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pcm: Return unsigned int if error occurs by `read` or `write` system call
by GitHub pull_request - opened 19 Jun '22

19 Jun '22

alsa-project/alsa-lib pull request #242 was opened from Korilakkuma: `read` or `write` system call returns `-1` if error occurs. Request URL : https://github.com/alsa-project/alsa-lib/pull/242 Patch URL : https://github.com/alsa-project/alsa-lib/pull/242.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] aplay: Support setting timestamp
by Pavel Hofman 18 Jun '22

18 Jun '22

To allow enabling timestamp and specify its type, a new option --tstamp-type=TYPE is added. Recognized values are none (default), gettimeofday, monotonic, monotonic-raw. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- aplay/aplay.1 | 4 ++++ aplay/aplay.c | 32 ++++++++++++++++++++++++++++++++ 2 files changed, 36 insertions(+) diff --git a/aplay/aplay.1 b/aplay/aplay.1 index 3bba59d..d3b7dce 100644 --- a/aplay/aplay.1 +++ b/aplay/aplay.1 @@ -207,6 +207,10 @@ sampling rates, numbers of channels, period and buffer bytes/sizes/times. For raw device hw:X this option basically lists hardware capabilities of the soundcard. .TP +\fI\-\-tstamp\-type=TYPE\fP +Specifies timestamp type inside the software configuration container. +Types are: none (default), gettimeofday, monotonic, monotonic\-raw. +.TP \fI\-\-fatal\-errors\fP Disables recovery attempts when errors (e.g. xrun) are encountered; the aplay process instead aborts immediately. diff --git a/aplay/aplay.c b/aplay/aplay.c index 63a4e34..5d15a32 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -139,6 +139,8 @@ static int use_strftime = 0; volatile static int recycle_capture_file = 0; static long term_c_lflag = -1; static int dump_hw_params = 0; +static int enable_tstamp = 0; +static snd_pcm_tstamp_type_t tstamp_type = SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY; static int fd = -1; static off64_t pbrec_count = LLONG_MAX, fdcount; @@ -244,6 +246,8 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --process-id-file write the process ID here\n" " --use-strftime apply the strftime facility to the output file name\n" " --dump-hw-params dump hw_params of the device\n" +" --tstamp-type=TYPE set timestamp (TYPE: none (default), gettimeofday,\n" +" monotonic, monotonic-raw)\n" " --fatal-errors treat all errors as fatal\n" ) , command); @@ -430,6 +434,7 @@ enum { OPT_PROCESS_ID_FILE, OPT_USE_STRFTIME, OPT_DUMP_HWPARAMS, + OPT_TSTAMP_TYPE, OPT_FATAL_ERRORS, }; @@ -517,6 +522,7 @@ int main(int argc, char *argv[]) {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, + {"tstamp-type", 1, 0, OPT_TSTAMP_TYPE}, {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, 0, 'm'}, @@ -799,6 +805,23 @@ int main(int argc, char *argv[]) case OPT_DUMP_HWPARAMS: dump_hw_params = 1; break; + case OPT_TSTAMP_TYPE: + if (strcasecmp(optarg, "gettimeofday") == 0) { + enable_tstamp = 1; + tstamp_type = SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY; + } else if (strcasecmp(optarg, "monotonic") == 0) { + enable_tstamp = 1; + tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC; + } else if (strcasecmp(optarg, "monotonic-raw") == 0) { + enable_tstamp = 1; + tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW; + } else if (strcasecmp(optarg, "none") == 0) + enable_tstamp = 0; + else { + error(_("unrecognized timestamp type %s"), optarg); + return 1; + } + break; case OPT_FATAL_ERRORS: fatal_errors = 1; break; @@ -1453,6 +1476,15 @@ static void set_params(void) stop_threshold = (double) rate * stop_delay / 1000000; err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold); assert(err >= 0); + if (enable_tstamp) { + err = snd_pcm_sw_params_set_tstamp_mode(handle, swparams, SND_PCM_TSTAMP_ENABLE); + assert(err >= 0); + err = snd_pcm_sw_params_set_tstamp_type(handle, swparams, tstamp_type); + if (err < 0) { + error(_("Unable to set the requested timestamp type.")); + prg_exit(EXIT_FAILURE); + } + } if (snd_pcm_sw_params(handle, swparams) < 0) { error(_("unable to install sw params:")); -- 2.25.1

3 6

[PATCH v2 0/3] firewire: fix minor issues
by Takashi Sakamoto 18 Jun '22

18 Jun '22

This second version of patchset is revised version of previous one[1] to fix mistake of macro usage pointed out by reviewer[2]. As I note, they are not so urgent changes, thus I don't mind postponing until next merge window. [1] https://lore.kernel.org/alsa-devel/20220512111756.103008-1-o-takashi@sakamo… [2] https://lore.kernel.org/alsa-devel/87o7yvpf4t.wl-tiwai@suse.de/ Jiapeng Chong (1): firewire: convert sysfs sprintf/snprintf family to sysfs_emit Lv Ruyi (1): firewire: Fix using uninitialized value Minghao Chi (CGEL ZTE) (1): firewire: use struct_size over open coded arithmetic drivers/firewire/core-device.c | 6 ++---- drivers/firewire/core-transaction.c | 3 ++- 2 files changed, 4 insertions(+), 5 deletions(-) -- 2.34.1

2 7

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 18 Jun '22

18 Jun '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - opened 18 Jun '22

18 Jun '22

alsa-project/alsa-ucm-conf issue #172 was opened from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ASoC: SOF/soundwire: use resume_and_get on component probe
by Pierre-Louis Bossart 18 Jun '22

18 Jun '22

While testing driver bind/unbind sequences, I stumbled on a corner case where the SoundWire bus can be suspended before the ASoC card registration happens. During the registration, register accesses would then lead to timeouts. This does not happen in regular usages where the card registration happens within the 3-second time window before suspend. Adding a simple pm_runtime_resume_and_get() on component probe solves the issue, but experiments showed it was too invasive to add at the ASoC core level, with multiple regressions reported by our CI. This patchset limits the additional resume to the SOF and SoundWire codec drivers. An additional patch for the soundwire/intel component will be sent separately. Pierre-Louis Bossart (2): ASoC: SOF: pcm: use pm_resume_and_get() on component probe ASoC: codecs: soundwire: call pm_runtime_resume() in component probe sound/soc/codecs/max98373.c | 14 +++++++++++++- sound/soc/codecs/rt1308-sdw.c | 12 ++++++++++++ sound/soc/codecs/rt1316-sdw.c | 12 ++++++++++++ sound/soc/codecs/rt700.c | 5 +++++ sound/soc/codecs/rt711-sdca.c | 5 +++++ sound/soc/codecs/rt711.c | 5 +++++ sound/soc/codecs/rt715-sdca.c | 12 ++++++++++++ sound/soc/codecs/rt715.c | 12 ++++++++++++ sound/soc/sof/pcm.c | 11 +++++++++++ 9 files changed, 87 insertions(+), 1 deletion(-) -- 2.34.1

2 9

[PATCH v1] ASoC: ops: Fix integer detection for when max possible values > 1
by Stefan Binding 18 Jun '22

18 Jun '22

The standard snd_soc_info_volsw() allows a two value control to be defined as an integer control only if the control name ends in "Volume". It achieves this by creating a substring if it contains " Volume", and ensuring this exists at the end of the name. The volume substring is then used to decide whether the type is a SNDRV_CTL_ELEM_TYPE_INTEGER or SNDRV_CTL_ELEM_TYPE_BOOLEAN. However this volume substring is only computed for a two value control. This means for controls where there are more than two possible values, the substring is never created, so in this case the substring remains NULL, and the condition yields SNDRV_CTL_ELEM_TYPE_BOOLEAN, even though there are more than 2 possible values. If there are more than 2 possible values for the control, then it should always be an integer control. Fixes: aa2a4b897132 ("ASoC: ops: Fix boolean/integer detection for simple controls") Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/soc/soc-ops.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 0267e39de2a8..1970bda074d8 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -183,17 +183,16 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, if (mc->platform_max && mc->platform_max < max) max = mc->platform_max; - /* Even two value controls ending in Volume should always be integer */ if (max == 1) { + /* Even two value controls ending in Volume should always be integer */ vol_string = strstr(kcontrol->id.name, " Volume"); - if (vol_string && strcmp(vol_string, " Volume")) - vol_string = NULL; - } - - if (!vol_string) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else + if (vol_string && !strcmp(vol_string, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + } else { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + } uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; -- 2.25.1

3 2

[PATCH v2] ASoC: mediatek: mt8186: Fix mutex double unlock in GPIO request
by Fei Shao 18 Jun '22

18 Jun '22

The lockdep mechanism revealed an unbalanced unlocking on MT8186: [ 2.993966] WARNING: bad unlock balance detected! [ 2.993978] ------------------------------------- [ 2.993983] kworker/u16:1/10 is trying to release lock (gpio_request_mutex) at: [ 2.993994] [<ffffffdcd9adebf8>] mt8186_afe_gpio_request+0xf8/0x210 [ 2.994012] but there are no more locks to release! The cause is that the mutex will be double unlocked if dai is unknown during GPIO selection, and this patch fixes it. Fixes: cfa9a966f12a ("ASoC: mediatek: mt8186: support gpio control in platform driver") Signed-off-by: Fei Shao <fshao(a)chromium.org> --- Changes in v2: - Trimmed the commit message sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c index 255ffba637d3..274c0c8ec2f2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -230,7 +230,6 @@ int mt8186_afe_gpio_request(struct device *dev, bool enable, sel = enable ? MT8186_AFE_GPIO_PCM_ON : MT8186_AFE_GPIO_PCM_OFF; break; default: - mutex_unlock(&gpio_request_mutex); dev_err(dev, "%s(), invalid dai %d\n", __func__, dai); goto unlock; } -- 2.36.1.476.g0c4daa206d-goog

2 1

[PATCH] ASoC: mediatek: mt8186: Fix mutex double unlock in GPIO request
by Fei Shao 18 Jun '22

18 Jun '22

The lockdep mechanism reveals an unbalanced unlocking on MT8186: [ 2.993966] WARNING: bad unlock balance detected! [ 2.993971] 5.15.46-421fef3b44d7-lockdep #16 Not tainted [ 2.993978] ------------------------------------- [ 2.993983] kworker/u16:1/10 is trying to release lock (gpio_request_mutex) at: [ 2.993994] [<ffffffdcd9adebf8>] mt8186_afe_gpio_request+0xf8/0x210 [ 2.994012] but there are no more locks to release! [ 2.994017] [ 2.994017] other info that might help us debug this: [ 2.994022] 3 locks held by kworker/u16:1/10: [ 2.994028] #0: ffffff80c004a148 ((wq_completion)events_unbound){+.+.}-{0:0}, at: process_one_work+0x1a8/0x364 [ 2.994054] #1: ffffffc0080d3d58 (deferred_probe_work){+.+.}-{0:0}, at: process_one_work+0x1d0/0x364 [ 2.994074] #2: ffffff80c12ab9a0 (&dev->mutex){....}-{3:3}, at: __device_attach+0x44/0x150 [ 2.994096] [ 2.994096] stack backtrace: [ 2.994102] CPU: 6 PID: 10 Comm: kworker/u16:1 Not tainted 5.15.46-421fef3b44d7-lockdep #16 a2af0e2c6b3eeab6799b5aa8091c1d05d3a7bbb1 [ 2.994113] Hardware name: Google Kingler board (DT) [ 2.994120] Workqueue: events_unbound deferred_probe_work_func [ 2.994130] Call trace: [ 2.994135] dump_backtrace+0x0/0x1d4 [ 2.994144] show_stack+0x20/0x2c [ 2.994152] dump_stack_lvl+0x78/0x9c [ 2.994163] dump_stack+0x18/0x44 [ 2.994172] print_unlock_imbalance_bug+0xf0/0xf4 [ 2.994184] lock_release+0x150/0x388 [ 2.994193] __mutex_unlock_slowpath+0x4c/0x1bc [ 2.994202] mutex_unlock+0x44/0x50 [ 2.994210] mt8186_afe_gpio_request+0xf8/0x210 [ 2.994220] mt8186_afe_gpio_init+0xcc/0x134 [ 2.994230] mt8186_mt6366_rt1019_rt5682s_dev_probe+0x14c/0x220 [ 2.994238] platform_probe+0xb0/0xd0 [ 2.994247] really_probe+0xcc/0x2c8 [ 2.994253] __driver_probe_device+0xbc/0xe8 [ 2.994264] driver_probe_device+0x48/0xf0 [ 2.994271] __device_attach_driver+0xa0/0xc8 [ 2.994278] bus_for_each_drv+0x8c/0xd8 [ 2.994288] __device_attach+0xc4/0x150 [ 2.994298] device_initial_probe+0x1c/0x28 [ 2.994308] bus_probe_device+0x3c/0xa0 [ 2.994318] deferred_probe_work_func+0xa0/0xe0 [ 2.994325] process_one_work+0x208/0x364 [ 2.994334] worker_thread+0x288/0x3fc [ 2.994343] kthread+0x140/0x160 [ 2.994351] ret_from_fork+0x10/0x20 The cause is that the mutex will be double unlocked if dai is unknown during GPIO selection, and this patch fixes it. Fixes: cfa9a966f12a ("ASoC: mediatek: mt8186: support gpio control in platform driver") Signed-off-by: Fei Shao <fshao(a)chromium.org> --- sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c index 255ffba637d3..274c0c8ec2f2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -230,7 +230,6 @@ int mt8186_afe_gpio_request(struct device *dev, bool enable, sel = enable ? MT8186_AFE_GPIO_PCM_ON : MT8186_AFE_GPIO_PCM_OFF; break; default: - mutex_unlock(&gpio_request_mutex); dev_err(dev, "%s(), invalid dai %d\n", __func__, dai); goto unlock; } -- 2.36.1.476.g0c4daa206d-goog

2 3

[PATCH 0/3] ASoC: SOF: Intel: fix resume from hibernate
by Pierre-Louis Bossart 18 Jun '22

18 Jun '22

The enablement of IMR-based DSP boot helped reduce resume latency, but unfortunately the context is not saved in S4 and S5 which leads to multiple reports of boot failures. This patchset forces a full firmware reload/reboot when resuming from S4/S5 and restores functionality. Pierre-Louis Bossart (3): ASoC: SOF: pm: add explicit behavior for ACPI S1 and S2 ASoC: SOF: pm: add definitions for S4 and S5 states ASoC: SOF: Intel: disable IMR boot when resuming from ACPI S4 and S5 states sound/soc/sof/intel/hda-loader.c | 3 ++- sound/soc/sof/pm.c | 21 ++++++++++++++++++++- sound/soc/sof/sof-priv.h | 2 ++ 3 files changed, 24 insertions(+), 2 deletions(-) -- 2.34.1

