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[PATCH -next] soundwire: intel: Make sdw_intel_init static
by YueHaibing 23 Apr '20

23 Apr '20

Fix sparse warning: drivers/soundwire/intel_init.c:193:6: warning: symbol 'sdw_intel_init' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- drivers/soundwire/intel_init.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 4b769409f6f8..ad7053463889 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -190,7 +190,8 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, * This scans the namespace and creates SoundWire link controller devices * based on the info queried. */ -void *sdw_intel_init(acpi_handle *parent_handle, struct sdw_intel_res *res) +static void *sdw_intel_init(acpi_handle *parent_handle, + struct sdw_intel_res *res) { acpi_status status; -- 2.17.1

4 3

sof-hda-dsp: fix speaker on ThinkPad X1C7
by GitHub pull_request - opened 23 Apr '20

23 Apr '20

alsa-project/alsa-ucm-conf pull request #24 was opened from chrisx8: With the current ucm2 profile at `sof-hda-dsp/HiFi.conf`, built-in speakers on the ThinkPad X1 Carbon Gen 7 sounds muffled/tinny. Muting 'Speaker' and setting 'Speaker' volume to 100 in `alsamixer` fixes this issue. I applied these fixes in the ucm2 profile, so they don't have to be applied manually. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/24 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/24.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda: Refactor Intel NHLT init
by Cezary Rojewski 23 Apr '20

23 Apr '20

NHLT fetch based on _DSM prevents ACPI table override mechanism from being utilized. Make use of acpi_get_table to enable it and get rid of redundant code. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/hda/intel-nhlt.c | 49 +++++++----------------------------------- 1 file changed, 8 insertions(+), 41 deletions(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 99a23fe7fab9..2f741d2792d8 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -4,58 +4,25 @@ #include <linux/acpi.h> #include <sound/intel-nhlt.h> -#define NHLT_ACPI_HEADER_SIG "NHLT" - -/* Unique identification for getting NHLT blobs */ -static const guid_t osc_guid = - GUID_INIT(0xA69F886E, 0x6CEB, 0x4594, - 0xA4, 0x1F, 0x7B, 0x5D, 0xCE, 0x24, 0xC5, 0x53); - struct nhlt_acpi_table *intel_nhlt_init(struct device *dev) { - acpi_handle handle; - union acpi_object *obj; - struct nhlt_resource_desc *nhlt_ptr; - struct nhlt_acpi_table *nhlt_table = NULL; - - handle = ACPI_HANDLE(dev); - if (!handle) { - dev_err(dev, "Didn't find ACPI_HANDLE\n"); - return NULL; - } + struct nhlt_acpi_table *nhlt; + acpi_status status; - obj = acpi_evaluate_dsm(handle, &osc_guid, 1, 1, NULL); - - if (!obj) - return NULL; - - if (obj->type != ACPI_TYPE_BUFFER) { - dev_dbg(dev, "No NHLT table found\n"); - ACPI_FREE(obj); + status = acpi_get_table(ACPI_SIG_NHLT, 0, + (struct acpi_table_header **)&nhlt); + if (ACPI_FAILURE(status)) { + dev_warn(dev, "NHLT table not found\n"); return NULL; } - nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - if (nhlt_ptr->length) - nhlt_table = (struct nhlt_acpi_table *) - memremap(nhlt_ptr->min_addr, nhlt_ptr->length, - MEMREMAP_WB); - ACPI_FREE(obj); - if (nhlt_table && - (strncmp(nhlt_table->header.signature, - NHLT_ACPI_HEADER_SIG, - strlen(NHLT_ACPI_HEADER_SIG)) != 0)) { - memunmap(nhlt_table); - dev_err(dev, "NHLT ACPI header signature incorrect\n"); - return NULL; - } - return nhlt_table; + return nhlt; } EXPORT_SYMBOL_GPL(intel_nhlt_init); void intel_nhlt_free(struct nhlt_acpi_table *nhlt) { - memunmap((void *)nhlt); + acpi_put_table((struct acpi_table_header *)nhlt); } EXPORT_SYMBOL_GPL(intel_nhlt_free); -- 2.17.1

3 7

[PATCH] ASoC: SOF: Add missing dependency on IMX_SCU
by Mark Brown 23 Apr '20

23 Apr '20

This broke PowerPC allyesconfig. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/imx/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5e0c68b12292..9a5ed1860e4e 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_SOF_IMX_OF config SND_SOC_SOF_IMX8_SUPPORT bool "SOF support for i.MX8" + depends on IMX_SCU help This adds support for Sound Open Firmware for NXP i.MX8 platforms Say Y if you have such a device. -- 2.20.1

3 5

[PATCH 0/4] ASoC:: don't use snd_soc_rtdcom_lookup()
by Kuninori Morimoto 23 Apr '20

23 Apr '20

Hi Mark Cc related engineer These patches are tring to not to use snd_soc_rtdcom_lookup() function on each drivers as much as possible, because we might have same name component under multi component situation. It can't find correct component in such case. I tried to add new feature on each drivers to not to use it, but I can't test. Thus, these patches should get Acked-by or Tested-by from each drivers user/maintenor. Please test these. After these patches, Intel / SOF drivers are still using snd_soc_rtdcom_lookup(). Because it is very complex, I couldn't try not to use it. If possible, each drivers should try to not use it, and it should be removed from ASoC. Kuninori Morimoto (4): ASoC: mediatek: don't use snd_soc_rtdcom_lookup() ASoC: intel: baytrail: don't use snd_soc_rtdcom_lookup() ASoC: intel: haswell: don't use snd_soc_rtdcom_lookup() ASoC: amd: don't use snd_soc_rtdcom_lookup() sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ------ sound/soc/intel/baytrail/sst-baytrail-pcm.c | 13 +++++++------ sound/soc/intel/haswell/sst-haswell-pcm.c | 4 +++- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 12 +++++------- sound/soc/mediatek/common/mtk-afe-fe-dai.h | 3 ++- sound/soc/mediatek/common/mtk-base-afe.h | 6 ++++-- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 6 ++++-- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 11 ++++------- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 7 ++++--- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 11 ++++------- 10 files changed, 37 insertions(+), 42 deletions(-) -- 2.17.1

3 10

[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

2 1

[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

2 1

[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

3 8

[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

3 5

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

1 0

New codec ALC245
by Kailang 23 Apr '20

23 Apr '20

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

2 1

[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

2 1

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.

3 5

[RFC PATCH 00/16] ASoC/SOF/clk/gpio/dt: add Hifiberry DAC+ PRO support
by Pierre-Louis Bossart 23 Apr '20

23 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

5 67

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

1 0

alsa-gobject: seq: Revert "seq: tstamp: use wrapper structure instead of union"
by GitHub pull_request - opened 22 Apr '20

22 Apr '20

alsa-project/alsa-gobject pull request #19 was opened from takaswie: This reverts commit 20fdc2d893ccba609cc786ebac733bf6914f9874. In the above commit, union type structure is wrapped with care of language bindings in which union-compatible feature is not expected. However, it's responsible for language bindings to parse gir and the care is not necessarily required in shared library side. In first place, ALSASeq.Tstamp is boxed type object. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-gobject/pull/19 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/19.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[GIT PULL] ASoC: tegra: Fixes for v5.7-rc3
by Thierry Reding 22 Apr '20

22 Apr '20

Hi Liam, Mark, The following changes since commit 8f3d9f354286745c751374f5f1fcafee6b3f3136: Linux 5.7-rc1 (2020-04-12 12:35:55 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/tegra/linux.git tags/tegra-for-5.7-asoc for you to fetch changes up to ff5d18cb04f4ecccbcf05b7f83ab6df2a0d95c16: ASoC: tegra: Enable audio mclk during tegra_asoc_utils_init() (2020-04-21 16:53:15 +0200) This is a set of patches that I was supposed to send out during the merge window[0], but I somehow messed up my scripts and then didn't notice that this was missing. I've rebased these patches on top of v5.7-rc1. The lack of these are now causing some issues with unbalanced clock reference counts. Is there any chance you can pick these up for v5.7-rc3? Thanks, Thierry [0]: https://lore.kernel.org/lkml/20200327154506.GF4437@sirena.org.uk/ ---------------------------------------------------------------- ASoC: tegra: Fixes for v5.7-rc3 This contains a couple of fixes that are needed to properly reconfigure the audio clocks on older Tegra devices. ---------------------------------------------------------------- Sowjanya Komatineni (4): ASoC: tegra: Use device managed resource APIs to get the clock ASoC: nau8825: Change Tegra clk_out_2 provider to PMC ASoC: tegra: Add audio mclk parent configuration ASoC: tegra: Enable audio mclk during tegra_asoc_utils_init() .../devicetree/bindings/sound/nau8825.txt | 2 +- sound/soc/tegra/tegra_alc5632.c | 7 +- sound/soc/tegra/tegra_asoc_utils.c | 113 ++++++++++----------- sound/soc/tegra/tegra_asoc_utils.h | 1 - sound/soc/tegra/tegra_max98090.c | 22 ++-- sound/soc/tegra/tegra_rt5640.c | 22 ++-- sound/soc/tegra/tegra_rt5677.c | 7 +- sound/soc/tegra/tegra_sgtl5000.c | 7 +- sound/soc/tegra/tegra_wm8753.c | 22 ++-- sound/soc/tegra/tegra_wm8903.c | 22 ++-- sound/soc/tegra/tegra_wm9712.c | 8 +- sound/soc/tegra/trimslice.c | 18 +--- 12 files changed, 90 insertions(+), 161 deletions(-)

2 1

[PATCH] ASoC: SOF: Fix build
by Mark Brown 22 Apr '20

22 Apr '20

The recent batch of SOF changes failed to build on some x86 configurations including an allmodconfig, revert the commits: e150ef4169a76 ASoC: SOF: Introduce extended manifest 371091417864b ASoC: SOF: ext_manifest: parse firmware version 7c024b948c206 ASoC: SOF: ext_manifest: parse compiler version 9e72f13ee541c ASoC: SOF: ext_manifest: parse windows to fix this. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/sof/ext_manifest.h | 91 ------------- sound/soc/sof/loader.c | 189 +------------------------- 2 files changed, 2 insertions(+), 278 deletions(-) delete mode 100644 include/uapi/sound/sof/ext_manifest.h diff --git a/include/uapi/sound/sof/ext_manifest.h b/include/uapi/sound/sof/ext_manifest.h deleted file mode 100644 index d49c47d08c7f..000000000000 --- a/include/uapi/sound/sof/ext_manifest.h +++ /dev/null @@ -1,91 +0,0 @@ -/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ -/* - * This file is provided under a dual BSD/GPLv2 license. When using or - * redistributing this file, you may do so under either license. - * - * Copyright(c) 2020 Intel Corporation. All rights reserved. - */ - -/* - * Extended manifest is a place to store metadata about firmware, known during - * compilation time - for example firmware version or used compiler. - * Given information are read on host side before firmware startup. - * This part of output binary is not signed. - */ - -#ifndef __SOF_FIRMWARE_EXT_MANIFEST_H__ -#define __SOF_FIRMWARE_EXT_MANIFEST_H__ - -#include <linux/const.h> -#include <sound/sof/info.h> - -/* In ASCII `XMan` */ -#define SOF_EXT_MAN_MAGIC_NUMBER 0x6e614d58 - -/* Build u32 number in format MMmmmppp */ -#define SOF_EXT_MAN_BUILD_VERSION(MAJOR, MINOR, PATH) ((uint32_t)( \ - ((MAJOR) << 24) | \ - ((MINOR) << 12) | \ - (PATH))) - -/* check extended manifest version consistency */ -#define SOF_EXT_MAN_VERSION_INCOMPATIBLE(host_ver, cli_ver) ( \ - ((host_ver) & GENMASK(31, 24)) != \ - ((cli_ver) & GENMASK(31, 24))) - -/* used extended manifest header version */ -#define SOF_EXT_MAN_VERSION SOF_EXT_MAN_BUILD_VERSION(1, 0, 0) - -/* extended manifest header, deleting any field breaks backward compatibility */ -struct sof_ext_man_header { - uint32_t magic; /*< identification number, */ - /*< EXT_MAN_MAGIC_NUMBER */ - uint32_t full_size; /*< [bytes] full size of ext_man, */ - /*< (header + content + padding) */ - uint32_t header_size; /*< [bytes] makes header extensionable, */ - /*< after append new field to ext_man header */ - /*< then backward compatible won't be lost */ - uint32_t header_version; /*< value of EXT_MAN_VERSION */ - /*< not related with following content */ - uint8_t elements[]; /*< list of ext_man_elem_* elements */ -} __packed; - -/* Now define extended manifest elements */ - -/* Extended manifest elements types */ -enum sof_ext_man_elem_type { - SOF_EXT_MAN_ELEM_FW_VERSION = 0, - SOF_EXT_MAN_ELEM_WINDOW = SOF_IPC_EXT_WINDOW, - SOF_EXT_MAN_ELEM_CC_VERSION = SOF_IPC_EXT_CC_INFO, -}; - -/* extended manifest element header */ -struct sof_ext_man_elem_header { - uint32_t type; /*< SOF_EXT_MAN_ELEM_ */ - uint32_t size; /*< in bytes, including header size */ - uint8_t blob[]; /*< type dependent content */ -} __packed; - -/* FW version */ -struct sof_ext_man_fw_version { - struct sof_ext_man_elem_header hdr; - /* use sof_ipc struct because of code re-use */ - struct sof_ipc_fw_version version; - uint32_t flags; -} __packed; - -/* extended data memory windows for IPC, trace and debug */ -struct sof_ext_man_window { - struct sof_ext_man_elem_header hdr; - /* use sof_ipc struct because of code re-use */ - struct sof_ipc_window ipc_window; -} __packed; - -/* Used C compiler description */ -struct sof_ext_man_cc_version { - struct sof_ext_man_elem_header hdr; - /* use sof_ipc struct because of code re-use */ - struct sof_ipc_cc_version cc_version; -} __packed; - -#endif /* __SOF_FIRMWARE_EXT_MANIFEST_H__ */ diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 8be30cd5e038..89f35db2577d 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -12,7 +12,6 @@ #include <linux/firmware.h> #include <sound/sof.h> -#include <uapi/sound/sof/ext_manifest.h> #include "ops.h" static int get_ext_windows(struct snd_sof_dev *sdev, @@ -20,21 +19,13 @@ static int get_ext_windows(struct snd_sof_dev *sdev, { const struct sof_ipc_window *w = container_of(ext_hdr, struct sof_ipc_window, ext_hdr); - size_t w_size = struct_size(w, window, w->num_windows); - - if (sdev->info_window) { - if (memcmp(sdev->info_window, w, w_size)) { - dev_err(sdev->dev, "error: mistmatch between window descriptor from extended manifest and mailbox"); - return -EINVAL; - } - return 0; - } if (w->num_windows == 0 || w->num_windows > SOF_IPC_MAX_ELEMS) return -EINVAL; /* keep a local copy of the data */ - sdev->info_window = kmemdup(w, w_size, GFP_KERNEL); + sdev->info_window = kmemdup(w, struct_size(w, window, w->num_windows), + GFP_KERNEL); if (!sdev->info_window) return -ENOMEM; @@ -49,14 +40,6 @@ static int get_cc_info(struct snd_sof_dev *sdev, const struct sof_ipc_cc_version *cc = container_of(ext_hdr, struct sof_ipc_cc_version, ext_hdr); - if (sdev->cc_version) { - if (memcmp(sdev->cc_version, cc, cc->ext_hdr.hdr.size)) { - dev_err(sdev->dev, "error: receive diverged cc_version descriptions"); - return -EINVAL; - } - return 0; - } - dev_dbg(sdev->dev, "Firmware info: used compiler %s %d:%d:%d%s used optimization flags %s\n", cc->name, cc->major, cc->minor, cc->micro, cc->desc, cc->optim); @@ -143,151 +126,6 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) } EXPORT_SYMBOL(snd_sof_fw_parse_ext_data); -static int ext_man_get_fw_version(struct snd_sof_dev *sdev, - const struct sof_ext_man_elem_header *hdr) -{ - const struct sof_ext_man_fw_version *v; - - v = container_of(hdr, struct sof_ext_man_fw_version, hdr); - - memcpy(&sdev->fw_ready.version, &v->version, sizeof(v->version)); - sdev->fw_ready.flags = v->flags; - - /* log ABI versions and check FW compatibility */ - return snd_sof_ipc_valid(sdev); -} - -static int ext_man_get_windows(struct snd_sof_dev *sdev, - const struct sof_ext_man_elem_header *hdr) -{ - const struct sof_ipc_ext_data_hdr *w_ipc; - const struct sof_ext_man_window *w; - - w = container_of(hdr, struct sof_ext_man_window, hdr); - w_ipc = (const struct sof_ipc_ext_data_hdr *)&w->ipc_window; - - return get_ext_windows(sdev, w_ipc); -} - -static int ext_man_get_cc_info(struct snd_sof_dev *sdev, - const struct sof_ext_man_elem_header *hdr) -{ - const struct sof_ext_man_cc_version *cc; - const struct sof_ipc_ext_data_hdr *cc_version; - - cc = container_of(hdr, struct sof_ext_man_cc_version, hdr); - cc_version = (const struct sof_ipc_ext_data_hdr *)&cc->cc_version; - - return get_cc_info(sdev, cc_version); -} - -static ssize_t snd_sof_ext_man_size(const struct firmware *fw) -{ - const struct sof_ext_man_header *head = (void *)fw->data; - - /* - * assert fw size is big enough to contain extended manifest header, - * it prevents from reading unallocated memory from `head` in following - * step. - */ - if (fw->size < sizeof(*head)) - return -EINVAL; - - /* - * When fw points to extended manifest, - * then first u32 must be equal SOF_EXT_MAN_MAGIC_NUMBER. - */ - if (head->magic == SOF_EXT_MAN_MAGIC_NUMBER) - return head->full_size; - - /* otherwise given fw don't have an extended manifest */ - return 0; -} - -/* parse extended FW manifest data structures */ -static int snd_sof_fw_ext_man_parse(struct snd_sof_dev *sdev, - const struct firmware *fw) -{ - const struct sof_ext_man_elem_header *elem_hdr; - const struct sof_ext_man_header *head; - ssize_t ext_man_size; - ssize_t remaining; - uintptr_t iptr; - int ret = 0; - - head = (struct sof_ext_man_header *)fw->data; - remaining = head->full_size - head->header_size; - ext_man_size = snd_sof_ext_man_size(fw); - - /* Assert firmware starts with extended manifest */ - if (ext_man_size < 0) { - dev_err(sdev->dev, "error: exception while reading firmware extended manifest, code %d\n", - (int)ext_man_size); - return ext_man_size; - } else if (!ext_man_size) { - dev_err(sdev->dev, "error: can't parse extended manifest when it's not present\n"); - return -EINVAL; - } - - /* incompatible version */ - if (SOF_EXT_MAN_VERSION_INCOMPATIBLE(SOF_EXT_MAN_VERSION, - head->header_version)) { - dev_err(sdev->dev, "error: extended manifest version 0x%X differ from used 0x%X\n", - head->header_version, SOF_EXT_MAN_VERSION); - return -EINVAL; - } - - /* get first extended manifest element header */ - iptr = (uintptr_t)fw->data + head->header_size; - - while (remaining > sizeof(*elem_hdr)) { - elem_hdr = (struct sof_ext_man_elem_header *)iptr; - - dev_dbg(sdev->dev, "found sof_ext_man header type %d size 0x%X\n", - elem_hdr->type, elem_hdr->size); - - if (elem_hdr->size < sizeof(*elem_hdr) || - elem_hdr->size > remaining) { - dev_err(sdev->dev, "error: invalid sof_ext_man header size, type %d size 0x%X\n", - elem_hdr->type, elem_hdr->size); - break; - } - - /* process structure data */ - switch (elem_hdr->type) { - case SOF_EXT_MAN_ELEM_FW_VERSION: - ret = ext_man_get_fw_version(sdev, elem_hdr); - break; - case SOF_EXT_MAN_ELEM_WINDOW: - ret = ext_man_get_windows(sdev, elem_hdr); - break; - case SOF_EXT_MAN_ELEM_CC_VERSION: - ret = ext_man_get_cc_info(sdev, elem_hdr); - break; - default: - dev_warn(sdev->dev, "warning: unknown sof_ext_man header type %d size 0x%X\n", - elem_hdr->type, elem_hdr->size); - break; - } - - if (ret < 0) { - dev_err(sdev->dev, "error: failed to parse sof_ext_man header type %d size 0x%X\n", - elem_hdr->type, elem_hdr->size); - break; - } - - remaining -= elem_hdr->size; - iptr += elem_hdr->size; - } - - if (remaining) { - dev_err(sdev->dev, "error: sof_ext_man header is inconsistent\n"); - ret = -EINVAL; - } - - return ret; -} - /* * IPC Firmware ready. */ @@ -635,7 +473,6 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) { struct snd_sof_pdata *plat_data = sdev->pdata; const char *fw_filename; - ssize_t ext_man_size; int ret; /* Don't request firmware again if firmware is already requested */ @@ -653,33 +490,11 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) if (ret < 0) { dev_err(sdev->dev, "error: request firmware %s failed err: %d\n", fw_filename, ret); - goto err; } else { dev_dbg(sdev->dev, "request_firmware %s successful\n", fw_filename); } - /* check for extended manifest */ - ext_man_size = snd_sof_ext_man_size(plat_data->fw); - if (ext_man_size > 0) { - ret = snd_sof_fw_ext_man_parse(sdev, plat_data->fw); - - /* when no error occurred, drop extended manifest */ - if (!ret) - plat_data->fw_offset = ext_man_size; - else - dev_err(sdev->dev, "error: firmware %s contains unsupported or invalid extended manifest: %d\n", - fw_filename, ret); - } else if (!ext_man_size) { - /* No extended manifest, so nothing to skip during FW load */ - dev_dbg(sdev->dev, "firmware doesn't contain extended manifest\n"); - } else { - ret = ext_man_size; - dev_err(sdev->dev, "error: firmware %s contains unsupported or invalid extended manifest: %d\n", - fw_filename, ret); - } - -err: kfree(fw_filename); return ret; -- 2.20.1

3 8

[PATCH] ASoC: soc-core: return true, false in snd_soc_volsw_is_stereo()
by Jason Yan 22 Apr '20

22 Apr '20

Fix the following coccicheck warning: include/sound/soc.h:1271:9-10: WARNING: return of 0/1 in function 'snd_soc_volsw_is_stereo' with return type bool Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- include/sound/soc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 946f88a6c63d..d1e9a526363e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1268,13 +1268,13 @@ static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card) static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) { if (mc->reg == mc->rreg && mc->shift == mc->rshift) - return 0; + return false; /* * mc->reg == mc->rreg && mc->shift != mc->rshift, or * mc->reg != mc->rreg means that the control is * stereo (bits in one register or in two registers) */ - return 1; + return true; } static inline unsigned int snd_soc_enum_val_to_item(struct soc_enum *e, -- 2.21.1

2 1

[PATCH] ASoC: dmic: Allow GPIO operations to sleep
by Mark Brown 22 Apr '20

22 Apr '20

If there is a power GPIO provided we control it from DAPM context so there is no problem with a sleeping GPIO, use the _cansleep() version of the API to allow this. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/dmic.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index f5560a49b9e5..5d079d90fd3b 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -59,14 +59,14 @@ static int dmic_aif_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: if (dmic->gpio_en) - gpiod_set_value(dmic->gpio_en, 1); + gpiod_set_value_cansleep(dmic->gpio_en, 1); if (dmic->wakeup_delay) msleep(dmic->wakeup_delay); break; case SND_SOC_DAPM_POST_PMD: if (dmic->gpio_en) - gpiod_set_value(dmic->gpio_en, 0); + gpiod_set_value_cansleep(dmic->gpio_en, 0); break; } -- 2.20.1

2 4

[PATCH 0/4] Add support for different DMIC configurations
by Mateusz Gorski 22 Apr '20

22 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 (4): ASoC: Intel: Skylake: Change the order of machine device and platform registration 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 include/uapi/sound/skl-tplg-interface.h | 2 + sound/soc/intel/skylake/skl-topology.c | 189 +++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 1 + sound/soc/intel/skylake/skl.c | 10 +- 4 files changed, 192 insertions(+), 10 deletions(-) -- 2.17.1

3 8

[PATCH] ASoC: SOF: ensure all fields in header are zero'd before copying back to userspace
by Colin King 22 Apr '20

22 Apr '20

From: Colin Ian King <colin.king(a)canonical.com> Field header.tlv is uninitialized and being copied back to userspace and hence leaking data from the stack to userspace. Fix this by ensuring the header structure is zero'd. Fixes: c3078f539704 ("ASoC: SOF: Add Sound Open Firmware KControl support") Addresses-Coverity: ("Uninitialized scalar variable") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sof/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index dfc412e2d956..97d5c1a4c1ff 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -362,7 +362,7 @@ int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, struct snd_sof_control *scontrol = be->dobj.private; struct snd_soc_component *scomp = scontrol->scomp; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; - struct snd_ctl_tlv header; + struct snd_ctl_tlv header = { }; struct snd_ctl_tlv __user *tlvd = (struct snd_ctl_tlv __user *)binary_data; int data_size; -- 2.25.1

3 2

[PATCH] ALSA: usb-audio: Add connector notifier delegation
by Takashi Iwai 22 Apr '20

22 Apr '20

It turned out that ALC1220-VB USB-audio device gives the interrupt event to some PCM terminals while those don't allow the connector state request but only the actual I/O terminals return the request. The recent commit 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals") excluded those phantom terminals, so those events are ignored, too. My first thought was that this could be easily deduced from the associated terminals, but some of them have even no associate terminal ID, hence it's not too trivial to figure out. Since the number of such terminals are small and limited, this patch implements another quirk table for the simple mapping of the connectors. It's not really scalable, but let's hope that there will be not many such funky devices in future. Fixes: 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 25 +++++++++++++++++++++++++ sound/usb/mixer.h | 10 ++++++++++ sound/usb/mixer_maps.c | 13 +++++++++++++ 3 files changed, 48 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ecd5036a0b44..a88d7854513b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3117,6 +3117,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (map->id == state.chip->usb_id) { state.map = map->map; state.selector_map = map->selector_map; + mixer->connector_map = map->connector_map; mixer->ignore_ctl_error |= map->ignore_ctl_error; break; } @@ -3198,10 +3199,32 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) return 0; } +static int delegate_notify(struct usb_mixer_interface *mixer, int unitid, + u8 *control, u8 *channel) +{ + const struct usbmix_connector_map *map = mixer->connector_map; + + if (!map) + return unitid; + + for (; map->id; map++) { + if (map->id == unitid) { + if (control && map->control) + *control = map->control; + if (channel && map->channel) + *channel = map->channel; + return map->delegated_id; + } + } + return unitid; +} + void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; + unitid = delegate_notify(mixer, unitid, NULL, NULL); + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info = mixer_elem_list_to_info(list); @@ -3271,6 +3294,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } + unitid = delegate_notify(mixer, unitid, &control, &channel); + for_each_mixer_elem(list, mixer, unitid) count++; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 65d6d08c96f5..41ec9dc4139b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -6,6 +6,13 @@ struct media_mixer_ctl; +struct usbmix_connector_map { + u8 id; + u8 delegated_id; + u8 control; + u8 channel; +}; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -18,6 +25,9 @@ struct usb_mixer_interface { /* the usb audio specification version this interface complies to */ int protocol; + /* optional connector delegation map */ + const struct usbmix_connector_map *connector_map; + /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; u32 rc_code; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index fb2c4a992951..0260c750e156 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -27,6 +27,7 @@ struct usbmix_ctl_map { u32 id; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; + const struct usbmix_connector_map *connector_map; int ignore_ctl_error; }; @@ -387,6 +388,15 @@ static const struct usbmix_name_map trx40_mobo_map[] = { {} }; +static const struct usbmix_connector_map trx40_mobo_connector_map[] = { + { 10, 16 }, /* (Back) Speaker */ + { 11, 17 }, /* Front Headphone */ + { 13, 7 }, /* Line */ + { 14, 8 }, /* Mic */ + { 15, 9 }, /* Front Mic */ + {} +}; + /* * Control map entries */ @@ -519,6 +529,7 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), @@ -531,10 +542,12 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { 0 } /* terminator */ }; -- 2.16.4

1 0

[PATCH] ALSA: oxygen: use true,false for bool variables
by Jason Yan 22 Apr '20

22 Apr '20

Fix the following coccicheck warning: sound/pci/oxygen/xonar_pcm179x.c:463:1-17: WARNING: Assignment of 0/1 to bool variable sound/pci/oxygen/xonar_pcm179x.c:505:1-17: WARNING: Assignment of 0/1 to bool variable Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/pci/oxygen/xonar_pcm179x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 6a0520c4fb5a..cf801a235df9 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -460,7 +460,7 @@ static void xonar_st_init(struct oxygen *chip) data->generic.anti_pop_delay = 100; data->h6 = chip->model.dac_channels_mixer > 2; - data->has_cs2000 = 1; + data->has_cs2000 = true; data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; data->broken_i2c = true; @@ -502,7 +502,7 @@ static void xonar_xense_init(struct oxygen *chip) xonar_init_ext_power(chip); data->generic.anti_pop_delay = 100; - data->has_cs2000 = 1; + data->has_cs2000 = true; data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, -- 2.21.1

2 1

[PATCH] ALSA: usb-audio: Fix a limit check in proc_dump_substream_formats()
by Dan Carpenter 22 Apr '20

22 Apr '20

This should be ARRAY_SIZE() instead of sizeof(). The sizeof() limit is too high so it doesn't work. Fixes: 093b8494f299 ("ALSA: usb-audio: Print more information in stream proc files") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/usb/proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 5a36e192ebb0..889c550c9f29 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -140,7 +140,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Channel map:"); for (c = 0; c < map->channels; c++) { - if (map->map[c] >= sizeof(channel_labels) || + if (map->map[c] >= ARRAY_SIZE(channel_labels) || !channel_labels[map->map[c]]) snd_iprintf(buffer, " --"); else -- 2.26.1

2 1

v2: Add support to SDM845 based platforms
by GitHub pull_request - opened 22 Apr '20

22 Apr '20

alsa-project/alsa-ucm-conf pull request #23 was opened from Srinivas-Kandagatla: Add Support to two codecs WCD934x and WSA881x and two platforms DragonBoard DB845c and Lenovo-YOGA-C630-13Q50 that use this codec. For ucm-validator to pass this PR depends on config files PR "v2: ucm-validator: add config files for DB845c and Lenovo-YOGA-C630-13Q50" (https://github.com/alsa-project/alsa-tests/pull/3) to alsa-tests Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/23 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/23.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-gobject: seq: some improvements and fixes
by GitHub pull_request - opened 22 Apr '20

22 Apr '20

alsa-project/alsa-gobject pull request #18 was opened from takaswie: In UAPI of ALSA Sequencer, some members in structures are defined as fixed-sized array. For accessor methods to the members, this patchset adds changes so that function annotations have 'not a variable length array type' in their prototype. In some cases, it's convenient to compare two instance of ALSASeq.Addr. This patchset adds a new API, 'ALSASeq.Addr.equal()' for this purpose. Finally, this patchset also adds some improvements and fixes. Request URL : https://github.com/alsa-project/alsa-gobject/pull/18 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/18.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

How to support multi FIFO address in ALSA & DMA Engine?
by Shengjiu Wang 22 Apr '20

22 Apr '20

Hi Experts Some cpu-dai devices support multi FIFO, so we need to tell DMA which fifos and how many fifos it should fill the data to. For example, when channels = 2, only one fifo is used. when channels = 4, then two fifos are used. or when channels = 2, two fifos are used, when channels= 4, four fifos are used. But in snd_dmaengine_dai_dma_data, and dma_slave_config, there is only one dma address. Do you have any idea to support multi FIFO? is there any implemented reference? Best regards Wang shengjiu

1 0

[GIT PULL] ASoC fix v5.7-rc2
by Mark Brown 21 Apr '20

21 Apr '20

The following changes since commit ccfc531695f3a4aada042f6bdb33ac6be24e1aec: ASoC: cs4270: pull reset GPIO low then high (2020-04-07 15:29:54 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.7-rc2 for you to fetch changes up to 1e060a453c8604311fb45ae2f84f67ed673329b4: ASoC: wm8960: Fix wrong clock after suspend & resume (2020-04-21 15:43:22 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.7 Quite a lot of fixes here, a lot of driver specific ones but the biggest one is the revert of changes to the startup and shutdown sequence for DAIs that went in during the merge window - they broke some older x86 platforms and attempts to fix them didn't succeed so it's safer to just roll them back and try to make sure those platforms are handled properly in any future attempt. The rockchip S/PDIF DT stuff was IIRC for validation issues. ---------------------------------------------------------------- Akshu Agrawal (1): ASoC: amd: Fix button configuration Amadeusz Sławiński (8): ASoC: topology: Add missing memory checks ASoC: topology: Check return value of soc_tplg_create_tlv ASoC: topology: Check return value of soc_tplg_*_create ASoC: topology: Check soc_tplg_add_route return value ASoC: topology: Check return value of pcm_new_ver ASoC: topology: Check return value of soc_tplg_dai_config ASoC: topology: Fix endianness issue ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry Bjorn Andersson (1): ASoC: qcom: common: Silence duplicate parse error messages Charles Keepax (4): ASoC: dapm: Fix regression introducing multiple copies of DAI widgets ASoC: dapm: Move error message to avoid some duplication ASoC: dapm: Remove dapm_connect_dai_link_widgets helper ASoC: madera: Remove a couple of stray blank lines Gyeongtaek Lee (1): ASoC: dapm: fixup dapm kcontrol widget Jason Yan (2): ASoC: intel: soc-acpi-intel-icl-match: remove useless 'rt1308_2_adr' ASoC: Intel: soc-acpi-intel-cml-match: remove useless 'rt1308_2_adr' Jerome Brunet (2): ASoC: meson: axg-card: fix codec-to-codec link setup ASoC: meson: gx-card: fix codec-to-codec link setup Johan Jonker (4): ASoC: convert rockchip spdif bindings to yaml ASoC: rockchip-spdif: add #sound-dai-cells property ASoC: rockchip-spdif: add power-domains property ASoC: rockchip-i2s: add power-domains property Krzysztof Kozlowski (1): ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume Mark Brown (3): Merge series "ASoC: rsnd: Fixes for multichannel HDMI audio output" from Matthias Blankertz <matthias.blankertz(a)cetitec.com>: Merge series "ASoC: meson: fix codec-to-codec link setup" from Jerome Brunet <jbrunet(a)baylibre.com>: Merge series "ASoC: rsnd: multi-SSI setup fixes" from Matthias Blankertz <matthias.blankertz(a)cetitec.com>: Matthias Blankertz (4): ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent ASoC: rsnd: Fix "status check failed" spam for multi-SSI Olivier Moysan (1): ASoC: stm32: sai: fix sai probe Philipp Puschmann (1): ASoC: tas571x: disable regulators on failed probe Pierre-Louis Bossart (2): ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell Sebastian Fricke (1): soc/stm/stm32_sub_sai: Add missing '\n' in log messages Sebastian Reichel (1): ASoC: sgtl5000: Fix VAG power-on handling Shengjiu Wang (1): ASoC: wm8960: Fix wrong clock after suspend & resume Srinivas Kandagatla (1): ASoC: wsa881x: mark read_only_wordlength flag Stephan Gerhold (2): ASoC: soc-pcm: dpcm: Only allow playback/capture if supported ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs YueHaibing (1): ASoC: wm89xx: Add missing dependency .../devicetree/bindings/sound/rockchip-i2s.yaml | 3 + .../devicetree/bindings/sound/rockchip-spdif.txt | 45 ------- .../devicetree/bindings/sound/rockchip-spdif.yaml | 101 ++++++++++++++ include/sound/soc-dai.h | 1 - include/sound/soc.h | 3 + sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/codecs/madera.c | 4 - sound/soc/codecs/sgtl5000.c | 34 +++++ sound/soc/codecs/sgtl5000.h | 1 + sound/soc/codecs/tas571x.c | 20 ++- sound/soc/codecs/wm8960.c | 3 +- sound/soc/codecs/wsa881x.c | 4 + sound/soc/intel/common/soc-acpi-intel-cml-match.c | 8 -- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 8 -- sound/soc/meson/axg-card.c | 4 +- sound/soc/meson/gx-card.c | 4 +- sound/soc/qcom/apq8096.c | 4 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 +++ sound/soc/qcom/sdm845.c | 4 +- sound/soc/samsung/s3c-i2s-v2.c | 57 -------- sound/soc/samsung/s3c2412-i2s.c | 56 ++++++++ sound/soc/sh/rcar/ssi.c | 11 +- sound/soc/sh/rcar/ssiu.c | 2 +- sound/soc/soc-dai.c | 11 +- sound/soc/soc-dapm.c | 147 +++++++++++---------- sound/soc/soc-pcm.c | 13 +- sound/soc/soc-topology.c | 115 ++++++++++++---- sound/soc/sof/intel/bdw.c | 16 +++ sound/soc/sof/intel/byt.c | 48 +++++++ sound/soc/stm/stm32_sai_sub.c | 14 +- 32 files changed, 509 insertions(+), 263 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.yaml

2 1

[PATCH v2] ASoC: intel/skl/hda - fix oops on systems without i915 audio codec
by Kai Vehmanen 21 Apr '20

21 Apr '20

Recent fix for jack detection caused a regression on systems with HDA audio codec but no HDMI/DP audio via i915 graphics, leading to a kernel oops at device probe. On these systems, HDA bus instance lookup fails, as the first ASoC runtime of the card is connected to a dummy codec (as no HDMI codec is present). Fixes: 3a24f135e6cc ("ASoC: intel/skl/hda - set autosuspend timeout for hda codecs") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 34 +++++++++++--------- 1 file changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 35f222ed9912..fc9290fb4e99 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -172,23 +172,25 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = - list_first_entry(&card->rtd_list, - struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_pcm_runtime *rtd; struct hdac_hda_priv *hda_pvt; - - if (!codec_dai) - return; - - /* - * all codecs are on the same bus, so it's sufficient - * to lookup the first runtime and its codec, and set - * power save defaults for all codecs on the bus - */ - hda_pvt = snd_soc_component_get_drvdata(codec_dai->component); - snd_hda_set_power_save(hda_pvt->codec.bus, - HDA_CODEC_AUTOSUSPEND_DELAY_MS); + struct snd_soc_dai *dai; + + for_each_card_rtds(card, rtd) { + if (!strstr(rtd->dai_link->codecs->name, "ehdaudio")) + continue; + dai = asoc_rtd_to_codec(rtd, 0); + hda_pvt = snd_soc_component_get_drvdata(dai->component); + if (hda_pvt) { + /* + * all codecs are on the same bus, so it's sufficient + * to look up only the first one + */ + snd_hda_set_power_save(hda_pvt->codec.bus, + HDA_CODEC_AUTOSUSPEND_DELAY_MS); + break; + } + } } static int skl_hda_audio_probe(struct platform_device *pdev) -- 2.26.0

