April 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH][next] ASoC: fsl_easrc: fix spelling mistake "prefitler" -> "prefilter"
by Colin King 23 Apr '20

23 Apr '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a deb_dbg message, fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/fsl_easrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 233f26ff885c..97658e1f4989 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1769,7 +1769,7 @@ static void fsl_easrc_dump_firmware(struct fsl_asrc *easrc) } dev_dbg(dev, "Firmware v%u dump:\n", firm->firmware_version); - dev_dbg(dev, "Num prefitler scenarios: %u\n", firm->prefil_scen); + dev_dbg(dev, "Num prefilter scenarios: %u\n", firm->prefil_scen); dev_dbg(dev, "Num interpolation scenarios: %u\n", firm->interp_scen); dev_dbg(dev, "\nInterpolation scenarios:\n"); -- 2.25.1

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[PATCH v1] ASoC: Intel: Skylake: Replace guid_copy() with import_guid()
by Andy Shevchenko 23 Apr '20

23 Apr '20

There is a specific API to treat raw data as GUID, i.e. import_guid(). Use it instead of guid_copy() with explicit casting. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-sst-utils.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index d43cbf4a71ef2..b233f89517c11 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -290,7 +290,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, goto free_uuid_list; } - guid_copy(&module->uuid, (guid_t *)&mod_entry->uuid); + import_guid(&module->uuid, mod_entry->uuid); module->id = (i | (index << 12)); module->is_loadable = mod_entry->type.load_type; diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2ac..43a159f23a10d 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1989,7 +1989,7 @@ static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) { if (uuid_tkn->token == SKL_TKN_UUID) { - guid_copy(guid, (guid_t *)&uuid_tkn->uuid); + import_guid(guid, uuid_tkn->uuid); return 0; } @@ -3376,8 +3376,8 @@ static int skl_tplg_get_manifest_tkn(struct device *dev, dev_err(dev, "Too many UUID tokens\n"); return -EINVAL; } - guid_copy(&skl->modules[uuid_index++]->uuid, - (guid_t *)&array->uuid->uuid); + import_guid(&skl->modules[uuid_index++]->uuid, + array->uuid->uuid); tuple_size += sizeof(*array->uuid); continue; -- 2.26.1

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[PATCH v2 0/3] Add support for different DMIC configurations
by Mateusz Gorski 23 Apr '20

23 Apr '20

Set of patches to enable DMIC capture on different hardware configurations. Information about supported DMIC configuration is read from NHLT and correct pipeline configuration is selected automatically. Also, adding additional option for topology binary name which is based on used machine driver. Mateusz Gorski (3): ASoC: Intel: Skylake: Add alternative topology binary name ASoC: Intel: Multiple I/O PCM format support for pipe ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT Changes in v2: - removed patch 1/4 swapping machine device and platform device registration order - alt_tplg_name creation now uses different field to read machine driver name - including of <sound/soc-acpi.h> moved to different patch include/uapi/sound/skl-tplg-interface.h | 2 + sound/soc/intel/skylake/skl-topology.c | 189 +++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 1 + 3 files changed, 187 insertions(+), 5 deletions(-) -- 2.17.1

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[PATCH] ALSA: hda: Always use jackpoll helper for jack update after resume
by Takashi Iwai 23 Apr '20