2 4

[PATCH] ASoC: ops: Fix multiple value control type
by Sameer Pujar 18 Jun '22

18 Jun '22

The commit aa2a4b897132("ASoC: ops: Fix boolean/integer detection for simple controls") fixes false positives with controls not ending in " Volume" string. But it now forces boolean type for the multi value controls. Fix this by adding a max check before assigning types. Fixes: aa2a4b897132("ASoC: ops: Fix boolean/integer detection for simple controls") Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 0267e39..21be8e8 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -190,7 +190,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, vol_string = NULL; } - if (!vol_string) + if (!vol_string && max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; -- 2.7.4

3 2

[PATCH v2 0/3] ASoC: rockchip: i2s: switch BCLK to GPIO
by Judy Hsiao 17 Jun '22

17 Jun '22

The patches series is to fix the unexpected large DC output voltage of Max98357a to burn the speakers on rockchip platform when BCLK and SD_MODE are ON but LRCLK is OFF. Changes Since V2: -- Add documents of i2s pinctrl-names. -- Fix dtsi syntax error. -- Include the dtsi change and the driver change in the same series. -- Ensure that driver gets both bclk_on and bclk_off states before using them. Judy Hsiao (3): ASoC: rockchip: i2s: switch BCLK to GPIO arm64: dts: rk3399: i2s: switch BCLK to GPIO ASoC: dt-bindings: rockchip: Document pinctrl-names for i2s .../bindings/sound/rockchip-i2s.yaml | 7 + .../boot/dts/rockchip/rk3399-gru-scarlet.dtsi | 10 ++ arch/arm64/boot/dts/rockchip/rk3399.dtsi | 25 +++- sound/soc/rockchip/rockchip_i2s.c | 134 ++++++++++++++---- 4 files changed, 149 insertions(+), 27 deletions(-) -- 2.36.1.476.g0c4daa206d-goog

4 14

[PATCH] ASoC: SOF: Intel: hda: Fix compressed stream position tracking
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Commit 288fad2f71fa ("ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information") modified the PCM path only, but left the compressed data patch using an obsolete option. Move the functionality in a helper that can be called for both PCM and compressed data. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Fixes: 288fad2f71fa ("ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-pcm.c | 74 +------------------------ sound/soc/sof/intel/hda-stream.c | 94 ++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda.h | 3 + 3 files changed, 94 insertions(+), 77 deletions(-) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index dc1f743730c0d..6888e0a4665d2 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -192,79 +192,7 @@ snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, goto found; } - switch (sof_hda_position_quirk) { - case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: - /* - * This legacy code, inherited from the Skylake driver, - * mixes DPIB registers and DPIB DDR updates and - * does not seem to follow any known hardware recommendations. - * It's not clear e.g. why there is a different flow - * for capture and playback, the only information that matters is - * what traffic class is used, and on all SOF-enabled platforms - * only VC0 is supported so the work-around was likely not necessary - * and quite possibly wrong. - */ - - /* DPIB/posbuf position mode: - * For Playback, Use DPIB register from HDA space which - * reflects the actual data transferred. - * For Capture, Use the position buffer for pointer, as DPIB - * is not accurate enough, its update may be completed - * earlier than the data written to DDR. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - } else { - /* - * For capture stream, we need more workaround to fix the - * position incorrect issue: - * - * 1. Wait at least 20us before reading position buffer after - * the interrupt generated(IOC), to make sure position update - * happens on frame boundary i.e. 20.833uSec for 48KHz. - * 2. Perform a dummy Read to DPIB register to flush DMA - * position value. - * 3. Read the DMA Position from posbuf. Now the readback - * value should be >= period boundary. - */ - usleep_range(20, 21); - snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - pos = snd_hdac_stream_get_pos_posbuf(hstream); - } - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: - /* - * In case VC1 traffic is disabled this is the recommended option - */ - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: - /* - * This is the recommended option when VC1 is enabled. - * While this isn't needed for SOF platforms it's added for - * consistency and debug. - */ - pos = snd_hdac_stream_get_pos_posbuf(hstream); - break; - default: - dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", - sof_hda_position_quirk); - pos = 0; - break; - } - - if (pos >= hstream->bufsize) - pos = 0; - + pos = hda_dsp_stream_get_position(hstream, substream->stream, true); found: pos = bytes_to_frames(substream->runtime, pos); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index daeb64c495e40..d95ae17e81cc4 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -707,12 +707,13 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) } static void -hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size) +hda_dsp_compr_bytes_transferred(struct hdac_stream *hstream, int direction) { + u64 buffer_size = hstream->bufsize; u64 prev_pos, pos, num_bytes; div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos); - pos = snd_hdac_stream_get_pos_posbuf(hstream); + pos = hda_dsp_stream_get_position(hstream, direction, false); if (pos < prev_pos) num_bytes = (buffer_size - prev_pos) + pos; @@ -748,8 +749,7 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) if (s->substream && sof_hda->no_ipc_position) { snd_sof_pcm_period_elapsed(s->substream); } else if (s->cstream) { - hda_dsp_set_bytes_transferred(s, - s->cstream->runtime->buffer_size); + hda_dsp_compr_bytes_transferred(s, s->cstream->direction); snd_compr_fragment_elapsed(s->cstream); } } @@ -1009,3 +1009,89 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev) devm_kfree(sdev->dev, hda_stream); } } + +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep) +{ + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + struct sof_intel_hda_stream *hda_stream = hstream_to_sof_hda_stream(hext_stream); + struct snd_sof_dev *sdev = hda_stream->sdev; + snd_pcm_uframes_t pos; + + switch (sof_hda_position_quirk) { + case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: + /* + * This legacy code, inherited from the Skylake driver, + * mixes DPIB registers and DPIB DDR updates and + * does not seem to follow any known hardware recommendations. + * It's not clear e.g. why there is a different flow + * for capture and playback, the only information that matters is + * what traffic class is used, and on all SOF-enabled platforms + * only VC0 is supported so the work-around was likely not necessary + * and quite possibly wrong. + */ + + /* DPIB/posbuf position mode: + * For Playback, Use DPIB register from HDA space which + * reflects the actual data transferred. + * For Capture, Use the position buffer for pointer, as DPIB + * is not accurate enough, its update may be completed + * earlier than the data written to DDR. + */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + } else { + /* + * For capture stream, we need more workaround to fix the + * position incorrect issue: + * + * 1. Wait at least 20us before reading position buffer after + * the interrupt generated(IOC), to make sure position update + * happens on frame boundary i.e. 20.833uSec for 48KHz. + * 2. Perform a dummy Read to DPIB register to flush DMA + * position value. + * 3. Read the DMA Position from posbuf. Now the readback + * value should be >= period boundary. + */ + if (can_sleep) + usleep_range(20, 21); + + snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + pos = snd_hdac_stream_get_pos_posbuf(hstream); + } + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: + /* + * In case VC1 traffic is disabled this is the recommended option + */ + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: + /* + * This is the recommended option when VC1 is enabled. + * While this isn't needed for SOF platforms it's added for + * consistency and debug. + */ + pos = snd_hdac_stream_get_pos_posbuf(hstream); + break; + default: + dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", + sof_hda_position_quirk); + pos = 0; + break; + } + + if (pos >= hstream->bufsize) + pos = 0; + + return pos; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index a3118499e34fc..dc713c20ba1d9 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -565,6 +565,9 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep); + struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag); -- 2.34.1

2 1

No sound on AMD audio device
by GitHub issues - edited 17 Jun '22

17 Jun '22

alsa-project/alsa-lib issue #241 was edited from chvmq: I have a laptop huawei 14 amd 2021 I don't know where write about it. maybe there I tried ubuntu 22.01 | arch-linux and no sound at all Issue URL : https://github.com/alsa-project/alsa-lib/issues/241 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib parser error: "sequence command 'devdisall' is ignored" when attempting alsactl restore, nrestore, init, etc.
by GitHub issues - opened 17 Jun '22

17 Jun '22

alsa-project/alsa-lib issue #240 was opened from salperinlea: This may play into a much larger issue I have been having with an Intel tiger lake soundcard, which cannot be activated after my machine has been suspended, but which always loads on boot. In an attempt to solve this error, I have tried to use alsactl restore, at which point I get the error: <code><pre> sudo alsactl restore alsa-lib parser.c:1194:(parse_sequence) error: sequence command 'devdisall' is ignored alsa-lib parser.c:1194:(parse_sequence) error: sequence command 'devdisall' is ignored </code></pre> I believe the reason that this repeats is that I have two soundcards; one is an Nvidia card, hard-wired to a dGPU and the HDMI port. Info: Arch Linux 251.2, Kernel 5.18.3, KDE 5.25 DE. <code><pre> lspci | grep 'Audio' 0000:00:1f.3 Multimedia audio controller: Intel Corporation Tiger Lake-H HD Audio Controller (rev 11) 0000:01:00.1 Audio device: NVIDIA Corporation Device 228e (rev a1) </code></pre> I am not sure what other info may be of use, but I will provide it upon request. Issue URL : https://github.com/alsa-project/alsa-lib/issues/240 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/13] ASoC: SOF/Intel: remove __func__ from dev_dbg()
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

Cleanups suggested by Greg KH during SoundWire reviews, since the __func__ information can be added with the dyndbg kernel parameter [1]. The first two patches change the error level in cases where the use of dev_dbg() was inconsistent with an aborted programming sequence. [1] https://lore.kernel.org/alsa-devel/20220610023537.27223-2-yung-chuan.liao@l… Pierre-Louis Bossart (13): ASoC: SOF: Intel: hda-dsp: report error on power-up/down ASoC: SOF: Intel: hda-stream: report error on stream not opened ASoC: SOF: Intel: hda-dai: remove use of __func__ in dev_dbg ASoC: SOF: Intel: hda-stream: remove use of __func__ in dev_dbg ASoC: SOF: Intel: mtl: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-dtrace: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-loader: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-topology: remove use of __func__ in dev_dbg ASoC: SOF: ipc4-topology remove use of __func__ in dev_dbg ASoC: SOF: sof-client: remove use of __func__ in dev_dbg ASoC: SOF: ipc4: remove use of __func__ in dev_dbg ASoC: Intel: boards: hda: remove use of __func__ in dev_dbg ASoC: Intel: boards: sof_sdw: remove use of __func__ in dev_dbg sound/soc/intel/boards/hda_dsp_common.c | 4 ++-- sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 ++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/sof/intel/hda-dai.c | 6 +++--- sound/soc/sof/intel/hda-dsp.c | 4 ++-- sound/soc/sof/intel/hda-stream.c | 6 +++--- sound/soc/sof/intel/mtl.c | 3 +-- sound/soc/sof/ipc3-dtrace.c | 5 ++--- sound/soc/sof/ipc3-loader.c | 11 +++++------ sound/soc/sof/ipc3-topology.c | 4 ++-- sound/soc/sof/ipc4-topology.c | 18 +++++++++--------- sound/soc/sof/ipc4.c | 2 +- sound/soc/sof/sof-client.c | 4 ++-- 13 files changed, 35 insertions(+), 38 deletions(-) -- 2.34.1

3 15

[PATCH 00/13] ASoC: Intel: Chromebooks: remap jack pins
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

This patchset adds the missing jack pin remapping needed by upstream UCM on Chromebooks, suggested by Jaroslav Kysela [1]. These patches were succesfully tested on GeminiLake and JasperLake platforms, and generalized to other Chromebooks. [1] https://mailman.alsa-project.org/pipermail/sound-open-firmware/2022-June/00… Pierre-Louis Bossart (13): ASoC: Intel: skl_nau88l25_max98357a: remap jack pins ASoC: Intel: skl_nau88l25_ssm4567: remap jack pins ASoC: Intel: kbl_rt5663_max98927: remap jack pins ASoC: Intel: kbl_da7219_max98357a: remap jack pins ASoC: Intel: kbl_da7219_max98927: remap jack pins ASoC: Intel: kbl_rt5663_rt5514_max98927: remap jack pins ASoC: Intel: bxt_da7219_max98357a: remap jack pins ASoC: Intel: glk_rt5682_max98357a: remap jack pins ASoC: Intel: cml_rt1011_rt5682: remap jack pins ASoC: Intel: sof_cs42l42: remap jack pins ASoC: Intel: sof_da7219_max98373: remap jack pins ASoC: Intel: sof_nau8825: remap jack pins ASoC: Intel: sof_rt5682: remap jack pins sound/soc/intel/boards/bxt_da7219_max98357a.c | 21 +++++++++++++---- sound/soc/intel/boards/cml_rt1011_rt5682.c | 23 +++++++++++++++---- sound/soc/intel/boards/glk_rt5682_max98357a.c | 21 +++++++++++++---- sound/soc/intel/boards/kbl_da7219_max98357a.c | 21 +++++++++++++---- sound/soc/intel/boards/kbl_da7219_max98927.c | 21 +++++++++++++---- sound/soc/intel/boards/kbl_rt5663_max98927.c | 21 +++++++++++++---- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 21 +++++++++++++---- .../soc/intel/boards/skl_nau88l25_max98357a.c | 19 ++++++++++++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 19 ++++++++++++--- sound/soc/intel/boards/sof_cs42l42.c | 23 +++++++++++++++---- sound/soc/intel/boards/sof_da7219_max98373.c | 23 +++++++++++++++---- sound/soc/intel/boards/sof_nau8825.c | 23 +++++++++++++++---- sound/soc/intel/boards/sof_rt5682.c | 23 +++++++++++++++---- 13 files changed, 224 insertions(+), 55 deletions(-) -- 2.34.1

2 14

[PATCH 0/2] ASoC: minor kernel-doc corrections
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

Remove warnings with make W=1 Pierre-Louis Bossart (2): ASoC: cs4270: update kernel-doc ASoC: sunxi: sun4i-i2s: update kernel-doc sound/soc/codecs/cs4270.c | 1 - sound/soc/sunxi/sun4i-i2s.c | 2 ++ 2 files changed, 2 insertions(+), 1 deletion(-) -- 2.34.1

2 3

[PATCH] ASoC: twl4030: Drop legacy, non DT boot support
by Peter Ujfalusi 17 Jun '22