3 4

[PATCH] ASoC: intel/skl/hda - fix oops on systems without i915 audio codec
by Kai Vehmanen 21 Apr '20

21 Apr '20

Recent fix for jack detection caused a regression on systems with HDA audio codec but no HDMI/DP audio via i915 graphics, leading to a kernel oops at device probe. On these systems, HDA bus instance lookup fails, as the first ASoC runtime of the card is connected to a dummy codec (as no HDMI codec is present). Fixes: 3a24f135e6cc ("ASoC: intel/skl/hda - set autosuspend timeout for hda codecs") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 34 +++++++++++--------- 1 file changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 35f222ed9912..fc9290fb4e99 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -172,23 +172,25 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = - list_first_entry(&card->rtd_list, - struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_pcm_runtime *rtd; struct hdac_hda_priv *hda_pvt; - - if (!codec_dai) - return; - - /* - * all codecs are on the same bus, so it's sufficient - * to lookup the first runtime and its codec, and set - * power save defaults for all codecs on the bus - */ - hda_pvt = snd_soc_component_get_drvdata(codec_dai->component); - snd_hda_set_power_save(hda_pvt->codec.bus, - HDA_CODEC_AUTOSUSPEND_DELAY_MS); + struct snd_soc_dai *dai; + + for_each_card_rtds(card, rtd) { + if (!strstr(rtd->dai_link->codecs->name, "ehdaudio")) + continue; + dai = asoc_rtd_to_codec(rtd, 0); + hda_pvt = snd_soc_component_get_drvdata(dai->component); + if (hda_pvt) { + /* + * all codecs are on the same bus, so it's sufficient + * to look up only the first one + */ + snd_hda_set_power_save(hda_pvt->codec.bus, + HDA_CODEC_AUTOSUSPEND_DELAY_MS); + break; + } + } } static int skl_hda_audio_probe(struct platform_device *pdev) -- 2.26.0

2 2

[PATCH] ASoC: txx9: add back the hack for a too small resource_size_t
by Christoph Hellwig 21 Apr '20

21 Apr '20

Looks like I misread the Kconfig magic and this driver can be compiled into 32-bit kernels. Add back the hack to extent the range of the resource_size_t, and include the header with the txx9-specific ioremap magic for that. Fixes: acfaaf52ebfd ("ASoC: txx9: don't work around too small resource_size_t") Signed-off-by: Christoph Hellwig <hch(a)lst.de> --- sound/soc/txx9/txx9aclc-ac97.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 7402448bdb09..d9e348444bd0 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -14,6 +14,7 @@ #include <linux/interrupt.h> #include <linux/io.h> #include <linux/gfp.h> +#include <asm/mach-tx39xx/ioremap.h> /* for TXX9_DIRECTMAP_BASE */ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -175,8 +176,6 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) int err; int irq; - BUILD_BUG_ON(sizeof(drvdata->physbase) > sizeof(r->start)); - irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; @@ -192,6 +191,10 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) platform_set_drvdata(pdev, drvdata); drvdata->physbase = r->start; + if (sizeof(drvdata->physbase) > sizeof(r->start) && + r->start >= TXX9_DIRECTMAP_BASE && + r->start < TXX9_DIRECTMAP_BASE + 0x400000) + drvdata->physbase |= 0xf00000000ull; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, 0, dev_name(&pdev->dev), drvdata); if (err < 0) -- 2.26.1

3 2

[PATCH v2 0/8] ASoC: add snd_compress_ops and replace
by Kuninori Morimoto 21 Apr '20

21 Apr '20

Hi Mark Current snd_soc_component_driver has compr_ops, but it is created for ALSA, not for ALSA SoC. This means it doesn't know the callback is for which component, thus, each callback needs to get component via lookup function. It is not good design, and will not work in the future, because ASoC can have multipul same name component which current lookup function can't handle. This v2 patch-set adds new snd_compress_ops and call callbacks with component. v1 is here Link: https://lore.kernel.org/alsa-devel/87blnqpuqp.wl-kuninori.morimoto.gx@renes… Kuninori Morimoto (8): ASoC: soc-compress: add snd_compress_ops ASoC: codec: wm_adsp: use snd_compress_ops ASoC: uniphier: use snd_compress_ops ASoC: qcom: q6sp6: use snd_compress_ops ASoC: intel: atom: use snd_compress_ops ASoC: sof: use snd_compress_ops ASoC: sprd: use snd_compress_ops ASoC: soc-compress: remove snd_compr_ops include/sound/soc-component.h | 40 ++++++++- sound/soc/codecs/cs47l15.c | 9 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 9 +- sound/soc/codecs/cs47l85.c | 9 +- sound/soc/codecs/cs47l90.c | 9 +- sound/soc/codecs/cs47l92.c | 9 +- sound/soc/codecs/wm5102.c | 9 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm_adsp.c | 18 ++-- sound/soc/codecs/wm_adsp.h | 18 ++-- .../intel/atom/sst-mfld-platform-compress.c | 43 +++++---- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 51 ++++++----- sound/soc/soc-compress.c | 89 ++++++++++--------- sound/soc/sof/compress.c | 7 +- sound/soc/sof/compress.h | 7 +- sound/soc/sof/pcm.c | 4 +- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 49 +++++----- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sprd/sprd-pcm-dma.h | 2 +- sound/soc/uniphier/aio-compress.c | 45 ++++++---- sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/uniphier/aio.h | 2 +- 26 files changed, 265 insertions(+), 190 deletions(-) -- 2.17.1

4 13

[PATCH v9 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 21 Apr '20

21 Apr '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v9 - use lowercase for dt example. changes in v8 - change get_pair_priv_size to pair_priv_size variable changes in v7 - updated according to Nicoin's comments. - add get_pair_priv_size to replace PAIR_PRIVATE_SIZE changes in v6 - updated according to Nicoin's and Rob's comments. changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 24 +- sound/soc/fsl/fsl_asrc.c | 305 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 106 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2114 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3212 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

2 8

[PATCH v2 1/2] dt-bindings: sound: add Microsemi ZL38060 binding
by Sven Van Asbroeck 21 Apr '20

21 Apr '20

Add devicetree binding for the Microsemi ZL38060 Connected Home Audio Processor. Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- Tree: v5.7-rc1 v1->v2: - removed all properties related to amp-en-gpios - removed 'mscc,load-firmware' property For further details, see v1->v2 notes in 2nd patch. .../devicetree/bindings/sound/zl38060.yaml | 69 +++++++++++++++++++ 1 file changed, 69 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zl38060.yaml diff --git a/Documentation/devicetree/bindings/sound/zl38060.yaml b/Documentation/devicetree/bindings/sound/zl38060.yaml new file mode 100644 index 000000000000..338e2a13c775 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zl38060.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/zl38060.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ZL38060 Connected Home Audio Processor from Microsemi. + +description: | + The ZL38060 is a "Connected Home Audio Processor" from Microsemi, + which consists of a Digital Signal Processor (DSP), several Digital + Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. + +maintainers: + - Jaroslav Kysela <perex(a)perex.cz> + - Takashi Iwai <tiwai(a)suse.com> + +properties: + compatible: + const: mscc,zl38060 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 24000000 + + reset-gpios: + description: + A GPIO line handling reset of the chip. As the line is active low, + it should be marked GPIO_ACTIVE_LOW (see ../gpio/gpio.txt) + maxItems: 1 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi0 { + #address-cells = <1>; + #size-cells = <0>; + + codec: zl38060@0 { + gpio-controller; + #gpio-cells = <2>; + #sound-dai-cells = <0>; + compatible = "mscc,zl38060"; + reg = <0>; + spi-max-frequency = <12000000>; + reset-gpios = <&gpio1 0 GPIO_ACTIVE_LOW>; + }; + }; -- 2.17.1

2 7

[PATCH v8 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 21 Apr '20

21 Apr '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v8 - change get_pair_priv_size to pair_priv_size variable changes in v7 - updated according to Nicoin's comments. - add get_pair_priv_size to replace PAIR_PRIVATE_SIZE changes in v6 - updated according to Nicoin's and Rob's comments. changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 24 +- sound/soc/fsl/fsl_asrc.c | 305 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 106 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2114 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3212 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

3 9

[PATCH] ASoC: wm8960: Fix wrong clock after suspend & resume
by Shengjiu Wang 21 Apr '20

21 Apr '20

After suspend & resume, wm8960_hw_params may be called when bias_level is not SND_SOC_BIAS_ON, then wm8960_configure_clocking is not called. But if sample rate is changed at that time, then the output clock rate will be not correct. So judgement of bias_level is SND_SOC_BIAS_ON in wm8960_hw_params is not necessary and it causes above issue. Fixes: 3176bf2d7ccd ("ASoC: wm8960: update pll and clock setting function") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 55112c1bba5e..6cf0f6612bda 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -860,8 +860,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, wm8960->is_stream_in_use[tx] = true; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON && - !wm8960->is_stream_in_use[!tx]) + if (!wm8960->is_stream_in_use[!tx]) return wm8960_configure_clocking(component); return 0; -- 2.21.0

3 2

[RFC PATCH v2 0/2] Introduce multi PM domains helpers
by Daniel Baluta 21 Apr '20

21 Apr '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Changes since v1: (addressed Ranjani's comments) - enhanced description for dev_multi_pm_attach return value - renamed exit_unroll_pm label to exit_detach_pm Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 60 ++++-------------------- 3 files changed, 121 insertions(+), 51 deletions(-) -- 2.17.1

4 6

[RFC PATCH 0/9] fix -Wempty-body build warnings
by Randy Dunlap 21 Apr '20

21 Apr '20

Hi, When -Wextra is used, gcc emits many warnings about an empty 'if' or 'else' body, like this: ../fs/posix_acl.c: In function ‘get_acl’: ../fs/posix_acl.c:127:22: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] /* fall through */ ; ^ To quieten these warnings, add a new macro "do_empty()". I originally wanted to use do_nothing(), but that's already in use. It would sorta be nice if "fallthrough" could be coerced for this instead of using something like do_empty(). Or should we just use "{}" in place of ";"? This causes some odd coding style issue IMO. E.g., see this change: original: if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) /* fall through */ ; with new macro: if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) do_empty(); /* fall through */ using {}: if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) {} /* fall through */ or { /* fall through */ } or even if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) { /* fall through */ } or if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) { } /* fall through */ drivers/base/devcoredump.c | 5 +++-- drivers/dax/bus.c | 5 +++-- drivers/input/mouse/synaptics.c | 3 ++- drivers/target/target_core_pscsi.c | 3 ++- drivers/usb/core/sysfs.c | 2 +- fs/nfsd/nfs4state.c | 3 ++- fs/posix_acl.c | 2 +- include/linux/kernel.h | 8 ++++++++ sound/drivers/vx/vx_core.c | 3 ++- 9 files changed, 24 insertions(+), 10 deletions(-)

14 38

[PATCH] ASoC: wm8962: restore the CLOCKING2 register in resume
by Shengjiu Wang 21 Apr '20

21 Apr '20

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); + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); @@ -3842,6 +3847,9 @@ static int wm8962_runtime_suspend(struct device *dev) WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, 0); + regmap_read(wm8962->regmap, WM8962_CLOCKING2, + &wm8962->regcache_clocking2); + regcache_cache_only(wm8962->regmap, true); regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), -- 2.21.0

2 1

[PATCH 1/2] ASoC: SOF: Fix "no reply expected" error during firmware-boot
by Hans de Goede 21 Apr '20

21 Apr '20

At least on Canon Lake each time the SOF firmware is booted, the following error is logged in dmesg: [ 36.711803] sof-audio-pci 0000:00:1f.3: error: no reply expected, received 0x0 Since the DSP is powered down when not in used this happens everytime e.g. a notification plays, polluting dmesg with these false-positive errors. Add a check to snd_sof_ipc_reply() which makes it return success when receiving an unexpected msg_id 0 during fw-boot, fixing these false-positive errors being logged. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/sof/ipc.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 78aa1da7c7a9..7303b3d42f12 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -312,6 +312,11 @@ int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) { struct snd_sof_ipc_msg *msg = &sdev->ipc->msg; + /* Ignore msg_id 0 being send during fw-boot */ + if (msg->ipc_complete && sdev->fw_state == SOF_FW_BOOT_IN_PROGRESS && + msg_id == 0) + return 0; + if (msg->ipc_complete) { dev_err(sdev->dev, "error: no reply expected, received 0x%x", msg_id); -- 2.26.0

4 11

Add support to SDM845 based platforms
by GitHub pull_request - opened 21 Apr '20

21 Apr '20

alsa-project/alsa-ucm-conf pull request #22 was opened from Srinivas-Kandagatla: Add Support to two codecs WCD934x and WSA881x and two platforms DragonBoard DB845c and Lenovo-YOGA-C630-13Q50 that use this codec. For ucm-validator to pass this PR depends on config files PR "ucm-validator: add config files for DB845c and Lenovo-YOGA-C630-13Q50" to alsa-tests Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/22 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/22.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: fsl_esai: Remove expensive print in irq handler
by Shengjiu Wang 21 Apr '20

21 Apr '20

Use dev_dbg instead of dev_err in irq handler, the dev_err is expensive, we don't need the message to be printed everytime, which is almost a debug option. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7a49d03463a..453392a9cd68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -94,10 +94,10 @@ static irqreturn_t esai_isr(int irq, void *devid) dev_dbg(&pdev->dev, "isr: Transmission Initialized\n"); if (esr & ESAI_ESR_RFF_MASK) - dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + dev_dbg(&pdev->dev, "isr: Receiving overrun\n"); if (esr & ESAI_ESR_TFE_MASK) - dev_warn(&pdev->dev, "isr: Transmission underrun\n"); + dev_dbg(&pdev->dev, "isr: Transmission underrun\n"); if (esr & ESAI_ESR_TLS_MASK) dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); -- 2.21.0

3 2

[PATCH v2 0/7] Add H6 I2S support
by Clément Péron 21 Apr '20

21 Apr '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Changes since v1: - Fix missing header in set sign extend sample Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 292 ++++++++++++++++-- 3 files changed, 301 insertions(+), 24 deletions(-) -- 2.20.1

2 11

Think I found an ALSA bug
by Ólafur Víðir Guðbjargarson 21 Apr '20

21 Apr '20

Dear receiver I think I found an ALSA bug. Yesterday my computer, which is running Manjaro 19.0.2 GNOME edition froze. After rebooting I checked the system log and found a message there to report to ALSA developers. This is just few lines from the log: apr 20 22:45:20 HP pulseaudio[1458]: E: [alsa-sink-ALC295 Analog] alsa- util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. apr 20 22:45:20 HP pulseaudio[1458]: E: [alsa-sink-ALC295 Analog] alsa- util.c: snd_pcm_avail() returned a value that is exceptionally large: 2271932 bytes (12879 ms). I can't wrap this into code block here. I have attached the log as txt file. If you need further information, I will do my best to provide them. -- Kær kveðja/best regards Ólafur Víðir Guðbjargarson Tæknifræðingur/engineer technical GSM: 692 6292 olividir(a)gmail.com

1 0

[PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2
by Gregor Pintar 21 Apr '20

21 Apr '20

Force it to use asynchronous playback. Same quirk has already been added for Focusrite Scarlett Solo (2nd gen) with a commit 46f5710f0b88 ("ALSA: usb-audio: Add quirk for Focusrite Scarlett Solo"). This also seems to prevent regular clicks when playing at 44100Hz on Scarlett 2i2 (2nd gen). I did not notice any side effects. Signed-off-by: Gregor Pintar <grpintar(a)gmail.com> --- sound/usb/quirks-table.h | 84 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 84 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e009d584e..bc936bf79 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2840,6 +2840,90 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Focusrite Scarlett 2i2 2nd generation + * Reports that playback should use Synch: Synchronous + * while still providing a feedback endpoint. Synchronous causes + * snapping on some sample rates. + * Force it to use Synch: Asynchronous. + */ + USB_DEVICE(0x1235, 0x8202), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .protocol = UAC_VERSION_2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 44100, + .rate_max = 192000, + .nr_rates = 6, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, + 96000, 176400, 192000 + }, + .clock = 41 + } + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .protocol = UAC_VERSION_2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 44100, + .rate_max = 192000, + .nr_rates = 6, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, + 96000, 176400, 192000 + }, + .clock = 41 + } + }, + { + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { -- 2.20.1

3 4

[PATCH 1/2] dt-bindings: ASoC: qdsp6: Suggest more generic node names
by Stephan Gerhold 21 Apr '20

21 Apr '20

Change the listed examples to use more generic node names, representing the class of the device nodes: - apr-service@<id> - dai@<id> Both names are already in use in arch/arm64/boot/dts/qcom/sdm845.dtsi. Also add #address-cells + #size-cells to the q6asm example, without them the example produces dtc warnings. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- .../devicetree/bindings/sound/qcom,q6adm.txt | 2 +- .../devicetree/bindings/sound/qcom,q6afe.txt | 46 +++++++++---------- .../devicetree/bindings/sound/qcom,q6asm.txt | 7 ++- .../devicetree/bindings/sound/qcom,q6core.txt | 2 +- 4 files changed, 30 insertions(+), 27 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt index bbae426cdfb1..15c353a20de8 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt @@ -29,7 +29,7 @@ used by the apr service device. Definition: Must be 0 = EXAMPLE -q6adm@8 { +apr-service@8 { compatible = "qcom,q6adm"; reg = <APR_SVC_ADM>; q6routing: routing { diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index d74888b9f1bb..4916dd6a0896 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -100,7 +100,7 @@ configuration of each dai. Must contain the following properties. = EXAMPLE -q6afe@4 { +apr-service@4 { compatible = "qcom,q6afe"; reg = <APR_SVC_AFE>; @@ -110,12 +110,12 @@ q6afe@4 { #address-cells = <1>; #size-cells = <0>; - hdmi@1 { - reg = <1>; + dai@1 { + reg = <HDMI_RX>; }; - tdm@24 { - reg = <24>; + dai@24 { + reg = <PRIMARY_TDM_RX_0>; qcom,tdm-sync-mode = <1>: qcom,tdm-sync-src = <1>; qcom,tdm-data-out = <0>; @@ -125,8 +125,8 @@ q6afe@4 { }; - tdm@25 { - reg = <25>; + dai@25 { + reg = <PRIMARY_TDM_TX_0>; qcom,tdm-sync-mode = <1>: qcom,tdm-sync-src = <1>; qcom,tdm-data-out = <0>; @@ -135,43 +135,43 @@ q6afe@4 { qcom,tdm-data-align = <0>; }; - prim-mi2s-rx@16 { - reg = <16>; + dai@16 { + reg = <PRIMARY_MI2S_RX>; qcom,sd-lines = <0 2>; }; - prim-mi2s-tx@17 { - reg = <17>; + dai@17 { + reg = <PRIMARY_MI2S_TX>; qcom,sd-lines = <1>; }; - sec-mi2s-rx@18 { - reg = <18>; + dai@18 { + reg = <SECONDARY_MI2S_RX>; qcom,sd-lines = <0 3>; }; - sec-mi2s-tx@19 { - reg = <19>; + dai@19 { + reg = <SECONDARY_MI2S_TX>; qcom,sd-lines = <1>; }; - tert-mi2s-rx@20 { - reg = <20>; + dai@20 { + reg = <TERTIARY_MI2S_RX>; qcom,sd-lines = <1 3>; }; - tert-mi2s-tx@21 { - reg = <21>; + dai@21 { + reg = <TERTIARY_MI2S_TX>; qcom,sd-lines = <0>; }; - quat-mi2s-rx@22 { - reg = <22>; + dai@22 { + reg = <QUATERNARY_MI2S_RX>; qcom,sd-lines = <0>; }; - quat-mi2s-tx@23 { - reg = <23>; + dai@23 { + reg = <QUATERNARY_MI2S_TX>; qcom,sd-lines = <1>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 9f5378c51686..6b9a88d0ea3f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -51,13 +51,16 @@ configuration of each dai. Must contain the following properties. = EXAMPLE -q6asm@7 { +apr-service@7 { compatible = "qcom,q6asm"; reg = <APR_SVC_ASM>; q6asmdai: dais { compatible = "qcom,q6asm-dais"; + #address-cells = <1>; + #size-cells = <0>; #sound-dai-cells = <1>; - mm@0 { + + dai@0 { reg = <0>; direction = <2>; is-compress-dai; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt index 7f36ff8bec18..5cd4cc9b1fde 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6core.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6core.txt @@ -15,7 +15,7 @@ used by the apr service device. example "qcom,q6core-v2.0" = EXAMPLE -q6core@3 { +apr-service@3 { compatible = "qcom,q6core"; reg = <APR_SVC_ADSP_CORE>; }; -- 2.26.1

5 7

[PATCH 1/2] dt-bindings: Clean-up schema indentation formatting
by Rob Herring 20 Apr '20

