms, hw_support_mmap() already returns true in anyway. That's the reason I didn't put Cc-to-stable mark here. This is only for any theoretical future extension. Fixes: 425da159707b ("ALSA: pcm: use dma_can_mmap() to check if a device supports dma_mmap_*") Fixes: 42e748a0b325 ("ALSA: memalloc: Add non-cached buffer type") Link: https://lore.kernel.org/r/20191104101115.27311-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 91c6ad58729fe..c3a139436ac26 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -222,7 +222,8 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) return false; if (substream->ops->mmap || - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV) + (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && + substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) return true; return dma_can_mmap(substream->dma_buffer.dev.dev); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 191/350] ALSA: bebob: expand sleep just after breaking connections for protocol version 1
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d3eabe939aee3ffd5b133766a932629a9746298c ] As long as I investigated, some devices with BeBoB protocol version 1 can be freezed during several hundreds milliseconds after breaking connections. When accessing during the freezed time, any transaction is corrupted. In the worst case, the device is going to reboot. I can see this issue in: * Roland FA-66 * M-Audio FireWire Solo This commit expands sleep just after breaking connections to avoid the freezed time as much as possible. I note that the freeze/reboot behaviour is similar to below models: * Focusrite Saffire Pro 10 I/O * Focusrite Saffire Pro 26 I/O The above models certainly reboot after breaking connections. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20191101131323.17300-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/bebob/bebob_stream.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 6c1497d9f52ba..ce07ea0d4e71d 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -415,15 +415,16 @@ static int make_both_connections(struct snd_bebob *bebob) return 0; } -static void -break_both_connections(struct snd_bebob *bebob) +static void break_both_connections(struct snd_bebob *bebob) { cmp_connection_break(&bebob->in_conn); cmp_connection_break(&bebob->out_conn); - /* These models seems to be in transition state for a longer time. */ - if (bebob->maudio_special_quirk != NULL) - msleep(200); + // These models seem to be in transition state for a longer time. When + // accessing in the state, any transactions is corrupted. In the worst + // case, the device is going to reboot. + if (bebob->version < 2) + msleep(600); } static int -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 179/350] ALSA: hda/hdmi - implement mst_no_extra_pcms flag
by Sasha Levin 10 Dec '19

10 Dec '19

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 2a2edfbbfee47947dd05f5860c66c0e80ee5e09d ] To support the DP-MST multiple streams via single connector feature, the HDMI driver was extended with the concept of backup PCMs. See commit 9152085defb6 ("ALSA: hda - add DP MST audio support"). This implementation works fine with snd_hda_intel.c as PCM topology is fully managed within the single driver. When the HDA codec driver is used from ASoC components, the concept of backup PCMs no longer fits. For ASoC topologies, the physical HDMI converters are presented as backend DAIs and these should match with hardware capabilities. The ASoC topology may define arbitrary PCMs (i.e. frontend DAIs) and have processing elements before eventual routing to the HDMI BE DAIs. With backup PCMs, the link between FE and BE DAIs would become dynamic and change when monitors are (un)plugged. This would lead to modifying the topology every time hotplug events happen, which is not currently possible in ASoC and there does not seem to be any obvious benefits from this design. To overcome above problems and enable the HDMI driver to be used from ASoC, this patch adds a new mode (mst_no_extra_pcms flags) to patch_hdmi.c. In this mode, the codec driver does not assume the backup PCMs to be created. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191029134017.18901-2-kai.vehmanen@linux.intel.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 19 ++++++++++++++----- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 9a0393cf024c2..ac18f428eda6c 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -254,6 +254,7 @@ struct hda_codec { unsigned int force_pin_prefix:1; /* Add location prefix */ unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ + unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d14f6684737d0..b8579cd218098 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2075,15 +2075,24 @@ static bool is_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx) static int generic_hdmi_build_pcms(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - int idx; + int idx, pcm_num; /* * for non-mst mode, pcm number is the same as before - * for DP MST mode, pcm number is (nid number + dev_num - 1) - * dev_num is the device entry number in a pin - * + * for DP MST mode without extra PCM, pcm number is same + * for DP MST mode with extra PCMs, pcm number is + * (nid number + dev_num - 1) + * dev_num is the device entry number in a pin */ - for (idx = 0; idx < spec->num_nids + spec->dev_num - 1; idx++) { + + if (codec->mst_no_extra_pcms) + pcm_num = spec->num_nids; + else + pcm_num = spec->num_nids + spec->dev_num - 1; + + codec_dbg(codec, "hdmi: pcm_num set to %d\n", pcm_num); + + for (idx = 0; idx < pcm_num; idx++) { struct hda_pcm *info; struct hda_pcm_stream *pstr; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 170/350] ALSA: hda - Fix pending unsol events at shutdown
by Sasha Levin 10 Dec '19

10 Dec '19

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ca58f55108fee41d87c9123f85ad4863e5de7f45 ] This is an alternative fix attemp for the issue reported in the commit caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") that was reverted later due to regressions. Instead of tweaking the hardware disablement order and the enforced irq flushing, do calling cancel_work_sync() of the unsol work early enough, and explicitly ignore the unsol events during the shutdown by checking the bus->shutdown flag. Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_bind.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 8272b50b83493..6a85645663759 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -43,6 +43,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + /* ignore unsol events during shutdown */ + if (codec->bus->shutdown) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c52419376c748..86a416cdeb296 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1382,8 +1382,11 @@ static int azx_free(struct azx *chip) static int azx_dev_disconnect(struct snd_device *device) { struct azx *chip = device->device_data; + struct hdac_bus *bus = azx_bus(chip); chip->bus.shutdown = 1; + cancel_work_sync(&bus->unsol_work); + return 0; } -- 2.20.1

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