17 Jun '22

Legacy or non DT boot is no longer possible on systems where the tw4030/5030 is used. Drop the support for handling legacy pdata and replace it with a local board_params struct to allow further cleanups on the mfd side. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> --- sound/soc/codecs/twl4030.c | 101 ++++++++++++++++++++----------------- 1 file changed, 54 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 0ba3546ef870..87b58017094b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -34,6 +34,14 @@ #define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) +struct twl4030_board_params { + unsigned int digimic_delay; /* in ms */ + unsigned int ramp_delay_value; + unsigned int offset_cncl_path; + unsigned int hs_extmute:1; + int hs_extmute_gpio; +}; + /* codec private data */ struct twl4030_priv { unsigned int codec_powered; @@ -58,7 +66,7 @@ struct twl4030_priv { u8 carkitl_enabled, carkitr_enabled; u8 ctl_cache[TWL4030_REG_PRECKR_CTL - TWL4030_REG_EAR_CTL + 1]; - struct twl4030_codec_data *pdata; + struct twl4030_board_params *board_params; }; static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030) @@ -193,73 +201,71 @@ static void twl4030_codec_enable(struct snd_soc_component *component, int enable udelay(10); } -static void twl4030_setup_pdata_of(struct twl4030_codec_data *pdata, - struct device_node *node) +static void +twl4030_get_board_param_values(struct twl4030_board_params *board_params, + struct device_node *node) { int value; - of_property_read_u32(node, "ti,digimic_delay", - &pdata->digimic_delay); - of_property_read_u32(node, "ti,ramp_delay_value", - &pdata->ramp_delay_value); - of_property_read_u32(node, "ti,offset_cncl_path", - &pdata->offset_cncl_path); + of_property_read_u32(node, "ti,digimic_delay", &board_params->digimic_delay); + of_property_read_u32(node, "ti,ramp_delay_value", &board_params->ramp_delay_value); + of_property_read_u32(node, "ti,offset_cncl_path", &board_params->offset_cncl_path); if (!of_property_read_u32(node, "ti,hs_extmute", &value)) - pdata->hs_extmute = value; + board_params->hs_extmute = value; - pdata->hs_extmute_gpio = of_get_named_gpio(node, - "ti,hs_extmute_gpio", 0); - if (gpio_is_valid(pdata->hs_extmute_gpio)) - pdata->hs_extmute = 1; + board_params->hs_extmute_gpio = of_get_named_gpio(node, "ti,hs_extmute_gpio", 0); + if (gpio_is_valid(board_params->hs_extmute_gpio)) + board_params->hs_extmute = 1; } -static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_component *component) +static struct twl4030_board_params* +twl4030_get_board_params(struct snd_soc_component *component) { - struct twl4030_codec_data *pdata = dev_get_platdata(component->dev); + struct twl4030_board_params *board_params = NULL; struct device_node *twl4030_codec_node = NULL; twl4030_codec_node = of_get_child_by_name(component->dev->parent->of_node, "codec"); - if (!pdata && twl4030_codec_node) { - pdata = devm_kzalloc(component->dev, - sizeof(struct twl4030_codec_data), - GFP_KERNEL); - if (!pdata) { + if (twl4030_codec_node) { + board_params = devm_kzalloc(component->dev, + sizeof(struct twl4030_board_params), + GFP_KERNEL); + if (!board_params) { of_node_put(twl4030_codec_node); return NULL; } - twl4030_setup_pdata_of(pdata, twl4030_codec_node); + twl4030_get_board_param_values(board_params, twl4030_codec_node); of_node_put(twl4030_codec_node); } - return pdata; + return board_params; } static void twl4030_init_chip(struct snd_soc_component *component) { - struct twl4030_codec_data *pdata; + struct twl4030_board_params *board_params; struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); u8 reg, byte; int i = 0; - pdata = twl4030_get_pdata(component); + board_params = twl4030_get_board_params(component); - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { int ret; - if (!pdata->hs_extmute_gpio) + if (!board_params->hs_extmute_gpio) dev_warn(component->dev, "Extmute GPIO is 0 is this correct?\n"); - ret = gpio_request_one(pdata->hs_extmute_gpio, + ret = gpio_request_one(board_params->hs_extmute_gpio, GPIOF_OUT_INIT_LOW, "hs_extmute"); if (ret) { dev_err(component->dev, "Failed to get hs_extmute GPIO\n"); - pdata->hs_extmute_gpio = -1; + board_params->hs_extmute_gpio = -1; } } else { u8 pin_mux; @@ -290,14 +296,14 @@ static void twl4030_init_chip(struct snd_soc_component *component) twl4030_write(component, TWL4030_REG_ARXR2_APGA_CTL, 0x32); /* Machine dependent setup */ - if (!pdata) + if (!board_params) return; - twl4030->pdata = pdata; + twl4030->board_params = board_params; reg = twl4030_read(component, TWL4030_REG_HS_POPN_SET); reg &= ~TWL4030_RAMP_DELAY; - reg |= (pdata->ramp_delay_value << 2); + reg |= (board_params->ramp_delay_value << 2); twl4030_write(component, TWL4030_REG_HS_POPN_SET, reg); /* initiate offset cancellation */ @@ -305,7 +311,7 @@ static void twl4030_init_chip(struct snd_soc_component *component) reg = twl4030_read(component, TWL4030_REG_ANAMICL); reg &= ~TWL4030_OFFSET_CNCL_SEL; - reg |= pdata->offset_cncl_path; + reg |= board_params->offset_cncl_path; twl4030_write(component, TWL4030_REG_ANAMICL, reg | TWL4030_CNCL_OFFSET_START); @@ -692,7 +698,7 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) { unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; /* Base values for ramp delay calculation: 2^19 - 2^26 */ unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, 8388608, 16777216, 33554432, 67108864}; @@ -705,9 +711,9 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) /* Enable external mute control, this dramatically reduces * the pop-noise */ - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { - gpio_set_value(pdata->hs_extmute_gpio, 1); + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { + gpio_set_value(board_params->hs_extmute_gpio, 1); } else { hs_pop |= TWL4030_EXTMUTE; twl4030_write(component, TWL4030_REG_HS_POPN_SET, hs_pop); @@ -741,9 +747,9 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) } /* Disable external mute */ - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { - gpio_set_value(pdata->hs_extmute_gpio, 0); + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { + gpio_set_value(board_params->hs_extmute_gpio, 0); } else { hs_pop &= ~TWL4030_EXTMUTE; twl4030_write(component, TWL4030_REG_HS_POPN_SET, hs_pop); @@ -806,10 +812,10 @@ static int digimic_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; - if (pdata && pdata->digimic_delay) - twl4030_wait_ms(pdata->digimic_delay); + if (board_params && board_params->digimic_delay) + twl4030_wait_ms(board_params->digimic_delay); return 0; } @@ -2168,10 +2174,11 @@ static int twl4030_soc_probe(struct snd_soc_component *component) static void twl4030_soc_remove(struct snd_soc_component *component) { struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; - if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio)) - gpio_free(pdata->hs_extmute_gpio); + if (board_params && board_params->hs_extmute && + gpio_is_valid(board_params->hs_extmute_gpio)) + gpio_free(board_params->hs_extmute_gpio); } static const struct snd_soc_component_driver soc_component_dev_twl4030 = { -- 2.36.1

2 1

ALSA 1.2.7.1 release
by Jaroslav Kysela 17 Jun '22

17 Jun '22

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-ucm-conf alsa-plugins Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7_v1.2.7.1 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[PATCH 0/5] ALSA: rawmidi: Make code robust for external calls
by Takashi Iwai 17 Jun '22

17 Jun '22

Hi, here is a small patch set for ALSA core rawmidi code to make the code a bit more robust, especially for the case where the exported functions get called from the external drivers. Currently most of those functions assume naively that they aren't called at a wrong timing. With the patch set, it tries to harden a bit not to hit serious breakage. Takashi === Takashi Iwai (5): ALSA: rawmidi: Make internal functions local static ALSA: rawmidi: Move lock to snd_rawmidi_substream ALSA: rawmidi: Take open_mutex around parameter changes ALSA: rawmidi: Check stream state at exported functions ALSA: rawmidi: Take buffer refcount while draining output include/sound/rawmidi.h | 6 +- sound/core/rawmidi.c | 274 +++++++++++++++++++++++++--------------- 2 files changed, 170 insertions(+), 110 deletions(-) -- 2.35.3

1 5

[PATCH] ALSA: hda/realtek: Add quirk for Clevo PD70PNT
by Tim Crawford 17 Jun '22

17 Jun '22

Fixes speaker output and headset detection on Clevo PD70PNT. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- 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 b937f63d0d09..ff9a09a670ed 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2634,6 +2634,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67f1, "Clevo PC70H[PRS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x67f5, "Clevo PD70PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170SM", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7715, "Clevo X170KM-G", ALC1220_FIXUP_CLEVO_PB51ED), -- 2.35.3

2 1

Silent speaker on ThinkPad T14s Gen 2i
by GitHub issues - opened 17 Jun '22

17 Jun '22

alsa-project/alsa-ucm-conf issue #171 was opened from joakim-tjernlund: Noted that speakers are silent and not sure when that happened. I note that pavucontrol shows sound playing on HDMI / Display port 2 (which is unplugged) when I test audio in MATE volume control. It seems that HDMI now is default even though it is unplugged ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/171 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Can anyone test with AMD onboard or HDMI/DP?
by Takashi Iwai 17 Jun '22

17 Jun '22

Hi, can anyone have an AMD onboard audio device and/or AMD HDMI/DP output for testing a patch below with 5.18.x or 5.19-rc kernels? It's a pending fix (for 5.18+), but currently it can't be verified whether it causes a regression on the actual audio I/O (while it fixes the kernel crash). If the generic allocator still doesn't work as expected here, it should show some audio stuttering or such effect. Thanks! Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: memalloc: Drop x86-specific hack for WC allocations The recent report for a crash on Haswell machines implied that the x86-specific (rather hackish) implementation for write-cache memory buffer allocation in ALSA core is buggy with the recent kernel in some corner cases. This patch drops the x86-specific implementation and uses the standard dma_alloc_wc() & co generically for avoiding the bug and also for simplification. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216112 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 23 +---------------------- 1 file changed, 1 insertion(+), 22 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 15dc7160ba34..8cfdaee77905 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -431,33 +431,17 @@ static const struct snd_malloc_ops snd_dma_iram_ops = { */ static void *snd_dma_dev_alloc(struct snd_dma_buffer *dmab, size_t size) { - void *p; - - p = dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); -#ifdef CONFIG_X86 - if (p && dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - set_memory_wc((unsigned long)p, PAGE_ALIGN(size) >> PAGE_SHIFT); -#endif - return p; + return dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); } static void snd_dma_dev_free(struct snd_dma_buffer *dmab) { -#ifdef CONFIG_X86 - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - set_memory_wb((unsigned long)dmab->area, - PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT); -#endif dma_free_coherent(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); } static int snd_dma_dev_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { -#ifdef CONFIG_X86 - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); -#endif return dma_mmap_coherent(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } @@ -471,10 +455,6 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ -#ifdef CONFIG_X86 -/* On x86, share the same ops as the standard dev ops */ -#define snd_dma_wc_ops snd_dma_dev_ops -#else /* CONFIG_X86 */ static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); @@ -497,7 +477,6 @@ static const struct snd_malloc_ops snd_dma_wc_ops = { .free = snd_dma_wc_free, .mmap = snd_dma_wc_mmap, }; -#endif /* CONFIG_X86 */ #ifdef CONFIG_SND_DMA_SGBUF static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); -- 2.35.3

2 3

[PATCH 00/23] tests: refine test implementation
by GitHub pull_request - opened 17 Jun '22

17 Jun '22

alsa-project/alsa-gobject pull request #76 was opened from takaswie: Current implementation of test just supports GObject-derived object, enumerations, and flags. It's not possible to test the other types of glib/gobject elements such as boxed structure. This patchset refines test implementation. Some helper functions for boxed structure and namespace/object functions are added. The existent helper function is renamed as 'test_object' and rewritten to walk through Python MRO hierarchy to check properties, virtual methods, and signals, defined by both GObject-derived objects and interfaces. I note that the tests are not done to execute actual symbols. They are just to check they are available or not via interface described in metadata of GObject Introspection. ``` Takashi Sakamoto (23): tests: rename helper function to test object interface tests: add helper function to test enumerations and flags tests: add test script for ALSACtl.ElemId boxed structure tests: add test script for ALSATimer.DeviceId boxed structure tests: add test script for ALSATimer.TickEvent boxed structure tests: add test script for ALSATimer.TstampEvent boxed structure tests: add test script for ALSASeq.Addr boxed structure tests: add test script for ALSASeq.EventCntr boxed structure tests: add test script for ALSASeq.Event boxed structure tests: add test script for ALSASeq.EventDataConnect boxed object tests: add test script for ALSASeq.EventDataCtl boxed structure tests: add test script for ALSASeq.EventDataNote boxed structure tests: add test script for ALSASeq.EventDataQueue boxed structure tests: add test script for ALSASeq.EventDataResult boxed structure tests: add test script for ALSASeq.RemoveFilter boxed structure tests: test object type instead of its instance tests: refine helper function to test object tests: test virtual functions in object tests: add test script for ALSACtl.ElemInfoCommon interface tests: add test script for ALSACtl.ElemInfoSingleArray interface tests: add test script for ALSAHwdep.DeviceCommon interface tests: add test script for ALSASeq.QueueTimerCommon interface tests: add test scripts for namespace functions tests/alsactl-card | 8 ++-- tests/alsactl-card-info | 7 +-- tests/alsactl-elem-id | 14 +++--- tests/alsactl-elem-info-boolean | 8 ++-- tests/alsactl-elem-info-bytes | 8 ++-- tests/alsactl-elem-info-common | 24 +++++++++++ tests/alsactl-elem-info-enumerated | 10 +++-- tests/alsactl-elem-info-iec60958 | 8 ++-- tests/alsactl-elem-info-integer | 14 +++--- tests/alsactl-elem-info-integer64 | 14 +++--- tests/alsactl-elem-info-single-array | 21 +++++++++ tests/alsactl-elem-value | 7 +-- tests/alsactl-enums | 11 ++--- tests/alsactl-functions | 26 +++++++++++ tests/alsahwdep-device-common | 32 ++++++++++++++ tests/alsahwdep-device-info | 7 +-- tests/alsahwdep-enums | 11 ++--- tests/alsahwdep-functions | 27 ++++++++++++ tests/alsarawmidi-enums | 11 ++--- tests/alsarawmidi-functions | 27 ++++++++++++ tests/alsarawmidi-stream-pair | 8 ++-- tests/alsarawmidi-substream-info | 7 +-- tests/alsarawmidi-substream-params | 7 +-- tests/alsarawmidi-substream-status | 7 +-- tests/alsaseq-addr | 21 +++++++++ tests/alsaseq-client-info | 7 +-- tests/alsaseq-client-pool | 7 +-- tests/alsaseq-enums | 13 +++--- tests/alsaseq-event | 61 ++++++++++++++++++++++++++ tests/alsaseq-event-cntr | 18 ++++++++ tests/alsaseq-event-data-connect | 21 +++++++++ tests/alsaseq-event-data-ctl | 23 ++++++++++ tests/alsaseq-event-data-note | 27 ++++++++++++ tests/alsaseq-event-data-queue | 33 ++++++++++++++ tests/alsaseq-event-data-result | 24 +++++++++++ tests/alsaseq-functions | 35 +++++++++++++++ tests/alsaseq-port-info | 7 +-- tests/alsaseq-queue-info | 7 +-- tests/alsaseq-queue-status | 7 +-- tests/alsaseq-queue-tempo | 7 +-- tests/alsaseq-queue-timer-alsa | 14 +++--- tests/alsaseq-queue-timer-common | 22 ++++++++++ tests/alsaseq-remove-filter | 21 +++++++++ tests/alsaseq-subscribe-data | 7 +-- tests/alsaseq-system-info | 7 +-- tests/alsaseq-user-client | 13 ++++-- tests/alsatimer-device-id | 18 +++----- tests/alsatimer-device-info | 7 +-- tests/alsatimer-device-params | 7 +-- tests/alsatimer-device-status | 7 +-- tests/alsatimer-enums | 10 ++--- tests/alsatimer-functions | 29 +++++++++++++ tests/alsatimer-instance-info | 7 +-- tests/alsatimer-instance-params | 7 +-- tests/alsatimer-instance-status | 7 +-- tests/alsatimer-tick-event | 19 +++++++++ tests/alsatimer-tstamp-event | 20 +++++++++ tests/alsatimer-user-instance | 14 +++--- tests/helper.py | 64 +++++++++++++++++++++++----- tests/meson.build | 22 ++++++++++ 60 files changed, 800 insertions(+), 164 deletions(-) create mode 100644 tests/alsactl-elem-info-common create mode 100644 tests/alsactl-elem-info-single-array create mode 100644 tests/alsactl-functions create mode 100644 tests/alsahwdep-device-common create mode 100644 tests/alsahwdep-functions create mode 100644 tests/alsarawmidi-functions create mode 100644 tests/alsaseq-addr create mode 100644 tests/alsaseq-event create mode 100644 tests/alsaseq-event-cntr create mode 100644 tests/alsaseq-event-data-connect create mode 100644 tests/alsaseq-event-data-ctl create mode 100644 tests/alsaseq-event-data-note create mode 100644 tests/alsaseq-event-data-queue create mode 100644 tests/alsaseq-event-data-result create mode 100644 tests/alsaseq-functions create mode 100644 tests/alsaseq-queue-timer-common create mode 100644 tests/alsaseq-remove-filter create mode 100644 tests/alsatimer-functions create mode 100644 tests/alsatimer-tick-event create mode 100644 tests/alsatimer-tstamp-event ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/76 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/76.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/2] ALSA: x86: intel_hdmi_audio: pm_runtime updates
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