20 Apr '20

Fix various inconsistencies in schema indentation. Most of these are list indentation which should be 2 spaces more than the start of the enclosing keyword. This doesn't matter functionally, but affects running scripts which do transforms on the schema files. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/arm/altera.yaml | 6 +- .../amlogic/amlogic,meson-gx-ao-secure.yaml | 2 +- .../devicetree/bindings/arm/bitmain.yaml | 2 +- .../devicetree/bindings/arm/nxp/lpc32xx.yaml | 9 ++- .../bindings/arm/socionext/uniphier.yaml | 26 ++++---- .../bindings/arm/stm32/st,mlahb.yaml | 2 +- .../bindings/arm/stm32/st,stm32-syscon.yaml | 6 +- .../bindings/ata/faraday,ftide010.yaml | 4 +- .../bindings/bus/allwinner,sun8i-a23-rsb.yaml | 4 +- .../clock/allwinner,sun4i-a10-gates-clk.yaml | 8 +-- .../devicetree/bindings/clock/fsl,plldig.yaml | 17 +++-- .../devicetree/bindings/clock/qcom,mmcc.yaml | 16 ++--- .../bindings/connector/usb-connector.yaml | 6 +- .../crypto/allwinner,sun4i-a10-crypto.yaml | 14 ++-- .../bindings/crypto/allwinner,sun8i-ce.yaml | 16 ++--- .../bindings/crypto/amlogic,gxl-crypto.yaml | 2 +- .../display/allwinner,sun4i-a10-hdmi.yaml | 40 ++++++------ .../display/allwinner,sun4i-a10-tcon.yaml | 58 ++++++++--------- .../display/allwinner,sun6i-a31-mipi-dsi.yaml | 28 ++++---- .../display/allwinner,sun8i-a83t-dw-hdmi.yaml | 10 +-- .../bindings/display/bridge/lvds-codec.yaml | 18 +++--- .../display/panel/sony,acx424akp.yaml | 2 +- .../display/panel/xinpeng,xpp055c272.yaml | 4 +- .../bindings/display/renesas,cmm.yaml | 16 ++--- .../devicetree/bindings/dma/ti/k3-udma.yaml | 8 +-- .../bindings/gpio/brcm,xgs-iproc-gpio.yaml | 2 +- .../bindings/gpu/arm,mali-midgard.yaml | 18 +++--- .../devicetree/bindings/gpu/vivante,gc.yaml | 2 +- .../devicetree/bindings/i2c/i2c-rk3x.yaml | 10 +-- .../bindings/iio/adc/adi,ad7124.yaml | 4 +- .../bindings/iio/adc/lltc,ltc2496.yaml | 6 +- .../input/allwinner,sun4i-a10-lradc-keys.yaml | 4 +- .../bindings/input/touchscreen/goodix.yaml | 2 +- .../bindings/interconnect/qcom,msm8916.yaml | 4 +- .../bindings/interconnect/qcom,msm8974.yaml | 4 +- .../bindings/interconnect/qcom,qcs404.yaml | 4 +- .../allwinner,sun7i-a20-sc-nmi.yaml | 12 ++-- .../intel,ixp4xx-interrupt.yaml | 8 +-- .../interrupt-controller/st,stm32-exti.yaml | 12 ++-- .../bindings/iommu/samsung,sysmmu.yaml | 10 +-- .../bindings/mailbox/st,stm32-ipcc.yaml | 2 +- .../media/allwinner,sun4i-a10-csi.yaml | 28 ++++---- .../bindings/media/amlogic,gx-vdec.yaml | 14 ++-- .../bindings/media/renesas,ceu.yaml | 28 ++++---- .../bindings/media/renesas,vin.yaml | 8 +-- .../devicetree/bindings/media/ti,vpe.yaml | 2 +- .../memory-controllers/fsl/imx8m-ddrc.yaml | 6 +- .../bindings/mfd/st,stm32-lptimer.yaml | 4 +- .../bindings/mfd/st,stm32-timers.yaml | 4 +- .../devicetree/bindings/mfd/syscon.yaml | 12 ++-- .../devicetree/bindings/mmc/cdns,sdhci.yaml | 2 +- .../bindings/mmc/rockchip-dw-mshc.yaml | 16 ++--- .../bindings/mmc/socionext,uniphier-sd.yaml | 14 ++-- .../devicetree/bindings/mtd/denali,nand.yaml | 4 +- .../net/allwinner,sun8i-a83t-emac.yaml | 4 +- .../bindings/net/can/bosch,m_can.yaml | 52 +++++++-------- .../bindings/net/renesas,ether.yaml | 4 +- .../bindings/net/ti,cpsw-switch.yaml | 12 ++-- .../bindings/net/ti,davinci-mdio.yaml | 27 ++++---- .../bindings/phy/intel,lgm-emmc-phy.yaml | 2 +- .../devicetree/bindings/pwm/pwm-samsung.yaml | 16 ++--- .../bindings/remoteproc/st,stm32-rproc.yaml | 2 +- .../reset/brcm,bcm7216-pcie-sata-rescal.yaml | 4 +- .../devicetree/bindings/rtc/st,stm32-rtc.yaml | 38 +++++------ .../bindings/serial/amlogic,meson-uart.yaml | 16 ++--- .../devicetree/bindings/serial/rs485.yaml | 17 ++--- .../bindings/soc/amlogic/amlogic,canvas.yaml | 10 +-- .../bindings/sound/renesas,fsi.yaml | 16 ++--- .../bindings/spi/qcom,spi-qcom-qspi.yaml | 10 +-- .../devicetree/bindings/spi/renesas,hspi.yaml | 4 +- .../devicetree/bindings/spi/spi-pl022.yaml | 2 +- .../bindings/spi/st,stm32-qspi.yaml | 4 +- .../allwinner,sun4i-a10-system-control.yaml | 64 +++++++++---------- .../bindings/thermal/amlogic,thermal.yaml | 10 +-- .../bindings/timer/arm,arch_timer.yaml | 4 +- .../bindings/timer/arm,arch_timer_mmio.yaml | 4 +- .../devicetree/bindings/usb/dwc2.yaml | 8 +-- 77 files changed, 450 insertions(+), 450 deletions(-) diff --git a/Documentation/devicetree/bindings/arm/altera.yaml b/Documentation/devicetree/bindings/arm/altera.yaml index 49e0362ddc11..b388c5aa7984 100644 --- a/Documentation/devicetree/bindings/arm/altera.yaml +++ b/Documentation/devicetree/bindings/arm/altera.yaml @@ -13,8 +13,8 @@ properties: compatible: items: - enum: - - altr,socfpga-cyclone5 - - altr,socfpga-arria5 - - altr,socfpga-arria10 + - altr,socfpga-cyclone5 + - altr,socfpga-arria5 + - altr,socfpga-arria10 - const: altr,socfpga ... diff --git a/Documentation/devicetree/bindings/arm/amlogic/amlogic,meson-gx-ao-secure.yaml b/Documentation/devicetree/bindings/arm/amlogic/amlogic,meson-gx-ao-secure.yaml index 66213bd95e6e..6cc74523ebfd 100644 --- a/Documentation/devicetree/bindings/arm/amlogic/amlogic,meson-gx-ao-secure.yaml +++ b/Documentation/devicetree/bindings/arm/amlogic/amlogic,meson-gx-ao-secure.yaml @@ -25,7 +25,7 @@ select: properties: compatible: - items: + items: - const: amlogic,meson-gx-ao-secure - const: syscon diff --git a/Documentation/devicetree/bindings/arm/bitmain.yaml b/Documentation/devicetree/bindings/arm/bitmain.yaml index 0efdb4ac028e..5cd5b36cff2d 100644 --- a/Documentation/devicetree/bindings/arm/bitmain.yaml +++ b/Documentation/devicetree/bindings/arm/bitmain.yaml @@ -13,6 +13,6 @@ properties: compatible: items: - enum: - - bitmain,sophon-edge + - bitmain,sophon-edge - const: bitmain,bm1880 ... diff --git a/Documentation/devicetree/bindings/arm/nxp/lpc32xx.yaml b/Documentation/devicetree/bindings/arm/nxp/lpc32xx.yaml index 07f39d3eee7e..f7f024910e71 100644 --- a/Documentation/devicetree/bindings/arm/nxp/lpc32xx.yaml +++ b/Documentation/devicetree/bindings/arm/nxp/lpc32xx.yaml @@ -17,9 +17,8 @@ properties: - nxp,lpc3230 - nxp,lpc3240 - items: - - enum: - - ea,ea3250 - - phytec,phy3250 - - const: nxp,lpc3250 - + - enum: + - ea,ea3250 + - phytec,phy3250 + - const: nxp,lpc3250 ... diff --git a/Documentation/devicetree/bindings/arm/socionext/uniphier.yaml b/Documentation/devicetree/bindings/arm/socionext/uniphier.yaml index 65ad6d8a3c99..113f93b9ae55 100644 --- a/Documentation/devicetree/bindings/arm/socionext/uniphier.yaml +++ b/Documentation/devicetree/bindings/arm/socionext/uniphier.yaml @@ -17,45 +17,45 @@ properties: - description: LD4 SoC boards items: - enum: - - socionext,uniphier-ld4-ref + - socionext,uniphier-ld4-ref - const: socionext,uniphier-ld4 - description: Pro4 SoC boards items: - enum: - - socionext,uniphier-pro4-ace - - socionext,uniphier-pro4-ref - - socionext,uniphier-pro4-sanji + - socionext,uniphier-pro4-ace + - socionext,uniphier-pro4-ref + - socionext,uniphier-pro4-sanji - const: socionext,uniphier-pro4 - description: sLD8 SoC boards items: - enum: - - socionext,uniphier-sld8-ref + - socionext,uniphier-sld8-ref - const: socionext,uniphier-sld8 - description: PXs2 SoC boards items: - enum: - - socionext,uniphier-pxs2-gentil - - socionext,uniphier-pxs2-vodka + - socionext,uniphier-pxs2-gentil + - socionext,uniphier-pxs2-vodka - const: socionext,uniphier-pxs2 - description: LD6b SoC boards items: - enum: - - socionext,uniphier-ld6b-ref + - socionext,uniphier-ld6b-ref - const: socionext,uniphier-ld6b - description: LD11 SoC boards items: - enum: - - socionext,uniphier-ld11-global - - socionext,uniphier-ld11-ref + - socionext,uniphier-ld11-global + - socionext,uniphier-ld11-ref - const: socionext,uniphier-ld11 - description: LD20 SoC boards items: - enum: - - socionext,uniphier-ld20-global - - socionext,uniphier-ld20-ref + - socionext,uniphier-ld20-global + - socionext,uniphier-ld20-ref - const: socionext,uniphier-ld20 - description: PXs3 SoC boards items: - enum: - - socionext,uniphier-pxs3-ref + - socionext,uniphier-pxs3-ref - const: socionext,uniphier-pxs3 diff --git a/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml b/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml index 55f7938c4826..9f276bc9efa0 100644 --- a/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml +++ b/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml @@ -20,7 +20,7 @@ description: | [2]: https://wiki.st.com/stm32mpu/wiki/STM32MP15_RAM_mapping allOf: - - $ref: /schemas/simple-bus.yaml# + - $ref: /schemas/simple-bus.yaml# properties: compatible: diff --git a/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml b/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml index baff80197d5a..cf5db5e273f3 100644 --- a/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml +++ b/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml @@ -14,9 +14,9 @@ properties: compatible: oneOf: - items: - - enum: - - st,stm32mp157-syscfg - - const: syscon + - enum: + - st,stm32mp157-syscfg + - const: syscon reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/ata/faraday,ftide010.yaml b/Documentation/devicetree/bindings/ata/faraday,ftide010.yaml index bfc6357476fd..6451928dd2ce 100644 --- a/Documentation/devicetree/bindings/ata/faraday,ftide010.yaml +++ b/Documentation/devicetree/bindings/ata/faraday,ftide010.yaml @@ -26,8 +26,8 @@ properties: oneOf: - const: faraday,ftide010 - items: - - const: cortina,gemini-pata - - const: faraday,ftide010 + - const: cortina,gemini-pata + - const: faraday,ftide010 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/bus/allwinner,sun8i-a23-rsb.yaml b/Documentation/devicetree/bindings/bus/allwinner,sun8i-a23-rsb.yaml index 80973619342d..32d33b983d66 100644 --- a/Documentation/devicetree/bindings/bus/allwinner,sun8i-a23-rsb.yaml +++ b/Documentation/devicetree/bindings/bus/allwinner,sun8i-a23-rsb.yaml @@ -21,8 +21,8 @@ properties: oneOf: - const: allwinner,sun8i-a23-rsb - items: - - const: allwinner,sun8i-a83t-rsb - - const: allwinner,sun8i-a23-rsb + - const: allwinner,sun8i-a83t-rsb + - const: allwinner,sun8i-a23-rsb reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-gates-clk.yaml b/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-gates-clk.yaml index ed1b2126a81b..9a37a357cb4e 100644 --- a/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-gates-clk.yaml +++ b/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-gates-clk.yaml @@ -52,12 +52,12 @@ properties: - const: allwinner,sun4i-a10-dram-gates-clk - items: - - const: allwinner,sun5i-a13-dram-gates-clk - - const: allwinner,sun4i-a10-gates-clk + - const: allwinner,sun5i-a13-dram-gates-clk + - const: allwinner,sun4i-a10-gates-clk - items: - - const: allwinner,sun8i-h3-apb0-gates-clk - - const: allwinner,sun4i-a10-gates-clk + - const: allwinner,sun8i-h3-apb0-gates-clk + - const: allwinner,sun4i-a10-gates-clk reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/clock/fsl,plldig.yaml b/Documentation/devicetree/bindings/clock/fsl,plldig.yaml index a203d5d498db..8141f22410dd 100644 --- a/Documentation/devicetree/bindings/clock/fsl,plldig.yaml +++ b/Documentation/devicetree/bindings/clock/fsl,plldig.yaml @@ -28,15 +28,14 @@ properties: const: 0 fsl,vco-hz: - description: Optional for VCO frequency of the PLL in Hertz. - The VCO frequency of this PLL cannot be changed during runtime - only at startup. Therefore, the output frequencies are very - limited and might not even closely match the requested frequency. - To work around this restriction the user may specify its own - desired VCO frequency for the PLL. - minimum: 650000000 - maximum: 1300000000 - default: 1188000000 + description: Optional for VCO frequency of the PLL in Hertz. The VCO frequency + of this PLL cannot be changed during runtime only at startup. Therefore, + the output frequencies are very limited and might not even closely match + the requested frequency. To work around this restriction the user may specify + its own desired VCO frequency for the PLL. + minimum: 650000000 + maximum: 1300000000 + default: 1188000000 required: - compatible diff --git a/Documentation/devicetree/bindings/clock/qcom,mmcc.yaml b/Documentation/devicetree/bindings/clock/qcom,mmcc.yaml index f684fe67db84..acc31b3991bd 100644 --- a/Documentation/devicetree/bindings/clock/qcom,mmcc.yaml +++ b/Documentation/devicetree/bindings/clock/qcom,mmcc.yaml @@ -15,15 +15,15 @@ description: | power domains. properties: - compatible : + compatible: enum: - - qcom,mmcc-apq8064 - - qcom,mmcc-apq8084 - - qcom,mmcc-msm8660 - - qcom,mmcc-msm8960 - - qcom,mmcc-msm8974 - - qcom,mmcc-msm8996 - - qcom,mmcc-msm8998 + - qcom,mmcc-apq8064 + - qcom,mmcc-apq8084 + - qcom,mmcc-msm8660 + - qcom,mmcc-msm8960 + - qcom,mmcc-msm8974 + - qcom,mmcc-msm8996 + - qcom,mmcc-msm8998 clocks: items: diff --git a/Documentation/devicetree/bindings/connector/usb-connector.yaml b/Documentation/devicetree/bindings/connector/usb-connector.yaml index 4638d7adb806..369c58e22a06 100644 --- a/Documentation/devicetree/bindings/connector/usb-connector.yaml +++ b/Documentation/devicetree/bindings/connector/usb-connector.yaml @@ -144,7 +144,7 @@ required: examples: # Micro-USB connector with HS lines routed via controller (MUIC). - - |+ + - | muic-max77843 { usb_con1: connector { compatible = "usb-b-connector"; @@ -156,7 +156,7 @@ examples: # USB-C connector attached to CC controller (s2mm005), HS lines routed # to companion PMIC (max77865), SS lines to USB3 PHY and SBU to DisplayPort. # DisplayPort video lines are routed to the connector via SS mux in USB3 PHY. - - |+ + - | ccic: s2mm005 { usb_con2: connector { compatible = "usb-c-connector"; @@ -190,7 +190,7 @@ examples: # USB-C connector attached to a typec port controller(ptn5110), which has # power delivery support and enables drp. - - |+ + - | #include <dt-bindings/usb/pd.h> typec: ptn5110 { usb_con3: connector { diff --git a/Documentation/devicetree/bindings/crypto/allwinner,sun4i-a10-crypto.yaml b/Documentation/devicetree/bindings/crypto/allwinner,sun4i-a10-crypto.yaml index 8b9a8f337f16..fc823572bcff 100644 --- a/Documentation/devicetree/bindings/crypto/allwinner,sun4i-a10-crypto.yaml +++ b/Documentation/devicetree/bindings/crypto/allwinner,sun4i-a10-crypto.yaml @@ -15,16 +15,16 @@ properties: oneOf: - const: allwinner,sun4i-a10-crypto - items: - - const: allwinner,sun5i-a13-crypto - - const: allwinner,sun4i-a10-crypto + - const: allwinner,sun5i-a13-crypto + - const: allwinner,sun4i-a10-crypto - items: - - const: allwinner,sun6i-a31-crypto - - const: allwinner,sun4i-a10-crypto + - const: allwinner,sun6i-a31-crypto + - const: allwinner,sun4i-a10-crypto - items: - - const: allwinner,sun7i-a20-crypto - - const: allwinner,sun4i-a10-crypto + - const: allwinner,sun7i-a20-crypto + - const: allwinner,sun4i-a10-crypto - items: - - const: allwinner,sun8i-a33-crypto + - const: allwinner,sun8i-a33-crypto reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml b/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml index 2c459b8c76ff..7a60d84289cc 100644 --- a/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml +++ b/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml @@ -50,16 +50,16 @@ if: const: allwinner,sun50i-h6-crypto then: properties: - clocks: - minItems: 3 - clock-names: - minItems: 3 + clocks: + minItems: 3 + clock-names: + minItems: 3 else: properties: - clocks: - maxItems: 2 - clock-names: - maxItems: 2 + clocks: + maxItems: 2 + clock-names: + maxItems: 2 required: - compatible diff --git a/Documentation/devicetree/bindings/crypto/amlogic,gxl-crypto.yaml b/Documentation/devicetree/bindings/crypto/amlogic,gxl-crypto.yaml index 5becc60a0e28..385b23d255c3 100644 --- a/Documentation/devicetree/bindings/crypto/amlogic,gxl-crypto.yaml +++ b/Documentation/devicetree/bindings/crypto/amlogic,gxl-crypto.yaml @@ -12,7 +12,7 @@ maintainers: properties: compatible: items: - - const: amlogic,gxl-crypto + - const: amlogic,gxl-crypto reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-hdmi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-hdmi.yaml index 5d4915aed1e2..75e6479397a5 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-hdmi.yaml @@ -21,8 +21,8 @@ properties: - const: allwinner,sun5i-a10s-hdmi - const: allwinner,sun6i-a31-hdmi - items: - - const: allwinner,sun7i-a20-hdmi - - const: allwinner,sun5i-a10s-hdmi + - const: allwinner,sun7i-a20-hdmi + - const: allwinner,sun5i-a10s-hdmi reg: maxItems: 1 @@ -33,32 +33,32 @@ properties: clocks: oneOf: - items: - - description: The HDMI interface clock - - description: The HDMI module clock - - description: The first video PLL - - description: The second video PLL + - description: The HDMI interface clock + - description: The HDMI module clock + - description: The first video PLL + - description: The second video PLL - items: - - description: The HDMI interface clock - - description: The HDMI module clock - - description: The HDMI DDC clock - - description: The first video PLL - - description: The second video PLL + - description: The HDMI interface clock + - description: The HDMI module clock + - description: The HDMI DDC clock + - description: The first video PLL + - description: The second video PLL clock-names: oneOf: - items: - - const: ahb - - const: mod - - const: pll-0 - - const: pll-1 + - const: ahb + - const: mod + - const: pll-0 + - const: pll-1 - items: - - const: ahb - - const: mod - - const: ddc - - const: pll-0 - - const: pll-1 + - const: ahb + - const: mod + - const: ddc + - const: pll-0 + - const: pll-1 resets: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml index e5344c4ae226..87cb77b32ee3 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml @@ -35,26 +35,26 @@ properties: - const: allwinner,sun9i-a80-tcon-tv - items: - - enum: - - allwinner,sun7i-a20-tcon0 - - allwinner,sun7i-a20-tcon1 - - const: allwinner,sun7i-a20-tcon + - enum: + - allwinner,sun7i-a20-tcon0 + - allwinner,sun7i-a20-tcon1 + - const: allwinner,sun7i-a20-tcon - items: - - enum: - - allwinner,sun50i-a64-tcon-lcd - - const: allwinner,sun8i-a83t-tcon-lcd + - enum: + - allwinner,sun50i-a64-tcon-lcd + - const: allwinner,sun8i-a83t-tcon-lcd - items: - - enum: - - allwinner,sun8i-h3-tcon-tv - - allwinner,sun50i-a64-tcon-tv - - const: allwinner,sun8i-a83t-tcon-tv + - enum: + - allwinner,sun8i-h3-tcon-tv + - allwinner,sun50i-a64-tcon-tv + - const: allwinner,sun8i-a83t-tcon-tv - items: - - enum: - - allwinner,sun50i-h6-tcon-tv - - const: allwinner,sun8i-r40-tcon-tv + - enum: + - allwinner,sun50i-h6-tcon-tv + - const: allwinner,sun8i-r40-tcon-tv reg: maxItems: 1 @@ -83,37 +83,37 @@ properties: resets: anyOf: - items: - - description: TCON Reset Line + - description: TCON Reset Line - items: - - description: TCON Reset Line - - description: TCON LVDS Reset Line + - description: TCON Reset Line + - description: TCON LVDS Reset Line - items: - - description: TCON Reset Line - - description: TCON eDP Reset Line + - description: TCON Reset Line + - description: TCON eDP Reset Line - items: - - description: TCON Reset Line - - description: TCON eDP Reset Line - - description: TCON LVDS Reset Line + - description: TCON Reset Line + - description: TCON eDP Reset Line + - description: TCON LVDS Reset Line reset-names: oneOf: - const: lcd - items: - - const: lcd - - const: lvds + - const: lcd + - const: lvds - items: - - const: lcd - - const: edp + - const: lcd + - const: edp - items: - - const: lcd - - const: edp - - const: lvds + - const: lcd + - const: edp + - const: lvds ports: type: object diff --git a/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml index 9e90c2b00960..eed05b26cdf3 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml @@ -76,28 +76,28 @@ required: allOf: - if: properties: - compatible: - contains: - const: allwinner,sun6i-a31-mipi-dsi + compatible: + contains: + const: allwinner,sun6i-a31-mipi-dsi then: - properties: - clocks: - minItems: 2 + properties: + clocks: + minItems: 2 - required: - - clock-names + required: + - clock-names - if: properties: - compatible: - contains: - const: allwinner,sun50i-a64-mipi-dsi + compatible: + contains: + const: allwinner,sun50i-a64-mipi-dsi then: - properties: - clocks: - minItems: 1 + properties: + clocks: + minItems: 1 additionalProperties: false diff --git a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml index 4d6795690ac3..fa4769a0b26e 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml @@ -29,11 +29,11 @@ properties: - const: allwinner,sun50i-h6-dw-hdmi - items: - - enum: - - allwinner,sun8i-h3-dw-hdmi - - allwinner,sun8i-r40-dw-hdmi - - allwinner,sun50i-a64-dw-hdmi - - const: allwinner,sun8i-a83t-dw-hdmi + - enum: + - allwinner,sun8i-h3-dw-hdmi + - allwinner,sun8i-r40-dw-hdmi + - allwinner,sun50i-a64-dw-hdmi + - const: allwinner,sun8i-a83t-dw-hdmi reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/bridge/lvds-codec.yaml b/Documentation/devicetree/bindings/display/bridge/lvds-codec.yaml index 8f373029f5d2..e737951f5873 100644 --- a/Documentation/devicetree/bindings/display/bridge/lvds-codec.yaml +++ b/Documentation/devicetree/bindings/display/bridge/lvds-codec.yaml @@ -32,17 +32,17 @@ properties: compatible: oneOf: - items: - - enum: - - ti,ds90c185 # For the TI DS90C185 FPD-Link Serializer - - ti,ds90c187 # For the TI DS90C187 FPD-Link Serializer - - ti,sn75lvds83 # For the TI SN75LVDS83 FlatLink transmitter - - const: lvds-encoder # Generic LVDS encoder compatible fallback + - enum: + - ti,ds90c185 # For the TI DS90C185 FPD-Link Serializer + - ti,ds90c187 # For the TI DS90C187 FPD-Link Serializer + - ti,sn75lvds83 # For the TI SN75LVDS83 FlatLink transmitter + - const: lvds-encoder # Generic LVDS encoder compatible fallback - items: - - enum: - - ti,ds90cf384a # For the DS90CF384A FPD-Link LVDS Receiver - - const: lvds-decoder # Generic LVDS decoders compatible fallback + - enum: + - ti,ds90cf384a # For the DS90CF384A FPD-Link LVDS Receiver + - const: lvds-decoder # Generic LVDS decoders compatible fallback - enum: - - thine,thc63lvdm83d # For the THC63LVDM83D LVDS serializer + - thine,thc63lvdm83d # For the THC63LVDM83D LVDS serializer ports: type: object diff --git a/Documentation/devicetree/bindings/display/panel/sony,acx424akp.yaml b/Documentation/devicetree/bindings/display/panel/sony,acx424akp.yaml index 185dcc8fd1f9..78d060097052 100644 --- a/Documentation/devicetree/bindings/display/panel/sony,acx424akp.yaml +++ b/Documentation/devicetree/bindings/display/panel/sony,acx424akp.yaml @@ -18,7 +18,7 @@ properties: reg: true reset-gpios: true vddi-supply: - description: regulator that supplies the vddi voltage + description: regulator that supplies the vddi voltage enforce-video-mode: true required: diff --git a/Documentation/devicetree/bindings/display/panel/xinpeng,xpp055c272.yaml b/Documentation/devicetree/bindings/display/panel/xinpeng,xpp055c272.yaml index d9fdb58e06b4..891de2256d22 100644 --- a/Documentation/devicetree/bindings/display/panel/xinpeng,xpp055c272.yaml +++ b/Documentation/devicetree/bindings/display/panel/xinpeng,xpp055c272.yaml @@ -19,9 +19,9 @@ properties: backlight: true reset-gpios: true iovcc-supply: - description: regulator that supplies the iovcc voltage + description: regulator that supplies the iovcc voltage vci-supply: - description: regulator that supplies the vci voltage + description: regulator that supplies the vci voltage required: - compatible diff --git a/Documentation/devicetree/bindings/display/renesas,cmm.yaml b/Documentation/devicetree/bindings/display/renesas,cmm.yaml index a57037b9e9ba..005406c89507 100644 --- a/Documentation/devicetree/bindings/display/renesas,cmm.yaml +++ b/Documentation/devicetree/bindings/display/renesas,cmm.yaml @@ -21,15 +21,15 @@ properties: compatible: oneOf: - items: - - enum: - - renesas,r8a7795-cmm - - renesas,r8a7796-cmm - - renesas,r8a77965-cmm - - renesas,r8a77990-cmm - - renesas,r8a77995-cmm - - const: renesas,rcar-gen3-cmm + - enum: + - renesas,r8a7795-cmm + - renesas,r8a7796-cmm + - renesas,r8a77965-cmm + - renesas,r8a77990-cmm + - renesas,r8a77995-cmm + - const: renesas,rcar-gen3-cmm - items: - - const: renesas,rcar-gen2-cmm + - const: renesas,rcar-gen2-cmm reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/dma/ti/k3-udma.yaml b/Documentation/devicetree/bindings/dma/ti/k3-udma.yaml index 39ea05e6e5ff..85056982a242 100644 --- a/Documentation/devicetree/bindings/dma/ti/k3-udma.yaml +++ b/Documentation/devicetree/bindings/dma/ti/k3-udma.yaml @@ -69,10 +69,10 @@ properties: maxItems: 3 reg-names: - items: - - const: gcfg - - const: rchanrt - - const: tchanrt + items: + - const: gcfg + - const: rchanrt + - const: tchanrt msi-parent: true diff --git a/Documentation/devicetree/bindings/gpio/brcm,xgs-iproc-gpio.yaml b/Documentation/devicetree/bindings/gpio/brcm,xgs-iproc-gpio.yaml index 5f1ed20e43ee..4f2cbd8307a7 100644 --- a/Documentation/devicetree/bindings/gpio/brcm,xgs-iproc-gpio.yaml +++ b/Documentation/devicetree/bindings/gpio/brcm,xgs-iproc-gpio.yaml @@ -27,7 +27,7 @@ properties: gpio-controller: true '#gpio-cells': - const: 2 + const: 2 ngpios: minimum: 0 diff --git a/Documentation/devicetree/bindings/gpu/arm,mali-midgard.yaml b/Documentation/devicetree/bindings/gpu/arm,mali-midgard.yaml index 0407e45eb8c4..a7a67e0a42e5 100644 --- a/Documentation/devicetree/bindings/gpu/arm,mali-midgard.yaml +++ b/Documentation/devicetree/bindings/gpu/arm,mali-midgard.yaml @@ -16,33 +16,33 @@ properties: oneOf: - items: - enum: - - samsung,exynos5250-mali + - samsung,exynos5250-mali - const: arm,mali-t604 - items: - enum: - - samsung,exynos5420-mali + - samsung,exynos5420-mali - const: arm,mali-t628 - items: - enum: - - allwinner,sun50i-h6-mali + - allwinner,sun50i-h6-mali - const: arm,mali-t720 - items: - enum: - - amlogic,meson-gxm-mali - - realtek,rtd1295-mali + - amlogic,meson-gxm-mali + - realtek,rtd1295-mali - const: arm,mali-t820 - items: - enum: - - arm,juno-mali + - arm,juno-mali - const: arm,mali-t624 - items: - enum: - - rockchip,rk3288-mali - - samsung,exynos5433-mali + - rockchip,rk3288-mali + - samsung,exynos5433-mali - const: arm,mali-t760 - items: - enum: - - rockchip,rk3399-mali + - rockchip,rk3399-mali - const: arm,mali-t860 # "arm,mali-t830" diff --git a/Documentation/devicetree/bindings/gpu/vivante,gc.yaml b/Documentation/devicetree/bindings/gpu/vivante,gc.yaml index 0bc4b38d5cbb..e1ac6ff5a230 100644 --- a/Documentation/devicetree/bindings/gpu/vivante,gc.yaml +++ b/Documentation/devicetree/bindings/gpu/vivante,gc.yaml @@ -9,7 +9,7 @@ title: Vivante GPU Bindings description: Vivante GPU core devices maintainers: - - Lucas Stach <l.stach(a)pengutronix.de> + - Lucas Stach <l.stach(a)pengutronix.de> properties: compatible: diff --git a/Documentation/devicetree/bindings/i2c/i2c-rk3x.yaml b/Documentation/devicetree/bindings/i2c/i2c-rk3x.yaml index 61eac76c84c4..790aa7218ee0 100644 --- a/Documentation/devicetree/bindings/i2c/i2c-rk3x.yaml +++ b/Documentation/devicetree/bindings/i2c/i2c-rk3x.yaml @@ -28,14 +28,14 @@ properties: - const: rockchip,rk3399-i2c - items: - enum: - - rockchip,rk3036-i2c - - rockchip,rk3368-i2c + - rockchip,rk3036-i2c + - rockchip,rk3368-i2c - const: rockchip,rk3288-i2c - items: - enum: - - rockchip,px30-i2c - - rockchip,rk3308-i2c - - rockchip,rk3328-i2c + - rockchip,px30-i2c + - rockchip,rk3308-i2c + - rockchip,rk3328-i2c - const: rockchip,rk3399-i2c reg: diff --git a/Documentation/devicetree/bindings/iio/adc/adi,ad7124.yaml b/Documentation/devicetree/bindings/iio/adc/adi,ad7124.yaml index f0934b295edc..97087a45ce54 100644 --- a/Documentation/devicetree/bindings/iio/adc/adi,ad7124.yaml +++ b/Documentation/devicetree/bindings/iio/adc/adi,ad7124.yaml @@ -72,8 +72,8 @@ patternProperties: The channel number. It can have up to 8 channels on ad7124-4 and 16 channels on ad7124-8, numbered from 0 to 15. items: - minimum: 0 - maximum: 15 + minimum: 0 + maximum: 15 adi,reference-select: description: | diff --git a/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml b/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml index 118809a03279..97f521d654ea 100644 --- a/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml +++ b/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml @@ -7,9 +7,9 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Linear Technology / Analog Devices LTC2496 ADC maintainers: - - Lars-Peter Clausen <lars(a)metafoo.de> - - Michael Hennerich <Michael.Hennerich(a)analog.com> - - Stefan Popa <stefan.popa(a)analog.com> + - Lars-Peter Clausen <lars(a)metafoo.de> + - Michael Hennerich <Michael.Hennerich(a)analog.com> + - Stefan Popa <stefan.popa(a)analog.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/input/allwinner,sun4i-a10-lradc-keys.yaml b/Documentation/devicetree/bindings/input/allwinner,sun4i-a10-lradc-keys.yaml index 5b3b71c9c018..512a6af5aa42 100644 --- a/Documentation/devicetree/bindings/input/allwinner,sun4i-a10-lradc-keys.yaml +++ b/Documentation/devicetree/bindings/input/allwinner,sun4i-a10-lradc-keys.yaml @@ -16,8 +16,8 @@ properties: - const: allwinner,sun4i-a10-lradc-keys - const: allwinner,sun8i-a83t-r-lradc - items: - - const: allwinner,sun50i-a64-lradc - - const: allwinner,sun8i-a83t-r-lradc + - const: allwinner,sun50i-a64-lradc + - const: allwinner,sun8i-a83t-r-lradc reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/input/touchscreen/goodix.yaml b/Documentation/devicetree/bindings/input/touchscreen/goodix.yaml index c8ea9434c9cc..e81cfa56f25a 100644 --- a/Documentation/devicetree/bindings/input/touchscreen/goodix.yaml +++ b/Documentation/devicetree/bindings/input/touchscreen/goodix.yaml @@ -63,7 +63,7 @@ required: - interrupts examples: -- | + - | i2c { #address-cells = <1>; #size-cells = <0>; diff --git a/Documentation/devicetree/bindings/interconnect/qcom,msm8916.yaml b/Documentation/devicetree/bindings/interconnect/qcom,msm8916.yaml index 4107e60cab12..e1009ae4e8f7 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,msm8916.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,msm8916.yaml @@ -10,8 +10,8 @@ maintainers: - Georgi Djakov <georgi.djakov(a)linaro.org> description: | - The Qualcomm MSM8916 interconnect providers support adjusting the - bandwidth requirements between the various NoC fabrics. + The Qualcomm MSM8916 interconnect providers support adjusting the + bandwidth requirements between the various NoC fabrics. properties: compatible: diff --git a/Documentation/devicetree/bindings/interconnect/qcom,msm8974.yaml b/Documentation/devicetree/bindings/interconnect/qcom,msm8974.yaml index 9af3c6e59cff..8004c4baf397 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,msm8974.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,msm8974.yaml @@ -10,8 +10,8 @@ maintainers: - Brian Masney <masneyb(a)onstation.org> description: | - The Qualcomm MSM8974 interconnect providers support setting system - bandwidth requirements between various network-on-chip fabrics. + The Qualcomm MSM8974 interconnect providers support setting system + bandwidth requirements between various network-on-chip fabrics. properties: reg: diff --git a/Documentation/devicetree/bindings/interconnect/qcom,qcs404.yaml b/Documentation/devicetree/bindings/interconnect/qcom,qcs404.yaml index 8d65c5f80679..3fbb8785fbc9 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,qcs404.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,qcs404.yaml @@ -10,8 +10,8 @@ maintainers: - Georgi Djakov <georgi.djakov(a)linaro.org> description: | - The Qualcomm QCS404 interconnect providers support adjusting the - bandwidth requirements between the various NoC fabrics. + The Qualcomm QCS404 interconnect providers support adjusting the + bandwidth requirements between the various NoC fabrics. properties: reg: diff --git a/Documentation/devicetree/bindings/interrupt-controller/allwinner,sun7i-a20-sc-nmi.yaml b/Documentation/devicetree/bindings/interrupt-controller/allwinner,sun7i-a20-sc-nmi.yaml index cf09055da78b..7cd6b8bacfa0 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/allwinner,sun7i-a20-sc-nmi.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/allwinner,sun7i-a20-sc-nmi.yaml @@ -27,15 +27,15 @@ properties: deprecated: true - const: allwinner,sun7i-a20-sc-nmi - items: - - const: allwinner,sun8i-a83t-r-intc - - const: allwinner,sun6i-a31-r-intc + - const: allwinner,sun8i-a83t-r-intc + - const: allwinner,sun6i-a31-r-intc - const: allwinner,sun9i-a80-sc-nmi - items: - - const: allwinner,sun50i-a64-r-intc - - const: allwinner,sun6i-a31-r-intc + - const: allwinner,sun50i-a64-r-intc + - const: allwinner,sun6i-a31-r-intc - items: - - const: allwinner,sun50i-h6-r-intc - - const: allwinner,sun6i-a31-r-intc + - const: allwinner,sun50i-h6-r-intc + - const: allwinner,sun6i-a31-r-intc reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/interrupt-controller/intel,ixp4xx-interrupt.yaml b/Documentation/devicetree/bindings/interrupt-controller/intel,ixp4xx-interrupt.yaml index ccc507f384d2..14dced11877b 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/intel,ixp4xx-interrupt.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/intel,ixp4xx-interrupt.yaml @@ -25,10 +25,10 @@ properties: compatible: items: - enum: - - intel,ixp42x-interrupt - - intel,ixp43x-interrupt - - intel,ixp45x-interrupt - - intel,ixp46x-interrupt + - intel,ixp42x-interrupt + - intel,ixp43x-interrupt + - intel,ixp45x-interrupt + - intel,ixp46x-interrupt reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml b/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml index 9e5c6608b4e3..2a5b29567926 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml @@ -14,13 +14,13 @@ properties: compatible: oneOf: - items: - - enum: - - st,stm32-exti - - st,stm32h7-exti + - enum: + - st,stm32-exti + - st,stm32h7-exti - items: - - enum: - - st,stm32mp1-exti - - const: syscon + - enum: + - st,stm32mp1-exti + - const: syscon "#interrupt-cells": const: 2 diff --git a/Documentation/devicetree/bindings/iommu/samsung,sysmmu.yaml b/Documentation/devicetree/bindings/iommu/samsung,sysmmu.yaml index 0e33cd9e010e..af51b91c893e 100644 --- a/Documentation/devicetree/bindings/iommu/samsung,sysmmu.yaml +++ b/Documentation/devicetree/bindings/iommu/samsung,sysmmu.yaml @@ -54,13 +54,13 @@ properties: clock-names: oneOf: - items: - - const: sysmmu + - const: sysmmu - items: - - const: sysmmu - - const: master + - const: sysmmu + - const: master - items: - - const: aclk - - const: pclk + - const: aclk + - const: pclk "#iommu-cells": const: 0 diff --git a/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml b/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml index 5b13d6672996..db851541d619 100644 --- a/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml +++ b/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml @@ -24,7 +24,7 @@ properties: maxItems: 1 clocks: - maxItems: 1 + maxItems: 1 interrupts: items: diff --git a/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-csi.yaml b/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-csi.yaml index 8453ee340b9f..09318830db47 100644 --- a/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-csi.yaml +++ b/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-csi.yaml @@ -20,11 +20,11 @@ properties: - const: allwinner,sun4i-a10-csi1 - const: allwinner,sun7i-a20-csi0 - items: - - const: allwinner,sun7i-a20-csi1 - - const: allwinner,sun4i-a10-csi1 + - const: allwinner,sun7i-a20-csi1 + - const: allwinner,sun4i-a10-csi1 - items: - - const: allwinner,sun8i-r40-csi0 - - const: allwinner,sun7i-a20-csi0 + - const: allwinner,sun8i-r40-csi0 + - const: allwinner,sun7i-a20-csi0 reg: maxItems: 1 @@ -35,24 +35,24 @@ properties: clocks: oneOf: - items: - - description: The CSI interface clock - - description: The CSI DRAM clock + - description: The CSI interface clock + - description: The CSI DRAM clock - items: - - description: The CSI interface clock - - description: The CSI ISP clock - - description: The CSI DRAM clock + - description: The CSI interface clock + - description: The CSI ISP clock + - description: The CSI DRAM clock clock-names: oneOf: - items: - - const: bus - - const: ram + - const: bus + - const: ram - items: - - const: bus - - const: isp - - const: ram + - const: bus + - const: isp + - const: ram resets: maxItems: 1 diff --git a/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml b/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml index 37d77e065491..5a1da4029c37 100644 --- a/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml +++ b/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml @@ -29,14 +29,14 @@ properties: compatible: oneOf: - items: - - enum: - - amlogic,gxbb-vdec # GXBB (S905) - - amlogic,gxl-vdec # GXL (S905X, S905D) - - amlogic,gxm-vdec # GXM (S912) - - const: amlogic,gx-vdec + - enum: + - amlogic,gxbb-vdec # GXBB (S905) + - amlogic,gxl-vdec # GXL (S905X, S905D) + - amlogic,gxm-vdec # GXM (S912) + - const: amlogic,gx-vdec - enum: - - amlogic,g12a-vdec # G12A (S905X2, S905D2) - - amlogic,sm1-vdec # SM1 (S905X3, S905D3) + - amlogic,g12a-vdec # G12A (S905X2, S905D2) + - amlogic,sm1-vdec # SM1 (S905X3, S905D3) interrupts: minItems: 2 diff --git a/Documentation/devicetree/bindings/media/renesas,ceu.yaml b/Documentation/devicetree/bindings/media/renesas,ceu.yaml index fcb5f13704a5..f2393458814e 100644 --- a/Documentation/devicetree/bindings/media/renesas,ceu.yaml +++ b/Documentation/devicetree/bindings/media/renesas,ceu.yaml @@ -32,23 +32,23 @@ properties: additionalProperties: false properties: - endpoint: - type: object - additionalProperties: false + endpoint: + type: object + additionalProperties: false # Properties described in # Documentation/devicetree/bindings/media/video-interfaces.txt - properties: - remote-endpoint: true - hsync-active: true - vsync-active: true - field-even-active: false - bus-width: - enum: [8, 16] - default: 8 - - required: - - remote-endpoint + properties: + remote-endpoint: true + hsync-active: true + vsync-active: true + field-even-active: false + bus-width: + enum: [8, 16] + default: 8 + + required: + - remote-endpoint required: - endpoint diff --git a/Documentation/devicetree/bindings/media/renesas,vin.yaml b/Documentation/devicetree/bindings/media/renesas,vin.yaml index 1ec947b4781f..ecc09f1124d4 100644 --- a/Documentation/devicetree/bindings/media/renesas,vin.yaml +++ b/Documentation/devicetree/bindings/media/renesas,vin.yaml @@ -261,13 +261,13 @@ properties: anyOf: - required: - - endpoint@0 + - endpoint@0 - required: - - endpoint@1 + - endpoint@1 - required: - - endpoint@2 + - endpoint@2 - required: - - endpoint@3 + - endpoint@3 additionalProperties: false diff --git a/Documentation/devicetree/bindings/media/ti,vpe.yaml b/Documentation/devicetree/bindings/media/ti,vpe.yaml index f3a8a350e85f..ef473f287399 100644 --- a/Documentation/devicetree/bindings/media/ti,vpe.yaml +++ b/Documentation/devicetree/bindings/media/ti,vpe.yaml @@ -17,7 +17,7 @@ description: |- properties: compatible: - const: ti,dra7-vpe + const: ti,dra7-vpe reg: items: diff --git a/Documentation/devicetree/bindings/memory-controllers/fsl/imx8m-ddrc.yaml b/Documentation/devicetree/bindings/memory-controllers/fsl/imx8m-ddrc.yaml index c9e6c22cb5be..445e46feda69 100644 --- a/Documentation/devicetree/bindings/memory-controllers/fsl/imx8m-ddrc.yaml +++ b/Documentation/devicetree/bindings/memory-controllers/fsl/imx8m-ddrc.yaml @@ -25,9 +25,9 @@ properties: compatible: items: - enum: - - fsl,imx8mn-ddrc - - fsl,imx8mm-ddrc - - fsl,imx8mq-ddrc + - fsl,imx8mn-ddrc + - fsl,imx8mm-ddrc + - fsl,imx8mq-ddrc - const: fsl,imx8m-ddrc reg: diff --git a/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml b/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml index ddf190cb800b..e675611f80d0 100644 --- a/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml +++ b/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml @@ -66,8 +66,8 @@ patternProperties: reg: description: Identify trigger hardware block. items: - minimum: 0 - maximum: 2 + minimum: 0 + maximum: 2 required: - compatible diff --git a/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml b/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml index 590849ee9f32..4acda7ce3b44 100644 --- a/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml +++ b/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml @@ -102,8 +102,8 @@ patternProperties: reg: description: Identify trigger hardware block. items: - minimum: 0 - maximum: 16 + minimum: 0 + maximum: 16 required: - compatible diff --git a/Documentation/devicetree/bindings/mfd/syscon.yaml b/Documentation/devicetree/bindings/mfd/syscon.yaml index 39375e4313d2..7a39486b215a 100644 --- a/Documentation/devicetree/bindings/mfd/syscon.yaml +++ b/Documentation/devicetree/bindings/mfd/syscon.yaml @@ -33,13 +33,13 @@ properties: compatible: anyOf: - items: - - enum: - - allwinner,sun8i-a83t-system-controller - - allwinner,sun8i-h3-system-controller - - allwinner,sun8i-v3s-system-controller - - allwinner,sun50i-a64-system-controller + - enum: + - allwinner,sun8i-a83t-system-controller + - allwinner,sun8i-h3-system-controller + - allwinner,sun8i-v3s-system-controller + - allwinner,sun50i-a64-system-controller - - const: syscon + - const: syscon - contains: const: syscon diff --git a/Documentation/devicetree/bindings/mmc/cdns,sdhci.yaml b/Documentation/devicetree/bindings/mmc/cdns,sdhci.yaml index 2f45dd0d04db..d43a0c557a44 100644 --- a/Documentation/devicetree/bindings/mmc/cdns,sdhci.yaml +++ b/Documentation/devicetree/bindings/mmc/cdns,sdhci.yaml @@ -17,7 +17,7 @@ properties: compatible: items: - enum: - - socionext,uniphier-sd4hc + - socionext,uniphier-sd4hc - const: cdns,sd4hc reg: diff --git a/Documentation/devicetree/bindings/mmc/rockchip-dw-mshc.yaml b/Documentation/devicetree/bindings/mmc/rockchip-dw-mshc.yaml index 89c3edd6a728..4ee3ed6efab4 100644 --- a/Documentation/devicetree/bindings/mmc/rockchip-dw-mshc.yaml +++ b/Documentation/devicetree/bindings/mmc/rockchip-dw-mshc.yaml @@ -30,21 +30,21 @@ properties: - items: - enum: # for Rockchip PX30 - - rockchip,px30-dw-mshc + - rockchip,px30-dw-mshc # for Rockchip RK3036 - - rockchip,rk3036-dw-mshc + - rockchip,rk3036-dw-mshc # for Rockchip RK322x - - rockchip,rk3228-dw-mshc + - rockchip,rk3228-dw-mshc # for Rockchip RK3308 - - rockchip,rk3308-dw-mshc + - rockchip,rk3308-dw-mshc # for Rockchip RK3328 - - rockchip,rk3328-dw-mshc + - rockchip,rk3328-dw-mshc # for Rockchip RK3368 - - rockchip,rk3368-dw-mshc + - rockchip,rk3368-dw-mshc # for Rockchip RK3399 - - rockchip,rk3399-dw-mshc + - rockchip,rk3399-dw-mshc # for Rockchip RV1108 - - rockchip,rv1108-dw-mshc + - rockchip,rv1108-dw-mshc - const: rockchip,rk3288-dw-mshc reg: diff --git a/Documentation/devicetree/bindings/mmc/socionext,uniphier-sd.yaml b/Documentation/devicetree/bindings/mmc/socionext,uniphier-sd.yaml index cdfac9b4411b..8d6413f48823 100644 --- a/Documentation/devicetree/bindings/mmc/socionext,uniphier-sd.yaml +++ b/Documentation/devicetree/bindings/mmc/socionext,uniphier-sd.yaml @@ -35,15 +35,15 @@ properties: oneOf: - const: host - items: - - const: host - - const: bridge + - const: host + - const: bridge - items: - - const: host - - const: hw + - const: host + - const: hw - items: - - const: host - - const: bridge - - const: hw + - const: host + - const: bridge + - const: hw resets: minItems: 1 diff --git a/Documentation/devicetree/bindings/mtd/denali,nand.yaml b/Documentation/devicetree/bindings/mtd/denali,nand.yaml index 46e6b6726bc0..c07b91592cbd 100644 --- a/Documentation/devicetree/bindings/mtd/denali,nand.yaml +++ b/Documentation/devicetree/bindings/mtd/denali,nand.yaml @@ -54,8 +54,8 @@ properties: reg: register reset oneOf: - items: - - const: nand - - const: reg + - const: nand + - const: reg - const: nand - const: reg diff --git a/Documentation/devicetree/bindings/net/allwinner,sun8i-a83t-emac.yaml b/Documentation/devicetree/bindings/net/allwinner,sun8i-a83t-emac.yaml index db36b4d86484..c7c9ad4e3f9f 100644 --- a/Documentation/devicetree/bindings/net/allwinner,sun8i-a83t-emac.yaml +++ b/Documentation/devicetree/bindings/net/allwinner,sun8i-a83t-emac.yaml @@ -19,8 +19,8 @@ properties: - const: allwinner,sun8i-v3s-emac - const: allwinner,sun50i-a64-emac - items: - - const: allwinner,sun50i-h6-emac - - const: allwinner,sun50i-a64-emac + - const: allwinner,sun50i-h6-emac + - const: allwinner,sun50i-a64-emac reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml b/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml index cccf8202c8f7..7a784dc4e513 100644 --- a/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml +++ b/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml @@ -9,7 +9,7 @@ title: Bosch MCAN controller Bindings description: Bosch MCAN controller for CAN bus maintainers: - - Sriram Dash <sriram.dash(a)samsung.com> + - Sriram Dash <sriram.dash(a)samsung.com> properties: compatible: @@ -51,31 +51,31 @@ properties: bosch,mram-cfg: description: | - Message RAM configuration data. - Multiple M_CAN instances can share the same Message RAM - and each element(e.g Rx FIFO or Tx Buffer and etc) number - in Message RAM is also configurable, so this property is - telling driver how the shared or private Message RAM are - used by this M_CAN controller. - - The format should be as follows: - <offset sidf_elems xidf_elems rxf0_elems rxf1_elems rxb_elems txe_elems txb_elems> - The 'offset' is an address offset of the Message RAM where - the following elements start from. This is usually set to - 0x0 if you're using a private Message RAM. The remain cells - are used to specify how many elements are used for each FIFO/Buffer. - - M_CAN includes the following elements according to user manual: - 11-bit Filter 0-128 elements / 0-128 words - 29-bit Filter 0-64 elements / 0-128 words - Rx FIFO 0 0-64 elements / 0-1152 words - Rx FIFO 1 0-64 elements / 0-1152 words - Rx Buffers 0-64 elements / 0-1152 words - Tx Event FIFO 0-32 elements / 0-64 words - Tx Buffers 0-32 elements / 0-576 words - - Please refer to 2.4.1 Message RAM Configuration in Bosch - M_CAN user manual for details. + Message RAM configuration data. + Multiple M_CAN instances can share the same Message RAM + and each element(e.g Rx FIFO or Tx Buffer and etc) number + in Message RAM is also configurable, so this property is + telling driver how the shared or private Message RAM are + used by this M_CAN controller. + + The format should be as follows: + <offset sidf_elems xidf_elems rxf0_elems rxf1_elems rxb_elems txe_elems txb_elems> + The 'offset' is an address offset of the Message RAM where + the following elements start from. This is usually set to + 0x0 if you're using a private Message RAM. The remain cells + are used to specify how many elements are used for each FIFO/Buffer. + + M_CAN includes the following elements according to user manual: + 11-bit Filter 0-128 elements / 0-128 words + 29-bit Filter 0-64 elements / 0-128 words + Rx FIFO 0 0-64 elements / 0-1152 words + Rx FIFO 1 0-64 elements / 0-1152 words + Rx Buffers 0-64 elements / 0-1152 words + Tx Event FIFO 0-32 elements / 0-64 words + Tx Buffers 0-32 elements / 0-576 words + + Please refer to 2.4.1 Message RAM Configuration in Bosch + M_CAN user manual for details. allOf: - $ref: /schemas/types.yaml#/definitions/int32-array - items: diff --git a/Documentation/devicetree/bindings/net/renesas,ether.yaml b/Documentation/devicetree/bindings/net/renesas,ether.yaml index 7f84df9790e2..2eaa8799e002 100644 --- a/Documentation/devicetree/bindings/net/renesas,ether.yaml +++ b/Documentation/devicetree/bindings/net/renesas,ether.yaml @@ -40,8 +40,8 @@ properties: reg: items: - - description: E-DMAC/feLic registers - - description: TSU registers + - description: E-DMAC/feLic registers + - description: TSU registers minItems: 1 interrupts: diff --git a/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml b/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml index 976f139bb66e..8fc8d3be303b 100644 --- a/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml +++ b/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml @@ -23,14 +23,14 @@ properties: oneOf: - const: ti,cpsw-switch - items: - - const: ti,am335x-cpsw-switch - - const: ti,cpsw-switch + - const: ti,am335x-cpsw-switch + - const: ti,cpsw-switch - items: - - const: ti,am4372-cpsw-switch - - const: ti,cpsw-switch + - const: ti,am4372-cpsw-switch + - const: ti,cpsw-switch - items: - - const: ti,dra7-cpsw-switch - - const: ti,cpsw-switch + - const: ti,dra7-cpsw-switch + - const: ti,cpsw-switch reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml b/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml index 242ac4935a4b..2ea14ab29254 100644 --- a/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml +++ b/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml @@ -18,25 +18,24 @@ allOf: properties: compatible: oneOf: - - const: ti,davinci_mdio - - items: - - const: ti,keystone_mdio - - const: ti,davinci_mdio - - items: - - const: ti,cpsw-mdio - - const: ti,davinci_mdio - - items: - - const: ti,am4372-mdio - - const: ti,cpsw-mdio - - const: ti,davinci_mdio + - const: ti,davinci_mdio + - items: + - const: ti,keystone_mdio + - const: ti,davinci_mdio + - items: + - const: ti,cpsw-mdio + - const: ti,davinci_mdio + - items: + - const: ti,am4372-mdio + - const: ti,cpsw-mdio + - const: ti,davinci_mdio reg: maxItems: 1 bus_freq: - maximum: 2500000 - description: - MDIO Bus frequency + maximum: 2500000 + description: MDIO Bus frequency ti,hwmods: description: TI hwmod name diff --git a/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml b/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml index 9a346d6290d9..77bb5309918e 100644 --- a/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml +++ b/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml @@ -23,7 +23,7 @@ description: |+ properties: compatible: - const: intel,lgm-emmc-phy + const: intel,lgm-emmc-phy "#phy-cells": const: 0 diff --git a/Documentation/devicetree/bindings/pwm/pwm-samsung.yaml b/Documentation/devicetree/bindings/pwm/pwm-samsung.yaml index ea7f32905172..4fe64f4dd594 100644 --- a/Documentation/devicetree/bindings/pwm/pwm-samsung.yaml +++ b/Documentation/devicetree/bindings/pwm/pwm-samsung.yaml @@ -49,17 +49,17 @@ properties: are available. oneOf: - items: - - const: timers + - const: timers - items: - - const: timers - - const: pwm-tclk0 + - const: timers + - const: pwm-tclk0 - items: - - const: timers - - const: pwm-tclk1 + - const: timers + - const: pwm-tclk1 - items: - - const: timers - - const: pwm-tclk0 - - const: pwm-tclk1 + - const: timers + - const: pwm-tclk0 + - const: pwm-tclk1 interrupts: description: diff --git a/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml b/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml index c0d83865e933..4ff4d3df0a06 100644 --- a/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml @@ -25,7 +25,7 @@ properties: maxItems: 3 resets: - maxItems: 1 + maxItems: 1 st,syscfg-holdboot: allOf: diff --git a/Documentation/devicetree/bindings/reset/brcm,bcm7216-pcie-sata-rescal.yaml b/Documentation/devicetree/bindings/reset/brcm,bcm7216-pcie-sata-rescal.yaml index 512a33bdb208..dfce6738b033 100644 --- a/Documentation/devicetree/bindings/reset/brcm,bcm7216-pcie-sata-rescal.yaml +++ b/Documentation/devicetree/bindings/reset/brcm,bcm7216-pcie-sata-rescal.yaml @@ -7,7 +7,9 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: BCM7216 RESCAL reset controller -description: This document describes the BCM7216 RESCAL reset controller which is responsible for controlling the reset of the SATA and PCIe0/1 instances on BCM7216. +description: This document describes the BCM7216 RESCAL reset controller + which is responsible for controlling the reset of the SATA and PCIe0/1 + instances on BCM7216. maintainers: - Florian Fainelli <f.fainelli(a)gmail.com> diff --git a/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml b/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml index 48c6cafca90c..57b087574aa1 100644 --- a/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml +++ b/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml @@ -38,10 +38,10 @@ properties: minItems: 3 maxItems: 3 description: | - Phandle/offset/mask triplet. The phandle to pwrcfg used to - access control register at offset, and change the dbp (Disable Backup - Protection) bit represented by the mask, mandatory to disable/enable backup - domain (RTC registers) write protection. + Phandle/offset/mask triplet. The phandle to pwrcfg used to + access control register at offset, and change the dbp (Disable Backup + Protection) bit represented by the mask, mandatory to disable/enable backup + domain (RTC registers) write protection. assigned-clocks: description: | @@ -78,14 +78,14 @@ allOf: const: st,stm32h7-rtc then: - properties: - clocks: - minItems: 2 - maxItems: 2 + properties: + clocks: + minItems: 2 + maxItems: 2 - required: - - clock-names - - st,syscfg + required: + - clock-names + - st,syscfg - if: properties: @@ -94,16 +94,16 @@ allOf: const: st,stm32mp1-rtc then: - properties: - clocks: - minItems: 2 - maxItems: 2 + properties: + clocks: + minItems: 2 + maxItems: 2 - assigned-clocks: false - assigned-clock-parents: false + assigned-clocks: false + assigned-clock-parents: false - required: - - clock-names + required: + - clock-names required: - compatible diff --git a/Documentation/devicetree/bindings/serial/amlogic,meson-uart.yaml b/Documentation/devicetree/bindings/serial/amlogic,meson-uart.yaml index d4178ab0d675..75ebc9952a99 100644 --- a/Documentation/devicetree/bindings/serial/amlogic,meson-uart.yaml +++ b/Documentation/devicetree/bindings/serial/amlogic,meson-uart.yaml @@ -24,18 +24,18 @@ properties: oneOf: - description: Always-on power domain UART controller items: - - enum: + - enum: + - amlogic,meson6-uart + - amlogic,meson8-uart + - amlogic,meson8b-uart + - amlogic,meson-gx-uart + - const: amlogic,meson-ao-uart + - description: Everything-Else power domain UART controller + enum: - amlogic,meson6-uart - amlogic,meson8-uart - amlogic,meson8b-uart - amlogic,meson-gx-uart - - const: amlogic,meson-ao-uart - - description: Everything-Else power domain UART controller - enum: - - amlogic,meson6-uart - - amlogic,meson8-uart - - amlogic,meson8b-uart - - amlogic,meson-gx-uart reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/serial/rs485.yaml b/Documentation/devicetree/bindings/serial/rs485.yaml index d4beaf11222d..2b8261ea6d9c 100644 --- a/Documentation/devicetree/bindings/serial/rs485.yaml +++ b/Documentation/devicetree/bindings/serial/rs485.yaml @@ -6,13 +6,12 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: RS485 serial communications Bindings -description: The RTS signal is capable of automatically controlling - line direction for the built-in half-duplex mode. - The properties described hereafter shall be given to a - half-duplex capable UART node. +description: The RTS signal is capable of automatically controlling line + direction for the built-in half-duplex mode. The properties described + hereafter shall be given to a half-duplex capable UART node. maintainers: - - Rob Herring <robh(a)kernel.org> + - Rob Herring <robh(a)kernel.org> properties: rs485-rts-delay: @@ -37,9 +36,11 @@ properties: $ref: /schemas/types.yaml#/definitions/flag linux,rs485-enabled-at-boot-time: - description: enables the rs485 feature at boot time. It can be disabled later with proper ioctl. + description: enables the rs485 feature at boot time. It can be disabled + later with proper ioctl. $ref: /schemas/types.yaml#/definitions/flag rs485-rx-during-tx: - description: enables the receiving of data even while sending data. - $ref: /schemas/types.yaml#/definitions/flag + description: enables the receiving of data even while sending data. + $ref: /schemas/types.yaml#/definitions/flag +... diff --git a/Documentation/devicetree/bindings/soc/amlogic/amlogic,canvas.yaml b/Documentation/devicetree/bindings/soc/amlogic/amlogic,canvas.yaml index cb008fd188d8..02b2d5ba01d6 100644 --- a/Documentation/devicetree/bindings/soc/amlogic/amlogic,canvas.yaml +++ b/Documentation/devicetree/bindings/soc/amlogic/amlogic,canvas.yaml @@ -26,11 +26,11 @@ properties: compatible: oneOf: - items: - - enum: - - amlogic,meson8-canvas - - amlogic,meson8b-canvas - - amlogic,meson8m2-canvas - - const: amlogic,canvas + - enum: + - amlogic,meson8-canvas + - amlogic,meson8b-canvas + - amlogic,meson8m2-canvas + - const: amlogic,canvas - const: amlogic,canvas # GXBB and newer SoCs reg: diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml index d1b65554e681..91cf4176abd5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -17,16 +17,16 @@ properties: oneOf: # for FSI2 SoC - items: - - enum: - - renesas,fsi2-sh73a0 - - renesas,fsi2-r8a7740 - - enum: - - renesas,sh_fsi2 + - enum: + - renesas,fsi2-sh73a0 + - renesas,fsi2-r8a7740 + - enum: + - renesas,sh_fsi2 # for Generic - items: - - enum: - - renesas,sh_fsi - - renesas,sh_fsi2 + - enum: + - renesas,sh_fsi + - renesas,sh_fsi2 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/spi/qcom,spi-qcom-qspi.yaml b/Documentation/devicetree/bindings/spi/qcom,spi-qcom-qspi.yaml index 0cf470eaf2a0..406286149a6b 100644 --- a/Documentation/devicetree/bindings/spi/qcom,spi-qcom-qspi.yaml +++ b/Documentation/devicetree/bindings/spi/qcom,spi-qcom-qspi.yaml @@ -8,12 +8,12 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: Qualcomm Quad Serial Peripheral Interface (QSPI) maintainers: - - Mukesh Savaliya <msavaliy(a)codeaurora.org> - - Akash Asthana <akashast(a)codeaurora.org> + - Mukesh Savaliya <msavaliy(a)codeaurora.org> + - Akash Asthana <akashast(a)codeaurora.org> -description: - The QSPI controller allows SPI protocol communication in single, dual, or quad - wire transmission modes for read/write access to slaves such as NOR flash. +description: The QSPI controller allows SPI protocol communication in single, + dual, or quad wire transmission modes for read/write access to slaves such + as NOR flash. allOf: - $ref: /spi/spi-controller.yaml# diff --git a/Documentation/devicetree/bindings/spi/renesas,hspi.yaml b/Documentation/devicetree/bindings/spi/renesas,hspi.yaml index c429cf4bea5b..f492cb9fea12 100644 --- a/Documentation/devicetree/bindings/spi/renesas,hspi.yaml +++ b/Documentation/devicetree/bindings/spi/renesas,hspi.yaml @@ -16,8 +16,8 @@ properties: compatible: items: - enum: - - renesas,hspi-r8a7778 # R-Car M1A - - renesas,hspi-r8a7779 # R-Car H1 + - renesas,hspi-r8a7778 # R-Car M1A + - renesas,hspi-r8a7779 # R-Car H1 - const: renesas,hspi reg: diff --git a/Documentation/devicetree/bindings/spi/spi-pl022.yaml b/Documentation/devicetree/bindings/spi/spi-pl022.yaml index dfb697c69341..22ba4e90655b 100644 --- a/Documentation/devicetree/bindings/spi/spi-pl022.yaml +++ b/Documentation/devicetree/bindings/spi/spi-pl022.yaml @@ -51,7 +51,7 @@ properties: pl022,rt: description: indicates the controller should run the message pump with realtime - priority to minimise the transfer latency on the bus (boolean) + priority to minimise the transfer latency on the bus (boolean) type: boolean dmas: diff --git a/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml b/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml index 3665a5fe6b7f..1a342ce1f798 100644 --- a/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml +++ b/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml @@ -24,8 +24,8 @@ properties: reg-names: items: - - const: qspi - - const: qspi_mm + - const: qspi + - const: qspi_mm clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sram/allwinner,sun4i-a10-system-control.yaml b/Documentation/devicetree/bindings/sram/allwinner,sun4i-a10-system-control.yaml index 4b5509436588..f5825935fd22 100644 --- a/Documentation/devicetree/bindings/sram/allwinner,sun4i-a10-system-control.yaml +++ b/Documentation/devicetree/bindings/sram/allwinner,sun4i-a10-system-control.yaml @@ -29,8 +29,8 @@ properties: - const: allwinner,sun4i-a10-system-control - const: allwinner,sun5i-a13-system-control - items: - - const: allwinner,sun7i-a20-system-control - - const: allwinner,sun4i-a10-system-control + - const: allwinner,sun7i-a20-system-control + - const: allwinner,sun4i-a10-system-control - const: allwinner,sun8i-a23-system-control - const: allwinner,sun8i-h3-system-control - const: allwinner,sun50i-a64-sram-controller @@ -38,11 +38,11 @@ properties: - const: allwinner,sun50i-a64-system-control - const: allwinner,sun50i-h5-system-control - items: - - const: allwinner,sun50i-h6-system-control - - const: allwinner,sun50i-a64-system-control + - const: allwinner,sun50i-h6-system-control + - const: allwinner,sun50i-a64-system-control - items: - - const: allwinner,suniv-f1c100s-system-control - - const: allwinner,sun4i-a10-system-control + - const: allwinner,suniv-f1c100s-system-control + - const: allwinner,sun4i-a10-system-control reg: maxItems: 1 @@ -69,44 +69,44 @@ patternProperties: - const: allwinner,sun4i-a10-sram-d - const: allwinner,sun50i-a64-sram-c - items: - - const: allwinner,sun5i-a13-sram-a3-a4 - - const: allwinner,sun4i-a10-sram-a3-a4 + - const: allwinner,sun5i-a13-sram-a3-a4 + - const: allwinner,sun4i-a10-sram-a3-a4 - items: - - const: allwinner,sun7i-a20-sram-a3-a4 - - const: allwinner,sun4i-a10-sram-a3-a4 + - const: allwinner,sun7i-a20-sram-a3-a4 + - const: allwinner,sun4i-a10-sram-a3-a4 - items: - - const: allwinner,sun5i-a13-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun5i-a13-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun7i-a20-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun7i-a20-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun8i-a23-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun8i-a23-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun8i-h3-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun8i-h3-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun50i-a64-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun50i-a64-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun50i-h5-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun50i-h5-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun50i-h6-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun50i-h6-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun5i-a13-sram-d - - const: allwinner,sun4i-a10-sram-d + - const: allwinner,sun5i-a13-sram-d + - const: allwinner,sun4i-a10-sram-d - items: - - const: allwinner,sun7i-a20-sram-d - - const: allwinner,sun4i-a10-sram-d + - const: allwinner,sun7i-a20-sram-d + - const: allwinner,sun4i-a10-sram-d - items: - - const: allwinner,suniv-f1c100s-sram-d - - const: allwinner,sun4i-a10-sram-d + - const: allwinner,suniv-f1c100s-sram-d + - const: allwinner,sun4i-a10-sram-d - items: - - const: allwinner,sun50i-h6-sram-c - - const: allwinner,sun50i-a64-sram-c + - const: allwinner,sun50i-h6-sram-c + - const: allwinner,sun50i-a64-sram-c required: - "#address-cells" diff --git a/Documentation/devicetree/bindings/thermal/amlogic,thermal.yaml b/Documentation/devicetree/bindings/thermal/amlogic,thermal.yaml index e43ec50bda37..999c6b365f1d 100644 --- a/Documentation/devicetree/bindings/thermal/amlogic,thermal.yaml +++ b/Documentation/devicetree/bindings/thermal/amlogic,thermal.yaml @@ -13,11 +13,11 @@ description: Binding for Amlogic Thermal properties: compatible: - items: - - enum: - - amlogic,g12a-cpu-thermal - - amlogic,g12a-ddr-thermal - - const: amlogic,g12a-thermal + items: + - enum: + - amlogic,g12a-cpu-thermal + - amlogic,g12a-ddr-thermal + - const: amlogic,g12a-thermal reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml b/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml index fa255672e8e5..135186f83925 100644 --- a/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml +++ b/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml @@ -28,10 +28,10 @@ properties: - arm,armv7-timer - items: - enum: - - arm,armv7-timer + - arm,armv7-timer - items: - enum: - - arm,armv8-timer + - arm,armv8-timer interrupts: items: diff --git a/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml b/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml index 582bbef62b95..6ff718ede184 100644 --- a/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml +++ b/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml @@ -20,7 +20,7 @@ properties: compatible: items: - enum: - - arm,armv7-timer-mem + - arm,armv7-timer-mem reg: maxItems: 1 @@ -77,7 +77,7 @@ patternProperties: - description: physical timer irq - description: virtual timer irq - reg : + reg: minItems: 1 maxItems: 2 items: diff --git a/Documentation/devicetree/bindings/usb/dwc2.yaml b/Documentation/devicetree/bindings/usb/dwc2.yaml index 0d6d850a7f17..fb2f62aef5fa 100644 --- a/Documentation/devicetree/bindings/usb/dwc2.yaml +++ b/Documentation/devicetree/bindings/usb/dwc2.yaml @@ -62,14 +62,14 @@ properties: resets: items: - - description: common reset - - description: ecc reset + - description: common reset + - description: ecc reset minItems: 1 reset-names: items: - - const: dwc2 - - const: dwc2-ecc + - const: dwc2 + - const: dwc2-ecc minItems: 1 phys: -- 2.20.1