23 Apr '20

HD-audio codec driver applies a tricky procedure to forcibly perform the runtime resume by mimicking the usage count even if the device has been runtime-suspended beforehand. This was needed to assure to trigger the jack detection update after the system resume. And recently we also applied the similar logic to the HD-audio controller side. However this seems leading to some inconsistency, and eventually PCI controller gets screwed up. This patch is an attempt to fix and clean up those behavior: instead of the tricky runtime resume procedure, the existing jackpoll work is scheduled when such a forced codec resume is required. The jackpoll work will power up the codec, and this alone should suffice for the jack status update in usual cases. If the extra polling is requested (by checking codec->jackpoll_interval), the manual update is invoked after that, and the codec is powered down again. Also, we filter the spurious wake up of the codec from the controller runtime resume by checking codec->relaxed_resume flag. If this flag is set, basically we don't need to wake up explicitly, but it's supposed to be done via the audio component notifier. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 28 +++++++++++++++++----------- sound/pci/hda/hda_intel.c | 17 ++--------------- 2 files changed, 19 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 86a632bf4d50..7e3ae4534df9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - snd_hda_jack_set_dirty_all(codec); - snd_hda_jack_poll_all(codec); + /* for non-polling trigger: we need nothing if already powered on */ + if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core)) + return; + + /* the power-up/down sequence triggers the runtime resume */ + snd_hda_power_up_pm(codec); + /* update jacks manually if polling is required, too */ + if (codec->jackpoll_interval) { + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_poll_all(codec); + } + snd_hda_power_down_pm(codec); if (!codec->jackpoll_interval) return; @@ -2951,18 +2961,14 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = hda_codec_need_resume(codec); int ret; - /* The get/put pair below enforces the runtime resume even if the - * device hasn't been used at suspend time. This trick is needed to - * update the jack state change during the sleep. - */ - if (forced_resume) - pm_runtime_get_noresume(dev); ret = pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); + /* schedule jackpoll work for jack detection update */ + if (codec->jackpoll_interval || + (pm_runtime_suspended(dev) && hda_codec_need_resume(codec))) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); return ret; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f995576cff1..0310193ea1bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1004,7 +1004,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) if (status && from_rt) { list_for_each_codec(codec, &chip->bus) - if (status & (1 << codec->addr)) + if (!codec->relaxed_resume && + (status & (1 << codec->addr))) schedule_delayed_work(&codec->jackpoll_work, codec->jackpoll_interval); } @@ -1044,9 +1045,7 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct hda_codec *codec; struct azx *chip; - bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1058,19 +1057,7 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - /* check for the forced resume */ - list_for_each_codec(codec, &chip->bus) { - if (hda_codec_need_resume(codec)) { - forced_resume = true; - break; - } - } - - if (forced_resume) - pm_runtime_get_noresume(dev); pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- 2.16.4

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Re: [PATCH] ASoC: wm8962: restore the CLOCKING2 register in resume
by S.j. Wang 23 Apr '20

23 Apr '20

> > On Tue, Apr 21, 2020 at 08:02:15PM +0800, Shengjiu Wang wrote: > > The CLOCKING2 is a volatile register, but some bits should be restored > > when resume, for example SYSCLK_SRC. otherwise the output clock is > > wrong > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/codecs/wm8962.c | 8 ++++++++ > > 1 file changed, 8 insertions(+) > > > > diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c > > index d9d59f45833f..6e96c0c5ad2a 100644 > > --- a/sound/soc/codecs/wm8962.c > > +++ b/sound/soc/codecs/wm8962.c > > @@ -82,6 +82,7 @@ struct wm8962_priv { #endif > > > > int irq; > > + u32 regcache_clocking2; > > }; > > > > /* We can't use the same notifier block for more than one supply and > > @@ -3813,6 +3814,10 @@ static int wm8962_runtime_resume(struct > device > > *dev) > > > > regcache_sync(wm8962->regmap); > > > > + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, > > + WM8962_SYSCLK_SRC_MASK, > > + wm8962->regcache_clocking2); > > + > > I wonder if it might just be better to make the register non-volatile? From > looking through the datasheet I am guessing this is volatile for the > CLASSD_CLK_DIV bits, which are controlled by the chip itself. But the > datasheet claims these are read only and protected by the security key, and > they are not read by the driver at all. > > Thanks, > Charles > Use non-volatile also can fix the issue, I will send v2 for it Best regards Wang shengjiu

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New codec ALC245
by Kailang 23 Apr '20

23 Apr '20

Hi Takashi, Attach was ALC245 supported patch. Many Thanks. BR, Kailang

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[PATCH] ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif
by Xiyu Yang 23 Apr '20