While cleaning up the code to use pm_runtime_resume_and_get(), I realized we never called pm_runtime_enable() for this driver. This is potentially very invasive and error-prone, so feedback and test results are welcome - all my BYT/CHT devices are not longer functional unfortunately, and our CI MinnowBoard uses the 'other' HDMI solution based on HDaudio. In addition, the resume was assumed to be always successful, which isn't a very good idea. Pierre-Louis Bossart (2): ALSA: x86: intel_hdmi_audio: enable pm_runtime and set autosuspend delay ALSA: x86: intel_hdmi_audio: use pm_runtime_resume_and_get() sound/x86/intel_hdmi_audio.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) -- 2.34.1

2 3

[PATCH] ALSA: hda: intel-nhlt: remove use of __func__ in dev_dbg
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH <gregkh(a)linuxfoundation.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/hda/intel-nhlt.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 4063da3782833..9db5ccd9aa2db 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -55,8 +55,8 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) /* find max number of channels based on format_configuration */ if (fmt_configs->fmt_count) { - dev_dbg(dev, "%s: found %d format definitions\n", - __func__, fmt_configs->fmt_count); + dev_dbg(dev, "found %d format definitions\n", + fmt_configs->fmt_count); for (i = 0; i < fmt_configs->fmt_count; i++) { struct wav_fmt_ext *fmt_ext; @@ -66,9 +66,9 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (fmt_ext->fmt.channels > max_ch) max_ch = fmt_ext->fmt.channels; } - dev_dbg(dev, "%s: max channels found %d\n", __func__, max_ch); + dev_dbg(dev, "max channels found %d\n", max_ch); } else { - dev_dbg(dev, "%s: No format information found\n", __func__); + dev_dbg(dev, "No format information found\n"); } if (cfg->device_config.config_type != NHLT_CONFIG_TYPE_MIC_ARRAY) { @@ -95,17 +95,16 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) } if (dmic_geo > 0) { - dev_dbg(dev, "%s: Array with %d dmics\n", __func__, dmic_geo); + dev_dbg(dev, "Array with %d dmics\n", dmic_geo); } if (max_ch > dmic_geo) { - dev_dbg(dev, "%s: max channels %d exceed dmic number %d\n", - __func__, max_ch, dmic_geo); + dev_dbg(dev, "max channels %d exceed dmic number %d\n", + max_ch, dmic_geo); } } } - dev_dbg(dev, "%s: dmic number %d max_ch %d\n", - __func__, dmic_geo, max_ch); + dev_dbg(dev, "dmic number %d max_ch %d\n", dmic_geo, max_ch); return dmic_geo; } -- 2.34.1

2 1

[PATCH] ALSA: hda: intel-dspcfg: use SOF for UpExtreme and UpExtreme11 boards
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

The UpExtreme BIOS reports microphones that are not physically present, so this module ends-up selecting SOF, while the UpExtreme11 BIOS does not report microphones so the snd-hda-intel driver is selected. For consistency use SOF unconditionally in autodetection mode. The use of the snd-hda-intel driver can still be enabled with 'options snd-intel-dspcfg dsp_driver=1' Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index a8fe01764b254..ec9cbb219bc14 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -196,6 +196,12 @@ static const struct config_entry config_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), } }, + { + .ident = "UP-WHL", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + } + }, {} } }, @@ -358,6 +364,12 @@ static const struct config_entry config_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), } }, + { + .ident = "UPX-TGL", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + } + }, {} } }, -- 2.34.1

2 1

[PATCH] firewire: cdev: fix potential leak of kernel stack due to uninitialized value
by Takashi Sakamoto 17 Jun '22

17 Jun '22

Recent change brings potential leak of value on kernel stack to userspace due to uninitialized value. This commit fixes the bug. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: baa914cd81f ("firewire: add kernel API to access CYCLE_TIME register") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- drivers/firewire/core-cdev.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/firewire/core-cdev.c b/drivers/firewire/core-cdev.c index 8e9670036e5c..032ee56c34a3 100644 --- a/drivers/firewire/core-cdev.c +++ b/drivers/firewire/core-cdev.c @@ -1211,7 +1211,7 @@ static int ioctl_get_cycle_timer2(struct client *client, union ioctl_arg *arg) struct fw_cdev_get_cycle_timer2 *a = &arg->get_cycle_timer2; struct fw_card *card = client->device->card; struct timespec64 ts = {0, 0}; - u32 cycle_time; + u32 cycle_time = 0; int ret = 0; local_irq_disable(); -- 2.34.1

2 4

Regression: Upgrade to 1.2.7 Causes No HDMI Audio Device on AMD Trinity APU
by GitHub issues - opened 17 Jun '22

17 Jun '22

alsa-project/alsa-lib issue #239 was opened from lovenemesis: After upgrading to alsa-lib 1.2.7, HDMI Audio from AMD A10-5800K "Trinity" series APU stopped working. HDMI Audio is no longer presented as a Output Device in GNOME Setting - Audio. aplay -l shows error for HDMI device: `aplay: device_list:304: control digital audio info (0): Input/output error` **Downgrading to alsa-lib to 1.2.6** restore the HDMI audio device on A10-5800K system. Below are some system info I collected when running alsa-lib-1.2.6 and alsa-lib-1.2.7 [alsa-lib-1.2.6.txt](https://github.com/alsa-project/alsa-lib/files/8923675/… [alsa-lib-1.2.7.txt](https://github.com/alsa-project/alsa-lib/files/8923676/… It appears to me that kernel, pipewire, wireplumber versions have no impact on the occurrence of the issue. It is also reported in [RedHat Bugzilla](https://bugzilla.redhat.com/show_bug.cgi?id=2097574) Issue URL : https://github.com/alsa-project/alsa-lib/issues/239 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: [PATCH v2] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Mark Brown 16 Jun '22

16 Jun '22

On Wed, Jun 15, 2022 at 10:09:55AM -0500, David Owens wrote: > Convert to managed versions of sysfs and clk allocation to simplify > unbinding and error handling in probe. Managed sysfs node > creation specifically addresses the following error seen the second time > probe is attempted after sdma_pcm_platform_register() previously requsted > probe deferral: This doesn't apply against current code (either for-5.19 or for-5.20), please check and resend.

1 0

Re: [PATCH v2] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Péter Ujfalusi 16 Jun '22

16 Jun '22

Hi David, On 6/15/22 18:09, David Owens wrote: > Convert to managed versions of sysfs and clk allocation to simplify > unbinding and error handling in probe. Managed sysfs node > creation specifically addresses the following error seen the second time > probe is attempted after sdma_pcm_platform_register() previously requsted > probe deferral: > > sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' I have tried to looked up devm_sysfs_remove_group(), which did not existed, but you found the correct devm ;) Looks good, thank you! Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> > Signed-off-by: David Owens <dowens(a)precisionplanting.com> > --- > sound/soc/ti/omap-mcbsp-priv.h | 2 -- > sound/soc/ti/omap-mcbsp-st.c | 21 ++++----------------- > sound/soc/ti/omap-mcbsp.c | 26 ++++---------------------- > 3 files changed, 8 insertions(+), 41 deletions(-) > > diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h > index 7865cda4bf0a..da519ea1f303 100644 > --- a/sound/soc/ti/omap-mcbsp-priv.h > +++ b/sound/soc/ti/omap-mcbsp-priv.h > @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, > > /* Sidetone specific API */ > int omap_mcbsp_st_init(struct platform_device *pdev); > -void omap_mcbsp_st_cleanup(struct platform_device *pdev); > - > int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); > int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); > > diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c > index 0bc7d26c660a..402a57a502e6 100644 > --- a/sound/soc/ti/omap-mcbsp-st.c > +++ b/sound/soc/ti/omap-mcbsp-st.c > @@ -292,14 +292,11 @@ static ssize_t st_taps_store(struct device *dev, > > static DEVICE_ATTR_RW(st_taps); > > -static const struct attribute *sidetone_attrs[] = { > +static struct attribute *sidetone_attrs[] = { > &dev_attr_st_taps.attr, > NULL, > }; > - > -static const struct attribute_group sidetone_attr_group = { > - .attrs = (struct attribute **)sidetone_attrs, > -}; > +ATTRIBUTE_GROUPS(sidetone); > > int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp) > { > @@ -347,7 +344,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) > if (!st_data) > return -ENOMEM; > > - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); > + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); > if (IS_ERR(st_data->mcbsp_iclk)) { > dev_warn(mcbsp->dev, > "Failed to get ick, sidetone might be broken\n"); > @@ -359,7 +356,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) > if (!st_data->io_base_st) > return -ENOMEM; > > - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); > + ret = devm_device_add_group(mcbsp->dev, &sidetone_group); > if (ret) > return ret; > > @@ -368,16 +365,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) > return 0; > } > > -void omap_mcbsp_st_cleanup(struct platform_device *pdev) > -{ > - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); > - > - if (mcbsp->st_data) { > - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); > - clk_put(mcbsp->st_data->mcbsp_iclk); > - } > -} > - > static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_info *uinfo) > { > diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c > index 4479d74f0a45..395493a2d965 100644 > --- a/sound/soc/ti/omap-mcbsp.c > +++ b/sound/soc/ti/omap-mcbsp.c > @@ -595,16 +595,13 @@ static ssize_t dma_op_mode_store(struct device *dev, > > static DEVICE_ATTR_RW(dma_op_mode); > > -static const struct attribute *additional_attrs[] = { > +static struct attribute *additional_attrs[] = { > &dev_attr_max_tx_thres.attr, > &dev_attr_max_rx_thres.attr, > &dev_attr_dma_op_mode.attr, > NULL, > }; > - > -static const struct attribute_group additional_attr_group = { > - .attrs = (struct attribute **)additional_attrs, > -}; > +ATTRIBUTE_GROUPS(additional); > > /* > * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. > @@ -702,8 +699,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) > mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; > mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; > > - ret = sysfs_create_group(&mcbsp->dev->kobj, > - &additional_attr_group); > + ret = devm_device_add_group(mcbsp->dev, &additional_group); > if (ret) { > dev_err(mcbsp->dev, > "Unable to create additional controls\n"); > @@ -711,16 +707,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) > } > } > > - ret = omap_mcbsp_st_init(pdev); > - if (ret) > - goto err_st; > - > - return 0; > - > -err_st: > - if (mcbsp->pdata->buffer_size) > - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); > - return ret; > + return omap_mcbsp_st_init(pdev); > } > > /* > @@ -1431,11 +1418,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) > if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) > cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); > > - if (mcbsp->pdata->buffer_size) > - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); > - > - omap_mcbsp_st_cleanup(pdev); > - > return 0; > } > > -- > 2.34.1 > > This email is intended solely for the use of the individual to whom it is addressed and may contain confidential and/or privileged material. Any views or opinions presented are solely those of the author and do not necessarily represent those of Precision Planting. If you are not the intended recipient, be advised that you have received this email in error and that any use, dissemination, forwarding, printing, or copying of this email is strictly prohibited. Neither AGCO nor the sender accepts any responsibility for viruses, and it is your responsibility to scan, and virus check the e-mail and its attachment(s) (if any). -- Péter

1 0

[PATCH] ASoC: SOF: Intel: IPC4: enable IMR boot
by Peter Ujfalusi 16 Jun '22

16 Jun '22

IPC4 based firmwares have unconditional support for IMR boot. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-loader.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 9e99f376f2b3..bca9dc5917f4 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -538,7 +538,8 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) /* Check if IMR boot is usable */ if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) && - sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT) + (sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT || + sdev->pdata->ipc_type == SOF_INTEL_IPC4)) hdev->imrboot_supported = true; } -- 2.36.1

2 1

[PATCH] ASoC: SOF: ipc4-topology: Fix error code in sof_ipc4_volume_put()
by Dan Carpenter 16 Jun '22