12 23

[PATCH] soundwire: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
by Kuninori Morimoto 20 Apr '20

20 Apr '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now ALSA SoC needs to use asoc_rtd_to_codec(), otherwise, it will be compile error. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 3c83e76c6bf9..ed8d576bf5dc 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -669,11 +669,11 @@ static int sdw_stream_setup(struct snd_pcm_substream *substream, /* Set stream pointer on all CODEC DAIs */ for (i = 0; i < rtd->num_codecs; i++) { - ret = snd_soc_dai_set_sdw_stream(rtd->codec_dais[i], sdw_stream, + ret = snd_soc_dai_set_sdw_stream(asoc_rtd_to_codec(rtd, i), sdw_stream, substream->stream); if (ret < 0) { dev_err(dai->dev, "failed to set stream pointer on codec dai %s", - rtd->codec_dais[i]->name); + asoc_rtd_to_codec(rtd, i)->name); goto release_stream; } } -- 2.17.1

3 6

[PATCH] ASoC: wm89xx: Fix build error without CONFIG_I2C
by YueHaibing 20 Apr '20

20 Apr '20

sound/soc/codecs/wm8900.o: In function `wm8900_i2c_probe': wm8900.c:(.text+0xa36): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8900.o: In function `wm8900_modinit': wm8900.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8900.o: In function `wm8900_exit': wm8900.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8988.o: In function `wm8988_i2c_probe': wm8988.c:(.text+0x857): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8988.o: In function `wm8988_modinit': wm8988.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8988.o: In function `wm8988_exit': wm8988.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8995.o: In function `wm8995_i2c_probe': wm8995.c:(.text+0x1c4f): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8995.o: In function `wm8995_modinit': wm8995.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8995.o: In function `wm8995_exit': wm8995.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' Add missing I2C dependency to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e6a0c5d05fa5..2d8d0bba0c0a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1525,6 +1525,7 @@ config SND_SOC_WM8804_SPI config SND_SOC_WM8900 tristate + depends on I2C config SND_SOC_WM8903 tristate "Wolfson Microelectronics WM8903 CODEC" @@ -1576,6 +1577,7 @@ config SND_SOC_WM8985 config SND_SOC_WM8988 tristate + depends on I2C config SND_SOC_WM8990 tristate @@ -1594,6 +1596,7 @@ config SND_SOC_WM8994 config SND_SOC_WM8995 tristate + depends on I2C config SND_SOC_WM8996 tristate -- 2.17.1

3 5

[PATCH] ASoC: wcd934x: remove unneeded semicolon
by Jason Yan 20 Apr '20

20 Apr '20

Fix the following coccicheck warning: sound/soc/codecs/wcd934x.c:3621:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:4270:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:4018:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:4043:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3804:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3866:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3596:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3542:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3673:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3723:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:2492:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:1790:3-4: Unneeded semicolon sound/soc/codecs/wcd934x.c:1806:3-4: Unneeded semicolon sound/soc/codecs/wcd934x.c:1812:2-3: Unneeded semicolon Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/codecs/wcd934x.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 5269857e2746..a514eb32e3e6 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1787,7 +1787,7 @@ static int wcd934x_hw_params(struct snd_pcm_substream *substream, params_rate(params)); return -EINVAL; - }; + } ret = wcd934x_set_decimator_rate(dai, tx_fs_rate, params_rate(params)); @@ -1803,13 +1803,13 @@ static int wcd934x_hw_params(struct snd_pcm_substream *substream, dev_err(wcd->dev, "Invalid format 0x%x\n", params_width(params)); return -EINVAL; - }; + } break; default: dev_err(wcd->dev, "Invalid stream type %d\n", substream->stream); return -EINVAL; - }; + } wcd->dai[dai->id].sconfig.rate = params_rate(params); wcd934x_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); @@ -2489,7 +2489,7 @@ static int wcd934x_compander_set(struct snd_kcontrol *kc, break; default: break; - }; + } return 0; } @@ -3539,7 +3539,7 @@ static int wcd934x_codec_enable_mix_path(struct snd_soc_dapm_widget *w, val += offset_val; snd_soc_component_write(comp, gain_reg, val); break; - }; + } return 0; } @@ -3593,7 +3593,7 @@ static int wcd934x_codec_enable_main_path(struct snd_soc_dapm_widget *w, snd_soc_component_write(comp, gain_reg, snd_soc_component_read32(comp, gain_reg)); break; - }; + } return 0; } @@ -3618,7 +3618,7 @@ static int wcd934x_codec_ear_dac_event(struct snd_soc_dapm_widget *w, wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, WCD_CLSH_STATE_EAR, CLS_H_NORMAL); break; - }; + } return 0; } @@ -3670,7 +3670,7 @@ static int wcd934x_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, break; default: break; - }; + } return 0; } @@ -3720,7 +3720,7 @@ static int wcd934x_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, break; default: break; - }; + } return 0; } @@ -3801,7 +3801,7 @@ static int wcd934x_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, */ usleep_range(20000, 20100); break; - }; + } return 0; } @@ -3863,7 +3863,7 @@ static int wcd934x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, */ usleep_range(20000, 20100); break; - }; + } return 0; } @@ -4015,7 +4015,7 @@ static int wcd934x_codec_enable_dmic(struct snd_soc_dapm_widget *w, dev_err(comp->dev, "%s: Invalid DMIC Selection\n", __func__); return -EINVAL; - }; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -4040,7 +4040,7 @@ static int wcd934x_codec_enable_dmic(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(comp, dmic_clk_reg, dmic_clk_en, 0); break; - }; + } return 0; } @@ -4267,7 +4267,7 @@ static int wcd934x_codec_enable_dec(struct snd_soc_dapm_widget *w, WCD934X_DEC_PWR_LVL_MASK, WCD934X_DEC_PWR_LVL_DF); break; - }; + } out: kfree(wname); return ret; -- 2.21.1

2 1

[PATCH 0/2] ASoC: Fix dependency issues of SND_SOC
by Wei Li 20 Apr '20

20 Apr '20

Fix dependency issues of SND_SOC introduced by commit ea00d95200d02 ("ASoC: Use imply for SND_SOC_ALL_CODECS"). Wei Li (2): ASoC: wm89xx: Fix build errors caused by I2C dependency ASoC: Fix wrong dependency of da7210 and wm8983 sound/soc/codecs/Kconfig | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) -- 2.17.1

2 3

[PATCH] ASoC: wcd934x: remove unnecessary comparisons to bool
by Jason Yan 20 Apr '20

20 Apr '20

Fix the following coccicheck warning: sound/soc/codecs/wcd934x.c:3881:8-17: WARNING: Comparison of 0/1 to bool variable Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index a514eb32e3e6..531b8b79e55f 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -3878,7 +3878,7 @@ static u32 wcd934x_get_dmic_sample_rate(struct snd_soc_component *comp, u16 adc_mux_ctl_reg, tx_fs_reg; u32 dmic_fs; - while (dec_found == 0 && adc_mux_index < WCD934X_MAX_VALID_ADC_MUX) { + while (!dec_found && adc_mux_index < WCD934X_MAX_VALID_ADC_MUX) { if (adc_mux_index < 4) { adc_mux_ctl_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0 + (adc_mux_index * 2); -- 2.21.1

2 1

[PATCH v3] ASoC: Intel: boards: split woofer and tweeter support
by mac.chiang＠intel.com 20 Apr '20

20 Apr '20

From: Mac Chiang <mac.chiang(a)intel.com> Support Woofer stereo speakers by default and optionally Tweeter stereo speakers with a DMI quirk Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v2: - considering woofer speakers settings as general. - separate out the tweeter speakers and consistency of woofer+tweeter when hardware-specific. - check NULL condition if devm_kzalloc are failed v3: - fixed multiple alignments - add invalid value return --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 222 +++++++++++++++++++++++------ 1 file changed, 180 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index ed6c26a..2872345 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -30,6 +30,36 @@ #define CML_RT5682_CODEC_DAI "rt5682-aif1" #define NAME_SIZE 32 +#define SOF_RT1011_SPEAKER_WL BIT(0) +#define SOF_RT1011_SPEAKER_WR BIT(1) +#define SOF_RT1011_SPEAKER_TL BIT(2) +#define SOF_RT1011_SPEAKER_TR BIT(3) +#define SPK_CH 4 + +/* Default: Woofer speakers */ +static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR; + +static int sof_rt1011_quirk_cb(const struct dmi_system_id *id) +{ + sof_rt1011_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_rt1011_quirk_table[] = { + { + .callback = sof_rt1011_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Helios"), + }, + .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR | + SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR), + }, + { + } +}; + static struct snd_soc_jack hdmi_jack[3]; struct hdmi_pcm { @@ -48,15 +78,16 @@ struct card_private { static const struct snd_kcontrol_new cml_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("TL Ext Spk"), - SOC_DAPM_PIN_SWITCH("TR Ext Spk"), SOC_DAPM_PIN_SWITCH("WL Ext Spk"), SOC_DAPM_PIN_SWITCH("WR Ext Spk"), }; +static const struct snd_kcontrol_new cml_rt1011_tt_controls[] = { + SOC_DAPM_PIN_SWITCH("TL Ext Spk"), + SOC_DAPM_PIN_SWITCH("TR Ext Spk"), +}; + static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { - SND_SOC_DAPM_SPK("TL Ext Spk", NULL), - SND_SOC_DAPM_SPK("TR Ext Spk", NULL), SND_SOC_DAPM_SPK("WL Ext Spk", NULL), SND_SOC_DAPM_SPK("WR Ext Spk", NULL), SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -64,10 +95,13 @@ static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; +static const struct snd_soc_dapm_widget cml_rt1011_tt_widgets[] = { + SND_SOC_DAPM_SPK("TL Ext Spk", NULL), + SND_SOC_DAPM_SPK("TR Ext Spk", NULL), +}; + static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { - /*speaker*/ - {"TL Ext Spk", NULL, "TL SPO"}, - {"TR Ext Spk", NULL, "TR SPO"}, + /*WL/WR speaker*/ {"WL Ext Spk", NULL, "WL SPO"}, {"WR Ext Spk", NULL, "WR SPO"}, @@ -82,6 +116,12 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { {"DMic", NULL, "SoC DMIC"}, }; +static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { + /*TL/TR speaker*/ + {"TL Ext Spk", NULL, "TL SPO" }, + {"TR Ext Spk", NULL, "TR SPO" }, +}; + static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -121,6 +161,35 @@ static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; }; +static int cml_rt1011_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_card *card = rtd->card; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + + ret = snd_soc_add_card_controls(card, cml_rt1011_tt_controls, + ARRAY_SIZE(cml_rt1011_tt_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, + cml_rt1011_tt_widgets, + ARRAY_SIZE(cml_rt1011_tt_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1011_tt_map, + ARRAY_SIZE(cml_rt1011_tt_map)); + + if (ret) + return ret; + } + + return ret; +} + static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -191,30 +260,38 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, * The feedback is captured for each codec individually. * Hence all 4 codecs use 1 Tx slot each for feedback. */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x4, 0x1, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x1, 0x1, 4, 24); - if (ret < 0) - break; + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } } - /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x8, 0x2, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x2, 0x2, 4, 24); - if (ret < 0) - break; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x1, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x2, 0x2, 4, 24); + if (ret < 0) + break; + } } } if (ret < 0) @@ -302,9 +379,7 @@ SND_SOC_DAILINK_DEF(ssp1_pin, SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY( /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), - /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI), - /* TL */ COMP_CODEC("i2c-10EC1011:02", CML_RT1011_CODEC_DAI), - /* TR */ COMP_CODEC("i2c-10EC1011:03", CML_RT1011_CODEC_DAI))); + /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -398,6 +473,7 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { .dpcm_playback = 1, .dpcm_capture = 1, /* Capture stream provides Feedback */ .no_pcm = 1, + .init = cml_rt1011_spk_init, .ops = &cml_rt1011_ops, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, @@ -412,14 +488,6 @@ static struct snd_soc_codec_conf rt1011_conf[] = { .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), .name_prefix = "WR", }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), - .name_prefix = "TR", - }, }; /* Cometlake audio machine driver for RT1011 and RT5682 */ @@ -441,10 +509,12 @@ static struct snd_soc_card snd_soc_card_cml = { static int snd_cml_rt1011_probe(struct platform_device *pdev) { + struct snd_soc_dai_link_component *rt1011_dais_components; + struct snd_soc_codec_conf *rt1011_dais_confs; struct card_private *ctx; struct snd_soc_acpi_mach *mach; const char *platform_name; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -455,6 +525,73 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; + dmi_check_system(sof_rt1011_quirk_table); + + dev_info(&pdev->dev, "sof_rt1011_quirk = %lx\n", sof_rt1011_quirk); + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + rt1011_dais_confs = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_codec_conf) * + SPK_CH, GFP_KERNEL); + + if (!rt1011_dais_confs) + return -ENOMEM; + + rt1011_dais_components = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link_component) * + SPK_CH, GFP_KERNEL); + + if (!rt1011_dais_components) + return -ENOMEM; + + for (i = 0; i < SPK_CH; i++) { + rt1011_dais_confs[i].dlc.name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + + if (!rt1011_dais_confs[i].dlc.name) + return -ENOMEM; + + switch (i) { + case 0: + rt1011_dais_confs[i].name_prefix = "WL"; + break; + case 1: + rt1011_dais_confs[i].name_prefix = "WR"; + break; + case 2: + rt1011_dais_confs[i].name_prefix = "TL"; + break; + case 3: + rt1011_dais_confs[i].name_prefix = "TR"; + break; + default: + return -EINVAL; + } + rt1011_dais_components[i].name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + if (!rt1011_dais_components[i].name) + return -ENOMEM; + + rt1011_dais_components[i].dai_name = CML_RT1011_CODEC_DAI; + } + + snd_soc_card_cml.codec_conf = rt1011_dais_confs; + snd_soc_card_cml.num_configs = SPK_CH; + + for (i = 0; i < ARRAY_SIZE(cml_rt1011_rt5682_dailink); i++) { + if (!strcmp(cml_rt1011_rt5682_dailink[i].codecs->dai_name, + CML_RT1011_CODEC_DAI)) { + cml_rt1011_rt5682_dailink[i].codecs = rt1011_dais_components; + cml_rt1011_rt5682_dailink[i].num_codecs = SPK_CH; + } + } + } + /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml, platform_name); @@ -482,5 +619,6 @@ MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1011 and RT5682 in I2S mo MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:cml_rt1011_rt5682"); -- 2.7.4

2 1

[PATCH] ASoC: wcd9335: remove unneeded semicolon
by Jason Yan 20 Apr '20

20 Apr '20

Fix the following coccicheck warning: sound/soc/codecs/wcd9335.c:2606:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3619:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2849:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2955:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2988:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3960:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3776:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3924:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3355:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3832:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3079:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3089:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3232:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3878:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3578:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3728:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:1922:3-4: Unneeded semicolon sound/soc/codecs/wcd9335.c:1938:3-4: Unneeded semicolon sound/soc/codecs/wcd9335.c:1944:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3144:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2568:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2219:2-3: Unneeded semicolon Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/codecs/wcd9335.c | 44 +++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 700cc1212770..fb073f4dc7ed 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1919,7 +1919,7 @@ static int wcd9335_hw_params(struct snd_pcm_substream *substream, __func__, params_rate(params)); return -EINVAL; - }; + } ret = wcd9335_set_decimator_rate(dai, tx_fs_rate, params_rate(params)); @@ -1935,13 +1935,13 @@ static int wcd9335_hw_params(struct snd_pcm_substream *substream, dev_err(wcd->dev, "%s: Invalid format 0x%x\n", __func__, params_width(params)); return -EINVAL; - }; + } break; default: dev_err(wcd->dev, "Invalid stream type %d\n", substream->stream); return -EINVAL; - }; + } wcd->dai[dai->id].sconfig.rate = params_rate(params); wcd9335_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); @@ -2216,7 +2216,7 @@ static int wcd9335_set_compander(struct snd_kcontrol *kc, break; default: break; - }; + } return 0; } @@ -2565,7 +2565,7 @@ static int wcd9335_micbias_control(struct snd_soc_component *component, 0xC0, 0x00); } break; - }; + } return 0; } @@ -2603,7 +2603,7 @@ static int __wcd9335_codec_enable_micbias(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: wcd9335_micbias_control(comp, micb_num, MICB_DISABLE, true); break; - }; + } return 0; } @@ -2846,7 +2846,7 @@ static int wcd9335_codec_enable_dec(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x00); break; - }; + } out: kfree(wname); return ret; @@ -2952,7 +2952,7 @@ static int wcd9335_codec_enable_dmic(struct snd_soc_dapm_widget *w, dev_err(comp->dev, "%s: Invalid DMIC Selection\n", __func__); return -EINVAL; - }; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -2985,7 +2985,7 @@ static int wcd9335_codec_enable_dmic(struct snd_soc_dapm_widget *w, dmic_rate_val << dmic_rate_shift); } break; - }; + } return 0; } @@ -3076,7 +3076,7 @@ static int wcd9335_codec_enable_mix_path(struct snd_soc_dapm_widget *w, dev_err(comp->dev, "%s: No gain register avail for %s\n", __func__, w->name); return 0; - }; + } switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -3086,7 +3086,7 @@ static int wcd9335_codec_enable_mix_path(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMD: break; - }; + } return 0; } @@ -3141,7 +3141,7 @@ static u16 wcd9335_interp_get_primary_reg(u16 reg, u16 *ind) prim_int_reg = WCD9335_CDC_RX8_RX_PATH_CTL; *ind = 8; break; - }; + } return prim_int_reg; } @@ -3229,7 +3229,7 @@ static int wcd9335_codec_enable_prim_interpolator( wcd9335_codec_hd2_control(comp, prim_int_reg, event); } break; - }; + } return 0; } @@ -3352,7 +3352,7 @@ static int wcd9335_codec_enable_interpolator(struct snd_soc_dapm_widget *w, wcd9335_config_compander(comp, w->shift, event); wcd9335_codec_enable_prim_interpolator(comp, reg, event); break; - }; + } return 0; } @@ -3575,7 +3575,7 @@ static int wcd9335_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, ((hph_mode == CLS_H_LOHIFI) ? CLS_H_HIFI : hph_mode)); break; - }; + } return 0; } @@ -3616,7 +3616,7 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, WCD_CLSH_STATE_EAR, CLS_H_NORMAL); break; - }; + } return 0; } @@ -3725,7 +3725,7 @@ static int wcd9335_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, WCD_CLSH_STATE_HPHR, ((hph_mode == CLS_H_LOHIFI) ? CLS_H_HIFI : hph_mode)); break; - }; + } return 0; } @@ -3773,7 +3773,7 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, */ usleep_range(5000, 5500); break; - }; + } return 0; } @@ -3829,7 +3829,7 @@ static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, */ usleep_range(5000, 5500); break; - }; + } return 0; } @@ -3875,7 +3875,7 @@ static int wcd9335_codec_enable_rx_bias(struct snd_soc_dapm_widget *w, WCD9335_ANA_RX_BIAS_ENABLE_MASK, WCD9335_ANA_RX_BIAS_DISABLE); break; - }; + } return 0; } @@ -3921,7 +3921,7 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, */ usleep_range(5000, 5500); break; - }; + } return 0; } @@ -3957,7 +3957,7 @@ static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, usleep_range(5000, 5500); break; - }; + } return 0; } -- 2.21.1

2 1

[PATCH] ASoC: topology: remove unneeded semicolon
by Jason Yan 20 Apr '20

20 Apr '20

Fix the following coccicheck warning: sound/soc/soc-topology.c:2344:2-3: Unneeded semicolon Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 33e8d189ba2f..a5cd73742dc5 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2390,7 +2390,7 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, tplg->pos += le32_to_cpu(hdr->size) + le32_to_cpu(hdr->payload_size); return 0; - }; + } /* check the element size and count */ link = (struct snd_soc_tplg_link_config *)tplg->pos; -- 2.21.1

2 1

[PATCH 0/2] ASoC: rsnd: multi-SSI setup fixes
by Matthias Blankertz 20 Apr '20

20 Apr '20

Fix rsnd_dai_call() operations being performed twice for the master SSI in multi-SSI setups, and fix the rsnd_ssi_stop operation for multi-SSI setups. The only visible effect of these issues was some "status check failed" spam when the rsnd_ssi_stop was called, but overall the code is cleaner now, and some questionable writes to the SSICR register which did not lead to any observable misbehaviour but were contrary to the datasheet are fixed. Mark: The first patch kind of reverts my "ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode" from a few days ago and achieves the same effect in a simpler fashion, if you would prefer a clean patch series based on v5.6 drop me a note. Greetings, Matthias Matthias Blankertz (2): ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent ASoC: rsnd: Fix "status check failed" spam for multi-SSI sound/soc/sh/rcar/ssi.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) base-commit: 15a5760cb8b6d5c1ebbf1d2e1f0b77380ab68a82 -- 2.26.1

3 4

[PATCH 0/2] ASoC: meson: fix codec-to-codec link setup
by Jerome Brunet 20 Apr '20

20 Apr '20

This patchset fixes the problem reported by Marc in this thread [0] The problem was due to an error in the meson card drivers which had the "no_pcm" dai_link property set on codec-to-codec links [0]: https://lore.kernel.org/r/20200417122732.GC5315@sirena.org.uk Jerome Brunet (2): ASoC: meson: axg-card: fix codec-to-codec link setup ASoC: meson: gx-card: fix codec-to-codec link setup sound/soc/meson/axg-card.c | 4 +++- sound/soc/meson/gx-card.c | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.25.2

2 3

[PATCH] ASoC: dapm: fixup dapm kcontrol widget
by Gyeongtaek Lee 20 Apr '20

20 Apr '20

snd_soc_dapm_kcontrol widget which is created by autodisable control should contain correct on_val, mask and shift because it is set when the widget is powered and changed value is applied on registers by following code in dapm_seq_run_coalesced(). mask |= w->mask << w->shift; if (w->power) value |= w->on_val << w->shift; else value |= w->off_val << w->shift; Shift on the mask in dapm_kcontrol_data_alloc() is removed to prevent double shift. And, on_val in dapm_kcontrol_set_value() is modified to get correct value in the dapm_seq_run_coalesced(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4143ca190d0..e2632841b321 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -423,7 +423,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, memset(&template, 0, sizeof(template)); template.reg = e->reg; - template.mask = e->mask << e->shift_l; + template.mask = e->mask; template.shift = e->shift_l; template.off_val = snd_soc_enum_item_to_val(e, 0); template.on_val = template.off_val; @@ -546,8 +546,22 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; - if (data->widget) - data->widget->on_val = value; + if (data->widget) { + switch (dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + data->widget->on_val = value & data->widget->mask; + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + data->widget->on_val = value >> data->widget->shift; + break; + default: + data->widget->on_val = value; + break; + } + } data->value = value; base-commit: ecdcf8a833d770347115f739a37583cdc6ddbf7a -- 2.21.0

2 1

[PATCH 0/2] ASoC: qdsp6: fix default FE dais and routings.
by Srinivas Kandagatla 20 Apr '20

20 Apr '20

QDSP6 Frontend dais can be configured to work in rx or tx or both rx/tx mode, however the default routing do not honour this DT configuration making sound card fail to probe. FE dais are also not fully honouring device tree configuration. Fix both of them. Originally issue was reported by Vinod Koul Srinivas Kandagatla (2): ASoC: qdsp6: q6asm-dai: only enable dais from device tree ASoC: qdsp6: q6routing: remove default routing sound/soc/qcom/qdsp6/q6asm-dai.c | 30 +++++++++++++++++++++++------- sound/soc/qcom/qdsp6/q6routing.c | 19 ------------------- 2 files changed, 23 insertions(+), 26 deletions(-) -- 2.21.0

4 11

wm8960 codec driver question
by Alexey Shinkin 20 Apr '20

20 Apr '20

Hi, I hope this is the right place to ask . In the driver ( soc/codecs/wm8960.c ) there is a function that sets pll - the wm8960_set_pll() , line 1185 . On exit , the function turns on the pll clocking like this : /* Turn it on */ snd_soc_component_update_bits(component, WM8960_POWER2, 0x1, 0x1); msleep(250); snd_soc_component_update_bits(component, WM8960_CLOCK1, 0x1, 0x1); What is the reason for the 250 ms delay ? I did not find anything related in the chip datasheet . I found the delay very annoying in my app that generates an audible feedback to touchscreen events. Looks like it is safe to comment out the msleep() call , could there be any implications ? Best regards, Alex Shinkin