23 Apr '20

snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which increases the refcount of the snd_usb_audio object "chip". When snd_microii_spdif_default_get() returns, local variable "chip" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in several exception handling paths of snd_microii_spdif_default_get(). When those error scenarios occur such as usb_ifnum_to_if() returns NULL, the function forgets to decrease the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak. Fix this issue by jumping to "end" label when those error scenarios occur. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> --- sound/usb/mixer_quirks.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index c237e24f08d9..0f072426b84c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1508,11 +1508,15 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, /* use known values for that card: interface#1 altsetting#1 */ iface = usb_ifnum_to_if(chip->dev, 1); - if (!iface || iface->num_altsetting < 2) - return -EINVAL; + if (!iface || iface->num_altsetting < 2) { + err = -EINVAL; + goto end; + } alts = &iface->altsetting[1]; - if (get_iface_desc(alts)->bNumEndpoints < 1) - return -EINVAL; + if (get_iface_desc(alts)->bNumEndpoints < 1) { + err = -EINVAL; + goto end; + } ep = get_endpoint(alts, 0)->bEndpointAddress; err = snd_usb_ctl_msg(chip->dev, -- 2.7.4

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Unrecoverable AER error when resuming from RAM (hda regression in 5.7-rc2)
by Alex Xu (Hello71) 23 Apr '20

23 Apr '20

With 5.7-rc2, after resuming from suspend to RAM, I get: [ 55.679382] pcieport 0000:00:03.1: AER: Multiple Uncorrected (Non-Fatal) error received: 0000:00:00.0 [ 55.679405] pcieport 0000:00:03.1: AER: PCIe Bus Error: severity=Uncorrected (Non-Fatal), type=Transaction Layer, (Requester ID) [ 55.679410] pcieport 0000:00:03.1: AER: device [1022:1453] error status/mask=00100000/04400000 [ 55.679414] pcieport 0000:00:03.1: AER: [20] UnsupReq (First) [ 55.679417] pcieport 0000:00:03.1: AER: TLP Header: 40000004 0a0000ff fffc0e80 00000000 [ 55.679423] amdgpu 0000:0a:00.0: AER: can't recover (no error_detected callback) [ 55.679425] snd_hda_intel 0000:0a:00.1: AER: can't recover (no error_detected callback) [ 55.679455] pcieport 0000:00:03.1: AER: device recovery failed Then the display freezes and the system basically falls apart (can't even sudo reboot -f, need to use magic sysrq). I bisected this to "ALSA: hda: Skip controller resume if not needed". Setting snd_hda_intel.power_save=0 resolves the issue. I am using an ASRock B450 Pro4 with Realtek HDA codec: [ 1.009400] snd_hda_intel 0000:0a:00.1: enabling device (0000 -> 0002) [ 1.009425] snd_hda_intel 0000:0a:00.1: Force to non-snoop mode [ 1.009653] snd_hda_intel 0000:0c:00.3: enabling device (0000 -> 0002) [ 1.021452] snd_hda_codec_generic hdaudioC0D0: ignore pin 0x7, too many assigned pins [ 1.021461] snd_hda_codec_generic hdaudioC0D0: ignore pin 0x9, too many assigned pins [ 1.021471] snd_hda_codec_generic hdaudioC0D0: ignore pin 0xb, too many assigned pins [ 1.021480] snd_hda_codec_generic hdaudioC0D0: ignore pin 0xd, too many assigned pins [ 1.021482] snd_hda_codec_generic hdaudioC0D0: autoconfig for Generic: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 1.021482] snd_hda_codec_generic hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 1.021483] snd_hda_codec_generic hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 1.021484] snd_hda_codec_generic hdaudioC0D0: mono: mono_out=0x0 [ 1.021484] snd_hda_codec_generic hdaudioC0D0: dig-out=0x3/0x5 [ 1.021485] snd_hda_codec_generic hdaudioC0D0: inputs: [ 1.046053] snd_hda_codec_realtek hdaudioC1D0: autoconfig for ALC892: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line [ 1.046054] snd_hda_codec_realtek hdaudioC1D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 1.046055] snd_hda_codec_realtek hdaudioC1D0: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 1.046055] snd_hda_codec_realtek hdaudioC1D0: mono: mono_out=0x0 [ 1.046056] snd_hda_codec_realtek hdaudioC1D0: inputs: [ 1.046057] snd_hda_codec_realtek hdaudioC1D0: Front Mic=0x19 [ 1.046058] snd_hda_codec_realtek hdaudioC1D0: Rear Mic=0x18 [ 1.046058] snd_hda_codec_realtek hdaudioC1D0: Line=0x1a I also have an ASUS RX 480 graphics card with HDMI audio output.