16 Jun '22

The sof_ipc4_volume_put() function returns type bool so returning -ENOENT means returning true. Return false instead. Fixes: 955e84fc0b6d ("ASoC: SOF: ipc4-topology: Add control IO ops") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/ipc4-control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c index 95ee121dd3cf..0d5a578c3496 100644 --- a/sound/soc/sof/ipc4-control.c +++ b/sound/soc/sof/ipc4-control.c @@ -142,7 +142,7 @@ static bool sof_ipc4_volume_put(struct snd_sof_control *scontrol, if (!widget_found) { dev_err(scomp->dev, "Failed to find widget for kcontrol %s\n", scontrol->name); - return -ENOENT; + return false; } ret = sof_ipc4_set_volume_data(sdev, swidget, scontrol); -- 2.35.1

3 2

[PATCH] ASoC: Intel: Skylake: remove redundant re-assignments to pointer array
by Colin Ian King 16 Jun '22

16 Jun '22

There are two occurrences where the pointer array is being assigned a value that is never read, the pointer gets updated in the next iteration of a loop. These assignments are redundant and can be removed. Cleans up clang scan-build warnings: sound/soc/intel/skylake/skl-topology.c:2953:3: warning: Value stored to 'array' is never read [deadcode.DeadStores] sound/soc/intel/skylake/skl-topology.c:3602:3: warning: Value stored to 'array' is never read [deadcode.DeadStores] Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/intel/skylake/skl-topology.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 9bdf020a2b64..e06eac592da1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2950,9 +2950,6 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, block_size = ret; off += array->size; - array = (struct snd_soc_tplg_vendor_array *) - (tplg_w->priv.data + off); - data = (tplg_w->priv.data + off); if (block_type == SKL_TYPE_TUPLE) { @@ -3599,9 +3596,6 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, block_size = ret; off += array->size; - array = (struct snd_soc_tplg_vendor_array *) - (manifest->priv.data + off); - data = (manifest->priv.data + off); if (block_type == SKL_TYPE_TUPLE) { -- 2.35.3

3 2

[PATCH] ASoC: tegra: Fix clock DAI format on Tegra210
by Sameer Pujar 16 Jun '22

16 Jun '22

I2S reset failures are seen on Tegra210 and later platforms. This indicates absence of I2S bit clock, which is required to perform the reset operation. Following failures are seen with I2S based tests on Tegra210 and later: tegra210-i2s 2901100.i2s: timeout: failed to reset I2S for playback tegra210-i2s 2901100.i2s: ASoC: PRE_PMU: I2S2 RX event failed: -110 tegra210-i2s 2901100.i2s: timeout: failed to reset I2S for capture tegra210-i2s 2901100.i2s: ASoC: PRE_PMU: I2S2 TX event failed: -110 The commit d92ad6633fa7 ("ASoC: tegra: Update to use set_fmt_new callback") regressed I2S functionality on Tegra platforms. Basically it flipped clock provider and consumer DAI formats. This configures Tegra I2S in consumer mode by default now and there is none to provide bit clock during loopback tests. The external codec based tests also fail because both Tegra I2S and codec I2S get configured in consumer mode. ASoC core flips the DAI format before calling set_fmt() for CPU DAIs. This is negated in above commit. Fix this by swapping SND_SOC_DAIFMT_BC_FC and SND_SOC_DAIFMT_BP_FP switch cases. Fixes: d92ad6633fa7 ("ASoC: tegra: Update to use set_fmt_new callback") Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/tegra/tegra210_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index a289458..01c76ba 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -215,10 +215,10 @@ static int tegra210_i2s_set_fmt(struct snd_soc_dai *dai, mask = I2S_CTRL_MASTER_EN_MASK; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_BC_FC: val = 0; break; - case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BP_FP: val = I2S_CTRL_MASTER_EN; break; default: -- 2.7.4

3 2

[linux-next:master] BUILD REGRESSION 8cb8311e95e3bb58bd84d6350365f14a718faa6d
by kernel test robot 16 Jun '22

16 Jun '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 8cb8311e95e3bb58bd84d6350365f14a718faa6d Add linux-next specific files for 20220525 Error/Warning reports: https://lore.kernel.org/linux-mm/202204291924.vTGZmerI-lkp@intel.com https://lore.kernel.org/linux-mm/202205031017.4TwMan3l-lkp@intel.com https://lore.kernel.org/linux-mm/202205041248.WgCwPcEV-lkp@intel.com https://lore.kernel.org/linux-mm/202205150051.3RzuooAG-lkp@intel.com https://lore.kernel.org/linux-mm/202205150117.sd6HzBVm-lkp@intel.com https://lore.kernel.org/lkml/202205100617.5UUm3Uet-lkp@intel.com https://lore.kernel.org/llvm/202205251645.gusu3spL-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) drivers/gpu/drm/amd/amdgpu/amdgpu_discovery.c:1364:5: warning: no previous prototype for 'amdgpu_discovery_get_mall_info' [-Wmissing-prototypes] drivers/gpu/drm/amd/amdgpu/soc21.c:171:6: warning: no previous prototype for 'soc21_grbm_select' [-Wmissing-prototypes] drivers/gpu/drm/solomon/ssd130x-spi.c:154:35: warning: 'ssd130x_spi_table' defined but not used [-Wunused-const-variable=] drivers/net/wireless/intel/iwlwifi/pcie/trans.c:1093:9: warning: 'CAUSE' macro redefined [-Wmacro-redefined] drivers/video/fbdev/omap/hwa742.c:492:5: warning: no previous prototype for 'hwa742_update_window_async' [-Wmissing-prototypes] fs/buffer.c:2254:5: warning: stack frame size (2144) exceeds limit (1024) in 'block_read_full_folio' [-Wframe-larger-than] fs/ntfs/aops.c:378:12: warning: stack frame size (2216) exceeds limit (1024) in 'ntfs_read_folio' [-Wframe-larger-than] Unverified Error/Warning (likely false positive, please contact us if interested): .__mulsi3.o.cmd: No such file or directory Makefile:686: arch/h8300/Makefile: No such file or directory Makefile:765: arch/h8300/Makefile: No such file or directory arch/Kconfig:10: can't open file "arch/h8300/Kconfig" arch/riscv/purgatory/kexec-purgatory.c:1860:9: sparse: sparse: trying to concatenate 29720-character string (8191 bytes max) drivers/gpu/drm/bridge/adv7511/adv7511.h:229:17: warning: 'ADV7511_REG_CEC_RX_FRAME_HDR' defined but not used [-Wunused-const-variable=] drivers/gpu/drm/bridge/adv7511/adv7511.h:235:17: warning: 'ADV7511_REG_CEC_RX_FRAME_LEN' defined but not used [-Wunused-const-variable=] drivers/infiniband/hw/hns/hns_roce_hw_v2.c:309:9: sparse: sparse: dubious: x & !y drivers/pinctrl/meson/pinctrl-meson8-pmx.c:60:25: warning: Value stored to 'func' during its initialization is never read [clang-analyzer-deadcode.DeadStores] drivers/staging/vt6655/card.c:758:16: sparse: sparse: cast to restricted __le64 drivers/vhost/vdpa.c:595 vhost_vdpa_unlocked_ioctl() warn: maybe return -EFAULT instead of the bytes remaining? kernel/bpf/helpers.c:1468:29: sparse: sparse: symbol 'bpf_dynptr_from_mem_proto' was not declared. Should it be static? kernel/bpf/helpers.c:1490:29: sparse: sparse: symbol 'bpf_dynptr_from_mem_proto' was not declared. Should it be static? kernel/bpf/helpers.c:1516:29: sparse: sparse: symbol 'bpf_dynptr_read_proto' was not declared. Should it be static? kernel/bpf/helpers.c:1542:29: sparse: sparse: symbol 'bpf_dynptr_write_proto' was not declared. Should it be static? kernel/bpf/helpers.c:1569:29: sparse: sparse: symbol 'bpf_dynptr_data_proto' was not declared. Should it be static? make[1]: *** No rule to make target 'arch/h8300/Makefile'. mm/shmem.c:1948 shmem_getpage_gfp() warn: should '(((1) << 12) / 512) << folio_order(folio)' be a 64 bit type? sound/soc/intel/avs/ipc.c:87:5-24: atomic_dec_and_test variation before object free at line 88. {standard input}:3488: Error: unknown pseudo-op: `.l28' Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- alpha-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- arc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- arm-allmodconfig | |-- arch-arm-mach-omap2-dma.c:Unneeded-variable:errata-Return-on-line | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- drivers-video-fbdev-omap-hwa742.c:warning:no-previous-prototype-for-hwa742_update_window_async | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static |-- arm-allyesconfig | |-- arch-arm-mach-omap2-dma.c:Unneeded-variable:errata-Return-on-line | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- drivers-video-fbdev-omap-hwa742.c:warning:no-previous-prototype-for-hwa742_update_window_async | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static |-- arm64-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- arm64-allyesconfig | |-- arch-arm64-kernel-signal.c:sparse:sparse:dereference-of-noderef-expression | |-- arch-arm64-kernel-signal.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-user_ctxs-noderef-__user-user-got-struct-user_ctxs | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_read_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_write_proto-was-not-declared.-Should-it-be-static | `-- kernel-stackleak.c:sparse:sparse:symbol-stackleak_erase_off_task_stack-was-not-declared.-Should-it-be-static |-- csky-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- csky-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- csky-randconfig-s032-20220524 | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_data_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_read_proto-was-not-declared.-Should-it-be-static | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_write_proto-was-not-declared.-Should-it-be-static |-- h8300-allmodconfig | |-- Makefile:arch-h8300-Makefile:No-such-file-or-directory | |-- arch-Kconfig:can-t-open-file-arch-h8300-Kconfig | `-- make:No-rule-to-make-target-arch-h8300-Makefile-. |-- h8300-allyesconfig | |-- Makefile:arch-h8300-Makefile:No-such-file-or-directory | |-- arch-Kconfig:can-t-open-file-arch-h8300-Kconfig | `-- make:No-rule-to-make-target-arch-h8300-Makefile-. |-- h8300-buildonly-randconfig-r004-20220524 | |-- Makefile:arch-h8300-Makefile:No-such-file-or-directory | |-- arch-Kconfig:can-t-open-file-arch-h8300-Kconfig | `-- make:No-rule-to-make-target-arch-h8300-Makefile-. |-- h8300-randconfig-r033-20220524 | |-- Makefile:arch-h8300-Makefile:No-such-file-or-directory | |-- arch-Kconfig:can-t-open-file-arch-h8300-Kconfig | `-- make:No-rule-to-make-target-arch-h8300-Makefile-. |-- i386-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used | `-- drivers-gpu-drm-solomon-ssd13-spi.c:warning:ssd13_spi_table-defined-but-not-used |-- i386-randconfig-a012 | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | `-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used |-- i386-randconfig-a014 | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | `-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used |-- i386-randconfig-m021 | `-- mm-shmem.c-shmem_getpage_gfp()-warn:should-((()-)-)-folio_order(folio)-be-a-bit-type |-- ia64-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- ia64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- ia64-randconfig-r036-20220524 | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- m68k-allyesconfig | |-- drivers-block-paride-bpck.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-comm.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-dstr.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-epat.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-epia.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-friq.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-frpw.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-kbic.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-on26.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-ppc6lnx.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-comedi-drivers-aio_aio12_8.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-comedi-drivers-das16m1.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-comedi-drivers-ni_at_ao.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-comedi-drivers-ni_daq_700.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-net-ethernet-apne.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-net-ethernet-xircom-xirc2ps_cs.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-tty-ipwireless-hardware.c:sparse:sparse:incorrect-type-in-initializer-(different-base-types)-expected-restricted-__le16-usertype-raw_data-got-int | `-- drivers-tty-ipwireless-hardware.c:sparse:sparse:incorrect-type-in-initializer-(different-base-types)-expected-unsigned-short-unused-usertype-__v-got-restricted-__le16-assigned-usertype-raw_data |-- microblaze-randconfig-m031-20220524 | `-- drivers-vhost-vdpa.c-vhost_vdpa_unlocked_ioctl()-warn:maybe-return-EFAULT-instead-of-the-bytes-remaining |-- mips-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- mips-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | `-- sound-soc-intel-avs-ipc.c:atomic_dec_and_test-variation-before-object-free-at-line-. |-- openrisc-randconfig-s032-20220524 | `-- __mulsi3.o.cmd:No-such-file-or-directory |-- parisc-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- parisc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- powerpc-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- powerpc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- riscv-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- riscv-allyesconfig | |-- arch-riscv-kernel-machine_kexec.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-const-got-void-noderef-__user-buf | |-- arch-riscv-purgatory-kexec-purgatory.c:sparse:sparse:trying-to-concatenate-character-string-(-bytes-max) | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- kernel-fork.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-atomic_t-usertype-lock-got-struct-atomic_t-noderef-__rcu | `-- kernel-seccomp.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-atomic_t-usertype-lock-got-struct-atomic_t-noderef-__rcu |-- riscv-randconfig-r042-20220524 | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- s390-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | `-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int |-- s390-randconfig-r014-20220524 | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- sh-buildonly-randconfig-r003-20220524 | `-- standard-input:Error:unknown-pseudo-op:l28 |-- sparc-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_read_proto-was-not-declared.-Should-it-be-static | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_write_proto-was-not-declared.-Should-it-be-static |-- sparc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_read_proto-was-not-declared.-Should-it-be-static | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_write_proto-was-not-declared.-Should-it-be-static |-- x86_64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used | `-- drivers-gpu-drm-solomon-ssd13-spi.c:warning:ssd13_spi_table-defined-but-not-used |-- x86_64-randconfig-a011 | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | `-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used |-- xtensa-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select `-- xtensa-allyesconfig |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select clang_recent_errors |-- arm-randconfig-c002-20220524 | `-- drivers-pinctrl-meson-pinctrl-meson8-pmx.c:warning:Value-stored-to-func-during-its-initialization-is-never-read-clang-analyzer-deadcode.DeadStores |-- hexagon-randconfig-r011-20220524 | `-- fs-buffer.c:warning:stack-frame-size-()-exceeds-limit-()-in-block_read_full_folio |-- hexagon-randconfig-r035-20220524 | |-- fs-buffer.c:warning:stack-frame-size-()-exceeds-limit-()-in-block_read_full_folio | `-- fs-ntfs-aops.c:warning:stack-frame-size-()-exceeds-limit-()-in-ntfs_read_folio `-- mips-randconfig-r022-20220524 `-- drivers-net-wireless-intel-iwlwifi-pcie-trans.c:warning:CAUSE-macro-redefined elapsed time: 858m configs tested: 94 configs skipped: 3 gcc tested configs: arm allmodconfig arm allyesconfig arm64 allyesconfig arm defconfig arm64 defconfig mips allyesconfig riscv allyesconfig um x86_64_defconfig riscv allmodconfig um i386_defconfig mips allmodconfig powerpc allmodconfig s390 allmodconfig m68k allmodconfig powerpc allyesconfig s390 allyesconfig m68k allyesconfig sparc allyesconfig parisc allyesconfig sh allmodconfig h8300 allyesconfig arc allyesconfig alpha allyesconfig nios2 allyesconfig m68k mac_defconfig arc nsimosci_hs_smp_defconfig sh ul2_defconfig mips maltaup_xpa_defconfig xtensa iss_defconfig ia64 defconfig ia64 allmodconfig ia64 allyesconfig m68k defconfig nios2 defconfig alpha defconfig csky defconfig xtensa allyesconfig arc defconfig parisc defconfig parisc64 defconfig s390 defconfig i386 debian-10.3-kselftests i386 debian-10.3 i386 defconfig i386 allyesconfig sparc defconfig powerpc allnoconfig i386 randconfig-a001 i386 randconfig-a003 i386 randconfig-a005 x86_64 randconfig-a013 x86_64 randconfig-a011 x86_64 randconfig-a015 i386 randconfig-a014 i386 randconfig-a012 riscv randconfig-r042-20220524 arc randconfig-r043-20220524 s390 randconfig-r044-20220524 x86_64 randconfig-a004 x86_64 randconfig-a002 x86_64 randconfig-a006 riscv allnoconfig riscv nommu_k210_defconfig riscv rv32_defconfig riscv nommu_virt_defconfig riscv defconfig x86_64 rhel-8.3-func x86_64 rhel-8.3-kselftests x86_64 rhel-8.3-kunit x86_64 rhel-8.3-syz x86_64 defconfig x86_64 kexec x86_64 rhel-8.3 x86_64 allyesconfig clang tested configs: mips bcm63xx_defconfig mips lemote2f_defconfig mips tb0287_defconfig arm ep93xx_defconfig arm am200epdkit_defconfig powerpc tqm5200_defconfig i386 randconfig-a002 i386 randconfig-a006 i386 randconfig-a004 x86_64 randconfig-a012 x86_64 randconfig-a014 x86_64 randconfig-a016 i386 randconfig-a013 i386 randconfig-a011 i386 randconfig-a015 hexagon randconfig-r045-20220524 hexagon randconfig-r041-20220524 x86_64 randconfig-a005 x86_64 randconfig-a001 x86_64 randconfig-a003 -- 0-DAY CI Kernel Test Service https://01.org/lkp