2 1

[PATCH] ALSA: usx2y: Fix potential NULL dereference
by Takashi Iwai 20 Apr '20

20 Apr '20

The error handling code in usX2Y_rate_set() may hit a potential NULL dereference when an error occurs before allocating all us->urb[]. Add a proper NULL check for fixing the corner case. Reported-by: Lin Yi <teroincn(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/usx2y/usbusx2yaudio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 37d290fe9d43..ecaf41265dcd 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -681,6 +681,8 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) us->submitted = 2*NOOF_SETRATE_URBS; for (i = 0; i < NOOF_SETRATE_URBS; ++i) { struct urb *urb = us->urb[i]; + if (!urb) + continue; if (urb->status) { if (!err) err = -ENODEV; -- 2.16.4

1 0

sound: usb: usx2y: is it a null pointer deference in function usX2Y_rate_set?
by 亿一 20 Apr '20

20 Apr '20

Hi, all: when reviewing function usX2Y_rate_set, here may exist a NULL pointer deference if kmalloc_array failed or usb_alloc_urb failed, in cleanup, we should judge whether us->urb[i] is NULL first. static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) { us = kzalloc(sizeof(*us) + sizeof(struct urb*) * NOOF_SETRATE_URBS, GFP_KERNEL); if (NULL == us) { err = -ENOMEM; goto cleanup; } usbdata = kmalloc_array(NOOF_SETRATE_URBS, sizeof(int), GFP_KERNEL); if (NULL == usbdata) { err = -ENOMEM; goto cleanup; } for (i = 0; i < NOOF_SETRATE_URBS; ++i) { if (NULL == (us->urb[i] = usb_alloc_urb(0, GFP_KERNEL))) { err = -ENOMEM; goto cleanup; } ... } ... cleanup: if (us) { us->submitted = 2*NOOF_SETRATE_URBS; for (i = 0; i < NOOF_SETRATE_URBS; ++i) { struct urb *urb = us->urb[i]; if (urb->status) { if (!err) err = -ENODEV; usb_kill_urb(urb); } usb_free_urb(urb);

2 2

PROBLEM: No sound on ASUS ROG Zenith Extreme in 5.6.5
by Johnathan Smithinovic 20 Apr '20

20 Apr '20

PROBLEM: No sound on ASUS ROG Zenith Extreme since 5.6.5 When I upgraded from 5.6.4 to 5.6.5 I noticed that my sound stopped working. My motherboard is a first generation "ASUS ROG Zenith Extreme". (I'm running Ubuntu 18.04 in case that matters.) I'm pretty sure it comes from stable kernel release commit 7f722d9aa00fde795834840cc84e13a94efb29ed (git://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git) which notes "commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab upstream." ("ALSA: hda: Add driver blacklist"). When I comment out 'SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0),' in sound/pci/hda/hda_intel.c my sound works again. (In case I chose incorrect places to write this mail to I'm sorry. I tried to follow https://www.kernel.org/doc/html/latest/admin-guide/reporting-bugs.html while keeping this report minimal.) Kind regards John

2 3

MIPS ioremap cleanups v2
by Christoph Hellwig 20 Apr '20

20 Apr '20

Hi Thomas, below is a bunch of cleanups for the MIPS ioremap code. Compile tested only. Changes since v1: - don't remove cpu_has_64bit_gp_regs - add a new prep patch to stop txx9aclc-ac97 from poking into ioremap internals

3 10

[PATCH 0/8] ASoC: add snd_compress_ops and replace
by Kuninori Morimoto 20 Apr '20

20 Apr '20

Hi Mark Current snd_soc_component_driver has compr_ops, but it is created for ALSA, not for ALSA SoC. This means it doesn't know the callback is for which component, thus, each callback needs to get component via lookup function. It is not good design, and will not work in the future, because ASoC can have multipul same name component which current lookup function can't handle. This patch-set adds new snd_compress_ops and call callbacks with component. Kuninori Morimoto (8): ASoC: soc-compress: add snd_compress_ops ASoC: codec: wm_adsp: use snd_compress_ops ASoC: uniphier: use snd_compress_ops ASoC: qcom: q6sp6: use snd_compress_ops ASoC: intel: atom: use snd_compress_ops ASoC: sof: use snd_compress_ops ASoC: sprd: use snd_compress_ops ASoC: soc-compress: remove snd_compr_ops include/sound/soc-component.h | 40 ++++++++- sound/soc/codecs/cs47l15.c | 9 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 9 +- sound/soc/codecs/cs47l85.c | 9 +- sound/soc/codecs/cs47l90.c | 9 +- sound/soc/codecs/cs47l92.c | 9 +- sound/soc/codecs/wm5102.c | 9 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm_adsp.c | 18 ++-- sound/soc/codecs/wm_adsp.h | 18 ++-- .../intel/atom/sst-mfld-platform-compress.c | 43 +++++---- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 51 ++++++----- sound/soc/soc-compress.c | 89 ++++++++++--------- sound/soc/sof/compress.c | 34 ++++--- sound/soc/sof/compress.h | 34 ++++--- sound/soc/sof/pcm.c | 4 +- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 49 +++++----- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sprd/sprd-pcm-dma.h | 2 +- sound/soc/uniphier/aio-compress.c | 45 ++++++---- sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/uniphier/aio.h | 2 +- 26 files changed, 299 insertions(+), 210 deletions(-) -- 2.17.1

2 10

[PATCH] ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos
by Takashi Iwai 20 Apr '20

20 Apr '20

TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need yet more quirks for the proper control names. This patch provides the mapping table for those boards, correcting the FU names for volume and mute controls as well as the terminal names for jack controls. It also improves build_connector_control() not to add the directional suffix blindly if the string is given from the mapping table. With this patch applied, the new UCM profiles will be effective. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 12 +++++++++--- sound/usb/mixer_maps.c | 24 +++++++++++++++++++++--- sound/usb/quirks-table.h | 14 ++++++++++++++ 3 files changed, 44 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e7b9040a54e6..ecd5036a0b44 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1776,8 +1776,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, { struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + const struct usbmix_name_map *map; - if (check_ignored_ctl(find_map(imap, term->id, 0))) + map = find_map(imap, term->id, 0); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1809,8 +1811,12 @@ static void build_connector_control(struct usb_mixer_interface *mixer, usb_mixer_elem_info_free(cval); return; } - get_connector_control_name(mixer, term, is_input, kctl->id.name, - sizeof(kctl->id.name)); + + if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) + strlcat(kctl->id.name, " Jack", sizeof(kctl->id.name)); + else + get_connector_control_name(mixer, term, is_input, kctl->id.name, + sizeof(kctl->id.name)); kctl->private_free = snd_usb_mixer_elem_free; snd_usb_mixer_add_control(&cval->head, kctl); } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index b4e77000f441..fb2c4a992951 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -369,6 +369,24 @@ static const struct usbmix_name_map asus_rog_map[] = { {} }; +/* TRX40 mobos with Realtek ALC1220-VB */ +static const struct usbmix_name_map trx40_mobo_map[] = { + { 18, NULL }, /* OT, IEC958 - broken response, disabled */ + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Speaker" }, /* OT */ + { 22, "Speaker Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 17, "Front Headphone" }, /* OT */ + { 23, "Front Headphone Playback" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + { 24, "IEC958 Playback" }, /* FU */ + {} +}; + /* * Control map entries */ @@ -500,7 +518,7 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), @@ -512,11 +530,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e009d584e7d0..7598d6d6740c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3635,4 +3635,18 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +#define ALC1220_VB_DESKTOP(vend, prod) { \ + USB_DEVICE(vend, prod), \ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + .vendor_name = "Realtek", \ + .product_name = "ALC1220-VB-DT", \ + .profile_name = "Realtek-ALC1220-VB-Desktop", \ + .ifnum = QUIRK_NO_INTERFACE \ + } \ +} +ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ +ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ +ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +#undef ALC1220_VB_DESKTOP + #undef USB_DEVICE_VENDOR_SPEC -- 2.16.4

1 0

[PATCH] ALSA: usb-audio: Print more information in stream proc files
by Takashi Iwai 20 Apr '20

20 Apr '20

For more debug and usability information, add the entry showing the DSD raw states and the channel mapping in each stream proc file. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/proc.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 4174ad11fca6..5a36e192ebb0 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -54,6 +54,38 @@ void snd_usb_audio_create_proc(struct snd_usb_audio *chip) proc_audio_usbid_read); } +static const char * const channel_labels[] = { + [SNDRV_CHMAP_NA] = "N/A", + [SNDRV_CHMAP_MONO] = "MONO", + [SNDRV_CHMAP_FL] = "FL", + [SNDRV_CHMAP_FR] = "FR", + [SNDRV_CHMAP_FC] = "FC", + [SNDRV_CHMAP_LFE] = "LFE", + [SNDRV_CHMAP_RL] = "RL", + [SNDRV_CHMAP_RR] = "RR", + [SNDRV_CHMAP_FLC] = "FLC", + [SNDRV_CHMAP_FRC] = "FRC", + [SNDRV_CHMAP_RC] = "RC", + [SNDRV_CHMAP_SL] = "SL", + [SNDRV_CHMAP_SR] = "SR", + [SNDRV_CHMAP_TC] = "TC", + [SNDRV_CHMAP_TFL] = "TFL", + [SNDRV_CHMAP_TFC] = "TFC", + [SNDRV_CHMAP_TFR] = "TFR", + [SNDRV_CHMAP_TRL] = "TRL", + [SNDRV_CHMAP_TRC] = "TRC", + [SNDRV_CHMAP_TRR] = "TRR", + [SNDRV_CHMAP_TFLC] = "TFLC", + [SNDRV_CHMAP_TFRC] = "TFRC", + [SNDRV_CHMAP_LLFE] = "LLFE", + [SNDRV_CHMAP_RLFE] = "RLFE", + [SNDRV_CHMAP_TSL] = "TSL", + [SNDRV_CHMAP_TSR] = "TSR", + [SNDRV_CHMAP_BC] = "BC", + [SNDRV_CHMAP_RLC] = "RLC", + [SNDRV_CHMAP_RRC] = "RRC", +}; + /* * proc interface for list the supported pcm formats */ @@ -97,6 +129,27 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); snd_iprintf(buffer, " Bits: %d\n", fp->fmt_bits); + + if (fp->dsd_raw) + snd_iprintf(buffer, " DSD raw: DOP=%d, bitrev=%d\n", + fp->dsd_dop, fp->dsd_bitrev); + + if (fp->chmap) { + const struct snd_pcm_chmap_elem *map = fp->chmap; + int c; + + snd_iprintf(buffer, " Channel map:"); + for (c = 0; c < map->channels; c++) { + if (map->map[c] >= sizeof(channel_labels) || + !channel_labels[map->map[c]]) + snd_iprintf(buffer, " --"); + else + snd_iprintf(buffer, " %s", + channel_labels[map->map[c]]); + } + snd_iprintf(buffer, "\n"); + } + // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } -- 2.16.4

1 0

No Sound Realtek ALC294ASUS UX534FT
by LACOUR François 19 Apr '20

19 Apr '20

Hi, I can't find a issue to this problem. Have you ? Thanks François Sent with [ProtonMail](https://protonmail.com) Secure Email.

1 0

[PATCH] ALSA: hda: Remove ASUS ROG Zenith from the blacklist
by Takashi Iwai 19 Apr '20

19 Apr '20

The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. This patch reverts the corresponding entry as a temporary solution. Although Zenith II and co will see get the empty HD-audio bus again, it'd be merely resource wastes and won't affect the functionality, so it's no end of the world. We'll need to address this later, e.g. by either switching to DMI string matching or using PCI ID & SSID pairs. Fixes: 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") Reported-by: Johnathan Smithinovic <johnathan.smithinovic(a)gmx.at> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a5fab12defde..d7adae316c0d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2092,7 +2092,6 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * should be ignored from the beginning. */ static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), {} -- 2.16.4

1 0

[PATCH 0/7] Add H6 I2S support
by Clément Péron 19 Apr '20

19 Apr '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 286 ++++++++++++++++-- 3 files changed, 295 insertions(+), 24 deletions(-) -- 2.20.1

1 8

[PATCH v2] ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
by Alexander Tsoy 18 Apr '20

18 Apr '20

Many Focusrite devices supports a limited set of sample rates per altsetting. These includes audio interfaces with ADAT ports: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - Scarlett 18i8 3rd gen, 18i20 3rd gen; - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Maximum rate is exposed in the last 4 bytes of Format Type descriptor which has a non-standard bLength = 10. Tested-by: Alexey Skobkin <skobkin-ru(a)ya.ru> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- Notes: v2: updated description sound/usb/format.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50e1874c847c..5ffb457cc88c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -277,6 +277,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, return false; } +/* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) -- 2.25.3

2 1

[PATCH] ALSA: hda/realtek - Fix unexpected init_amp override
by Takashi Iwai 18 Apr '20

18 Apr '20

The commit 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") changed the way to assign spec->init_amp field that specifies the way to initialize the amp. Along with the change, the commit also replaced a few fixups that set spec->init_amp in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather aligning to the other fixups, and not supposed to change the actual behavior. However, this change turned out to cause a regression on FSC S7020, which hit exactly the above. The reason was that there is still one place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE call, namely in alc_ssid_check(). This patch fixes the regression by adding the proper spec->init_amp override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED. Fixes: 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc5557d79c43..54e1c9c0a33f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -797,9 +797,11 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) { if (!alc_subsystem_id(codec, ports)) { struct alc_spec *spec = codec->spec; - codec_dbg(codec, - "realtek: Enable default setup for auto mode as fallback\n"); - spec->init_amp = ALC_INIT_DEFAULT; + if (spec->init_amp == ALC_INIT_UNDEFINED) { + codec_dbg(codec, + "realtek: Enable default setup for auto mode as fallback\n"); + spec->init_amp = ALC_INIT_DEFAULT; + } } } -- 2.25.0

1 0

[PATCH] ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
by Alexander Tsoy 18 Apr '20

18 Apr '20

Many Focusrite devices supports a limited set of sample rates per altsetting. This includes interfaces with ADAT ports and interfaces with dedicated feedback channels: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - all Scarlett 3rd gen devices (feedback channels works only up to 96 kHz); - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Tested-by: Alexey Skobkin <skobkin-ru(a)ya.ru> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/format.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50e1874c847c..5ffb457cc88c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -277,6 +277,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, return false; } +/* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) -- 2.25.3

1 1

[PATCH trivial 0/6] Fix misspellings of "Analog Devices"
by Geert Uytterhoeven 18 Apr '20

18 Apr '20

Hi all, In several files the company also known as ADI is spelled as "Analog Device". However, according to https://www.analog.com/, the company name is spelled "Analog Devices". Hence this patch series, one per subsystem, fixes these misspellings. Thanks for your comments! Geert Uytterhoeven (6): dt-bindings: Fix misspellings of "Analog Devices" dma: Fix misspelling of "Analog Devices" drm: Fix misspellings of "Analog Devices" iio: Fix misspellings of "Analog Devices" ALSA: Fix misspellings of "Analog Devices" ASoC: Fix misspellings of "Analog Devices" .../devicetree/bindings/display/bridge/adi,adv7123.txt | 4 ++-- .../devicetree/bindings/display/bridge/adi,adv7511.txt | 4 ++-- Documentation/devicetree/bindings/dma/adi,axi-dmac.txt | 2 +- Documentation/devicetree/bindings/iio/dac/ad5755.txt | 2 +- drivers/dma/Kconfig | 2 +- drivers/gpu/drm/bridge/adv7511/Kconfig | 2 +- drivers/gpu/drm/drm_fb_cma_helper.c | 2 +- drivers/gpu/drm/tegra/fb.c | 2 +- drivers/iio/adc/ad7791.c | 2 +- drivers/iio/trigger/iio-trig-hrtimer.c | 2 +- drivers/staging/iio/Documentation/overview.txt | 2 +- sound/isa/ad1816a/ad1816a.c | 2 +- sound/pci/ac97/ac97_patch.c | 2 +- sound/pci/hda/Kconfig | 4 ++-- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 2 +- sound/soc/codecs/wm8782.c | 2 +- 17 files changed, 20 insertions(+), 20 deletions(-) -- 2.17.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

8 14

[PATCH AUTOSEL 4.4 19/19] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-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_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eacf..7ccfb09535e14 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -310,8 +310,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.4 18/19] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2697402b51957..41f6450a2539f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -569,7 +569,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -583,7 +583,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 4.4 08/19] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index d55388e082e18..43c960bb4b211 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -963,7 +963,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

1 0

[PATCH AUTOSEL 4.4 06/19] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7cffa98ec313c..d4bf3dc6b0158 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1951,7 +1951,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -1981,7 +1982,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

1 0

[PATCH AUTOSEL 4.9 22/23] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-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_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eacf..7ccfb09535e14 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -310,8 +310,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.9 21/23] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2697402b51957..41f6450a2539f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -569,7 +569,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -583,7 +583,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 4.9 09/23] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0838478c4c3f9..92aa1518b7723 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -976,7 +976,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

1 0

[PATCH AUTOSEL 4.9 07/23] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9df0c8102dc03..b67d105b76e46 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2062,7 +2062,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2092,7 +2093,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 26/28] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-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_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eacf..7ccfb09535e14 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -310,8 +310,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 25/28] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2697402b51957..41f6450a2539f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -569,7 +569,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -583,7 +583,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 10/28] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0f3604b559424..b04853bdaf377 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -974,7 +974,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 08/28] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e75822dd9930f..fd4b71729eedd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2048,7 +2048,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2078,7 +2079,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 36/47] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e58240e18b301..f29014a7d6723 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -588,6 +588,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 34/47] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-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_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 066b5b59c4d7c..0224011a240fb 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -297,8 +297,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 33/47] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index c97b5528e4b80..317bbb725b299 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -564,7 +564,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 30/47] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 23aab2fdac468..72d3b74a42cbb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2444,6 +2444,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

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[PATCH AUTOSEL 4.19 12/47] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 3672d36b4b66f..d32d48e82a2af 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -974,7 +974,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

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[PATCH AUTOSEL 4.19 10/47] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 356d4e754561b..a0d1ce0edaf9a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2266,7 +2266,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2296,7 +2297,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

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[PATCH AUTOSEL 5.4 68/78] ASoC: SOF: trace: fix unconditional free in trace release
by Sasha Levin 18 Apr '20

18 Apr '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit e6110114d18d330c05fd6de9f31283fd086a5a3a ] Check if DMA pages were successfully allocated in initialization before calling free. For many types of memory (like sgbufs) the extra free is harmless, but not all backends track allocation state, so add an explicit check. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200124213625.30186-5-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/trace.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index 4c3cff031fd66..fd6f5913782bf 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -328,7 +328,10 @@ void snd_sof_free_trace(struct snd_sof_dev *sdev) { snd_sof_release_trace(sdev); - snd_dma_free_pages(&sdev->dmatb); - snd_dma_free_pages(&sdev->dmatp); + if (sdev->dma_trace_pages) { + snd_dma_free_pages(&sdev->dmatb); + snd_dma_free_pages(&sdev->dmatp); + sdev->dma_trace_pages = 0; + } } EXPORT_SYMBOL(snd_sof_free_trace); -- 2.20.1

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[PATCH AUTOSEL 5.4 59/78] ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ca707b3f00b4f31a6e1eb37e8ae99f15f2bb1fe5 ] The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable(a)vger.kernel.org Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 190c1aa6a4e62..7c327a37eba10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7300,6 +7300,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.20.1

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[PATCH AUTOSEL 5.4 58/78] ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk
by Sasha Levin 18 Apr '20

18 Apr '20

From: František Kučera <franta-linux(a)frantovo.cz> [ Upstream commit 73d8c94084341e2895169a0462dbc18167f01683 ] Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE. Input is not working. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db0..dcaf9eed9a415 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.20.1

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[PATCH AUTOSEL 5.4 56/78] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 243f683bc02a7..e62e1d7815aa9 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

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[PATCH AUTOSEL 5.4 54/78] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-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_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index b7d9160ed8688..c6e1e03a5e4da 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -290,8 +290,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

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[PATCH AUTOSEL 5.4 53/78] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 9d71e9d5c9a06..3cf41c11a4050 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -536,7 +536,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

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[PATCH AUTOSEL 5.4 49/78] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 128db2e6bc64e..190c1aa6a4e62 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

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[PATCH AUTOSEL 5.4 48/78] ALSA: hda: Add driver blacklist
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab ] The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-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_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 85beb172d8108..1db9d0579c72a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2024,6 +2024,17 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #endif } +/* Blacklist for skipping the whole probe: + * some HD-audio PCI entries are exposed without any codecs, and such devices + * should be ignored from the beginning. + */ +static const struct snd_pci_quirk driver_blacklist[] = { + SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), + SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), + SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), + {} +}; + static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, .pcm_mmap_prepare = pcm_mmap_prepare, @@ -2059,6 +2070,11 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; + if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + dev_info(&pci->dev, "Skipping the blacklisted device\n"); + return -ENODEV; + } + if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { -- 2.20.1

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[PATCH AUTOSEL 5.4 19/78] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f19..2c3798034b1de 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -966,7 +966,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

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[PATCH AUTOSEL 5.4 16/78] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d978df95c5c65..cc4e9aa80fb0d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2222,7 +2222,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2252,7 +2253,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

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[PATCH AUTOSEL 5.5 70/75] ASoC: SOF: trace: fix unconditional free in trace release
by Sasha Levin 18 Apr '20

18 Apr '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit e6110114d18d330c05fd6de9f31283fd086a5a3a ] Check if DMA pages were successfully allocated in initialization before calling free. For many types of memory (like sgbufs) the extra free is harmless, but not all backends track allocation state, so add an explicit check. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200124213625.30186-5-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/trace.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index b0e4556c8536a..5bf63fa67fb09 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -343,7 +343,10 @@ void snd_sof_free_trace(struct snd_sof_dev *sdev) snd_sof_release_trace(sdev); - snd_dma_free_pages(&sdev->dmatb); - snd_dma_free_pages(&sdev->dmatp); + if (sdev->dma_trace_pages) { + snd_dma_free_pages(&sdev->dmatb); + snd_dma_free_pages(&sdev->dmatp); + sdev->dma_trace_pages = 0; + } } EXPORT_SYMBOL(snd_sof_free_trace); -- 2.20.1

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[PATCH AUTOSEL 5.5 64/75] ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ca707b3f00b4f31a6e1eb37e8ae99f15f2bb1fe5 ] The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable(a)vger.kernel.org Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62eac992ebd55..e2c77d2ab26e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7300,6 +7300,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.20.1

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[PATCH AUTOSEL 5.5 63/75] ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk
by Sasha Levin 18 Apr '20

18 Apr '20

From: František Kučera <franta-linux(a)frantovo.cz> [ Upstream commit 73d8c94084341e2895169a0462dbc18167f01683 ] Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE. Input is not working. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db0..dcaf9eed9a415 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.20.1

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[PATCH AUTOSEL 5.5 61/75] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 243f683bc02a7..e62e1d7815aa9 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

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[PATCH AUTOSEL 5.5 59/75] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-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_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index b7d9160ed8688..c6e1e03a5e4da 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -290,8 +290,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

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[PATCH AUTOSEL 5.5 58/75] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 9d71e9d5c9a06..3cf41c11a4050 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -536,7 +536,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

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[PATCH AUTOSEL 5.5 54/75] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f44d8e2589337..62eac992ebd55 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

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[PATCH AUTOSEL 5.5 53/75] ALSA: hda: Add driver blacklist
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab ] The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-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_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9d3b28bcba9b4..d2d5a19bdecce 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2074,6 +2074,17 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #endif } +/* Blacklist for skipping the whole probe: + * some HD-audio PCI entries are exposed without any codecs, and such devices + * should be ignored from the beginning. + */ +static const struct snd_pci_quirk driver_blacklist[] = { + SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), + SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), + SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), + {} +}; + static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, .pcm_mmap_prepare = pcm_mmap_prepare, @@ -2090,6 +2101,11 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; + if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + dev_info(&pci->dev, "Skipping the blacklisted device\n"); + return -ENODEV; + } + if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { -- 2.20.1

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[PATCH AUTOSEL 5.5 20/75] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f19..2c3798034b1de 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -966,7 +966,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

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[PATCH AUTOSEL 5.5 17/75] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8de29f48442f6..e4ff0796526c7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2256,7 +2256,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2286,7 +2287,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

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[PATCH AUTOSEL 5.6 68/73] ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ca707b3f00b4f31a6e1eb37e8ae99f15f2bb1fe5 ] The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable(a)vger.kernel.org Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5cced684cf41f..3f3484455080d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7300,6 +7300,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.20.1

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[PATCH AUTOSEL 5.6 67/73] ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk
by Sasha Levin 18 Apr '20

18 Apr '20

From: František Kučera <franta-linux(a)frantovo.cz> [ Upstream commit 73d8c94084341e2895169a0462dbc18167f01683 ] Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE. Input is not working. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db0..dcaf9eed9a415 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.20.1

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[PATCH AUTOSEL 5.6 65/73] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6bd9ae813be28..d14d5f7db1689 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

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[PATCH AUTOSEL 5.6 63/73] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-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_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index f5fd62ed4df5b..841523f6b88de 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -290,8 +290,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

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[PATCH AUTOSEL 5.6 62/73] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 91f83cef0e56c..9aa12a67d3705 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -536,7 +536,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

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[PATCH AUTOSEL 5.6 58/73] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63e1a56f705b7..5cced684cf41f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

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[PATCH AUTOSEL 5.6 57/73] ALSA: hda: Add driver blacklist
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab ] The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-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_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 92a042e34d3e5..bd093593f8fbd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2076,6 +2076,17 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #endif } +/* Blacklist for skipping the whole probe: + * some HD-audio PCI entries are exposed without any codecs, and such devices + * should be ignored from the beginning. + */ +static const struct snd_pci_quirk driver_blacklist[] = { + SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), + SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), + SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), + {} +}; + static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, .pcm_mmap_prepare = pcm_mmap_prepare, @@ -2092,6 +2103,11 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; + if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + dev_info(&pci->dev, "Skipping the blacklisted device\n"); + return -ENODEV; + } + if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { -- 2.20.1

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[PATCH AUTOSEL 5.6 22/73] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f19..2c3798034b1de 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -966,7 +966,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

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[PATCH AUTOSEL 5.6 19/73] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca01..8f6f0ad502880 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

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[PATCH AUTOSEL 5.6 18/73] ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag
by Sasha Levin 18 Apr '20

18 Apr '20

From: Stephan Gerhold <stephan(a)gerhold.net> [ Upstream commit 7f2430cda819a9ecb1df5a0f3ef4f1c20db3f811 ] At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200330175210.47518-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c0d422d0ab94f..d7dc80ede8927 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -73,7 +73,7 @@ struct q6asm_dai_data { }; static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -95,7 +95,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { }; static struct snd_pcm_hardware q6asm_dai_hardware_playback = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | -- 2.20.1

1 0

State of hda-analyser.py
by Moosa Mahsoom 18 Apr '20

18 Apr '20

Hi, I am a final year student from BITS Pilani, India. I have been facing some issues with my laptop audio and hence wanted to use the hda_analyser script. However, I noticed that the python script doesn't seem to compatible with the current pygtk libraries. If possible,I would like to contribute towards bringing the scripts to parity. Yours Sincerely, Moosa Mahsoom

1 0

Bug WD19 dockstation
by Gregorio Rodrigo 17 Apr '20

17 Apr '20

Hi everyone, I am not sure if this is the proper way to submit a bug/ticket/question. I posted a message here https://github.com/edrose/dell-dock-audio-fix/issues/2#issuecomment-6152907… and got the suggestion to write to the alsa mail list. I have a WD19 docking station and the lineout is not recognized. It seems it is recognized as an input. However it works if I run the speaker-test command. Anyone knows if I can apply those UCM settings or how could I do it? I copy here the message. I have a WD19 docking station and i am facing the same problem. Debian GNU/Linux bullseye/sid \n \l Linux xps 5.5.0-1-amd64 #1 <https://github.com/edrose/dell-dock-audio-fix/issues/1> SMP Debian 5.5.13-2 (2020-03-30) x86_64 GNU/Linux pactl list cards Card #1 <https://github.com/edrose/dell-dock-audio-fix/issues/1> Name: alsa_card.usb-Generic_USB_Audio_200901010001-00 Driver: module-alsa-card.c Owner Module: 25 Properties: alsa.card = "2" alsa.card_name = "WD19 Dock" alsa.long_card_name = "Dell-WD15-Dock" alsa.driver_name = "snd_usb_audio" device.bus_path = "pci-0000:00:14.0-usb-0:9.3.4:1.0" sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb3/3-9/3-9.3/3-9.3.4/3-9.3.4:1.0/sound/card2" udev.id = "usb-Generic_USB_Audio_200901010001-00" device.bus = "usb" device.vendor.id = "0bda" device.vendor.name = "Realtek Semiconductor Corp." device.product.id = "402e" device.product.name = "USB Audio" device.serial = "Generic_USB_Audio_200901010001" device.string = "2" device.description = "USB Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-usb" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo+input:analog-stereo Ports: analog-input-mic: Microphone (priority: 8700, latency offset: 0 usec) Properties: device.icon_name = "audio-input-microphone" Part of profile(s): input:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-headphones: Headphones (priority: 9900, latency offset: 0 usec) Properties: device.icon_name = "audio-headphones" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo http://alsa-project.org/db/?f=616054544b8748431c354f1affbfd705c88fad37 If i run manualy speaker-test and select the proper subdevice, i get sound over the lineoutput. I cannot set it as default (or i dont know how to do it). It seems that if i am able to set the UCM2 configuration from WD15 should work. Any advise? Thank you in advance.

1 0

[PATCH] ASoC: stm32: sai: fix sai probe
by Olivier Moysan 17 Apr '20

17 Apr '20

pcm config must be set before snd_dmaengine_pcm_register() call. Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 34a7c3d6fb91..41f01c3e639e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1547,6 +1547,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + conf = &stm32_sai_pcm_config_spdif; + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { if (ret != -EPROBE_DEFER) @@ -1556,15 +1559,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) { + if (ret) snd_dmaengine_pcm_unregister(&pdev->dev); - return ret; - } - - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - conf = &stm32_sai_pcm_config_spdif; - return 0; + return ret; } static int stm32_sai_sub_remove(struct platform_device *pdev) -- 2.17.1

2 1

[PATCH] ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
by Pierre-Louis Bossart 17 Apr '20

17 Apr '20

Major regressions were detected by SOF CI on CherryTrail and Broadwell: [ 25.705750] SSP2-Codec: ASoC: no backend playback stream [ 27.923378] SSP2-Codec: ASoC: no users playback at close - state This is root-caused to the introduction of the DAI capability checks with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a requirement for all DAIs to report at least a non-zero min_channels field. For some reason the SSP structures used for SKL+ did provide this information but legacy platforms didn't. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/bdw.c | 16 +++++++++++++ sound/soc/sof/intel/byt.c | 48 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 6c23c5769330..a32a3ef78ec5 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -567,9 +567,25 @@ static void bdw_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver bdw_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index f84391294f12..29fd1d86156c 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -459,21 +459,69 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver byt_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp2-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + } }, { .name = "ssp3-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp4-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp5-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; -- 2.20.1

2 1

[PATCH 0/2] ASoC: rsnd: Fixes for multichannel HDMI audio output
by Matthias Blankertz 17 Apr '20

17 Apr '20

This fixes two issues in the snd-soc-rcar driver blocking multichannel HDMI audio out: The parent SSI in a multi-SSI configuration is not correctly set up and started, and the SSI->HDMI channel mapping is wrong. With these patches, the following device tree snippet can be used on an r8a7795-based platform (Salvator-X) to enable multichannel HDMI audio on HDMI0: rsnd_port1: port@1 { rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; dai-format = "i2s"; bitclock-master = <&rsnd_endpoint1>; frame-master = <&rsnd_endpoint1>; playback = <&ssi0 &ssi1 &ssi2 &ssi9>; }; }; With a capable receiver attached, all of 2ch (stereo), 6ch (e.g. 5.1) and 8ch audio output should work. Matthias Blankertz (2): ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode sound/soc/sh/rcar/ssi.c | 8 ++++---- sound/soc/sh/rcar/ssiu.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) base-commit: 7111951b8d4973bda27ff663f2cf18b663d15b48 -- 2.26.0

3 6

Question about snd_soc_card_register()
by Sridharan, Ranjani 17 Apr '20

17 Apr '20

Hi Takashi, While working on implementing the probes features in SOF using a separate card for the probe DAI links, I noticed that calling snd_soc_register_card() results in incrementing the usage_count for the device that registers the card by 2 and it is not decremented until the card is freed. Is this the expected behaviour? Typically, we register a separate platform device for the Intel machines which in turn register the card and none of them ever enable runtime PM. So this has no impact on the parent device's runtime PM status. I'd like to avoid creating a separate platform device just to register the card if possible while also enabling runtime PM . But when I do this today, the device cannot enter runtime suspend at all. Could you please shed some light on this? Thanks, Ranjani

3 6

Re: next/master bisection: baseline.dmesg.alert on meson-sm1-sei610
by Mark Brown 17 Apr '20

17 Apr '20

On Fri, Apr 17, 2020 at 02:53:36AM -0700, kernelci.org bot wrote: > next/master bisection: baseline.dmesg.alert on meson-sm1-sei610 > Summary: > Start: a3ca59b9af21e Add linux-next specific files for 20200416 > Plain log: https://storage.kernelci.org//next/master/next-20200416/arm64/defconfig+CON… > HTML log: https://storage.kernelci.org//next/master/next-20200416/arm64/defconfig+CON… > Result: 9b5db059366ae ASoC: soc-pcm: dpcm: Only allow playback/capture if supported This change to check the playback and capture constraints is also causing issues on at least Meson - I'm a bit worried that this is also causing oopses here, not just audio problems so copying in Morimoto-san too. We should fix the things that are broken by this change since it's a thing we should be able to rely on and do enforce for non-DPCm links but I'm wondering if we should revert for 5.7 and reapply for 5.8. Including complete report so people have it. > Checks: > revert: PASS > verify: PASS > Parameters: > Tree: next > URL: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git > Branch: master > Target: meson-sm1-sei610 > CPU arch: arm64 > Lab: lab-baylibre > Compiler: gcc-8 > Config: defconfig+CONFIG_RANDOMIZE_BASE=y > Test case: baseline.dmesg.alert > > Breaking commit found: > > ------------------------------------------------------------------------------- > commit 9b5db059366ae2087e07892b5fc108f81f4ec189 > Author: Stephan Gerhold <stephan(a)gerhold.net> > Date: Wed Apr 15 12:49:28 2020 +0200 > > ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > > At the moment, PCM devices for DPCM are only created based on the > dpcm_playback/capture parameters of the DAI link, without considering > if the CPU/FE DAI is actually capable of playback/capture. > > Normally the dpcm_playback/capture parameter should match the > capabilities of the CPU DAI. However, there is no way to set that > parameter from the device tree (e.g. with simple-audio-card or > qcom sound cards). dpcm_playback/capture are always both set to 1. > > This causes problems when the CPU DAI does only support playback > or capture. Attemting to open that PCM device with an unsupported > stream type then results in a null pointer dereference: > > Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 > Internal error: Oops: 96000044 [#1] PREEMPT SMP > CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 > pc : invalidate_paths_ep+0x30/0xe0 > lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 > Call trace: > invalidate_paths_ep+0x30/0xe0 > snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 > dpcm_path_get+0x38/0xd0 > dpcm_fe_dai_open+0x70/0x920 > snd_pcm_open_substream+0x564/0x840 > snd_pcm_open+0xfc/0x228 > snd_pcm_capture_open+0x4c/0x78 > snd_open+0xac/0x1a8 > ... > > ... because the DAI playback/capture_widget is not set in that case. > > We could add checks there to fix the problem (maybe we should > anyway), but much easier is to not expose the device as > playback/capture in the first place. Attemting to use that > device would always fail later anyway. > > Add checks for snd_soc_dai_stream_valid() to the DPCM case > to avoid exposing playback/capture if it is not supported. > > Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> > Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c > index 289aebc155293..1f302de440525 100644 > --- a/sound/soc/soc-pcm.c > +++ b/sound/soc/soc-pcm.c > @@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) > int i; > > if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { > - playback = rtd->dai_link->dpcm_playback; > - capture = rtd->dai_link->dpcm_capture; > + cpu_dai = asoc_rtd_to_cpu(rtd, 0); > + if (rtd->num_cpus > 1) { > + dev_err(rtd->dev, > + "DPCM doesn't support Multi CPU yet\n"); > + return -EINVAL; > + } > + > + playback = rtd->dai_link->dpcm_playback && > + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); > + capture = rtd->dai_link->dpcm_capture && > + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); > } else { > /* Adapt stream for codec2codec links */ > int cpu_capture = rtd->dai_link->params ? > ------------------------------------------------------------------------------- > > > Git bisection log: > > ------------------------------------------------------------------------------- > git bisect start > # good: [87b0f983f66f23762921129fd35966eddc3f2dae] net: mscc: ocelot: fix untagged packet drops when enslaving to vlan aware bridge > git bisect good 87b0f983f66f23762921129fd35966eddc3f2dae > # bad: [a3ca59b9af21e68069555ffff1ad89bd2a7c40fc] Add linux-next specific files for 20200416 > git bisect bad a3ca59b9af21e68069555ffff1ad89bd2a7c40fc > # bad: [feb09551bfe34ccf0ba462188a1aee651be0f2c3] Merge remote-tracking branch 'i2c/i2c/for-next' > git bisect bad feb09551bfe34ccf0ba462188a1aee651be0f2c3 > # good: [a4721ced760684d1776bf31f7925aa41bb3f4846] Merge v5.7-rc1 into drm-misc-fixes > git bisect good a4721ced760684d1776bf31f7925aa41bb3f4846 > # bad: [e75043c693af6a10c9e2087adeef243cf05ce3bd] Merge remote-tracking branch 'rockchip/for-next' > git bisect bad e75043c693af6a10c9e2087adeef243cf05ce3bd > # bad: [08a315860a1b462fef5969d2e6ab4cdab26df8e3] Merge remote-tracking branch 'arm-soc/for-next' > git bisect bad 08a315860a1b462fef5969d2e6ab4cdab26df8e3 > # bad: [80dbfa57508c36c2b3798b387ef3eb92954b9c09] Merge remote-tracking branch 'slave-dma-fixes/fixes' > git bisect bad 80dbfa57508c36c2b3798b387ef3eb92954b9c09 > # good: [a723380c6f0a3723db72da6c6097b32b23f5564c] Merge remote-tracking branch 'sound-current/for-linus' > git bisect good a723380c6f0a3723db72da6c6097b32b23f5564c > # bad: [5bd70440cb0a6f5c6a84019bb2aa93ab8310a5cd] ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence > git bisect bad 5bd70440cb0a6f5c6a84019bb2aa93ab8310a5cd > # good: [4d1a015a203c0249e3332ea217a38ec978118daa] ASoC: convert rockchip spdif bindings to yaml > git bisect good 4d1a015a203c0249e3332ea217a38ec978118daa > # good: [0f2a3b02274c02eb97697c4d89c019d1d21ac225] ASoC: wsa881x: mark read_only_wordlength flag > git bisect good 0f2a3b02274c02eb97697c4d89c019d1d21ac225 > # good: [aa7812737f2877e192d57626cbe8825cc7cf6de9] ASoC: sgtl5000: Fix VAG power-on handling > git bisect good aa7812737f2877e192d57626cbe8825cc7cf6de9 > # bad: [0c824ec094b5cda766c80d88c2036e28c24a4cb1] ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs > git bisect bad 0c824ec094b5cda766c80d88c2036e28c24a4cb1 > # bad: [9b5db059366ae2087e07892b5fc108f81f4ec189] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > git bisect bad 9b5db059366ae2087e07892b5fc108f81f4ec189 > # first bad commit: [9b5db059366ae2087e07892b5fc108f81f4ec189] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > -------------------------------------------------------------------------------