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[RFC PATCH 00/16] ASoC/SOF/clk/gpio/dt: add Hifiberry DAC+ PRO support
by Pierre-Louis Bossart 22 Apr '20

22 Apr '20

The Hifiberry DAC+ / DAC+ PRO is supported in the Raspberry PI tree but until now not in the mainline and not for ACPI platforms. This patchset implements the recommendations suggested by Mark Brown back in 2018: first add a gpiochip in the PCM512x codec driver, then use these gpios from a clock driver and the machine driver. Since this patchset relies on different subsystems, sending as RFC for now. I chose to import the original code from the Raspberry PI tree as is, and add my changes on top. If there is a preference to squash the changes that'd be fine. I also don't know if I should split this series in two, one for ASoC and one for clk changes? This patchset does not add changes to the sound/soc/bcm machine drivers, but that should be trivial once all the gpio/clock is available. Thanks to Andy Shevchenko for his help navigating the gpio subsystem and flagging mistakes in the use of lookup tables, and to Rob Herring for pointers on the DT bindings verification tools. Daniel Matuschek (1): clk: hifiberry-dacpro: initial import Pierre-Louis Bossart (15): ASoC: pcm512x: expose 6 GPIOs ASoC: pcm512x: use "sclk" string to retrieve clock ASoC: Intel: sof-pcm512x: use gpiod for LED ASoC: Intel: sof-pcm512x: detect Hifiberry DAC+ PRO ASoC: Intel: sof-pcm512x: reconfigure sclk in hw_params if needed ASoC: Intel: sof-pcm512x: select HIFIBERRY_DACPRO clk clk: hifiberry-dacpro: update SDPX/copyright clk: hifiberry-dacpro: style cleanups, use devm_ clk: hifiberry-dacpro: add OF dependency clk: hifiberry-dacpro: transition to _hw functions clk: hifiberry-dacpro: add ACPI support clk: hifiberry-dacpro: add "sclk" lookup clk: hifiberry-dacpro: toggle GPIOs on prepare/unprepare clk: hifiberry-dacpro: add delay on clock prepare/deprepare ASoC: dt-bindings: add document for Hifiberry DAC+ PRO clock .../bindings/sound/hifiberry-dacpro.yaml | 38 +++ drivers/clk/Kconfig | 4 + drivers/clk/Makefile | 1 + drivers/clk/clk-hifiberry-dacpro.c | 284 ++++++++++++++++++ sound/soc/codecs/pcm512x.c | 134 ++++++++- sound/soc/intel/boards/Kconfig | 2 + sound/soc/intel/boards/sof_pcm512x.c | 190 +++++++++++- 7 files changed, 635 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/hifiberry-dacpro.yaml create mode 100644 drivers/clk/clk-hifiberry-dacpro.c base-commit: dd8e871d4e560eeb8d22af82dde91457ad835a63 -- 2.20.1

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sound:core:oss: what determines the value of plugin->extra_data in rate_src_frames()?
by Xingyu Li 22 Apr '20

22 Apr '20

Hi all, I encountered a problem about ALSA subsystem when I try to reproduce a bug https://syzkaller.appspot.com/bug?id=54576f48248190caa29334a754275ce5e1d8be… . I run the PoC https://syzkaller.appspot.com/text?tag=ReproC&x=1275090ee00000, it is supposed to crash the kernel. But it sometimes does not. Then I found the point is rate_src_frames() in sound/core/oss/rate.c. When I can not reproduce the bug,data->old_src_frames variable in rate_src_frames() and the passed arguments frames are both 11. When it can reproduce, the value of data->old_src_frames is 85. The variable data is one member extra_data in plugin variable. But I can not find where it write to plugin->extra_data. So I want to what determines the value of plugin->extra_data? Why is the value of it different at different time? Thanks

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