7 10

[PATCH] ASoC: SOF: mediatek: Fix error code in probe
by Dan Carpenter 15 Jun '22

15 Jun '22

This should return PTR_ERR() instead of IS_ERR(). Fixes: e0100bfd383c ("ASoC: SOF: mediatek: Add mt8186 ipc support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 3333a0634e29..e006532caf2f 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -392,7 +392,7 @@ static int mt8186_dsp_probe(struct snd_sof_dev *sdev) PLATFORM_DEVID_NONE, pdev, sizeof(*pdev)); if (IS_ERR(priv->ipc_dev)) { - ret = IS_ERR(priv->ipc_dev); + ret = PTR_ERR(priv->ipc_dev); dev_err(sdev->dev, "failed to create mtk-adsp-ipc device\n"); goto err_adsp_off; } -- 2.35.1

5 4

USB-Audio: no Audient iD4 profile for USB ID 2708:0003
by GitHub issues - opened 15 Jun '22

15 Jun '22

alsa-project/alsa-ucm-conf issue #170 was opened from tiran: PR https://github.com/alsa-project/alsa-ucm-conf/pull/151 added a profile for Audient iD4 audio interface with USB ID ``2708:0009``. I own a version of the device with ``2708:0003``. The configuration for ``2708:0009`` does not work for version ``2708:0003``. I hacked the config file from ``alsa-ucm-1.2.7-1.fc36.noarch`` package and ``systemctl --user restart pipewire pipewire-pulse``. After restart the audio interface no longer shows up in KDE. ``pw-dump`` shows only a disabled/off profile for the device. @robertgzr @perexg Also see https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1119 [alsactl.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8910772/al… [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8910773/lsus… [pw-dump.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8910774/pw… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/170 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ASoC: SOF: Intel: Add support for MeteorLake
by Bard Liao 15 Jun '22

15 Jun '22

Add platform abstraction for the Meteor Lake platform. This platform has significant differences compared to the TGL/ADL generation: it relies on new hardware using the code name 'ACE' and only supports the INTEL_IPC4 protocol and firmware architecture based on the Zephyr RTOS Bard Liao (2): ASoC: SOC: Intel: introduce cl_init callback ASoC: SOF: Intel: Add support for MeteorLake (MTL) include/linux/soundwire/sdw_intel.h | 2 + sound/soc/sof/intel/Kconfig | 16 + sound/soc/sof/intel/Makefile | 4 +- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 2 + sound/soc/sof/intel/hda-loader.c | 15 +- sound/soc/sof/intel/hda.h | 4 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/mtl.c | 800 ++++++++++++++++++++++++++++ sound/soc/sof/intel/mtl.h | 76 +++ sound/soc/sof/intel/pci-mtl.c | 71 +++ sound/soc/sof/intel/shim.h | 2 + sound/soc/sof/intel/tgl.c | 4 + 13 files changed, 994 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/intel/mtl.c create mode 100644 sound/soc/sof/intel/mtl.h create mode 100644 sound/soc/sof/intel/pci-mtl.c -- 2.17.1

2 3

[PATCH] ASoC: tegra: Fix MBDRC bypass mode check
by Sameer Pujar 15 Jun '22

15 Jun '22

MBDRC supports different modes of operation. There is no configuration required for bypass mode. The hw_params() call does not filter bypass mode correctly and it leads to following Smatch static checker warning: sound/soc/tegra/tegra210_mbdrc.c:778 tegra210_mbdrc_hw_params() warn: bitwise AND condition is false here Fix this condition by using proper mode mask and just return for bypass mode. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 7358a803c778 ("ASoC: tegra: Add Tegra210 based OPE driver") Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/tegra/tegra210_mbdrc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra210_mbdrc.c b/sound/soc/tegra/tegra210_mbdrc.c index 7d9da33..d786daa 100644 --- a/sound/soc/tegra/tegra210_mbdrc.c +++ b/sound/soc/tegra/tegra210_mbdrc.c @@ -775,7 +775,9 @@ int tegra210_mbdrc_hw_params(struct snd_soc_component *cmpnt) regmap_read(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, &val); - if (val & TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) + val &= TEGRA210_MBDRC_CFG_MBDRC_MODE_MASK; + + if (val == TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) return 0; for (i = 0; i < MBDRC_NUM_BAND; i++) { -- 2.7.4

2 1

[PATCH v2 0/2] Add pinctrl support adsp based platforms
by Srinivasa Rao Mandadapu 15 Jun '22

15 Jun '22

This patch set is to add pinctrl support adsp enabled sc7280 platforms. Changes Since V1: -- Update commit message. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add compatible string for adsp based platforms pinctrl: qcom: sc7280: Add lpi pinctrl variant data for adsp based targets .../bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 4 +++- drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 14 ++++++++++++++ 2 files changed, 17 insertions(+), 1 deletion(-) -- 2.7.4

3 8

[v1] ASoC: rockchip: i2s: switch BCLK to GPIO
by Judy Hsiao 15 Jun '22

15 Jun '22

We discoverd that the state of BCLK on, LRCLK off and SD_MODE on may cause the speaker melting issue. Removing LRCLK while BCLK is present can cause unexpected output behavior including a large DC output voltage as described in the Max98357a datasheet. In order to: 1. prevent BCLK from turning on by other component. 2. keep BCLK and LRCLK being present at the same time This patch switches BCLK to GPIO func before LRCLK output, and configures BCLK func back during LRCLK is output. Without this fix, BCLK is turned on 11 ms earlier than LRCK by the da7219. With this fix, BCLK is turned on only 0.4 ms earlier than LRCK by the rockchip codec. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/rockchip/rockchip_i2s.c | 162 ++++++++++++++++++++++++------ 1 file changed, 130 insertions(+), 32 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 47a3971a9ce1..cecaadd54b48 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -13,6 +13,7 @@ #include <linux/of_gpio.h> #include <linux/of_device.h> #include <linux/clk.h> +#include <linux/pinctrl/consumer.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/spinlock.h> @@ -54,8 +55,40 @@ struct rk_i2s_dev { const struct rk_i2s_pins *pins; unsigned int bclk_ratio; spinlock_t lock; /* tx/rx lock */ + struct pinctrl *pinctrl; + struct pinctrl_state *bclk_on; + struct pinctrl_state *bclk_off; }; +static int i2s_pinctrl_select_bclk_on(struct rk_i2s_dev *i2s) +{ + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_on)) + ret = pinctrl_select_state(i2s->pinctrl, + i2s->bclk_on); + + if (ret) + dev_err(i2s->dev, "bclk enable failed %d\n", ret); + + return ret; +} + +static int i2s_pinctrl_select_bclk_off(struct rk_i2s_dev *i2s) +{ + + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_off)) + ret = pinctrl_select_state(i2s->pinctrl, + i2s->bclk_off); + + if (ret) + dev_err(i2s->dev, "bclk disable failed %d\n", ret); + + return ret; +} + static int i2s_runtime_suspend(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); @@ -92,38 +125,49 @@ static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) return snd_soc_dai_get_drvdata(dai); } -static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + if (ret < 0) + goto end; - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->tx_start = true; } else { i2s->tx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + ret = regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); @@ -138,44 +182,57 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } - spin_unlock(&i2s->lock); +end: + spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } -static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); + if (ret < 0) + goto end; - regmap_update_bits(i2s->regmap, I2S_XFER, + ret = regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->rx_start = true; } else { i2s->rx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, + ret = regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_STOP | I2S_XFER_RXS_STOP); - + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, + ret = regmap_update_bits(i2s->regmap, I2S_CLR, I2S_CLR_TXC | I2S_CLR_RXC, I2S_CLR_TXC | I2S_CLR_RXC); - + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ while (val) { regmap_read(i2s->regmap, I2S_CLR, &val); @@ -187,7 +244,12 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } +end: spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -425,17 +487,26 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 1); + ret = rockchip_snd_rxctrl(i2s, 1); else - rockchip_snd_txctrl(i2s, 1); + ret = rockchip_snd_txctrl(i2s, 1); + /* Do not turn on bclk if lrclk open fails. */ + if (ret < 0) + return ret; + i2s_pinctrl_select_bclk_on(i2s); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 0); - else - rockchip_snd_txctrl(i2s, 0); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!i2s->tx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_rxctrl(i2s, 0); + } else { + if (!i2s->rx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_txctrl(i2s, 0); + } break; default: ret = -EINVAL; @@ -736,6 +807,33 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } i2s->bclk_ratio = 64; + i2s->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(i2s->pinctrl)) + dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + + i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, + "bclk_on"); + if (IS_ERR_OR_NULL(i2s->bclk_on)) + dev_err(&pdev->dev, "failed to find i2s default state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk state\n"); + + i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, + "bclk_off"); + if (IS_ERR_OR_NULL(i2s->bclk_off)) + dev_err(&pdev->dev, "failed to find i2s gpio state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); + + i2s_pinctrl_select_bclk_off(i2s); + + i2s->playback_dma_data.addr = res->start + I2S_TXDR; + i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->playback_dma_data.maxburst = 4; + + i2s->capture_dma_data.addr = res->start + I2S_RXDR; + i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->capture_dma_data.maxburst = 4; dev_set_drvdata(&pdev->dev, i2s); -- 2.36.1.476.g0c4daa206d-goog

2 1

ACPI REV override for Dell XPS 13 9343
by Cezary Rojewski 15 Jun '22

15 Jun '22

Hello, It's been a while since catpt-driver [1] has been introduced to provide full support for Broadwell (BDW) machines with Intel DSP. For BDW, audio devices can make use of DSP only in I2S mode. In 2015 Rafael and Dominik provided quirk [2] for Dell XPS 13 9343. Given the description: _For example, based on what ACPI exports as the supported revision, Dell XPS 13 (2015) configures its audio device to either work in HDA mode or in I2S mode, where the former is supposed to be used on Linux until the latter is fully supported (in the kernel as well as in user space)._ It's clear that such configuration was not fully supported back then. I believe now it is. Perhaps it is time to let the quirk in mention go? By that I mean just the relevant entry, not the ACPI_REV_OVERRIDE_POSSIBLE functionality as a whole. [1]: https://lore.kernel.org/all/20200929141247.8058-1-cezary.rojewski@intel.com/ [2]: https://lore.kernel.org/alsa-devel/5115175.tbCl1JQVaV@vostro.rjw.lan/ Regards, Czarek

3 5

[bug report] ASoC: tegra: Add Tegra210 based OPE driver
by Dan Carpenter 15 Jun '22

15 Jun '22

Hello Sameer Pujar, The patch 7358a803c778: "ASoC: tegra: Add Tegra210 based OPE driver" from Jun 3, 2022, leads to the following Smatch static checker warning: sound/soc/tegra/tegra210_mbdrc.c:778 tegra210_mbdrc_hw_params() warn: bitwise AND condition is false here sound/soc/tegra/tegra210_mbdrc.c 769 int tegra210_mbdrc_hw_params(struct snd_soc_component *cmpnt) 770 { 771 struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); 772 const struct tegra210_mbdrc_config *conf = &mbdrc_init_config; 773 u32 val = 0; 774 unsigned int i; 775 776 regmap_read(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, &val); 777 --> 778 if (val & TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS is zero so this can't be true. #define TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS (0 << TEGRA210_MBDRC_CFG_MBDRC_MODE_SHIFT) The common false positive with this warning is that the macro is something which depends on the .config and in that case I just add it to the list in smatch_data/kernel.unconstant_macros. But in this case the macro is just always zero... Is there a plan to make it configurable or something? 779 return 0; 780 781 for (i = 0; i < MBDRC_NUM_BAND; i++) { 782 const struct tegra210_mbdrc_band_params *params = 783 &conf->band_params[i]; 784 785 u32 reg_off = i * TEGRA210_MBDRC_FILTER_PARAM_STRIDE; 786 787 tegra210_mbdrc_write_ram(ope->mbdrc_regmap, 788 reg_off + TEGRA210_MBDRC_CFG_RAM_CTRL, 789 reg_off + TEGRA210_MBDRC_CFG_RAM_DATA, 790 0, (u32 *)&params->biquad_params[0], 791 TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5); 792 } 793 return 0; 794 } regards, dan carpenter