3 4

[PATCH] ASoC: Intel: atom: remove unused 'sst_mix_voip_controls'
by Jason Yan 17 Apr '20

17 Apr '20

Fix the following gcc warning: sound/soc/intel/atom/sst-atom-controls.c:710:30: warning: ‘sst_mix_voip_controls’ defined but not used [-Wunused-const-variable=] SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls); ^ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/intel/atom/sst-atom-controls.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 69f3af4524ab..541f0059ba7a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -707,7 +707,6 @@ SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls); -SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_modem_controls); -- 2.21.1

2 1

[PATCH v2] ASoC: bdw-rt5650: remove 3-channel capture support
by Brent Lu 17 Apr '20

17 Apr '20

Implement a constrain to exclude 3-channel capture since only 2 and 4 channel capture are supported on the platform. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5650.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index af2f502..eedbdad 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -83,6 +83,36 @@ static struct snd_soc_jack_pin mic_jack_pin = { .mask = SND_JACK_MICROPHONE, }; +static const unsigned int channels[] = { + 2, 4, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int bdw_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 2 or 4 channel capture + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return 0; +} + +static const struct snd_soc_ops bdw_rt5650_fe_ops = { + .startup = bdw_fe_startup, +}; + static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -234,6 +264,7 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = { .name = "System PCM", .stream_name = "System Playback", .dynamic = 1, + .ops = &bdw_rt5650_fe_ops, #if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) .init = bdw_rt5650_rtd_init, #endif -- 2.7.4

4 6

[PATCH alsa-lib] conf: USB-Audio: Fix S/PDIF output of ASUS Xonar AE
by Ivan Mironov 17 Apr '20

17 Apr '20

ASUS Xonar AE is a PCI-Express card containing USB controller: USB controller [0c03]: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller [1b21:1142] and the actual USB sound card: ID 0b05:180f ASUSTek Computer, Inc. XONAR SOUND CARD As other Xonar USB sound cards, it uses second device for digital output. Signed-off-by: Ivan Mironov <mironov.ivan(a)gmail.com> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 89811086db1d..3f4c421b7e2c 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -40,6 +40,7 @@ USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 "Xonar U7" 1 "ASUS XONAR U5" 1 + "XONAR SOUND CARD" 1 # The below don't have digital in/out, so prevent them from being opened. "Andrea PureAudio USB-SA Headset" 999 -- 2.25.1

2 1

alsa / dma: Sound stops playing after xrun
by Martin Fuzzey 17 Apr '20

17 Apr '20

Hi all, I am seeing a sound problem leading to no more sound after an XRUN . I see more or less what is happening and have a work around but am not entirely sure of the correct solution. The problem occurs on an i.MX6DL using mainline kernel 5.4 with the fsl_ssi driver and a SGTL5000 audio codec connected over I2S. The userspace is Android 8 using tinnyhal + tinyalsa. This uses ioctl to push samples rather than mmap. The scenario is: 1) A buffer underrun occurs, causing -EPIPE to be returned to userspace. 2) Userpsapce (the tinyalsa component) does a pcm_prepare() to recover (which completes OK). 3) Userspace starts sending data again and then, when the start threshold is exceeded this kernel log is generated fsl-ssi-dai 2028000.ssi: Timeout waiting TX FIFO filling 4) From this point on no more sound is played, further writes timeout after 10s and return -EIO. My analsysis is as follows: When the underrun occurs snd_dmaengine_pcm_trigger(SNDRV_PCM_TRIGGER_STOP) is performed which does dmaengine_terminate_async() When the stream is restarted snd_dmaengine_pcm_trigger(SNDRV_PCM_TRIGGER_START does dmaengine_submit() dma_async_issue_pending() Because dmaengine_terminate_async() is asynchronus it sometimes completes after the dmaengine_submit(), causing the new DMA request to be cancelled before it has started. This results in the "Timeout waiting TX FIFO filling" message (in fsl_ssi_config_enable()) because it enables the SSI and expects data to be transfered to the FIFO by DMA. The message is only a warning, (void function with no error return) So without DMA running the buffer quicky fills up, resulting in all future writes timeouting waiting for buffer space. Where I'm not sure is if this is an ALSA bug or a bug in the i.MX6 SDMA controller driver. IE is it OK to do dmaengine_terminate_async() and later dmaengine_submit() on the same DMA channel without waiting for the terminate to complete? My workaround is to wait for the terminate to complete before returning -EPIPE (just after releasing the lock which prevents sleeping earlier). Ie this: diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2236b5e..b03dac3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -18,6 +18,9 @@ #include <sound/pcm_params.h> #include <sound/timer.h> +#include <linux/dmaengine.h> +#include <sound/dmaengine_pcm.h> + #include "pcm_local.h" #ifdef CONFIG_SND_PCM_XRUN_DEBUG @@ -2239,6 +2242,10 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, } } _end_unlock: + /* MF: Workaround for broken sound after XRUN. */ + if (err == -EPIPE) + dmaengine_synchronize(snd_dmaengine_pcm_get_chan(substream)); + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); Regards, Martin

1 0

[PATCH] slimbus: ngd: remove redundant assignment
by Srinivas Kandagatla 17 Apr '20

17 Apr '20

remove redundant assignment, as this is already done fewlines lines before. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 7426b5884218..743ee7b4e63f 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1365,7 +1365,6 @@ static int of_qcom_slim_ngd_register(struct device *parent, platform_device_add(ngd->pdev); ngd->base = ctrl->base + ngd->id * data->offset + (ngd->id - 1) * data->size; - ctrl->ngd = ngd; return 0; } -- 2.21.0

1 0

[PATCH] slimbus: ngd: get drvdata from correct device
by Srinivas Kandagatla 17 Apr '20

17 Apr '20

Get drvdata directly from parent instead of ngd dev, as ngd dev can probe defer and previously set drvdata will become null. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index fc2575fef51b..7426b5884218 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1361,7 +1361,6 @@ static int of_qcom_slim_ngd_register(struct device *parent, ngd->pdev->driver_override = QCOM_SLIM_NGD_DRV_NAME; ngd->pdev->dev.of_node = node; ctrl->ngd = ngd; - platform_set_drvdata(ngd->pdev, ctrl); platform_device_add(ngd->pdev); ngd->base = ctrl->base + ngd->id * data->offset + @@ -1376,12 +1375,13 @@ static int of_qcom_slim_ngd_register(struct device *parent, static int qcom_slim_ngd_probe(struct platform_device *pdev) { - struct qcom_slim_ngd_ctrl *ctrl = platform_get_drvdata(pdev); struct device *dev = &pdev->dev; + struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev->parent); int ret; ctrl->ctrl.dev = dev; + platform_set_drvdata(pdev, ctrl); pm_runtime_use_autosuspend(dev); pm_runtime_set_autosuspend_delay(dev, QCOM_SLIM_NGD_AUTOSUSPEND); pm_runtime_set_suspended(dev); -- 2.21.0

1 0

[PATCH v2] slimbus: core: Set fwnode for a device when setting of_node
by Saravana Kannan 17 Apr '20

17 Apr '20

When setting the of_node for a newly created device, also set the fwnode. This allows fw_devlink feature to work for slimbus devices. Signed-off-by: Saravana Kannan <saravanak(a)google.com> --- drivers/slimbus/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 526e3215d8fe..44228a5b246d 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -163,8 +163,10 @@ static int slim_add_device(struct slim_controller *ctrl, INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - if (node) + if (node) { sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); + } dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, -- 2.25.1.696.g5e7596f4ac-goog

3 3

[PATCH v1] slimbus: core: Fix mismatch in of_node_get/put
by Saravana Kannan 17 Apr '20

17 Apr '20

Also, remove some unnecessary NULL checks. The functions in question already do NULL checks. Signed-off-by: Saravana Kannan <saravanak(a)google.com> --- drivers/slimbus/core.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 44228a5b246d..ae1e248a8fb8 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -162,11 +162,8 @@ static int slim_add_device(struct slim_controller *ctrl, sbdev->ctrl = ctrl; INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - - if (node) { - sbdev->dev.of_node = of_node_get(node); - sbdev->dev.fwnode = of_fwnode_handle(node); - } + sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, @@ -285,6 +282,7 @@ EXPORT_SYMBOL_GPL(slim_register_controller); /* slim_remove_device: Remove the effect of slim_add_device() */ static void slim_remove_device(struct slim_device *sbdev) { + of_node_put(sbdev->dev.of_node); device_unregister(&sbdev->dev); } -- 2.25.1.696.g5e7596f4ac-goog

2 1

[PATCH] ALSA: au88x0: remove some dead code
by Jason Yan 17 Apr '20

17 Apr '20

Fix the following gcc warning: sound/pci/au88x0/au88x0_a3ddata.c:62:25: warning: ‘A3dHrirDImpulse’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirDImpulse = { ^~~~~~~~~~~~~~~ sound/pci/au88x0/au88x0_a3ddata.c:50:25: warning: ‘A3dHrirSatTest’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirSatTest = { ^~~~~~~~~~~~~~ sound/pci/au88x0/au88x0_a3ddata.c:33:25: warning: ‘A3dHrirOnes’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirOnes = { ^~~~~~~~~~~ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/pci/au88x0/au88x0_a3d.c | 23 ------------------- sound/pci/au88x0/au88x0_a3ddata.c | 38 ------------------------------- 2 files changed, 61 deletions(-) diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 2db183f8826a..4d3b79a34c69 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -512,31 +512,8 @@ static void a3dsrc_ProgramPipe(a3dsrc_t * a) /* Single spike leads to identity transfer function. */ a3dsrc_SetHrtfCurrent(a, A3dHrirImpulse, A3dHrirImpulse); a3dsrc_SetHrtfTarget(a, A3dHrirImpulse, A3dHrirImpulse); - - /* Test: Sounds saturated. */ - //a3dsrc_SetHrtfCurrent(a, A3dHrirSatTest, A3dHrirSatTest); - //a3dsrc_SetHrtfTarget(a, A3dHrirSatTest, A3dHrirSatTest); } -/* VDB = Vortex audio Dataflow Bus */ -#if 0 -static void a3dsrc_ClearVDBData(a3dsrc_t * a, unsigned long aa) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - - // ((aa >> 2) << 8) - (aa >> 2) - hwwrite(vortex->mmio, - a3d_addrS(a->slice, A3D_SLICE_VDBDest) + (a->source << 2), 0); - hwwrite(vortex->mmio, - a3d_addrS(a->slice, - A3D_SLICE_VDBDest + 4) + (a->source << 2), 0); - /* - hwwrite(vortex->mmio, 0x19c00 + (((aa>>2)*255*4)+aa)*8, 0); - hwwrite(vortex->mmio, 0x19c04 + (((aa>>2)*255*4)+aa)*8, 0); - */ -} -#endif - /* A3D HwSource stuff. */ static void vortex_A3dSourceHw_Initialize(vortex_t * v, int source, int slice) diff --git a/sound/pci/au88x0/au88x0_a3ddata.c b/sound/pci/au88x0/au88x0_a3ddata.c index 18623cb6bc52..cc41ea67a6d3 100644 --- a/sound/pci/au88x0/au88x0_a3ddata.c +++ b/sound/pci/au88x0/au88x0_a3ddata.c @@ -30,44 +30,6 @@ static const a3d_Hrtf_t A3dHrirImpulse = { 0, 0, 0 }; -static const a3d_Hrtf_t A3dHrirOnes = { - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff -}; - -static const a3d_Hrtf_t A3dHrirSatTest = { - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, - 0x8001, - 0x8001, - 0x7fff, 0x0000, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 -}; - -static const a3d_Hrtf_t A3dHrirDImpulse = { - 0, 0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0 -}; - static const a3d_ItdDline_t A3dItdDlineZeros = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 -- 2.21.1

2 2

[PATCH] fix sw_params handling of timestamp types in direct plugins
by sylvain.bertrand＠gmail.com 17 Apr '20

17 Apr '20

In pcms using direct plugins (dmix/dsnoop/dshare), the timestamp type could be different from the terminating hw plugin, then the kernel driver. Be sure such pcms have plugins using consistently the same timestamp type. signed-off-by: Sylvain Bertrand <sylvain.bertrand(a)legeek.net> --- src/conf/alsa.conf | 1 + src/conf/pcm/dmix.conf | 4 ++++ src/conf/pcm/dsnoop.conf | 4 ++++ src/pcm/pcm_direct.c | 36 +++++++++++++++++++++++++++++++++++- src/pcm/pcm_direct.h | 2 ++ src/pcm/pcm_dmix.c | 4 ++++ src/pcm/pcm_dshare.c | 4 ++++ src/pcm/pcm_dsnoop.c | 4 ++++ src/pcm/pcm_local.h | 2 ++ 9 files changed, 60 insertions(+), 1 deletion(-) diff --git a/src/conf/alsa.conf b/src/conf/alsa.conf index a091b810..0e01c887 100644 --- a/src/conf/alsa.conf +++ b/src/conf/alsa.conf @@ -69,6 +69,7 @@ defaults.pcm.minperiodtime 5000 # in us defaults.pcm.ipc_key 5678293 defaults.pcm.ipc_gid audio defaults.pcm.ipc_perm 0660 +defaults.pcm.tstamp_type "default" defaults.pcm.dmix.max_periods 0 defaults.pcm.dmix.channels 2 defaults.pcm.dmix.rate 48000 diff --git a/src/conf/pcm/dmix.conf b/src/conf/pcm/dmix.conf index 7fa5c8b2..50e573da 100644 --- a/src/conf/pcm/dmix.conf +++ b/src/conf/pcm/dmix.conf @@ -56,6 +56,10 @@ pcm.!dmix { @func refer name defaults.pcm.ipc_perm } + tstamp_type { + @func refer + name defaults.pcm.tstamp_type + } slave { pcm { type hw diff --git a/src/conf/pcm/dsnoop.conf b/src/conf/pcm/dsnoop.conf index abbd44f7..f4336e5f 100644 --- a/src/conf/pcm/dsnoop.conf +++ b/src/conf/pcm/dsnoop.conf @@ -49,6 +49,10 @@ pcm.!dsnoop { @func refer name defaults.pcm.ipc_perm } + tstamp_type { + @func refer + name defaults.pcm.tstamp_type + } slave { pcm { type hw diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 54d99005..aa60a477 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -991,8 +991,11 @@ int snd_pcm_direct_hw_free(snd_pcm_t *pcm ATTRIBUTE_UNUSED) return 0; } -int snd_pcm_direct_sw_params(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_sw_params_t * params ATTRIBUTE_UNUSED) +int snd_pcm_direct_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) { + if (params->tstamp_type != pcm->tstamp_type) + return -EINVAL; + /* values are cached in the pcm structure */ return 0; } @@ -1318,6 +1321,15 @@ int snd_pcm_direct_initialize_slave(snd_pcm_direct_t *dmix, snd_pcm_t *spcm, str return ret; } + if (dmix->tstamp_type != -1) { + ret = snd_pcm_sw_params_set_tstamp_type(spcm, &sw_params, + dmix->tstamp_type); + if (ret < 0) { + SNDERR("unable to set tstamp type"); + return ret; + } + } + if (dmix->type != SND_PCM_TYPE_DMIX && dmix->type != SND_PCM_TYPE_DSHARE) goto __skip_silencing; @@ -1878,6 +1890,7 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, rec->var_periodsize = 0; rec->direct_memory_access = 1; rec->hw_ptr_alignment = SND_PCM_HW_PTR_ALIGNMENT_AUTO; + rec->tstamp_type = -1; /* read defaults */ if (snd_config_search(root, "defaults.pcm.dmix_max_periods", &n) >= 0) { @@ -1941,6 +1954,27 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, continue; } + if (strcmp(id, "tstamp_type") == 0) { + const char *str; + err = snd_config_get_string(n, &str); + if (err < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + if (strcmp(str, "default") == 0) + rec->tstamp_type = -1; + else if (strcmp(str, "gettimeofday") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY; + else if (strcmp(str, "monotonic") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC; + else if (strcmp(str, "monotonic_raw") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW; + else { + SNDERR("The field tstamp_type is invalid : %s", str); + return -EINVAL; + } + continue; + } if (strcmp(id, "ipc_gid") == 0) { char *group; char *endp; diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 221edbe1..8a236970 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -173,6 +173,7 @@ struct snd_pcm_direct { unsigned int recoveries; /* mirror of executed recoveries on slave */ int direct_memory_access; /* use arch-optimized buffer RW */ snd_pcm_direct_hw_ptr_alignment_t hw_ptr_alignment; + int tstamp_type; /* cached from conf, can be -1(default) on top of real types */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -357,6 +358,7 @@ struct snd_pcm_direct_open_conf { int var_periodsize; int direct_memory_access; snd_pcm_direct_hw_ptr_alignment_t hw_ptr_alignment; + int tstamp_type; snd_config_t *slave; snd_config_t *bindings; }; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index d533f40c..843fa316 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -1038,6 +1038,7 @@ int snd_pcm_dmix_open(snd_pcm_t **pcmp, const char *name, dmix->ipc_key = opts->ipc_key; dmix->ipc_perm = opts->ipc_perm; dmix->ipc_gid = opts->ipc_gid; + dmix->tstamp_type = opts->tstamp_type; dmix->semid = -1; dmix->shmid = -1; @@ -1237,6 +1238,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 59448cfb..6a99452b 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -723,6 +723,7 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, dshare->ipc_key = opts->ipc_key; dshare->ipc_perm = opts->ipc_perm; dshare->ipc_gid = opts->ipc_gid; + dshare->tstamp_type = opts->tstamp_type; dshare->semid = -1; dshare->shmid = -1; @@ -929,6 +930,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 24f472c7..c64df381 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -591,6 +591,7 @@ int snd_pcm_dsnoop_open(snd_pcm_t **pcmp, const char *name, dsnoop->ipc_key = opts->ipc_key; dsnoop->ipc_perm = opts->ipc_perm; dsnoop->ipc_gid = opts->ipc_gid; + dsnoop->tstamp_type = opts->tstamp_type; dsnoop->semid = -1; dsnoop->shmid = -1; @@ -780,6 +781,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 05ed935f..89d4125b 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -928,6 +928,8 @@ int INTERNAL(snd_pcm_hw_params_set_buffer_size_last)(snd_pcm_t *pcm, snd_pcm_hw_ int snd_pcm_sw_params_set_tstamp_mode(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_tstamp_t val); int INTERNAL(snd_pcm_sw_params_get_tstamp_mode)(const snd_pcm_sw_params_t *params, snd_pcm_tstamp_t *val); +int snd_pcm_sw_params_set_tstamp_type(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_tstamp_type_t val); +int snd_pcm_sw_params_get_tstamp_type(const snd_pcm_sw_params_t *params, snd_pcm_tstamp_type_t *val); int snd_pcm_sw_params_set_avail_min(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val); int INTERNAL(snd_pcm_sw_params_get_avail_min)(const snd_pcm_sw_params_t *params, snd_pcm_uframes_t *val); int snd_pcm_sw_params_set_start_threshold(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);

2 1

[PATCH] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported
by Stephan Gerhold 17 Apr '20

17 Apr '20

At the moment, PCM devices for DPCM are only created based on the dpcm_playback/capture parameters of the DAI link, without considering if the CPU/FE DAI is actually capable of playback/capture. Normally the dpcm_playback/capture parameter should match the capabilities of the CPU DAI. However, there is no way to set that parameter from the device tree (e.g. with simple-audio-card or qcom sound cards). dpcm_playback/capture are always both set to 1. This causes problems when the CPU DAI does only support playback or capture. Attemting to open that PCM device with an unsupported stream type then results in a null pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... ... because the DAI playback/capture_widget is not set in that case. We could add checks there to fix the problem (maybe we should anyway), but much easier is to not expose the device as playback/capture in the first place. Attemting to use that device would always fail later anyway. Add checks for snd_soc_dai_stream_valid() to the DPCM case to avoid exposing playback/capture if it is not supported. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- This can be reproduced with q6asm-dai with a device tree configuration like: &q6asmdai { dai@0 { reg = <0>; direction = <Q6ASM_DAI_RX>; }; }; Then a simple "arecord output.wav" will result in the crash above, since it attempts to use that FE DAI for capture (even though it is limited to playback). --- sound/soc/soc-pcm.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 454735f8fa92..77a680da366f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - playback = rtd->dai_link->dpcm_playback; - capture = rtd->dai_link->dpcm_capture; + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU yet\n"); + return -EINVAL; + } + + playback = rtd->dai_link->dpcm_playback && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); + capture = rtd->dai_link->dpcm_capture && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? -- 2.26.1

3 4

sof-hda-dsp: Reverse order and priority of headset and headphone mic
by GitHub pull_request - opened 17 Apr '20

17 Apr '20

alsa-project/alsa-ucm-conf pull request #21 was opened from juimonen: Headset mic is the analog input source in 99% of the cases compared to headphone mic, so change it's priority higher. Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/21 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/21.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: qcom: lpass-cpu: support full duplex operation
by Srinivas Kandagatla 16 Apr '20

16 Apr '20

From: Takahide Higuchi <takahidehiguchi(a)gmail.com> This patch fixes a bug where playback on bidirectional I2S interface stops when we start recording on the same interface. We use regmap_update_bits instead of regmap_write so that we will not clear SPKEN and SPKMODE bits when we start/stop recording. Signed-off-by: Takahide Higuchi <takahidehiguchi(a)gmail.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 ++ sound/soc/qcom/lpass-cpu.c | 24 ++++++++++++++++++------ sound/soc/qcom/lpass-lpaif-reg.h | 2 +- 3 files changed, 21 insertions(+), 7 deletions(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 6575da549237..85079c697faa 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -121,6 +121,8 @@ static struct snd_soc_dai_driver apq8016_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .symmetric_samplebits = 1, + .symmetric_rates = 1, }, }; diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index dbce7e92baf3..dc8acb380b6f 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -72,6 +72,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t format = params_format(params); unsigned int channels = params_channels(params); unsigned int rate = params_rate(params); + unsigned int mask; unsigned int regval; int bitwidth, ret; @@ -81,6 +82,9 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return bitwidth; } + mask = LPAIF_I2SCTL_LOOPBACK_MASK | + LPAIF_I2SCTL_WSSRC_MASK | + LPAIF_I2SCTL_BITWIDTH_MASK; regval = LPAIF_I2SCTL_LOOPBACK_DISABLE | LPAIF_I2SCTL_WSSRC_INTERNAL; @@ -100,6 +104,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mask |= LPAIF_I2SCTL_SPKMODE_MASK | LPAIF_I2SCTL_SPKMONO_MASK; switch (channels) { case 1: regval |= LPAIF_I2SCTL_SPKMODE_SD0; @@ -127,6 +132,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } else { + mask |= LPAIF_I2SCTL_MICMODE_MASK | LPAIF_I2SCTL_MICMONO_MASK; switch (channels) { case 1: regval |= LPAIF_I2SCTL_MICMODE_SD0; @@ -155,9 +161,9 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, } } - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - regval); + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), + mask, regval); if (ret) { dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); return ret; @@ -178,11 +184,17 @@ static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + unsigned int mask; int ret; - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mask = LPAIF_I2SCTL_SPKMODE_MASK; + else + mask = LPAIF_I2SCTL_MICMODE_MASK; + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), + mask, 0); if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 3d74ae123e9d..7a2b9cf99976 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -56,7 +56,7 @@ #define LPAIF_I2SCTL_MICMODE_6CH (7 << LPAIF_I2SCTL_MICMODE_SHIFT) #define LPAIF_I2SCTL_MICMODE_8CH (8 << LPAIF_I2SCTL_MICMODE_SHIFT) -#define LPAIF_I2SCTL_MIMONO_MASK GENMASK(3, 3) +#define LPAIF_I2SCTL_MICMONO_MASK GENMASK(3, 3) #define LPAIF_I2SCTL_MICMONO_SHIFT 3 #define LPAIF_I2SCTL_MICMONO_STEREO (0 << LPAIF_I2SCTL_MICMONO_SHIFT) #define LPAIF_I2SCTL_MICMONO_MONO (1 << LPAIF_I2SCTL_MICMONO_SHIFT) -- 2.21.0

3 3

[PATCH v1 1/2] dt-bindings: sound: add Microsemi ZL38060 binding
by Sven Van Asbroeck 16 Apr '20

16 Apr '20

Add devicetree binding for the Microsemi ZL38060 Connected Home Audio Processor. Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- Tree: v5.7-rc1 .../devicetree/bindings/sound/zl38060.yaml | 90 +++++++++++++++++++ 1 file changed, 90 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zl38060.yaml diff --git a/Documentation/devicetree/bindings/sound/zl38060.yaml b/Documentation/devicetree/bindings/sound/zl38060.yaml new file mode 100644 index 000000000000..8e1976bee76b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zl38060.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/zl38060.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ZL38060 Connected Home Audio Processor from Microsemi. + +description: | + The ZL38060 is a "Connected Home Audio Processor" from Microsemi, + which consists of a Digital Signal Processor (DSP), several Digital + Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. + +maintainers: + - Jaroslav Kysela <perex(a)perex.cz> + - Takashi Iwai <tiwai(a)suse.com> + +properties: + compatible: + const: mscc,zl38060 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 24000000 + + reset-gpios: + description: + A GPIO line handling reset of the chip. As the line is active low, + it should be marked GPIO_ACTIVE_LOW (see ../gpio/gpio.txt) + maxItems: 1 + + mscc,load-firmware: + description: + If present, the driver will attempt to load firmware into the chip via + the SPI interface. + type: boolean + + '#gpio-cells': + const: 2 + + gpio-controller: true + + '#sound-dai-cells': + const: 0 + + amp-en-gpios: + description: + An optional GPIO line which is active when the codec is playing back + audio. Can be used to control an external amplifier. + maxItems: 1 + + mscc,amp-startup-delay-ms: + description: + An optional delay between amp-en-gpio becoming active, and the start + of audio playback. In milliseconds. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi0 { + #address-cells = <1>; + #size-cells = <0>; + + codec: zl38060@0 { + gpio-controller; + #gpio-cells = <2>; + #sound-dai-cells = <0>; + compatible = "mscc,zl38060"; + reg = <0>; + spi-max-frequency = <12000000>; + reset-gpios = <&gpio1 0 GPIO_ACTIVE_LOW>; + amp-en-gpios = <&codec 3 GPIO_ACTIVE_HIGH>; + mscc,amp-startup-delay-ms = <12>; + mscc,load-firmware; + }; + }; -- 2.17.1

2 4

[alsa-devel] [PATCH] ASoC: stm32: sai: fix master clock management
by Olivier Moysan 16 Apr '20

16 Apr '20

When master clock is used, master clock rate is set exclusively. Parent clocks of master clock cannot be changed after a call to clk_set_rate_exclusive(). So the parent clock of SAI kernel clock must be set before. Ensure also that exclusive rate operations are balanced in STM32 SAI driver. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 64 +++++++++++++++++++++++++++++++------------ 1 file changed, 47 insertions(+), 17 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3dd54bc54fa1..77aa38183955 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -70,6 +70,7 @@ #define SAI_IEC60958_STATUS_BYTES 24 #define SAI_MCLK_NAME_LEN 32 +#define SAI_RATE_11K 11025 /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) @@ -311,6 +312,25 @@ static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, return ret; } +static int stm32_sai_set_parent_clock(struct stm32_sai_sub_data *sai, + unsigned int rate) +{ + struct platform_device *pdev = sai->pdev; + struct clk *parent_clk = sai->pdata->clk_x8k; + int ret; + + if (!(rate % SAI_RATE_11K)) + parent_clk = sai->pdata->clk_x11k; + + ret = clk_set_parent(sai->sai_ck, parent_clk); + if (ret) + dev_err(&pdev->dev, " Error %d setting sai_ck parent clock. %s", + ret, ret == -EBUSY ? + "Active stream rates conflict\n" : "\n"); + + return ret; +} + static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, unsigned long *prate) { @@ -492,25 +512,29 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - if (dir == SND_SOC_CLOCK_OUT) { + if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, (unsigned int)~SAI_XCR1_NODIV); if (ret < 0) return ret; - dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); - sai->mclk_rate = freq; + /* If master clock is used, set parent clock now */ + ret = stm32_sai_set_parent_clock(sai, freq); + if (ret) + return ret; - if (sai->sai_mclk) { - ret = clk_set_rate_exclusive(sai->sai_mclk, - sai->mclk_rate); - if (ret) { - dev_err(cpu_dai->dev, - "Could not set mclk rate\n"); - return ret; - } + ret = clk_set_rate_exclusive(sai->sai_mclk, freq); + if (ret) { + dev_err(cpu_dai->dev, + ret == -EBUSY ? + "Active streams have incompatible rates" : + "Could not set mclk rate\n"); + return ret; } + + dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); + sai->mclk_rate = freq; } return 0; @@ -917,11 +941,13 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, int div = 0, cr1 = 0; int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); + int ret; - if (!(rate % 11025)) - clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k); - else - clk_set_parent(sai->sai_ck, sai->pdata->clk_x8k); + if (!sai->sai_mclk) { + ret = stm32_sai_set_parent_clock(sai, rate); + if (ret) + return ret; + } sai_clk_rate = clk_get_rate(sai->sai_ck); if (STM_SAI_IS_F4(sai->pdata)) { @@ -1080,9 +1106,13 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, SAI_XCR1_NODIV); - clk_disable_unprepare(sai->sai_ck); + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } - clk_rate_exclusive_put(sai->sai_mclk); + clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); sai->substream = NULL; -- 2.7.4

2 3

[PATCH 1/3] ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry
by Amadeusz Sławiński 16 Apr '20

16 Apr '20

If we don't find any pcm, pcm will point at address at an offset from the the list head and not a meaningful structure. Fix this by returning correct pcm if found and NULL if not. Found with coccinelle. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/codecs/hdac_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index fba9b749839d..f26b77faed59 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { - struct hdac_hdmi_pcm *pcm = NULL; + struct hdac_hdmi_pcm *pcm; list_for_each_entry(pcm, &hdmi->pcm_list, head) { if (pcm->cvt == cvt) - break; + return pcm; } - return pcm; + return NULL; } static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, -- 2.17.1

2 4

Volume is not restored after boot
by GitHub issues - opened 16 Apr '20

16 Apr '20

alsa-project/alsa-utils issue #39 was opened from Oscar65: I have archlinux updated system last sunday. ``` $ uname -a Linux mypc 5.6.3-arch1-1 #1 SMP PREEMPT Wed, 08 Apr 2020 07:47:16 +0000 x86_64 GNU/Linux ``` I checked with alsa-restore.service and alsa-state.service. None of them work as expected. My sound card is: ``` # lshw -C sound *-multimedia description: Audio device product: 82801H (ICH8 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 02 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:29 memory:ffaf8000-ffafbfff ``` Every time that my system boots the volume levels are different than last time a powered off. Issue URL : https://github.com/alsa-project/alsa-utils/issues/39 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsactl: Wrong line if (!sched_setscheduler(0, SCHED_IDLE, &sched_param))
by GitHub pull_request - opened 16 Apr '20

16 Apr '20

alsa-project/alsa-utils pull request #38 was opened from Oscar65: As man page says: "If successful, the sched_setparam() function shall return zero." So the if must be without ! Without update I got this output in the syslog (journalctl): abr 16 09:25:30 mypc alsactl[1652]: alsactl 1.2.2 daemon started abr 16 09:25:30 mypc alsactl[1652]: /usr/bin/alsactl: do_nice:165sched_setparam failed: No such file or directory That I think that is a wrong message. The call to sched_setparam is correct. This is the output with strace: getpriority(PRIO_PROCESS, 0) = 20 setpriority(PRIO_PROCESS, 0, 19) = 0 getpriority(PRIO_PROCESS, 0) = 1 sched_getparam(0, [0]) = 0 sched_setscheduler(0, SCHED_IDLE, [0]) = 0 write(2, "/usr/bin/alsactl: do_nice:165: ", 31/usr/bin/alsactl: do_nice:165: ) = 31 write(2, "sched_setparam failed: No such f"..., 48sched_setparam failed: No such file or directory) = 48 write(2, "\n", 1) = 1 Call to sched_setscheduler returns 0, so it means that the call was successful. Request URL : https://github.com/alsa-project/alsa-utils/pull/38 Patch URL : https://github.com/alsa-project/alsa-utils/pull/38.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RFC TEST] ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence
by Pierre-Louis Bossart 16 Apr '20

16 Apr '20

On Baytrail/Cherrytrail, the Atom/SST driver fails miserably: [ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed [ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed Commit b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") was the initial problematic commit. Commit 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") was an attempt to fix things but it does not work on Baytrail, reverting all changes seems necessary for now. Fixes: 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Sending as RFC since I don't have a good understanding of the root-cause and for others to confirm my findings. Tested on top of v5.7-rc1. include/sound/soc-dai.h | 1 - sound/soc/soc-dai.c | 11 ++--------- 2 files changed, 2 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index d4825b82c7a3..b33abe93b905 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -351,7 +351,6 @@ struct snd_soc_dai { /* bit field */ unsigned int probed:1; - unsigned int started[SNDRV_PCM_STREAM_LAST + 1]; }; static inline struct snd_soc_pcm_stream * diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 8f3cad8db89a..31c41559034b 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,24 +295,17 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (!dai->started[substream->stream] && - dai->driver->ops->startup) + if (dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); - if (ret == 0) - dai->started[substream->stream] = 1; - return ret; } void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->started[substream->stream] && - dai->driver->ops->shutdown) + if (dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); - - dai->started[substream->stream] = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, -- 2.20.1

3 6

[PATCH v2 0/5] Add support for SOF on i.MX8M
by Daniel Baluta 16 Apr '20

16 Apr '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for SOF on i.MX8M family. First board from this family that has a DSP is i.MX8MP. First 2 patches are trying to fix some compilation issues, the next two are adding the imx8m support and the last one adds the devicetree binding. Changes since v2: - add reviewed by from Rob to DT patch - fix ownership for patch 2 Daniel Baluta (3): ASoC: SOF: imx: Add i.MX8M HW support ASoC: SOF: Add i.MX8MP device descriptor dt-bindings: dsp: fsl: Add fsl,imx8mp-dsp entry Pierre-Louis Bossart (1): ASoC: SOF: imx: fix undefined reference issue YueHaibing (1): ASoC: SOF: imx8: Fix randbuild error .../devicetree/bindings/dsp/fsl,dsp.yaml | 2 + sound/soc/sof/imx/Kconfig | 32 +- sound/soc/sof/imx/Makefile | 2 + sound/soc/sof/imx/imx8m.c | 279 ++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 14 + 5 files changed, 325 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/imx/imx8m.c -- 2.17.1

2 6

[PATCH 00/24] ASoC: SOF: topology and firmware IPC updates for 5.8
by Pierre-Louis Bossart 16 Apr '20

16 Apr '20

Long series made of a relatively small changes from multiple SOF contributors. I didn't find a good way to split this series since it tracks SOF minor ABI changes (backwards-compatible with older firmware files) and needs to be kept in-order. Future series should be much shorter. The main addition is support for an extended firmware manifest, which helps retrieve capabilities directly from the firmware file instead of the current IPC mechanism (still supported but will be deprecated). The IPC is realigned with the firmware, along with type cleanups, and the DMIC interface is simplified. The topology changes are mainly about a multi-cpu DAI fix, a new DC blocking component, better parsing of tuples and new parameters for ALH (SoundWire) and HDaudio DAIs. New tokens are also added to clarify the firmware behavior in the case of dependent pipelines, e.g. for echo reference generation. Artur Kloniecki (1): ASoC: SOF: Add XRUN flags field to struct sof_ipc_buffer. Bard Liao (5): ASoC: SOF: topology: fix: handle DAI widget connections properly with multiple CPU DAI's ASoC: SOF: align sof_ipc_dai_alh_params with FW ASoC: SOF: topology: Get ALH rate amd channels from topology ASoC: SOF: topology: fix: parse hda_tokens to &config->hda ASoC: SOF: topology: Get HDA rate and channels from topology Jaska Uimonen (2): ASoC: SOF: topology: stop parsing when all tokens have been found ASoC: SOF: topology: handle multiple sets of tuple arrays Karol Trzcinski (6): ASoC: SOF: Mark get_ext* function ext_hdr arguments as const ASoC: SOF: Introduce offset in firmware data ASoC: SOF: Introduce extended manifest ASoC: SOF: ext_manifest: parse firmware version ASoC: SOF: ext_manifest: parse windows ASoC: SOF: ext_manifest: parse compiler version Pan Xiuli (6): ASoC: SOF: add probe support extend data ASoC: SOF: add debug ABI version ASoC: SOF: change type char to uint8_t in info.h ASoC: SOF: change type char to uint8_t in trace.h ASoC: SOF: change type char to uint8_t in topology.h ASoC: SOF: make sof_ipc_cc_version to fixed length Sebastiano Carlucci (1): ASoC: SOF: topology: Add support for DC Blocker Seppo Ingalsuo (3): ASoC: SOF: Intel: Fix typo in header file comment text ASoC: SOF: Intel: Change DMIC load IPC to fixed length ASoC: SOF: Intel: Rename deprecated DMIC IPC struct field include/sound/sof.h | 3 + include/sound/sof/dai-intel.h | 20 +- include/sound/sof/info.h | 26 ++- include/sound/sof/topology.h | 16 +- include/sound/sof/trace.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/ext_manifest.h | 91 ++++++++ include/uapi/sound/sof/tokens.h | 8 + sound/soc/sof/intel/hda-loader.c | 9 +- sound/soc/sof/loader.c | 226 ++++++++++++++++-- sound/soc/sof/topology.c | 323 ++++++++++++++++---------- 11 files changed, 568 insertions(+), 158 deletions(-) create mode 100644 include/uapi/sound/sof/ext_manifest.h base-commit: 83b35f4586e235bfb785a7947b555ad8f3d96887 -- 2.20.1

2 25

[alsa-devel] [PATCH] ASoC: pcm: fix error handling when try_module_get() fails.
by Ranjani Sridharan 16 Apr '20

16 Apr '20

Handle error before returning when try_module_get() fails to prevent inconsistent mutex lock/unlock. Fixes: 52034add7 (ASoC: pcm: update module refcount if module_get_upon_open is set) Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d21247546f7f..be80a12fba27 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -518,8 +518,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) continue; if (component->driver->module_get_upon_open && - !try_module_get(component->dev->driver->owner)) - return -ENODEV; + !try_module_get(component->dev->driver->owner)) { + ret = -ENODEV; + goto module_err; + } ret = component->driver->ops->open(substream); if (ret < 0) { @@ -636,7 +638,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) component_err: soc_pcm_components_close(substream, component); - +module_err: if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: -- 2.17.1

2 5

[alsa-devel] [PATCH] ASoC: wcd9335: Fix missing regmap requirement
by Marc Gonzalez 16 Apr '20

16 Apr '20

wcd9335.c: undefined reference to 'devm_regmap_add_irq_chip' Signed-off-by: Marc Gonzalez <marc.w.gonzalez(a)free.fr> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 419142111b6d..33778dc99108 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1162,6 +1162,7 @@ config SND_SOC_WCD9335 tristate "WCD9335 Codec" depends on SLIMBUS select REGMAP_SLIMBUS + select REGMAP_IRQ help The WCD9335 is a standalone Hi-Fi audio CODEC IC, supports Qualcomm Technologies, Inc. (QTI) multimedia solutions, -- 2.17.1

3 4

Baytrail audio not working on next branch
by Jaime Perez 15 Apr '20

15 Apr '20

tested 20200409 20200413 20200414 tags maybe related issue https://bugzilla.kernel.org/show_bug.cgi?id=201787

2 2

[PATCH 1/2] ASoC: topology: Remove unneeded semicolon
by Amadeusz Sławiński 15 Apr '20

15 Apr '20

There is unnecessary semicolon after last bracket of if statement, remove it. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 33e8d189ba2f..a5cd73742dc5 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2390,7 +2390,7 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, tplg->pos += le32_to_cpu(hdr->size) + le32_to_cpu(hdr->payload_size); return 0; - }; + } /* check the element size and count */ link = (struct snd_soc_tplg_link_config *)tplg->pos; -- 2.17.1

2 3

[PATCH] ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs
by Stephan Gerhold 15 Apr '20

15 Apr '20

For some reason, the MI2S DAIs do not have channels_min/max defined. This means that snd_soc_dai_stream_valid() returns false, i.e. the DAIs have neither valid playback nor capture stream. It's quite surprising that this ever worked correctly, but in 5.7-rc1 this is now failing badly: :) Commit 0e9cf4c452ad ("ASoC: pcm: check if cpu-dai supports a given stream") introduced a check for snd_soc_dai_stream_valid() before calling hw_params(), which means that the q6i2s_hw_params() function was never called, eventually resulting in: qcom-q6afe aprsvc:q6afe:4:4: no line is assigned ... even though "qcom,sd-lines" is set in the device tree. Commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") now even avoids creating PCM devices if the stream is not supported, which means that it is failing even earlier with e.g.: Primary MI2S: ASoC: no backend playback stream Avoid all that trouble by adding channels_min/max for the MI2S DAIs. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index c1a7624eaf17..2a5302f1db98 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -902,6 +902,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -917,6 +919,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -931,6 +935,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -946,6 +952,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -960,6 +968,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -975,6 +985,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -989,6 +1001,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -1004,6 +1018,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, -- 2.26.1

3 2

[PATCH v4 0/9] x86: Easy way of detecting MS Surface 3
by Andy Shevchenko 15 Apr '20

15 Apr '20

While working on RTC regression, I noticed that we are using the same DMI check over and over in the drivers for MS Surface 3 platform. This series dedicated for making it easier in the same way how it's done for Apple machines. Changelog v4: - rebase on top of v5.7-rc1 Changelog v3: - fixed typo in patch 5 (Jonathan) - returned back to if {} else {} condition in ASoC driver (Mark) - added Mark's Ack tag Changelog v2: - removed RTC patches for now (the fix will be independent to this series) - added couple more clean ups to arch/x86/kernel/quirks.c - redone DMI quirk to use driver_data instead of callback - simplified check in soc-acpi-intel-cht-match.c to be oneliner - added a new patch to cover rt5645 codec driver Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Andy Shevchenko (9): x86/platform: Rename x86/apple.h -> x86/machine.h x86/quirks: Add missed include to satisfy static checker x86/quirks: Introduce hpet_dev_print_force_hpet_address() helper x86/quirks: Join string literals back x86/quirks: Convert DMI matching to use a table x86/quirks: Add a DMI quirk for Microsoft Surface 3 platform/x86: surface3_wmi: Switch DMI table match to a test of variable ASoC: rt5645: Switch DMI table match to a test of variable ASoC: Intel: Switch DMI table match to a test of variable arch/x86/kernel/quirks.c | 91 +++++++++++++------ drivers/platform/x86/surface3-wmi.c | 16 +--- include/linux/platform_data/x86/apple.h | 14 +-- include/linux/platform_data/x86/machine.h | 20 ++++ sound/soc/codecs/rt5645.c | 14 ++- .../intel/common/soc-acpi-intel-cht-match.c | 28 +----- 6 files changed, 93 insertions(+), 90 deletions(-) create mode 100644 include/linux/platform_data/x86/machine.h -- 2.25.1

2 5

[PATCH] ALSA: hda/hdmi: Add module option to disable audio component binding
by Takashi Iwai 15 Apr '20

15 Apr '20

As the recent regression showed, we want sometimes to turn off the audio component binding just for debugging. This patch adds the module option to control it easily without compilation. Fixes: ade49db337a9 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bb287a916dae..4eff16053bd5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,10 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +static bool enable_acomp = true; +module_param(enable_acomp, bool, 0444); +MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)"); + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -2505,6 +2509,11 @@ static void generic_acomp_init(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; + if (!enable_acomp) { + codec_info(codec, "audio component disabled by module option\n"); + return; + } + spec->port2pin = port2pin; setup_drm_audio_ops(codec, ops); if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops, -- 2.25.0

1 0

[PATCH] bytcht-es8316: fix Speaker or Headphone no sound if enable Mic or Headset
by youling257 15 Apr '20

15 Apr '20

alsaucm -c bytcht-es8316 set _verb HiFi set _enadev Mic, Speaker will no sound. alsaucm -c bytcht-es8316 set _verb HiFi set _enadev Headset, Headphone will no sound. Signed-off-by: youling257 <youling257(a)gmail.com> --- ucm2/codecs/es8316/EnableSeq.conf | 2 -- 1 file changed, 2 deletions(-) diff --git a/ucm2/codecs/es8316/EnableSeq.conf b/ucm2/codecs/es8316/EnableSeq.conf index 551b79b..53bb115 100644 --- a/ucm2/codecs/es8316/EnableSeq.conf +++ b/ucm2/codecs/es8316/EnableSeq.conf @@ -1,6 +1,4 @@ # Disable all inputs / outputs -cset "name='Speaker Switch' off" -cset "name='Headphone Switch' off" cset "name='Headset Mic Switch' off" cset "name='Internal Mic Switch' off" -- 2.25.1

4 11

Debugging dmix?
by Mark Hills 15 Apr '20

15 Apr '20

Recommendations for debugging dmix? For a long time I have a problem where audio output regresses to glitches, only solved by closing all applications using audio and reopening. Especially common when Chromium is used: 3-4 times a day, frustrating. I have a very explicit dmix configuration (see below); no PulseAudio. Reproducable on latest alsa-lib (fb48ad9e; later than v1.2.2) I can rule out _some_ aspects of hardware/drivers (Layla3G); eg. when the fault occurs, other soundcard channels are fine. Any way to understand what dmix is doing? Clients? Control process? Here's a recording taken from the analogue output (L+R chanel get zero'd): http://www.pogo.org.uk/~mark/tmp/dmix-glitch.png http://www.pogo.org.uk/~mark/tmp/dmix-glitch.flac Some process overwriting the ring buffer? Not high enough thread priority? Seems like it might be more prevalant with more applications using sound. Maybe why Chromium makes it worse; several threads/sandboxes, perhaps. dmix has all the trappings of something fiendishly clever and concise, but quite opaque. I'd need time to study the code (it has few comments, a lot of assumed knowledge) to understand its IPC. Can anyone give a head start? Specifically, how does the final mix get written to the output? eg. Thread priorities, if that process goes away etc. Thanks --- pcm.layla_multi { type route slave.pcm { type dmix ipc_key 833282 ipc_key_add_uid true slave { pcm "hw:Layla3G,0,0" period_size 256 buffer_size 32768 channels 4 rate 48000 format S24_3LE } } ttable { # Headphones 0.0 1 1.1 1 # Speakers 0.2 1 1.3 1 } } pcm.!default { type plug slave.pcm layla_multi } -- Mark

1 0

[PATCH v6 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 15 Apr '20

15 Apr '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v6 - updated according to Nicoin's and Rob's comments. changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 21 +- sound/soc/fsl/fsl_asrc.c | 303 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 105 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2114 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3206 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

4 22

[PATCH] ASoC: Intel: skl_hda_generic: remove rtd->codec_dai
by Pierre-Louis Bossart 15 Apr '20

15 Apr '20

Use macro and solve compilation issues Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 64197b010e7d..35f222ed9912 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -175,7 +175,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct hdac_hda_priv *hda_pvt; if (!codec_dai) -- 2.20.1

2 1

[PATCH] ASoC: fsl_micfil: Omit superfluous error message in fsl_micfil_probe()
by Tang Bin 15 Apr '20

15 Apr '20

In the function fsl_micfil_probe(), when get irq failed, the function platform_get_irq() logs an error message, so remove redundant message here. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> Signed-off-by: Shengju Zhang <zhangshengju(a)cmss.chinamobile.com> --- sound/soc/fsl/fsl_micfil.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index f7f2d29f1..e73bd6570 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -702,10 +702,8 @@ static int fsl_micfil_probe(struct platform_device *pdev) for (i = 0; i < MICFIL_IRQ_LINES; i++) { micfil->irq[i] = platform_get_irq(pdev, i); dev_err(&pdev->dev, "GET IRQ: %d\n", micfil->irq[i]); - if (micfil->irq[i] < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + if (micfil->irq[i] < 0) return micfil->irq[i]; - } } if (of_property_read_bool(np, "fsl,shared-interrupt")) -- 2.20.1.windows.1

2 1

[PATCH] ASoC: sgtl5000: Fix VAG power-on handling
by Sebastian Reichel 15 Apr '20

15 Apr '20

As mentioned slightly out of patch context in the code, there is no reset routine for the chip. On boards where the chip is supplied by a fixed regulator, it might not even be resetted during (e.g. watchdog) reboot and can be in any state. If the device is probed with VAG enabled, the driver's probe routine will generate a loud pop sound when ANA_POWER is being programmed. Avoid this by properly disabling just the VAG bit and waiting the required power down time. Signed-off-by: Sebastian Reichel <sebastian.reichel(a)collabora.com> --- sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 1 + 2 files changed, 35 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d5130193b4a2..e8a8bf7b4ffe 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1653,6 +1653,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "Error %d initializing CHIP_CLK_CTRL\n", ret); + /* Mute everything to avoid pop from the following power-up */ + ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_CHIP_ANA_CTRL_DEFAULT); + if (ret) { + dev_err(&client->dev, + "Error %d muting outputs via CHIP_ANA_CTRL\n", ret); + goto disable_clk; + } + + /* + * If VAG is powered-on (e.g. from previous boot), it would be disabled + * by the write to ANA_POWER in later steps of the probe code. This + * may create a loud pop even with all outputs muted. The proper way + * to circumvent this is disabling the bit first and waiting the proper + * cool-down time. + */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value); + if (ret) { + dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret); + goto disable_clk; + } + if (value & SGTL5000_VAG_POWERUP) { + ret = regmap_update_bits(sgtl5000->regmap, + SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, + 0); + if (ret) { + dev_err(&client->dev, "Error %d disabling VAG\n", ret); + goto disable_clk; + } + + msleep(SGTL5000_VAG_POWERDOWN_DELAY); + } + /* Follow section 2.2.1.1 of AN3663 */ ana_pwr = SGTL5000_ANA_POWER_DEFAULT; if (sgtl5000->num_supplies <= VDDD) { diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index a4bf4bca95bf..56ec5863f250 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -233,6 +233,7 @@ /* * SGTL5000_CHIP_ANA_CTRL */ +#define SGTL5000_CHIP_ANA_CTRL_DEFAULT 0x0133 #define SGTL5000_LINE_OUT_MUTE 0x0100 #define SGTL5000_HP_SEL_MASK 0x0040 #define SGTL5000_HP_SEL_SHIFT 6 -- 2.25.1

3 2

[PATCH 1/2] dt-bindings: sound: lpass-cpu: Document DAI subnodes
by Stephan Gerhold 15 Apr '20

15 Apr '20

The lpass-cpu driver now allows configuring the MI2S SD lines by defining subnodes for one of the DAIs. Document this in the device tree bindings. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- .../bindings/sound/qcom,lpass-cpu.txt | 23 +++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt index 21c648328be9..df53a10502f7 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -30,6 +30,8 @@ Required properties: - reg : Must contain an address for each entry in reg-names. - reg-names : A list which must include the following entries: * "lpass-lpaif" +- #address-cells : Must be 1 +- #size-cells : Must be 0 @@ -37,6 +39,18 @@ Optional properties: - qcom,adsp : Phandle for the audio DSP node +By default, the driver uses up to 4 MI2S SD lines, for a total of 8 channels. +The SD lines to use can be configured by adding subnodes for each of the DAIs. + +Required properties for each DAI (represented by a subnode): +- reg : Must be one of the DAI IDs + (usually part of dt-bindings header) +- qcom,playback-sd-lines: List of serial data lines (0-3) to use for playback +- qcom,capture-sd-lines : List of serial data lines (0-3) to use for capture + +Note that adding a subnode changes the default to "no lines configured", +so both playback and capture lines should be configured when a subnode is added. + Example: lpass@28100000 { @@ -51,4 +65,13 @@ lpass@28100000 { reg = <0x28100000 0x10000>; reg-names = "lpass-lpaif"; qcom,adsp = <&adsp>; + + #address-cells = <1>; + #size-cells = <0>; + + /* Optional to set different MI2S SD lines */ + mi2s-quaternary@3 { + reg = <MI2S_QUATERNARY>; + qcom,playback-sd-lines = <0 1>; + }; }; -- 2.26.0

2 3

[PATCH] ALSA: usb-audio: RME Babyface Pro mixer patch
by Thomas Ebeling 15 Apr '20

15 Apr '20

Added mixer quirks to allow controlling the internal DSP of the RME Babyface Pro and its successor Babyface Pro FS. Signed-off-by: Thomas Ebeling <penguins(a)bollie.de> --- sound/usb/mixer_quirks.c | 418 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 418 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 02b036b2aefb..f0302ad7f5a0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2181,6 +2181,421 @@ static int snd_rme_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * RME Babyface Pro (FS) + * + * These devices exposes a couple of DSP functions via request to EP0. + * Switches are available via control registers, while routing is controlled + * by controlling the volume on each possible crossing point. + * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 46341) with + * 0dB being at dec. 32768. + */ +enum { + SND_BBFPRO_CTL_REG1 = 0, + SND_BBFPRO_CTL_REG2 +}; + +#define SND_BBFPRO_CTL_REG_MASK 1 +#define SND_BBFPRO_CTL_IDX_MASK 0xff +#define SND_BBFPRO_CTL_IDX_SHIFT 1 +#define SND_BBFPRO_CTL_VAL_MASK 1 +#define SND_BBFPRO_CTL_VAL_SHIFT 9 +#define SND_BBFPRO_CTL_REG1_CLK_MASTER 0 +#define SND_BBFPRO_CTL_REG1_CLK_OPTICAL 1 +#define SND_BBFPRO_CTL_REG1_SPDIF_PRO 7 +#define SND_BBFPRO_CTL_REG1_SPDIF_EMPH 8 +#define SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL 10 +#define SND_BBFPRO_CTL_REG2_48V_AN1 0 +#define SND_BBFPRO_CTL_REG2_48V_AN2 1 +#define SND_BBFPRO_CTL_REG2_SENS_IN3 2 +#define SND_BBFPRO_CTL_REG2_SENS_IN4 3 +#define SND_BBFPRO_CTL_REG2_PAD_AN1 4 +#define SND_BBFPRO_CTL_REG2_PAD_AN2 5 + +#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff +#define SND_BBFPRO_MIXER_VAL_SHIFT 9 +#define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf +#define SND_BBFPRO_MIXER_VAL_MAX 46341 // +6dB + +#define SND_BBFPRO_USBREQ_CTL_REG1 0x10 +#define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_MIXER 0x12 + +static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, + u8 index, u8 value) +{ + int err; + u16 usb_req, usb_idx, usb_val; + struct snd_usb_audio *chip = mixer->chip; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + if (reg == SND_BBFPRO_CTL_REG1) { + usb_req = SND_BBFPRO_USBREQ_CTL_REG1; + if (index == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) { + usb_idx = 3; + usb_val = value ? 3 : 0; + } else { + usb_idx = 1 << index; + usb_val = value ? usb_idx : 0; + } + } else { + usb_req = SND_BBFPRO_USBREQ_CTL_REG2; + usb_idx = 1 << index; + usb_val = value ? usb_idx : 0; + } + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), usb_req, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + usb_val, usb_idx, 0, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 reg, idx, val; + int pv; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + val = kcontrol->private_value >> SND_BBFPRO_CTL_VAL_SHIFT; + + if ((reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) || + (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) { + ucontrol->value.enumerated.item[0] = val; + } else { + ucontrol->value.integer.value[0] = val; + } + return 0; +} + +static int snd_bbfpro_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + u8 reg, idx; + int pv; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + + if (reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) { + static const char * const texts[2] = { + "AutoSync", + "Internal" + }; + return snd_ctl_enum_info(uinfo, 1, 2, texts); + } else if (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4)) { + static const char * const texts[2] = { + "-10dBV", + "+4dBu" + }; + return snd_ctl_enum_info(uinfo, 1, 2, texts); + } + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + return 0; +} + +static int snd_bbfpro_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int err; + u8 reg, idx; + int old_value, pv, val; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + old_value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK; + + if ((reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) || + (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) { + val = ucontrol->value.enumerated.item[0]; + } else { + val = ucontrol->value.integer.value[0]; + } + + if (val > 1) + return -EINVAL; + + if (val == old_value) + return 0; + + kcontrol->private_value = reg + | ((idx & SND_BBFPRO_CTL_IDX_MASK) << SND_BBFPRO_CTL_IDX_SHIFT) + | ((val & SND_BBFPRO_CTL_VAL_MASK) << SND_BBFPRO_CTL_VAL_SHIFT); + + err = snd_bbfpro_ctl_update(mixer, reg, idx, val); + return err < 0 ? err : 1; +} + +static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) +{ + u8 reg, idx; + int value, pv; + + pv = list->kctl->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK; + + return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); +} + +static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, + u32 value) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + u16 idx; + u16 usb_idx, usb_val; + u32 v; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + idx = index & SND_BBFPRO_MIXER_IDX_MASK; + // 18 bit linear volume, split so 2 bits end up in index. + v = value & SND_BBFPRO_MIXER_VAL_MASK; + usb_idx = idx | (v & 0x3) << 14; + usb_val = (v >> 2) & 0xffff; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_MIXER, + USB_DIR_OUT | USB_TYPE_VENDOR | + USB_RECIP_DEVICE, + usb_val, usb_idx, 0, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = + kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT; + return 0; +} + +static int snd_bbfpro_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_MIXER_VAL_MIN; + uinfo->value.integer.max = SND_BBFPRO_MIXER_VAL_MAX; + return 0; +} + +static int snd_bbfpro_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int err; + u16 idx; + u32 new_val, old_value, uvalue; + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + uvalue = ucontrol->value.integer.value[0]; + idx = kcontrol->private_value & SND_BBFPRO_MIXER_IDX_MASK; + old_value = kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT; + + if (uvalue > SND_BBFPRO_MIXER_VAL_MAX) + return -EINVAL; + + if (uvalue == old_value) + return 0; + + new_val = uvalue & SND_BBFPRO_MIXER_VAL_MASK; + + kcontrol->private_value = idx + | (new_val << SND_BBFPRO_MIXER_VAL_SHIFT); + + err = snd_bbfpro_vol_update(mixer, idx, new_val); + return err < 0 ? err : 1; +} + +static int snd_bbfpro_vol_resume(struct usb_mixer_elem_list *list) +{ + int pv = list->kctl->private_value; + u16 idx = pv & SND_BBFPRO_MIXER_IDX_MASK; + u32 val = (pv >> SND_BBFPRO_MIXER_VAL_SHIFT) + & SND_BBFPRO_MIXER_VAL_MASK; + return snd_bbfpro_vol_update(list->mixer, idx, val); +} + +// Predfine elements +static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_ctl_info, + .get = snd_bbfpro_ctl_get, + .put = snd_bbfpro_ctl_put +}; + +static const struct snd_kcontrol_new snd_bbfpro_vol_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_vol_info, + .get = snd_bbfpro_vol_get, + .put = snd_bbfpro_vol_put +}; + +static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, + u8 index, char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_ctl_control; + + knew.name = name; + knew.private_value = (reg & SND_BBFPRO_CTL_REG_MASK) + | ((index & SND_BBFPRO_CTL_IDX_MASK) + << SND_BBFPRO_CTL_IDX_SHIFT); + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_ctl_resume, + &knew, NULL); +} + +static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_vol_control; + + knew.name = name; + knew.private_value = index & SND_BBFPRO_MIXER_IDX_MASK; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_vol_resume, + &knew, NULL); +} + +static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i, o; + char name[48]; + + static const char * const input[] = { + "AN1", "AN2", "IN3", "IN4", "AS1", "AS2", "ADAT3", + "ADAT4", "ADAT5", "ADAT6", "ADAT7", "ADAT8"}; + + static const char * const output[] = { + "AN1", "AN2", "PH3", "PH4", "AS1", "AS2", "ADAT3", "ADAT4", + "ADAT5", "ADAT6", "ADAT7", "ADAT8"}; + + for (o = 0 ; o < 12 ; ++o) { + for (i = 0 ; i < 12 ; ++i) { + // Line routing + snprintf(name, sizeof(name), + "%s-%s-%s Playback Volume", + (i < 2 ? "Mic" : "Line"), + input[i], output[o]); + err = snd_bbfpro_vol_add(mixer, (26 * o + i), name); + if (err < 0) + return err; + + // PCM routing... yes, it is output remapping + snprintf(name, sizeof(name), + "PCM-%s-%s Playback Volume", + output[i], output[o]); + err = snd_bbfpro_vol_add(mixer, (26 * o + 12 + i), + name); + if (err < 0) + return err; + } + } + + // Control Reg 1 + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_CLK_OPTICAL, + "Sample Clock Source"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_PRO, + "IEC958 Pro Mask"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_EMPH, + "IEC958 Emphasis"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL, + "IEC958 Switch"); + if (err < 0) + return err; + + // Control Reg 2 + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_48V_AN1, + "Mic-AN1 48V"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_48V_AN2, + "Mic-AN2 48V"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_SENS_IN3, + "Line-IN3 Sens."); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_SENS_IN4, + "Line-IN4 Sens."); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_PAD_AN1, + "Mic-AN1 PAD"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_PAD_AN2, + "Mic-AN2 PAD"); + if (err < 0) + return err; + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2282,6 +2697,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ err = snd_sc1810_init_mixer(mixer); break; + case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ + err = snd_bbfpro_controls_create(mixer); + break; } return err; -- 2.26.0

2 1

monotonic raw setup seems buggy
by sylvain.bertrand＠gmail.com 15 Apr '20

15 Apr '20

Hi, I am writting an alsa application and while working on pcm status timings, I noticed that setting monotonic_raw timings in sw_params while the hardware device plugin is already running with monotonic timings did not produce an error. Follow the pcm dump which shows monotonic_raw being set on all plugin instances but the already running hw plugin instance. Is this a bug, or am I doing something the wrong way? Plug PCM: Linear Integer <-> Linear Float conversion PCM (S32_LE) Its setup is: stream : PLAYBACK access : RW_NONINTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC_RAW period_step : 1 avail_min : 1024 period_event : 1 start_threshold : 1 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC_RAW period_step : 1 avail_min : 1024 period_event : 1 start_threshold : 1 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 0 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 1484839806 cheers, -- Sylvain

5 20

[PATCH] ALSA: hda: enable the runtime_pm for non-vgaswitcheroo hda controllers
by Hui Wang 14 Apr '20

14 Apr '20

Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbib() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled even it calls put_autosuspend(). Need to call runtime_allow() for those controllers in the hda driver. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8519051a426e..779705bef88b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2356,6 +2356,8 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + if (!use_vga_switcheroo(chip)) + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.17.1

4 11

[FRUSTRATED] Missing Sound Codecs for Lenovo ThinkCentre M710e
by Michael 14 Apr '20

14 Apr '20

Hi All, Technically this probably should go to a support channnel, and not a development list, but at my wits end. Kernel: 4.15.0-91-generic #92~16.04.1-Ubuntu SMP dmesg | grep snd [ 12.656843] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 12.807521] snd_hda_intel 0000:00:1f.3: CORB reset timeout#1, CORBRP = 0 [ 12.810429] snd_hda_intel 0000:00:1f.3: no codecs found! I have been playing with settings in the modprobe directory for Also with no effect.. current settings are.. options snd-hda-intel single_cmd=1 options snd-hda-intel probe_mask=1 options snd-hda-intel model=thinkpad options snd-hda-intel position_fix=3 Have tried model=generic, and various other suggestions from online, but at a loss. Don't see anything in: https://github.com/torvalds/linux/blob/master/Documentation/sound/hd-audio/… Your ALSA information is located at http://alsa-project.org/db/?f=7898b30139e9a91fecec22f3ee46049f33248b85 Can anyone point me to how to debug this further, or any way to force detection of a proper codec for this model? 00:1f.3 Audio device [0403]: Intel Corporation 200 Series PCH HD Audio [8086:a2f0] Subsystem: Lenovo Device [17aa:313c] Kernel driver in use: snd_hda_intel description: Motherboard product: 313C vendor: LENOVO physical id: 0 version: SDK0J40697 WIN 3305181519567 slot: Default string Booting in legacy OS mode, not EUFI.. -- "Catch the Magic of Linux..." ------------------------------------------------------------------------ Michael Peddemors, President/CEO LinuxMagic Inc. Visit us at http://www.linuxmagic.com @linuxmagic A Wizard IT Company - For More Info http://www.wizard.ca "MagicSpam" is a Registered TradeMark of Wizard Tower TechnoServices Ltd. ------------------------------------------------------------------------ 604-682-0300 Beautiful British Columbia, Canada This email and any electronic data contained are confidential and intended solely for the use of the individual or entity to which they are addressed. Please note that any views or opinions presented in this email are solely those of the author and are not intended to represent those of the company.

3 5

[PATCH 00/13] ASoC: Intel: machine drivers update for 5.8
by Pierre-Louis Bossart 14 Apr '20

14 Apr '20

This patchset a) adds support for the Intel ElkhartLake platforms b)aligns the HDaudio mic detection with the snd-hda-intel driver c) correct DMIC missing configurations d) fixes initialization/compilation problems for SoundWire platforms d) completes the removal of codec_dais missing in Morimoto-san's series. Bard Liao (1): ASoC: intel: sof_sdw: init all aggregated codecs Hui Wang (1): ASoC: intel/skl/hda - set autosuspend timeout for hda codecs Keyon Jie (1): ASoC: Intel: sof-da7219-max98373: add DMIC widget and route Libin Yang (3): ALSA: hda: Add ElkhartLake HDMI codec vid ASoC: SOF: Intel: add PCI ID for ElkhartLake ASoC: Intel: boards: support Elkhart Lake with rt5660 Pierre-Louis Bossart (6): ASoC: Intel: sof_sdw_hdmi: fix compilation issue in fallback mode ASoC: Intel: sof_sdw_hdmi: remove codec_dai use ASoC: Intel: sof_sdw_rt1308: remove codec dai use ASoC: Intel: sof_sdw_rt5682: remove codec_dai use ASoC: Intel: sof_sdw_rt700: remove codec_dai use ASoC: Intel: sof_sdw_rt711: remove codec_dai use Yong Zhi (1): ASoC: Intel: sof_da7219_max98373: Add BE dailink for dmic16k sound/pci/hda/patch_hdmi.c | 1 + sound/soc/intel/boards/Kconfig | 15 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/ehl_rt5660.c | 323 ++++++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 29 +- sound/soc/intel/boards/sof_da7219_max98373.c | 21 ++ sound/soc/intel/boards/sof_sdw.c | 41 ++- sound/soc/intel/boards/sof_sdw_hdmi.c | 4 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 3 +- sound/soc/intel/boards/sof_sdw_rt700.c | 3 +- sound/soc/intel/boards/sof_sdw_rt711.c | 3 +- .../intel/common/soc-acpi-intel-ehl-match.c | 7 + sound/soc/sof/sof-pci-dev.c | 2 + 14 files changed, 434 insertions(+), 22 deletions(-) create mode 100644 sound/soc/intel/boards/ehl_rt5660.c base-commit: dd8e871d4e560eeb8d22af82dde91457ad835a63 -- 2.20.1

4 30

[PATCH v2] ALSA: hda: call runtime_allow() for all hda controllers
by Hui Wang 14 Apr '20

14 Apr '20

Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbid() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled because the runtime_allow() is not called. Since it is no harm calling runtime_allow() twice, here let hda driver call runtime_allow() for all controllers. Then the runtime_pm is enabled on all controllers after the put_autosuspend() is called. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8519051a426e..a5fab12defde 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2356,6 +2356,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.17.1

2 1

[PATCH 0/2] ASoC: SOF: trivial code cleanups
by Pierre-Louis Bossart 14 Apr '20

14 Apr '20

Fix warnings reported by tools - no functionality change. Payal Kshirsagar (2): ASoC: SOF: remove unneeded variables ASoC: SOF: Intel: hda: remove unnecessary parentheses sound/soc/sof/intel/hda-codec.c | 5 +---- sound/soc/sof/intel/hda-dsp.c | 8 ++++---- sound/soc/sof/nocodec.c | 6 ++---- 3 files changed, 7 insertions(+), 12 deletions(-) base-commit: dd8e871d4e560eeb8d22af82dde91457ad835a63 -- 2.20.1

2 4

[PATCH 0/4] ASoC: SOF: adjust dmesg verbosity
by Pierre-Louis Bossart 14 Apr '20

14 Apr '20

Decrease the dmesg verbosity to remove unnecessary logs on SoundWire platforms, and conversely add more information to help the community and downstream distros with HDaudio/SOF support (DMIC detection and card instanciation are the most prevalent issues on GitHub). Pierre-Louis Bossart (3): ASoC: codecs: rt1308-sdw: reduce verbosity ASoC: SOF: Intel: hda: reduce verbosity on SoundWire detection ASoC: SOF: Intel: hda: log number of microphones detected in NHLT tables Ranjani Sridharan (1): ASoC: soc-core: Add dynamic debug logs in soc_dai_link_sanity_check() sound/soc/codecs/rt1308-sdw.c | 4 ++-- sound/soc/soc-core.c | 18 +++++++++++++++--- sound/soc/sof/intel/hda.c | 10 ++++++---- 3 files changed, 23 insertions(+), 9 deletions(-) base-commit: dd8e871d4e560eeb8d22af82dde91457ad835a63 -- 2.20.1

2 8

Applied "ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer 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 c2233a266178f8937cc26a84cd7672334b5424b7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 30 Mar 2020 10:47:37 +0900 Subject: [PATCH] ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87imimboli.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 12 ++++----- sound/soc/soc-compress.c | 36 +++++++++++++-------------- sound/soc/soc-core.c | 16 ++++++------ sound/soc/soc-dapm.c | 4 +-- sound/soc/soc-generic-dmaengine-pcm.c | 6 ++--- sound/soc/soc-pcm.c | 30 +++++++++++----------- 6 files changed, 52 insertions(+), 52 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 13458e4fbb13..4dff6745b067 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1179,16 +1179,16 @@ struct snd_soc_pcm_runtime { (i)++) #define for_each_rtd_cpu_dais(rtd, i, dai) \ for ((i) = 0; \ - ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]); \ + ((i) < rtd->num_cpus) && ((dai) = asoc_rtd_to_cpu(rtd, i)); \ (i)++) #define for_each_rtd_cpu_dais_rollback(rtd, i, dai) \ - for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);) + for (; (--(i) >= 0) && ((dai) = asoc_rtd_to_cpu(rtd, i));) #define for_each_rtd_codec_dais(rtd, i, dai) \ for ((i) = 0; \ - ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ + ((i) < rtd->num_codecs) && ((dai) = asoc_rtd_to_codec(rtd, i)); \ (i)++) #define for_each_rtd_codec_dais_rollback(rtd, i, dai) \ - for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) + for (; (--(i) >= 0) && ((dai) = asoc_rtd_to_codec(rtd, i));) #define for_each_rtd_dais(rtd, i, dai) \ for ((i) = 0; \ ((i) < (rtd)->num_cpus + (rtd)->num_codecs) && \ @@ -1381,8 +1381,8 @@ struct snd_soc_dai *snd_soc_card_get_codec_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd; list_for_each_entry(rtd, &card->rtd_list, list) { - if (!strcmp(rtd->codec_dai->name, dai_name)) - return rtd->codec_dai; + if (!strcmp(asoc_rtd_to_codec(rtd, 0)->name, dai_name)) + return asoc_rtd_to_codec(rtd, 0); } return NULL; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 50062eb79adb..3f77c0208e09 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -73,7 +73,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component, *save = NULL; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret, i; for_each_rtd_components(rtd, i, component) { @@ -141,7 +141,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) struct snd_pcm_substream *fe_substream = fe->pcm->streams[cstream->direction].substream; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); struct snd_soc_dpcm *dpcm; struct snd_soc_dapm_widget_list *list; int stream; @@ -230,8 +230,8 @@ static int soc_compr_free(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int stream, i; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -274,7 +274,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) static int soc_compr_free_fe(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *fe = cstream->private_data; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); struct snd_soc_dpcm *dpcm; int stream, ret; @@ -343,8 +343,8 @@ static int soc_compr_components_trigger(struct snd_compr_stream *cstream, static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -373,7 +373,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) { struct snd_soc_pcm_runtime *fe = cstream->private_data; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); int ret, stream; if (cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN || @@ -446,7 +446,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, struct snd_compr_params *params) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -500,7 +500,7 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *fe = cstream->private_data; struct snd_pcm_substream *fe_substream = fe->pcm->streams[cstream->direction].substream; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); int ret, stream; if (cstream->direction == SND_COMPRESS_PLAYBACK) @@ -558,7 +558,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -632,7 +632,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -664,7 +664,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; int i, ret = 0; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -711,7 +711,7 @@ static int soc_compr_set_metadata(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, ret; if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_metadata) { @@ -739,7 +739,7 @@ static int soc_compr_get_metadata(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, ret; if (cpu_dai->driver->cops && cpu_dai->driver->cops->get_metadata) { @@ -801,8 +801,8 @@ static struct snd_compr_ops soc_compr_dyn_ops = { int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_component *component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_compr *compr; struct snd_pcm *be_pcm; char new_name[64]; @@ -891,7 +891,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) ret = snd_compress_new(rtd->card->snd_card, num, direction, new_name, compr); if (ret < 0) { - component = rtd->codec_dai->component; + component = asoc_rtd_to_codec(rtd, 0)->component; dev_err(component->dev, "Compress ASoC: can't create compress for codec %s: %d\n", component->name, ret); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 843b8b1c89d4..4ad64b96d893 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -364,7 +364,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); */ void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int playback = SNDRV_PCM_STREAM_PLAYBACK; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -991,13 +991,13 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, rtd->num_cpus = dai_link->num_cpus; for_each_link_cpus(dai_link, i, cpu) { - rtd->cpu_dais[i] = snd_soc_find_dai(cpu); - if (!rtd->cpu_dais[i]) { + asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai(cpu); + if (!asoc_rtd_to_cpu(rtd, i)) { dev_info(card->dev, "ASoC: CPU DAI %s not registered\n", cpu->dai_name); goto _err_defer; } - snd_soc_rtd_add_component(rtd, rtd->cpu_dais[i]->component); + snd_soc_rtd_add_component(rtd, asoc_rtd_to_cpu(rtd, i)->component); } /* Single cpu links expect cpu and cpu_dai in runtime data */ @@ -1006,14 +1006,14 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, /* Find CODEC from registered CODECs */ rtd->num_codecs = dai_link->num_codecs; for_each_link_codecs(dai_link, i, codec) { - rtd->codec_dais[i] = snd_soc_find_dai(codec); - if (!rtd->codec_dais[i]) { + asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec); + if (!asoc_rtd_to_codec(rtd, i)) { dev_info(card->dev, "ASoC: CODEC DAI %s not registered\n", codec->dai_name); goto _err_defer; } - snd_soc_rtd_add_component(rtd, rtd->codec_dais[i]->component); + snd_soc_rtd_add_component(rtd, asoc_rtd_to_codec(rtd, i)->component); } /* Single codec links expect codec and codec_dai in runtime data */ @@ -1062,7 +1062,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai_link *dai_link = rtd->dai_link; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_component *component; int ret, num, i; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 679ed60d850e..8f260994b149 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4370,11 +4370,11 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, if (rtd->num_cpus == 1) { for_each_rtd_codec_dais(rtd, i, codec_dai) dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[0]); + asoc_rtd_to_cpu(rtd, 0)); } else if (rtd->num_codecs == rtd->num_cpus) { for_each_rtd_codec_dais(rtd, i, codec_dai) dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[i]); + asoc_rtd_to_cpu(rtd, i)); } else { dev_err(card->dev, "N cpus to M codecs link is not supported yet\n"); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index facf1922a714..f728309a0833 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -68,7 +68,7 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, return -EINVAL; } - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); if (ret) @@ -134,7 +134,7 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); memset(&hw, 0, sizeof(hw)); hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -203,7 +203,7 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( return NULL; } - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 289aebc15529..454735f8fa92 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -136,7 +136,7 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, return -ENOMEM; for_each_pcm_streams(stream) - if (snd_soc_dai_stream_valid(fe->cpu_dai, stream)) + if (snd_soc_dai_stream_valid(asoc_rtd_to_cpu(fe, 0), stream)) offset += dpcm_show_state(fe, stream, buf + offset, out_count - offset); @@ -836,10 +836,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) soc_pcm_init_runtime_hw(substream); if (rtd->num_codecs == 1) - codec_dai_name = rtd->codec_dai->name; + codec_dai_name = asoc_rtd_to_codec(rtd, 0)->name; if (rtd->num_cpus == 1) - cpu_dai_name = rtd->cpu_dai->name; + cpu_dai_name = asoc_rtd_to_cpu(rtd, 0)->name; if (soc_pcm_has_symmetry(substream)) runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -1483,7 +1483,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list) { - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0); int paths; if (fe->num_cpus > 1) { @@ -1842,7 +1842,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, * DAIs connected to a single CPU DAI, use CPU DAI's directly */ if (be->num_codecs == 1) { - codec_stream = snd_soc_dai_get_pcm_stream(be->codec_dais[0], stream); + codec_stream = snd_soc_dai_get_pcm_stream(asoc_rtd_to_codec(be, 0), stream); *channels_min = max(*channels_min, codec_stream->channels_min); @@ -2759,7 +2759,7 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) return 0; /* only check active links */ - if (!fe->cpu_dai->active) + if (!asoc_rtd_to_cpu(fe, 0)->active) return 0; /* DAPM sync will call this to update DSP paths */ @@ -2769,13 +2769,13 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) for_each_pcm_streams(stream) { /* skip if FE doesn't have playback/capture capability */ - if (!snd_soc_dai_stream_valid(fe->cpu_dai, stream) || - !snd_soc_dai_stream_valid(fe->codec_dai, stream)) + if (!snd_soc_dai_stream_valid(asoc_rtd_to_cpu(fe, 0), stream) || + !snd_soc_dai_stream_valid(asoc_rtd_to_codec(fe, 0), stream)) continue; /* skip if FE isn't currently playing/capturing */ - if (!fe->cpu_dai->stream_active[stream] || - !fe->codec_dai->stream_active[stream]) + if (!asoc_rtd_to_cpu(fe, 0)->stream_active[stream] || + !asoc_rtd_to_codec(fe, 0)->stream_active[stream]) continue; paths = dpcm_path_get(fe, stream, &list); @@ -2922,9 +2922,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) for_each_rtd_codec_dais(rtd, i, codec_dai) { if (rtd->num_cpus == 1) { - cpu_dai = rtd->cpu_dais[0]; + cpu_dai = asoc_rtd_to_cpu(rtd, 0); } else if (rtd->num_cpus == rtd->num_codecs) { - cpu_dai = rtd->cpu_dais[i]; + cpu_dai = asoc_rtd_to_cpu(rtd, i); } else { dev_err(rtd->card->dev, "N cpus to M codecs link is not supported yet\n"); @@ -2971,7 +2971,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, (rtd->num_codecs > 1) ? - "multicodec" : rtd->codec_dai->name, num); + "multicodec" : asoc_rtd_to_codec(rtd, 0)->name, num); ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, capture, &pcm); @@ -3050,8 +3050,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->no_device_suspend = true; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", - (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, - (rtd->num_cpus > 1) ? "multicpu" : rtd->cpu_dai->name); + (rtd->num_codecs > 1) ? "multicodec" : asoc_rtd_to_codec(rtd, 0)->name, + (rtd->num_cpus > 1) ? "multicpu" : asoc_rtd_to_cpu(rtd, 0)->name); return ret; } -- 2.20.1