2 3

pcm: Add error handler for `fgets`
by GitHub pull_request - opened 15 Jun '22

15 Jun '22

alsa-project/alsa-lib pull request #238 was opened from Korilakkuma: `fgets` function returns `NULL` if error occured. Therefore, end loop in the case. Request URL : https://github.com/alsa-project/alsa-lib/pull/238 Patch URL : https://github.com/alsa-project/alsa-lib/pull/238.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Any way to fuse static libasound with alsa-plugins so no dlopens are performed?
by GitHub issues - opened 14 Jun '22

14 Jun '22

alsa-project/alsa-plugins issue #45 was opened from MIvanchev: See title. I have a project that might benefit from such a mix. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/45 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ASoC: cs35l45: Add endianness flag in snd_soc_component_driver
by Charles Keepax 14 Jun '22

14 Jun '22

The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. Fixes: 0d463d016000 ("ASoC: cs35l45: Add driver for Cirrus Logic CS35L45 Smart Amp") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index c94edfce4b720..d15b3b77c7eb0 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -500,6 +500,8 @@ static const struct snd_soc_component_driver cs35l45_component = { .num_controls = ARRAY_SIZE(cs35l45_controls), .name = "cs35l45", + + .endianness = 1, }; static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) -- 2.30.2

2 1

[PATCH] ASoC: ops: Fix boolean/integer detection for simple controls
by Mark Brown 14 Jun '22

14 Jun '22

The standard snd_soc_info_volsw() detects if a control is a volume control and needs to be reported as an integer even if it only has two values by looking for the string " Volume" in the control name. This results in false positives if the control has a name like "HP Volume Ramp Switch" since any " Volume" is matched, not just a trailing one. Fix this by making sure that we only match at the end of the control name. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e11109dcaa50..0267e39de2a8 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,13 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + const char *vol_string = NULL; int max; max = uinfo->value.integer.max = mc->max - mc->min; if (mc->platform_max && mc->platform_max < max) max = mc->platform_max; - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + /* Even two value controls ending in Volume should always be integer */ + if (max == 1) { + vol_string = strstr(kcontrol->id.name, " Volume"); + if (vol_string && strcmp(vol_string, " Volume")) + vol_string = NULL; + } + + if (!vol_string) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; -- 2.30.2

1 1

Re: [PATCH] ASoC: ti: omap-mcbsp: duplicate sysfs failure after PROBE_DEFER
by Mark Brown 14 Jun '22

14 Jun '22

On Tue, Jun 14, 2022 at 10:59:31AM -0500, David Owens wrote: > The call to sdma_pcm_platform_register() can return PROBE_DEFER, leading > to omap_mcbsp_init() being called multiple times. sysfs node creation > fails in subsequent calls to omap_mcbsp_init(), which prevents > the driver from ever successfully probing. The resulting errors can be > seen during boot: > > [ 1.749328] sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' > [ 1.759643] CPU: 0 PID: 6 Comm: kworker/u2:0 Not tainted 5.18.0-yocto-standard #1 > [ 1.767181] Hardware name: Generic OMAP36xx (Flattened Device Tree) > [ 1.773498] Workqueue: events_unbound deferred_probe_work_func > [ 1.779449] unwind_backtrace from show_stack+0x10/0x14 > [ 1.784729] show_stack from sysfs_warn_dup+0x4c/0x60 Please think hard before including complete backtraces in upstream reports, they are very large and contain almost no useful information relative to their size so often obscure the relevant content in your message. If part of the backtrace is usefully illustrative (it often is for search engines if nothing else) then it's usually better to pull out the relevant sections. > +++ b/sound/soc/ti/omap-mcbsp.c > @@ -1403,6 +1403,10 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) > mcbsp->dev = &pdev->dev; > platform_set_drvdata(pdev, mcbsp); > > + ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx"); > + if (ret) > + return ret; > + > ret = omap_mcbsp_init(pdev); > if (ret) > return ret; > @@ -1412,13 +1416,9 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) > omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; > } > > - ret = devm_snd_soc_register_component(&pdev->dev, > + return devm_snd_soc_register_component(&pdev->dev, > &omap_mcbsp_component, > &omap_mcbsp_dai, 1); > - if (ret) > - return ret; > - > - return sdma_pcm_platform_register(&pdev->dev, "tx", "rx"); > } It's not clear to me how this fixes the problem, your commit message doesn't mention how? I was expecting to see more error handling paths being added to unwind the sysfs allocation, or a conversion to devm. As things stand it's not clear to me that the error won't persist in the case where we defer registering the component.

1 0

[PATCH 0/2] ASoC/SoundWire: Intel: add sdw BE dai trigger
by Bard Liao 14 Jun '22

14 Jun '22

For SOF IPC4, we need to set pipeline state in BE DAI trigger. @Vinod, Could you ack if the soundwire patch looks good to you? And we can go through the ASoC tree since the change is mainly on ASoC. Bard Liao (2): soundwire: Intel: add trigger callback ASoC: SOF: Intel: add trigger callback into sdw_callback drivers/soundwire/intel.c | 8 ++++++++ include/linux/soundwire/sdw_intel.h | 1 + sound/soc/sof/intel/hda-dai.c | 15 ++++++++++++--- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 1 + 5 files changed, 23 insertions(+), 4 deletions(-) -- 2.17.1

3 6

[PATCH] ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support
by Bard Liao 14 Jun '22

14 Jun '22

Some SoundWire hardware topologies rely on different amplifiers or capture devices connected on different links. These devices need to be 'aggregated', remain synchronized and be handled as a single logical device. In the IPC3 solution, the aggregation for amplifiers was handled by a firmware 'demux' component. In the IPC4 solution, the demux component is not needed, the gateway component can handle multiple ALH/DMA transfers at the same time. This change makes the topology slightly more complicated in that only one ALH DAI will be connected in the topology with the gateway. The other DAIs that are part of the 'aggregated' dailink are not shown in the DAPM graph as connected to the gateway, but they will however be activated thanks to a feature in soc-dapm.c where events are forwarded to all DAIs in the dailink (see soc_dapm_stream_event). The topology also sets the same stream name for all widgets, dais and dailinks, so a search for the stream name helps identify cases where SoundWire/ALH aggregation is needed. Signed-off-by: Bard Liao <yung-chuan.liao(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/sof/ipc4-topology.c | 68 ++++++++++++++++++++++++++++++++--- sound/soc/sof/ipc4-topology.h | 11 ++++++ 2 files changed, 74 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index cb0f0823b8eb..3c949298e007 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -19,6 +19,8 @@ #define SOF_IPC4_GAIN_PARAM_ID 0 #define SOF_IPC4_TPLG_ABI_SIZE 6 +static DEFINE_IDA(alh_group_ida); + static const struct sof_topology_token ipc4_sched_tokens[] = { {SOF_TKN_SCHED_LP_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pipeline, lp_mode)} @@ -478,7 +480,9 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_ALH: { + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct sof_ipc4_alh_configuration_blob *blob; + struct snd_sof_widget *w; blob = kzalloc(sizeof(*blob), GFP_KERNEL); if (!blob) { @@ -486,6 +490,14 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) goto err; } + list_for_each_entry(w, &sdev->widget_list, list) { + if (w->widget->sname && + strcmp(w->widget->sname, swidget->widget->sname)) + continue; + + blob->alh_cfg.count++; + } + ipc4_copier->copier_config = (uint32_t *)blob; ipc4_copier->data.gtw_cfg.config_length = sizeof(*blob) >> 2; break; @@ -844,6 +856,17 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) struct snd_sof_dai *dai = swidget->private; ipc4_copier = dai->private; + if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { + struct sof_ipc4_alh_configuration_blob *blob; + unsigned int group_id; + + blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; + if (blob->alh_cfg.count > 1) { + group_id = SOF_IPC4_NODE_INDEX(ipc4_copier->data.gtw_cfg.node_id) - + ALH_MULTI_GTW_BASE; + ida_free(&alh_group_ida, group_id); + } + } } if (ipc4_copier) { @@ -973,6 +996,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct sof_ipc4_copier_data *copier_data; struct snd_pcm_hw_params *ref_params; struct sof_ipc4_copier *ipc4_copier; + struct snd_sof_dai *dai; struct snd_mask *fmt; int out_sample_valid_bits; size_t ref_audio_fmt_size; @@ -1022,7 +1046,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: { - struct snd_sof_dai *dai = swidget->private; + dai = swidget->private; ipc4_copier = (struct sof_ipc4_copier *)dai->private; copier_data = &ipc4_copier->data; @@ -1077,22 +1101,56 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, */ if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { struct sof_ipc4_alh_configuration_blob *blob; + struct sof_ipc4_copier_data *alh_data; + struct sof_ipc4_copier *alh_copier; + struct snd_sof_widget *w; + u32 ch_mask = 0; u32 ch_map; int i; blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; - /* TODO: add aggregation mode support */ - blob->alh_cfg.count = 1; - blob->alh_cfg.mapping[0].alh_id = copier_data->gtw_cfg.node_id; + blob->gw_attr.lp_buffer_alloc = 0; /* Get channel_mask from ch_map */ ch_map = copier_data->base_config.audio_fmt.ch_map; for (i = 0; ch_map; i++) { if ((ch_map & 0xf) != 0xf) - blob->alh_cfg.mapping[0].channel_mask |= BIT(i); + ch_mask |= BIT(i); ch_map >>= 4; } + + /* + * Set each gtw_cfg.node_id to blob->alh_cfg.mapping[] + * for all widgets with the same stream name + */ + i = 0; + list_for_each_entry(w, &sdev->widget_list, list) { + if (w->widget->sname && + strcmp(w->widget->sname, swidget->widget->sname)) + continue; + + dai = w->private; + alh_copier = (struct sof_ipc4_copier *)dai->private; + alh_data = &alh_copier->data; + blob->alh_cfg.mapping[i].alh_id = alh_data->gtw_cfg.node_id; + blob->alh_cfg.mapping[i].channel_mask = ch_mask; + i++; + } + if (blob->alh_cfg.count > 1) { + int group_id; + + group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT, + GFP_KERNEL); + + if (group_id < 0) + return group_id; + + /* add multi-gateway base */ + group_id += ALH_MULTI_GTW_BASE; + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(group_id); + } } } } diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 1a9c0627bae9..3bc2fe38c733 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -42,6 +42,17 @@ #define ALH_MAX_NUMBER_OF_GTW 16 +/* + * The base of multi-gateways. Multi-gateways addressing starts from + * ALH_MULTI_GTW_BASE and there are ALH_MULTI_GTW_COUNT multi-sources + * and ALH_MULTI_GTW_COUNT multi-sinks available. + * Addressing is continuous from ALH_MULTI_GTW_BASE to + * ALH_MULTI_GTW_BASE + ALH_MULTI_GTW_COUNT - 1. + */ +#define ALH_MULTI_GTW_BASE 0x50 +/* A magic number from FW */ +#define ALH_MULTI_GTW_COUNT 8 + /** * struct sof_ipc4_pipeline - pipeline config data * @priority: Priority of this pipeline -- 2.17.1

2 1

[PATCH] ALSA: pcm: Test for "silence" field in struct "pcm_format_data"
by Fabio M. De Francesco 14 Jun '22

14 Jun '22

Syzbot reports "KASAN: null-ptr-deref Write in snd_pcm_format_set_silence".[1] It is due to missing validation of the "silence" field of struct "pcm_format_data" in "pcm_formats" array. Add a test for valid "pat" and, if it is not so, return -EINVAL. [1] https://lore.kernel.org/lkml/000000000000d188ef05dc2c7279@google.com/ Reported-and-tested-by: syzbot+205eb15961852c2c5974(a)syzkaller.appspotmail.com Signed-off-by: Fabio M. De Francesco <fmdefrancesco(a)gmail.com> --- I wasn't able to figure out the commit for the "Fixes:" tag. If this patch is good, can someone please help with providing this missing information? sound/core/pcm_misc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 4866aed97aac..5588b6a1ee8b 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -433,7 +433,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; width = pcm_formats[(INT)format].phys; /* physical width */ pat = pcm_formats[(INT)format].silence; - if (! width) + if (!width || !pat) return -EINVAL; /* signed or 1 byte data */ if (pcm_formats[(INT)format].signd == 1 || width <= 8) { -- 2.34.1

3 8

seq: add boxed structure for event
by GitHub pull_request - opened 14 Jun '22

14 Jun '22

alsa-project/alsa-gobject pull request #75 was opened from takaswie: ``` Current implementation provides ALSASeq.EventCntr to handle batch of sequencer events. Parameters of each event are set/get by public methods of it with index of event. But it's necesarrily convenient. This patchset adds boxed structure to express each of the event. Some public methods are rewritten according to it, with code refactoring. ALSASeq.EventCntr is largely rewritten with serializer/deserializer for flattened event buffer. Takashi Sakamoto (12): update README seq: event-cntr/event-data-queue: obsolete timestamp functions by adding alternatives seq: tstamp: remove unused boxed structure seq: remove-filter: remove unused error arguments seq: enums: rename to tstamp mode seq: port-info: rename properties for time stamp seq: enums: obsolete ALSASeqSubscribeFlag enumerations seq: event: add boxed structure for event seq: user-client: add public API to schedule list of events seq: user-client: rewrite public API to schedule single event seq: event-cntr: rewrite as boxed structure seq: event-cntr: add internal API to serialize list of events README.rst | 18 +- samples/seq | 96 +- src/seq/alsaseq-enum-types.h | 49 +- src/seq/alsaseq.h | 3 +- src/seq/alsaseq.map | 133 +-- src/seq/event-cntr.c | 1709 ++-------------------------------- src/seq/event-cntr.h | 115 +-- src/seq/event-data-queue.c | 52 +- src/seq/event-data-queue.h | 13 +- src/seq/event.c | 1687 +++++++++++++++++++++++++++++++++ src/seq/event.h | 112 +++ src/seq/meson.build | 4 +- src/seq/port-info.c | 118 ++- src/seq/privates.h | 5 + src/seq/remove-filter.c | 31 +- src/seq/remove-filter.h | 17 +- src/seq/subscribe-data.c | 115 ++- src/seq/tstamp.c | 81 -- src/seq/tstamp.h | 24 - src/seq/user-client.c | 153 ++- src/seq/user-client.h | 18 +- tests/alsaseq-enums | 20 +- tests/alsaseq-event-cntr | 60 -- tests/alsaseq-port-info | 4 +- tests/alsaseq-subscribe-data | 4 +- tests/alsaseq-user-client | 1 + tests/meson.build | 1 - 27 files changed, 2452 insertions(+), 2191 deletions(-) create mode 100644 src/seq/event.c create mode 100644 src/seq/event.h delete mode 100644 src/seq/tstamp.c delete mode 100644 src/seq/tstamp.h delete mode 100644 tests/alsaseq-event-cntr ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/75 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/75.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ASoC: SOF: Compile and runtime IPC version selection
by Peter Ujfalusi 14 Jun '22