1 0

Applied "ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime()" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() 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 49648d02008dd75ddd84b16f9abf29f24c383519 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 30 Mar 2020 10:47:54 +0900 Subject: [PATCH] ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() rtd->cpu_dais/codec_dais are set at soc_new_pcm_runtime(). rtd->num_cpus/codecs should be set there. This patch do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87h7y6bol1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4ad64b96d893..8cef4fd6df36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -490,6 +490,8 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( */ rtd->cpu_dais = &rtd->dais[0]; rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; + rtd->num_cpus = dai_link->num_cpus; + rtd->num_codecs = dai_link->num_codecs; /* * rtd remaining settings @@ -989,7 +991,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, if (!rtd) return -ENOMEM; - rtd->num_cpus = dai_link->num_cpus; for_each_link_cpus(dai_link, i, cpu) { asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai(cpu); if (!asoc_rtd_to_cpu(rtd, i)) { @@ -1004,7 +1005,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, rtd->cpu_dai = rtd->cpu_dais[0]; /* Find CODEC from registered CODECs */ - rtd->num_codecs = dai_link->num_codecs; for_each_link_codecs(dai_link, i, codec) { asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec); if (!asoc_rtd_to_codec(rtd, i)) { -- 2.20.1

1 0

Applied "ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups 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 01faf501eefa86fa6f334bb9c71a48498de653e6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 30 Mar 2020 10:48:14 +0900 Subject: [PATCH] ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups soc_new_pcm_runtime() setups rtd, but code is very random. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87ftdqbokh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8cef4fd6df36..4d617c495864 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -492,17 +492,12 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; rtd->num_cpus = dai_link->num_cpus; rtd->num_codecs = dai_link->num_codecs; - - /* - * rtd remaining settings - */ - rtd->card = card; - rtd->dai_link = dai_link; + rtd->card = card; + rtd->dai_link = dai_link; + rtd->num = card->num_rtd++; /* see for_each_card_rtds */ list_add_tail(&rtd->list, &card->rtd_list); - rtd->num = card->num_rtd; - card->num_rtd++; return rtd; -- 2.20.1

1 0

Applied "ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais 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 1729025b04b9f242dca37f50dba8dd3705eb1ea1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 30 Mar 2020 10:48:27 +0900 Subject: [PATCH] ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais No-one is using cpu_dai/codec_dai/cpu_dais/codec_dais. Let's remove these from snd_soc_pcm_runtime Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87eetabok4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 14 ++++++++------ sound/soc/soc-core.c | 11 +++-------- 2 files changed, 11 insertions(+), 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4dff6745b067..a904086b8ef6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1143,14 +1143,16 @@ struct snd_soc_pcm_runtime { /* runtime devices */ struct snd_pcm *pcm; struct snd_compr *compr; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; - struct snd_soc_dai **dais; - struct snd_soc_dai **codec_dais; + /* + * dais = cpu_dai + codec_dai + * see + * soc_new_pcm_runtime() + * asoc_rtd_to_cpu() + * asoc_rtd_to_codec() + */ + struct snd_soc_dai **dais; unsigned int num_codecs; - - struct snd_soc_dai **cpu_dais; unsigned int num_cpus; struct delayed_work delayed_work; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d617c495864..7ea9725e33b3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -487,9 +487,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( * dais = [][][][][][][][][][][][][][][][][][] * ^cpu_dais ^codec_dais * |--- num_cpus ---|--- num_codecs --| + * see + * asoc_rtd_to_cpu() + * asoc_rtd_to_codec() */ - rtd->cpu_dais = &rtd->dais[0]; - rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; rtd->num_cpus = dai_link->num_cpus; rtd->num_codecs = dai_link->num_codecs; rtd->card = card; @@ -996,9 +997,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, snd_soc_rtd_add_component(rtd, asoc_rtd_to_cpu(rtd, i)->component); } - /* Single cpu links expect cpu and cpu_dai in runtime data */ - rtd->cpu_dai = rtd->cpu_dais[0]; - /* Find CODEC from registered CODECs */ for_each_link_codecs(dai_link, i, codec) { asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec); @@ -1011,9 +1009,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, snd_soc_rtd_add_component(rtd, asoc_rtd_to_codec(rtd, i)->component); } - /* Single codec links expect codec and codec_dai in runtime data */ - rtd->codec_dai = rtd->codec_dais[0]; - /* Find PLATFORM from registered PLATFORMs */ for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { -- 2.20.1

1 0

[PATCH v1 0/2] Support headset on Tegra boards that use WM8903
by Dmitry Osipenko 14 Apr '20

14 Apr '20

Hello, Some devices have a 4-pin jack instead of a 3-pin and currently the WM8903 configuration is hardcoded to the case of 3-pin jack in the Tegra's ASoC driver. A new device-tree property is required in order to convey that hardware has a 4-pin jack, and thus, microphone's detection needs to be done in a different way. In particular this is needed for Acer A500 tablet device that has a 4-pin headset jack, otherwise userspace sees headset instead of headphones and internal microphone isn't enabled by ALSA UCM rule when it should be. Please review and apply, thanks in advance. Dmitry Osipenko (2): dt-bindings: sound: tegra-wm8903: Document new nvidia,headset property ASoC: tegra: tegra_wm8903: Support nvidia,headset property .../devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 + sound/soc/tegra/tegra_wm8903.c | 6 +++++- 2 files changed, 6 insertions(+), 1 deletion(-) -- 2.25.1

4 10

[PATCH 0/4] ASoC: fix some compile warnings
by Jason Yan 14 Apr '20

14 Apr '20

Remove some defined but not used symbols to fix the gcc warnings. Jason Yan (4): ASoC: wm8994: remove wm1811_snd_controls and mixin_boost_tlv ASoC: wm8991: remove defined but not used 'wm8991_dapm_rxvoice_controls' ASoC: wm8990: remove some defined but unused symbols ASoC: wm8900: remove some defined but not used symbols sound/soc/codecs/wm8900.c | 6 --- sound/soc/codecs/wm8990.c | 98 --------------------------------------- sound/soc/codecs/wm8991.c | 8 ---- sound/soc/codecs/wm8994.c | 8 ---- 4 files changed, 120 deletions(-) -- 2.17.2

2 8

[PATCH 1/2] ASoC: intel: soc-acpi-intel-icl-match: remove useless 'rt1308_2_adr'
by Jason Yan 14 Apr '20

14 Apr '20

Fix the following gcc warning: sound/soc/intel/common/soc-acpi-intel-icl-match.c:90:45: warning: ‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=] static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { ^~~~~~~~~~~~ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index ef8500349f2f..16ec9f382b0f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -87,14 +87,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { } }; -static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { - { - .adr = 0x000210025D130800, - .num_endpoints = 1, - .endpoints = &single_endpoint, - } -}; - static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { { .adr = 0x000110025D130800, -- 2.17.2

3 4

[PATCH v3 1/3] dt-bindings: sound: convert rockchip spdif bindings to yaml
by Johan Jonker 14 Apr '20

14 Apr '20

Current dts files with 'spdif' nodes are manually verified. In order to automate this process rockchip-spdif.txt has to be converted to yaml. Also rk3188.dtsi, rk3288.dtsi use an extra fallback string, so change this in the documentation. Changed: "rockchip,rk3188-spdif", "rockchip,rk3066-spdif" "rockchip,rk3288-spdif", "rockchip,rk3066-spdif" Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changes v3: Add reviewed by Fix irq.h already included in arm-gic.h Change rk3188-cru-common.h to rk3188-cru.h Changed v2: dmas and dma-names layout --- .../devicetree/bindings/sound/rockchip-spdif.txt | 45 ----------- .../devicetree/bindings/sound/rockchip-spdif.yaml | 93 ++++++++++++++++++++++ 2 files changed, 93 insertions(+), 45 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt deleted file mode 100644 index ec20c1271..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ /dev/null @@ -1,45 +0,0 @@ -* Rockchip SPDIF transceiver - -The S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-spdif" - - "rockchip,rk3188-spdif" - - "rockchip,rk3228-spdif" - - "rockchip,rk3288-spdif" - - "rockchip,rk3328-spdif" - - "rockchip,rk3366-spdif" - - "rockchip,rk3368-spdif" - - "rockchip,rk3399-spdif" -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the SPDIF interrupt. -- dmas: DMA specifiers for tx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should be "tx" -- clocks: a list of phandle + clock-specifier pairs, one for each entry - in clock-names. -- clock-names: should contain following: - - "hclk": clock for SPDIF controller - - "mclk" : clock for SPDIF bus - -Required properties on RK3288: - - rockchip,grf: the phandle of the syscon node for the general register - file (GRF) - -Example for the rk3188 SPDIF controller: - -spdif: spdif@1011e000 { - compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; - reg = <0x1011e000 0x2000>; - interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dmac1_s 8>; - dma-names = "tx"; - clock-names = "hclk", "mclk"; - clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000..bfd44fd8e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,93 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip SPDIF transceiver + +description: + The S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via a coaxial or + fibre cable. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-spdif + - const: rockchip,rk3228-spdif + - const: rockchip,rk3328-spdif + - const: rockchip,rk3366-spdif + - const: rockchip,rk3368-spdif + - const: rockchip,rk3399-spdif + - items: + - enum: + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - const: rockchip,rk3066-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF bus + - description: clock for SPDIF controller + + clock-names: + items: + - const: mclk + - const: hclk + + dmas: + maxItems: 1 + + dma-names: + const: tx + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property on RK3288. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + +then: + required: + - rockchip,grf + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + }; -- 2.11.0

2 5

Applied "ASoC: rockchip-spdif: add #sound-dai-cells property" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: rockchip-spdif: add #sound-dai-cells property 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 f0c0f5a4541cb5dee1f16cebcc4e11d952394c53 Mon Sep 17 00:00:00 2001 From: Johan Jonker <jbx6244(a)gmail.com> Date: Sat, 4 Apr 2020 13:52:24 +0200 Subject: [PATCH] ASoC: rockchip-spdif: add #sound-dai-cells property '#sound-dai-cells' is required to properly interpret the list of DAI specified in the 'sound-dai' property, so add them to 'rockchip-spdif.yaml' Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20200404115225.4314-2-jbx6244@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index bfd44fd8ecbc..f381dbbf5de6 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -57,6 +57,9 @@ properties: The phandle of the syscon node for the GRF register. Required property on RK3288. + "#sound-dai-cells": + const: 0 + required: - compatible - reg @@ -65,6 +68,7 @@ required: - clock-names - dmas - dma-names + - "#sound-dai-cells" if: properties: @@ -90,4 +94,5 @@ examples: clock-names = "mclk", "hclk"; dmas = <&dmac1_s 8>; dma-names = "tx"; + #sound-dai-cells = <0>; }; -- 2.20.1

1 0

Applied "ASoC: rockchip-spdif: add power-domains property" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: rockchip-spdif: add power-domains property 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 65f1d3d9cee7e10d105756b6ef678885a23a31d7 Mon Sep 17 00:00:00 2001 From: Johan Jonker <jbx6244(a)gmail.com> Date: Sat, 4 Apr 2020 13:52:25 +0200 Subject: [PATCH] ASoC: rockchip-spdif: add power-domains property In the old txt situation we add/describe only properties that are used by the driver/hardware itself. With yaml it also filters things in a node that are used by other drivers like 'power-domains' for rk3399, so add it to 'rockchip-spdif.yaml'. Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20200404115225.4314-3-jbx6244@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index f381dbbf5de6..c467152656f7 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -51,6 +51,9 @@ properties: dma-names: const: tx + power-domains: + maxItems: 1 + rockchip,grf: $ref: /schemas/types.yaml#/definitions/phandle description: -- 2.20.1

1 0

[PATCH 1/3] ASoC: dapm: Fix regression introducing multiple copies of DAI widgets
by Charles Keepax 14 Apr '20

14 Apr '20

Refactoring was done to factor out the linking of DAI widgets into a helper function, dapm_add_valid_dai_widget. However when this was done, a regression was introduced for CODEC to CODEC links. It was over looked that the playback and capture variables persisted across all CODEC DAIs being processed, which ensured that the special DAI widget that is added for CODEC to CODEC links was only created once. This bug causes kernel panics during DAPM shutdown. To stick with the spirit of the original refactoring whilst fixing the issue, variables to hold the DAI widgets are added to snd_soc_dai_link. Furthermore the dapm_add_valid_dai_widget function is renamed to dapm_connect_dai_pair, the function only adds DAI widgets in the CODEC to CODEC case and its primary job is to add routes connecting two DAI widgets, making the original name quite misleading. Fixes: 6c4b13b51aa3 ("ASoC: Add dapm_add_valid_dai_widget helper") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc.h | 3 ++ sound/soc/soc-dapm.c | 91 +++++++++++++++++++++++++++------------------------- 2 files changed, 51 insertions(+), 43 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 13458e4fbb132..946f88a6c63d1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -790,6 +790,9 @@ struct snd_soc_dai_link { const struct snd_soc_pcm_stream *params; unsigned int num_params; + struct snd_soc_dapm_widget *playback_widget; + struct snd_soc_dapm_widget *capture_widget; + unsigned int dai_fmt; /* format to set on init */ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 679ed60d850ec..fe907f0cc709b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4283,52 +4283,63 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void dapm_add_valid_dai_widget(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - struct snd_soc_dai *codec_dai, - struct snd_soc_dai *cpu_dai) +static void dapm_connect_dai_routes(struct snd_soc_dapm_context *dapm, + struct snd_soc_dai *src_dai, + struct snd_soc_dapm_widget *src, + struct snd_soc_dapm_widget *dai, + struct snd_soc_dai *sink_dai, + struct snd_soc_dapm_widget *sink) { - struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; - struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; + dev_dbg(dapm->dev, "connected DAI link %s:%s -> %s:%s\n", + src_dai->component->name, src->name, + sink_dai->component->name, sink->name); + + if (dai) { + snd_soc_dapm_add_path(dapm, src, dai, NULL, NULL); + src = dai; + } + + snd_soc_dapm_add_path(dapm, src, sink, NULL, NULL); +} + +static void dapm_connect_dai_pair(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_dapm_widget *dai, *codec, *playback_cpu, *capture_cpu; struct snd_pcm_substream *substream; struct snd_pcm_str *streams = rtd->pcm->streams; - if (rtd->dai_link->params) { + if (dai_link->params) { playback_cpu = cpu_dai->capture_widget; capture_cpu = cpu_dai->playback_widget; } else { - playback = cpu_dai->playback_widget; - capture = cpu_dai->capture_widget; - playback_cpu = playback; - capture_cpu = capture; + playback_cpu = cpu_dai->playback_widget; + capture_cpu = cpu_dai->capture_widget; } /* connect BE DAI playback if widgets are valid */ codec = codec_dai->playback_widget; if (playback_cpu && codec) { - if (!playback) { + if (dai_link->params && !dai_link->playback_widget) { substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - playback = snd_soc_dapm_new_dai(card, substream, - "playback"); - if (IS_ERR(playback)) { + dai = snd_soc_dapm_new_dai(card, substream, "playback"); + if (IS_ERR(dai)) { dev_err(rtd->dev, "ASoC: Failed to create DAI %s: %ld\n", codec_dai->name, - PTR_ERR(playback)); + PTR_ERR(dai)); goto capture; } - - snd_soc_dapm_add_path(&card->dapm, playback_cpu, - playback, NULL, NULL); + dai_link->playback_widget = dai; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, playback_cpu->name, - codec_dai->component->name, codec->name); - - snd_soc_dapm_add_path(&card->dapm, playback, codec, - NULL, NULL); + dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu, + dai_link->playback_widget, + codec_dai, codec); } capture: @@ -4336,28 +4347,22 @@ static void dapm_add_valid_dai_widget(struct snd_soc_card *card, codec = codec_dai->capture_widget; if (codec && capture_cpu) { - if (!capture) { + if (dai_link->params && !dai_link->capture_widget) { substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; - capture = snd_soc_dapm_new_dai(card, substream, - "capture"); - if (IS_ERR(capture)) { + dai = snd_soc_dapm_new_dai(card, substream, "capture"); + if (IS_ERR(dai)) { dev_err(rtd->dev, "ASoC: Failed to create DAI %s: %ld\n", codec_dai->name, - PTR_ERR(capture)); + PTR_ERR(dai)); return; } - - snd_soc_dapm_add_path(&card->dapm, capture, - capture_cpu, NULL, NULL); + dai_link->capture_widget = dai; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, codec->name, - cpu_dai->component->name, capture_cpu->name); - - snd_soc_dapm_add_path(&card->dapm, codec, capture, - NULL, NULL); + dapm_connect_dai_routes(&card->dapm, codec_dai, codec, + dai_link->capture_widget, + cpu_dai, capture_cpu); } } @@ -4369,12 +4374,12 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, if (rtd->num_cpus == 1) { for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[0]); + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[0]); } else if (rtd->num_codecs == rtd->num_cpus) { for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[i]); + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[i]); } else { dev_err(card->dev, "N cpus to M codecs link is not supported yet\n"); -- 2.11.0

2 5

[PATCH] soc/stm/stm32_sub_sai: Add missing '\n' in log messages
by Sebastian Fricke 14 Apr '20

14 Apr '20

Message logged by 'dev_xxx()' or 'pr_xxx()' should end with a '\n'. Fixes: 3e086ed("ASoC: stm32: add SAI drivers") Signed-off-by: Sebastian Fricke <sebastian.fricke.linux(a)gmail.com> --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 0d0c9afd8791..34a7c3d6fb91 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -837,7 +837,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_32); break; default: - dev_err(cpu_dai->dev, "Data format not supported"); + dev_err(cpu_dai->dev, "Data format not supported\n"); return -EINVAL; } -- 2.20.1

2 1

[PATCH] ASoC: qcom: common: Silence duplicate parse error messages
by Bjorn Andersson 14 Apr '20

14 Apr '20

All error paths in qcom_snd_parse_of() prints more specific error messages, so silence the one in apq8096_platform_probe() and sdm845_snd_platform_probe() to avoid spamming the kernel log. Signed-off-by: Bjorn Andersson <bjorn.andersson(a)linaro.org> --- sound/soc/qcom/apq8096.c | 4 +--- sound/soc/qcom/sdm845.c | 4 +--- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index d55e3ad96716..287ad2aa27f3 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -116,10 +116,8 @@ static int apq8096_platform_probe(struct platform_device *pdev) card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); - if (ret) { - dev_err(dev, "Error parsing OF data\n"); + if (ret) goto err; - } apq8096_add_be_ops(card); ret = snd_soc_register_card(card); diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index b2de65c7f95c..68e9388ff46f 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -559,10 +559,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); - if (ret) { - dev_err(dev, "Error parsing OF data\n"); + if (ret) goto parse_dt_fail; - } data->card = card; snd_soc_card_set_drvdata(card, data); -- 2.24.0

2 1

Applied "ASoC: wsa881x: mark read_only_wordlength flag" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: wsa881x: mark read_only_wordlength flag 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 0f2a3b02274c02eb97697c4d89c019d1d21ac225 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Tue, 14 Apr 2020 12:03:47 +0100 Subject: [PATCH] ASoC: wsa881x: mark read_only_wordlength flag WSA881x works in PDM mode so the wordlength is fixed, which also makes the only field "WordLength" in DPN_BlockCtrl1 register a read-only. Writing to this register will throw up errors with Qualcomm Controller. So use ro_blockctrl1_reg flag to mark this field as read-only so that core will not write to this register. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/20200414110347.23829-1-srinivas.kandagatla@linaro… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wsa881x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index f2d6f2f81f14..d39d479e2378 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* COMP */ .num = 2, @@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* BOOST */ .num = 3, @@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* VISENSE */ .num = 4, @@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, } }; -- 2.20.1

1 0

[PATCH] ASoC: madera: Remove a couple of stray blank lines
by Charles Keepax 14 Apr '20

14 Apr '20

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/madera.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 40de9d7811d1f..a448d2a2918a9 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -1903,7 +1903,6 @@ const struct soc_enum madera_isrc_fsh[] = { MADERA_ISRC4_FSH_SHIFT, 0xf, MADERA_RATE_ENUM_SIZE, madera_rate_text, madera_rate_val), - }; EXPORT_SYMBOL_GPL(madera_isrc_fsh); @@ -1924,7 +1923,6 @@ const struct soc_enum madera_isrc_fsl[] = { MADERA_ISRC4_FSL_SHIFT, 0xf, MADERA_RATE_ENUM_SIZE, madera_rate_text, madera_rate_val), - }; EXPORT_SYMBOL_GPL(madera_isrc_fsl); @@ -1938,7 +1936,6 @@ const struct soc_enum madera_asrc1_rate[] = { MADERA_ASYNC_RATE_ENUM_SIZE, madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE, madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE), - }; EXPORT_SYMBOL_GPL(madera_asrc1_rate); @@ -1964,7 +1961,6 @@ const struct soc_enum madera_asrc2_rate[] = { MADERA_ASYNC_RATE_ENUM_SIZE, madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE, madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE), - }; EXPORT_SYMBOL_GPL(madera_asrc2_rate); -- 2.11.0

2 1

[PATCH v2 1/3] dt-bindings: sound: convert rockchip i2s bindings to yaml
by Johan Jonker 14 Apr '20

14 Apr '20

Current dts files with 'i2s' nodes are manually verified. In order to automate this process rockchip-i2s.txt has to be converted to yaml. Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changed V2: Add Reviewed-by --- .../devicetree/bindings/sound/rockchip-i2s.txt | 49 ---------- .../devicetree/bindings/sound/rockchip-i2s.yaml | 106 +++++++++++++++++++++ 2 files changed, 106 insertions(+), 49 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt deleted file mode 100644 index 54aefab71..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ /dev/null @@ -1,49 +0,0 @@ -* Rockchip I2S controller - -The I2S bus (Inter-IC sound bus) is a serial link for digital -audio data transfer between devices in the system. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-i2s": for rk3066 - - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 - - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 - - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328 - - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366 - - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 - - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the I2S interrupt. -- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should contain the following: - - "i2s_hclk": clock for I2S BUS - - "i2s_clk" : clock for I2S controller -- rockchip,playback-channels: max playback channels, if not set, 8 channels default. -- rockchip,capture-channels: max capture channels, if not set, 2 channels default. - -Required properties for controller which support multi channels -playback/capture: - -- rockchip,grf: the phandle of the syscon node for GRF register. - -Example for rk3288 I2S controller: - -i2s@ff890000 { - compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; - reg = <0xff890000 0x10000>; - interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&pdma1 0>, <&pdma1 1>; - dma-names = "tx", "rx"; - clock-names = "i2s_hclk", "i2s_clk"; - clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; - rockchip,playback-channels = <8>; - rockchip,capture-channels = <2>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml new file mode 100644 index 000000000..eff06b4b5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -0,0 +1,106 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S controller + +description: + The I2S bus (Inter-IC sound bus) is a serial link for digital + audio data transfer between devices in the system. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-i2s + - items: + - enum: + - rockchip,px30-i2s + - rockchip,rk3036-i2s + - rockchip,rk3188-i2s + - rockchip,rk3228-i2s + - rockchip,rk3288-i2s + - rockchip,rk3328-i2s + - rockchip,rk3366-i2s + - rockchip,rk3368-i2s + - rockchip,rk3399-i2s + - const: rockchip,rk3066-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for I2S controller + - description: clock for I2S BUS + + clock-names: + items: + - const: i2s_clk + - const: i2s_hclk + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + rockchip,capture-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 2 + description: + Max capture channels, if not set, 2 channels default. + + rockchip,playback-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 8 + description: + Max playback channels, if not set, 8 channels default. + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property for controllers which support multi channel + playback/capture. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3288-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>; + clock-names = "i2s_clk", "i2s_hclk"; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "tx", "rx"; + rockchip,capture-channels = <2>; + rockchip,playback-channels = <8>; + }; -- 2.11.0

3 4

Regression on Intel Atom/SST/Baytrail devices w/ v5.7-rc1
by Pierre-Louis Bossart 14 Apr '20

14 Apr '20

Hi, With Mark's for-next tree and v5.7-rc1 I get these really bad errors with the Atom/SST driver on a Baytrail device - found by accident while trying to validate UCM changes. [ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed [ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed Trying to bisect, the errors change at some point: [c77b8317ee3ab43634421afb73fdb1ea253d3d47] ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL in TDM out [ 9.387540] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for Headset Playback [ 9.387551] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for media1_in [ 9.387556] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for media0_out mix 0 [ 9.387561] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for media0_out [ 9.387566] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for pcm0_in [ 9.387570] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for modem_out mix 0 [ 9.387575] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for modem_out [ 9.387580] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for codec_out0 mix 0 [ 9.387584] bytcr_rt5640 bytcr_rt5640: ASoC: can't get playback BE for codec_out0 And then from [5d3c644773925c3568617435e42a9404a114c428] drm/mediatek: support HDMI jack status reporting to v5.6-rc1 I get those time outs... [ 21.031638] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 21.031661] intel_sst_acpi 80860F28:00: fw returned err -16 [ 21.031692] sst-mfld-platform sst-mfld-platform: ASoC: PRE_PMD: pcm0_in event failed: -16 Both 5.5 and 5.6 work though, as if we are missing a set of dependencies? I am running out of time this evening, but I'd be good to see if anyone else reproduces my findings. Thanks -Pierre

2 2

[alsa-devel] [PATCH] alsactl: avoid needless wakeups in monitor loop.
by Zev Weiss 14 Apr '20

14 Apr '20

The timeout wasn't really being used for anything; disabling it should reduce idle energy consumption slightly. Signed-off-by: Zev Weiss <zev(a)bewilderbeest.net> --- alsactl/monitor.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsactl/monitor.c b/alsactl/monitor.c index 6b090e4f5c92..6d9328d899a4 100644 --- a/alsactl/monitor.c +++ b/alsactl/monitor.c @@ -340,7 +340,7 @@ static int run_dispatcher(int epfd, int sigfd, int infd, struct list_head *srcs, int count; int i; - count = epoll_wait(epfd, epev, max_ev_count, 200); + count = epoll_wait(epfd, epev, max_ev_count, -1); if (count < 0) { err = count; break; -- 2.23.0

3 3

[alsa-devel] [PATCH] alsactl: don't exit on EINTR from epoll_wait().
by Zev Weiss 14 Apr '20

14 Apr '20

Previously, things like attaching strace to a running 'alsactl monitor' process would cause it to exit. Signed-off-by: Zev Weiss <zev(a)bewilderbeest.net> --- alsactl/monitor.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsactl/monitor.c b/alsactl/monitor.c index 6b090e4f5c92..cf4167bee165 100644 --- a/alsactl/monitor.c +++ b/alsactl/monitor.c @@ -342,6 +342,8 @@ static int run_dispatcher(int epfd, int sigfd, int infd, struct list_head *srcs, count = epoll_wait(epfd, epev, max_ev_count, 200); if (count < 0) { + if (errno == EINTR) + continue; err = count; break; } -- 2.23.0

3 2

[PATCH] ASoC: tas571x: disable regulators on failed probe
by Philipp Puschmann 14 Apr '20

14 Apr '20

If probe fails after enabling the regulators regulator_put is called for each supply without having them disabled before. This produces some warnings like WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c [<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14) [<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4) [<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4) [<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c) [<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38) [<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38) [<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c) [<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c) [<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c) [<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60) [<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc) [<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0) [<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0) [<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108) [<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88) [<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250) [<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244) [<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540) [<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8) [<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20) Fixes: 3fd6e7d9a146 (ASoC: tas571x: New driver for TI TAS571x power amplifiers) Signed-off-by: Philipp Puschmann <p.puschmann(a)pironex.de> --- sound/soc/codecs/tas571x.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 1554631cb397..5b7f9fcf6cbf 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, priv->regmap = devm_regmap_init(dev, NULL, client, priv->chip->regmap_config); - if (IS_ERR(priv->regmap)) - return PTR_ERR(priv->regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + goto disable_regs; + } priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW); if (IS_ERR(priv->pdn_gpio)) { @@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client, ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0); if (ret) - return ret; + goto disable_regs; usleep_range(50000, 60000); @@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client, */ ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0); if (ret) - return ret; + goto disable_regs; } - return devm_snd_soc_register_component(&client->dev, + ret = devm_snd_soc_register_component(&client->dev, &priv->component_driver, &tas571x_dai, 1); + if (ret) + goto disable_regs; + + return ret; + +disable_regs: + regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies); + return ret; } static int tas571x_i2c_remove(struct i2c_client *client) -- 2.26.0

2 2

Applied "ASoC: tas571x: disable regulators on failed probe" to the asoc tree
by Mark Brown 14 Apr '20

14 Apr '20

The patch ASoC: tas571x: disable regulators on failed probe 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 9df8ba7c63073508e5aa677dade48fcab6a6773e Mon Sep 17 00:00:00 2001 From: Philipp Puschmann <p.puschmann(a)pironex.de> Date: Tue, 14 Apr 2020 13:27:54 +0200 Subject: [PATCH] ASoC: tas571x: disable regulators on failed probe If probe fails after enabling the regulators regulator_put is called for each supply without having them disabled before. This produces some warnings like WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c [<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14) [<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4) [<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4) [<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c) [<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38) [<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38) [<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c) [<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c) [<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c) [<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60) [<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc) [<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0) [<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0) [<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108) [<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88) [<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250) [<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244) [<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540) [<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8) [<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20) Fixes: 3fd6e7d9a146 (ASoC: tas571x: New driver for TI TAS571x power amplifiers) Signed-off-by: Philipp Puschmann <p.puschmann(a)pironex.de> Link: https://lore.kernel.org/r/20200414112754.3365406-1-p.puschmann@pironex.de Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas571x.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 1554631cb397..5b7f9fcf6cbf 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, priv->regmap = devm_regmap_init(dev, NULL, client, priv->chip->regmap_config); - if (IS_ERR(priv->regmap)) - return PTR_ERR(priv->regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + goto disable_regs; + } priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW); if (IS_ERR(priv->pdn_gpio)) { @@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client, ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0); if (ret) - return ret; + goto disable_regs; usleep_range(50000, 60000); @@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client, */ ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0); if (ret) - return ret; + goto disable_regs; } - return devm_snd_soc_register_component(&client->dev, + ret = devm_snd_soc_register_component(&client->dev, &priv->component_driver, &tas571x_dai, 1); + if (ret) + goto disable_regs; + + return ret; + +disable_regs: + regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies); + return ret; } static int tas571x_i2c_remove(struct i2c_client *client) -- 2.20.1

1 0

[PATCH] ASoC: amd: Fix button configuration
by Akshu Agrawal 14 Apr '20

14 Apr '20

RT5682 buttons were incorrectly mapped. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 024a7ee54cd5..e499c00e0c66 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -89,9 +89,9 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) } snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ret = snd_soc_component_set_jack(component, &pco_jack, NULL); if (ret) { -- 2.20.1

2 1

[RESEND PATCH] ASoC: wsa881x: mark read_only_wordlength flag
by Srinivas Kandagatla 14 Apr '20

14 Apr '20

WSA881x works in PDM mode so the wordlength is fixed, which also makes the only field "WordLength" in DPN_BlockCtrl1 register a read-only. Writing to this register will throw up errors with Qualcomm Controller. So use ro_blockctrl1_reg flag to mark this field as read-only so that core will not write to this register. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Hi Mark, For some reason this patch was missed during last merge window, Other patch in this series is already in mainline. Without this patch audio is not functional on DB845c and other SDM845 based platforms. Can you please take this for next possible rc. Thanks, srini sound/soc/codecs/wsa881x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index f2d6f2f81f14..d39d479e2378 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* COMP */ .num = 2, @@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* BOOST */ .num = 3, @@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* VISENSE */ .num = 4, @@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, } }; -- 2.21.0

2 1

[PATCH] ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume
by Krzysztof Kozlowski 14 Apr '20

14 Apr '20

Commit 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume") removed the DAI side suspend/resume hooks and switched entirely to component suspend/resume. However the Samsung SoC s3c-i2s-v2 driver was not updated. Move the suspend/resume hooks from s3c-i2s-v2.c to s3c2412-i2s.c while changing dai to component which allows to keep the struct snd_soc_component_driver const. This fixes build errors: sound/soc/samsung/s3c-i2s-v2.c: In function ‘s3c_i2sv2_register_component’: sound/soc/samsung/s3c-i2s-v2.c:730:9: error: ‘struct snd_soc_dai_driver’ has no member named ‘suspend’ dai_drv->suspend = s3c2412_i2s_suspend; Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume") Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Only compile tested. --- sound/soc/samsung/s3c-i2s-v2.c | 57 --------------------------------- sound/soc/samsung/s3c2412-i2s.c | 56 ++++++++++++++++++++++++++++++++ 2 files changed, 56 insertions(+), 57 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 358887848293..5e95c30fb2ba 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -656,60 +656,6 @@ void s3c_i2sv2_cleanup(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup); -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warn("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warn("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warn("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - int s3c_i2sv2_register_component(struct device *dev, int id, const struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) @@ -727,9 +673,6 @@ int s3c_i2sv2_register_component(struct device *dev, int id, if (!ops->delay) ops->delay = s3c2412_i2s_delay; - dai_drv->suspend = s3c2412_i2s_suspend; - dai_drv->resume = s3c2412_i2s_resume; - return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 787a3f6e9f24..b35d828c1cfe 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -117,6 +117,60 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + u32 iismod; + + if (component->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warn("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warn("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warn("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + + pr_info("component_active %d, IISMOD %08x, IISCON %08x\n", + component->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (component->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + #define S3C2412_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -146,6 +200,8 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static const struct snd_soc_component_driver s3c2412_i2s_component = { .name = "s3c2412-i2s", + .suspend = s3c2412_i2s_suspend, + .resume = s3c2412_i2s_resume, }; static int s3c2412_iis_dev_probe(struct platform_device *pdev) -- 2.17.1

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[RFC v2] ALSA: control: fix a error handling exist in snd_ctl_elem_add
by yangerkun 14 Apr '20

14 Apr '20

CVE-2020-11725 report that 'count = info->owner' may result a SIZE_OVERFLOW. 'info->owner' represent a pid, and actually, we should use info->count. Signed-off-by: yangerkun <yangerkun(a)huawei.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) v1->v2: reword the patch head diff --git a/sound/core/control.c b/sound/core/control.c index aa0c0cf182af..c77ca7f39637 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1431,7 +1431,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, return -ENOMEM; /* Check the number of elements for this userspace control. */ - count = info->owner; + count = info->count; if (count == 0) count = 1; -- 2.21.1

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[RFC] ALSA: control: fix a mishandled exits in snd_ctl_elem_add
by yangerkun 14 Apr '20

14 Apr '20

CVE-2020-11725 report that 'count = info->owner' may result a SIZE_OVERFLOW. 'info->owner' represent a pid, and actually, we should use info->count. Signed-off-by: yangerkun <yangerkun(a)huawei.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index aa0c0cf182af..c77ca7f39637 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1431,7 +1431,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, return -ENOMEM; /* Check the number of elements for this userspace control. */ - count = info->owner; + count = info->count; if (count == 0) count = 1; -- 2.21.1

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[PATCH v7 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 14 Apr '20

14 Apr '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v7 - updated according to Nicoin's comments. - add get_pair_priv_size to replace PAIR_PRIVATE_SIZE changes in v6 - updated according to Nicoin's and Rob's comments. changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 24 +- sound/soc/fsl/fsl_asrc.c | 310 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 106 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2119 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3222 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

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