14 Jun '22

The new IPC4 version is only supported by Intel platforms, iMX, AMD and MediaTek only uses the standard SOF IPC. There is no need for these platforms to build kernel support for IPC4 as it is just dead code for them. SND_SOC_SOF_IPC3 and SND_SOC_SOF_INTEL_IPC4 is introduced to allow compile time selection and exclusion of IPC implementations. To avoid randconfig failures add also support for runtime selection of the IPC ops in ipc.c based on sdev->pdata->ipc_type Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/Kconfig | 7 +++++++ sound/soc/sof/Makefile | 16 ++++++++++++---- sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/imx/Kconfig | 1 + sound/soc/sof/intel/Kconfig | 11 +++++++++++ sound/soc/sof/ipc.c | 24 ++++++++++++++++++------ sound/soc/sof/mediatek/Kconfig | 1 + 7 files changed, 51 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 4542868cd730..e90f173d067c 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -252,6 +252,13 @@ config SND_SOC_SOF_PROBE_WORK_QUEUE When selected, the probe is handled in two steps, for example to avoid lockdeps if request_module is used in the probe. +# Supported IPC versions +config SND_SOC_SOF_IPC3 + bool + +config SND_SOC_SOF_INTEL_IPC4 + bool + source "sound/soc/sof/amd/Kconfig" source "sound/soc/sof/imx/Kconfig" source "sound/soc/sof/intel/Kconfig" diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 2fa8088707a8..9a74ed116ed9 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -1,10 +1,18 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ - control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\ - ipc3-topology.o ipc3-control.o ipc3.o ipc3-pcm.o ipc3-loader.o\ - ipc3-dtrace.o\ - ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o + control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o + +# IPC implementations +ifneq ($(CONFIG_SND_SOC_SOF_IPC3),) +snd-sof-objs += ipc3.o ipc3-loader.o ipc3-topology.o ipc3-control.o ipc3-pcm.o\ + ipc3-dtrace.o +endif +ifneq ($(CONFIG_SND_SOC_SOF_INTEL_IPC4),) +snd-sof-objs += ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o +endif + +# SOF client support ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) snd-sof-objs += sof-client.o endif diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 085232e04582..190c85d57047 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -17,6 +17,7 @@ if SND_SOC_SOF_AMD_TOPLEVEL config SND_SOC_SOF_AMD_COMMON tristate select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_PCI_DEV select SND_AMD_ACP_CONFIG select SND_SOC_ACPI if ACPI diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 9b8d5bb1e449..cc6e695f913a 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_SOF_IMX_COMMON tristate select SND_SOC_SOF_OF_DEV select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_XTENSA select SND_SOC_SOF_COMPRESS help diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 0def2aa5581d..80cdc3788bbe 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -40,6 +40,7 @@ if SND_SOC_SOF_ACPI config SND_SOC_SOF_BAYTRAIL tristate "SOF support for Baytrail, Braswell and Cherrytrail" default SND_SOC_SOF_ACPI + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_ATOM_HIFI_EP select SND_SOC_SOF_ACPI_DEV @@ -60,6 +61,7 @@ config SND_SOC_SOF_BAYTRAIL config SND_SOC_SOF_BROADWELL tristate "SOF support for Broadwell" default SND_SOC_SOF_ACPI + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_HIFI_EP_IPC select SND_SOC_SOF_ACPI_DEV @@ -85,6 +87,7 @@ config SND_SOC_SOF_MERRIFIELD tristate "SOF support for Tangier/Merrifield" default SND_SOC_SOF_PCI select SND_SOC_SOF_PCI_DEV + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_ATOM_HIFI_EP help This adds support for Sound Open Firmware for Intel(R) platforms @@ -95,6 +98,8 @@ config SND_SOC_SOF_MERRIFIELD config SND_SOC_SOF_INTEL_APL tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_APOLLOLAKE tristate "SOF support for Apollolake" @@ -120,6 +125,8 @@ config SND_SOC_SOF_INTEL_CNL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_CANNONLAKE tristate "SOF support for Cannonlake" @@ -154,6 +161,8 @@ config SND_SOC_SOF_INTEL_ICL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_ICELAKE tristate "SOF support for Icelake" @@ -179,6 +188,8 @@ config SND_SOC_SOF_INTEL_TGL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_TIGERLAKE tristate "SOF support for Tigerlake" diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index c5aef5fc056b..6ed3f9b6a0c4 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -155,12 +155,22 @@ struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) init_waitqueue_head(&msg->waitq); - /* - * Use IPC3 ops as it is the only available version now. With the addition of new IPC - * versions, this will need to be modified to use the selected version at runtime. - */ - ipc->ops = &ipc3_ops; - ops = ipc->ops; + switch (sdev->pdata->ipc_type) { +#if defined(CONFIG_SND_SOC_SOF_IPC3) + case SOF_IPC: + ops = &ipc3_ops; + break; +#endif +#if defined(CONFIG_SND_SOC_SOF_INTEL_IPC4) + case SOF_INTEL_IPC4: + ops = &ipc4_ops; + break; +#endif + default: + dev_err(sdev->dev, "Not supported IPC version: %d\n", + sdev->pdata->ipc_type); + return NULL; + } /* check for mandatory ops */ if (!ops->tx_msg || !ops->rx_msg || !ops->set_get_data || !ops->get_reply) { @@ -190,6 +200,8 @@ struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) return NULL; } + ipc->ops = ops; + return ipc; } EXPORT_SYMBOL(snd_sof_ipc_init); diff --git a/sound/soc/sof/mediatek/Kconfig b/sound/soc/sof/mediatek/Kconfig index a149dd1b3f44..4a2eddf6009a 100644 --- a/sound/soc/sof/mediatek/Kconfig +++ b/sound/soc/sof/mediatek/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_SOF_MTK_COMMON tristate select SND_SOC_SOF_OF_DEV select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_XTENSA select SND_SOC_SOF_COMPRESS help -- 2.36.1

2 1

[PATCH] ASoC: SOF: topology: add code to parse config params for ACPDMIC dai
by Peter Ujfalusi 14 Jun '22

14 Jun '22

From: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Add sof_ipc_dai_acpdmic_params and tokens to parse dmic channels and rate params from topology file Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- include/sound/sof/dai-amd.h | 7 +++++++ include/sound/sof/dai.h | 2 +- include/uapi/sound/sof/tokens.h | 4 ++++ sound/soc/sof/ipc3-pcm.c | 8 ++++---- sound/soc/sof/ipc3-topology.c | 25 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 4 ++++ 7 files changed, 40 insertions(+), 11 deletions(-) diff --git a/include/sound/sof/dai-amd.h b/include/sound/sof/dai-amd.h index 90d09dbdd709..92f45c180b7c 100644 --- a/include/sound/sof/dai-amd.h +++ b/include/sound/sof/dai-amd.h @@ -18,4 +18,11 @@ struct sof_ipc_dai_acp_params { uint32_t fsync_rate; /* FSYNC frequency in Hz */ uint32_t tdm_slots; } __packed; + +/* ACPDMIC Configuration Request - SOF_IPC_DAI_AMD_CONFIG */ +struct sof_ipc_dai_acpdmic_params { + uint32_t pdm_rate; + uint32_t pdm_ch; +} __packed; + #endif diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index a818a0f0a226..21d98f31a9ca 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -111,7 +111,7 @@ struct sof_ipc_dai_config { struct sof_ipc_dai_sai_params sai; struct sof_ipc_dai_acp_params acpbt; struct sof_ipc_dai_acp_params acpsp; - struct sof_ipc_dai_acp_params acpdmic; + struct sof_ipc_dai_acpdmic_params acpdmic; struct sof_ipc_dai_mtk_afe_params afe; }; } __packed; diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index f7b2019065ad..5caf75cadaf8 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -157,6 +157,10 @@ /* MIXER */ #define SOF_TKN_MIXER_TYPE 1700 +/* ACPDMIC */ +#define SOF_TKN_AMD_ACPDMIC_RATE 1800 +#define SOF_TKN_AMD_ACPDMIC_CH 1801 + /* CAVS AUDIO FORMAT */ #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_RATE 1900 #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_BIT_DEPTH 1901 diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index c8774a891d6f..b97e63d3724a 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -344,10 +344,10 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, channels->min, channels->max); break; case SOF_DAI_AMD_DMIC: - rate->min = private->dai_config->acpdmic.fsync_rate; - rate->max = private->dai_config->acpdmic.fsync_rate; - channels->min = private->dai_config->acpdmic.tdm_slots; - channels->max = private->dai_config->acpdmic.tdm_slots; + rate->min = private->dai_config->acpdmic.pdm_rate; + rate->max = private->dai_config->acpdmic.pdm_rate; + channels->min = private->dai_config->acpdmic.pdm_ch; + channels->max = private->dai_config->acpdmic.pdm_ch; dev_dbg(component->dev, "AMD_DMIC rate_min: %d rate_max: %d\n", rate->min, rate->max); diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index d20e71be8b8e..28d3c1414572 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -266,6 +266,16 @@ static const struct sof_topology_token afe_tokens[] = { offsetof(struct sof_ipc_dai_mtk_afe_params, format)}, }; +/* ACPDMIC */ +static const struct sof_topology_token acpdmic_tokens[] = { + {SOF_TKN_AMD_ACPDMIC_RATE, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acpdmic_params, pdm_rate)}, + {SOF_TKN_AMD_ACPDMIC_CH, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acpdmic_params, pdm_ch)}, +}; + /* Core tokens */ static const struct sof_topology_token core_tokens[] = { {SOF_TKN_COMP_CORE_ID, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, @@ -300,6 +310,7 @@ static const struct sof_token_info ipc3_token_list[SOF_TOKEN_COUNT] = { [SOF_ESAI_TOKENS] = {"ESAI tokens", esai_tokens, ARRAY_SIZE(esai_tokens)}, [SOF_SAI_TOKENS] = {"SAI tokens", sai_tokens, ARRAY_SIZE(sai_tokens)}, [SOF_AFE_TOKENS] = {"AFE tokens", afe_tokens, ARRAY_SIZE(afe_tokens)}, + [SOF_ACPDMIC_TOKENS] = {"ACPDMIC tokens", acpdmic_tokens, ARRAY_SIZE(acpdmic_tokens)}, }; /** @@ -1120,20 +1131,22 @@ static int sof_link_acp_dmic_load(struct snd_soc_component *scomp, struct snd_so struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); + int ret; /* handle master/slave and inverted clocks */ sof_dai_set_format(hw_config, config); - /* init IPC */ - memset(&config->acpdmic, 0, sizeof(config->acpdmic)); config->hdr.size = size; - config->acpdmic.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->acpdmic.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + /* parse the required set of ACPDMIC tokens based on num_hw_cfgs */ + ret = sof_update_ipc_object(scomp, &config->acpdmic, SOF_ACPDMIC_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; dev_info(scomp->dev, "ACP_DMIC config ACP%d channel %d rate %d\n", - config->dai_index, config->acpdmic.tdm_slots, - config->acpdmic.fsync_rate); + config->dai_index, config->acpdmic.pdm_ch, + config->acpdmic.pdm_rate); dai->number_configs = 1; dai->current_config = 0; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 79486266081f..4284ea2f3a1f 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -236,6 +236,7 @@ enum sof_tokens { SOF_AUDIO_FMT_NUM_TOKENS, SOF_COPIER_FORMAT_TOKENS, SOF_GAIN_TOKENS, + SOF_ACPDMIC_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 1893c590f2f0..7e54eb1bf77b 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1739,6 +1739,10 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ token_id = SOF_AFE_TOKENS; num_tuples += token_list[SOF_AFE_TOKENS].count; break; + case SOF_DAI_AMD_DMIC: + token_id = SOF_ACPDMIC_TOKENS; + num_tuples += token_list[SOF_ACPDMIC_TOKENS].count; + break; default: break; } -- 2.36.1

2 1

[PATCH 1/2] ASoC: sun8i-codec: Partial revert to fix clock specifiers
by Charles Keepax 14 Jun '22

14 Jun '22

Recent updates accidentally updated the clock producer/consumer specifiers on this device as part of refactoring the CPU side of the DAI links. However, this device sits on the CODEC side and shouldn't have been updated. Partially revert the changes keeping the switch to the new clock terminology but going back to the CODEC defines. Fixes: 7cc3965fde74 ("ASoC: sunxi: Update to use set_fmt_new callback") Reported-by: Samuel Holland <samuel(a)sholland.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/sunxi/sun8i-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 90d74a2d53f38..f797c535f2983 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -287,10 +287,10 @@ static int sun8i_codec_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* clock masters */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: /* Codec slave, DAI master */ + case SND_SOC_DAIFMT_CBC_CFC: /* Codec slave, DAI master */ value = 0x1; break; - case SND_SOC_DAIFMT_BC_FC: /* Codec Master, DAI slave */ + case SND_SOC_DAIFMT_CBP_CFP: /* Codec Master, DAI slave */ value = 0x0; break; default: -- 2.30.2

3 4

[PATCH] ASoC: SOF: reduce default verbosity of IPC logs
by Pierre-Louis Bossart 14 Jun '22

14 Jun '22

We currently log the initiation of an IPC as well at its success. [ 3906.106987] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc tx: 0x80010000: GLB_DAI_MSG: CONFIG [ 3906.107189] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc tx succeeded: 0x80010000: GLB_DAI_MSG: CONFIG This is overkill in most cases, we already have a message thrown in case of errors and have tracepoints enabled to check for IPC duration. The only case where this might be useful is to check if there is an interleaved IPC RX. Add a flag and only print those logs if enabled. In addition, the DMA_POSITION_UPDATE for traces brings limited information in most cases and pollutes the logs for no good reason. [ 3906.322256] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.322308] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.822261] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.822319] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3907.822261] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3907.822319] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3908.822251] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3908.822309] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION This information is only helpful when debugging the trace support, not when using the trace. Add a flag to only print DMA position update logs if enabled. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc3.c | 5 ++++- sound/soc/sof/sof-priv.h | 6 ++++++ 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index dff5feaad3703..7608a3b0aeb21 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -147,6 +147,8 @@ static void ipc3_log_header(struct device *dev, u8 *text, u32 cmd) case SOF_IPC_TRACE_DMA_PARAMS: str2 = "DMA_PARAMS"; break; case SOF_IPC_TRACE_DMA_POSITION: + if (!sof_debug_check_flag(SOF_DBG_PRINT_DMA_POSITION_UPDATE_LOGS)) + return; str2 = "DMA_POSITION"; break; case SOF_IPC_TRACE_DMA_PARAMS_EXT: str2 = "DMA_PARAMS_EXT"; break; @@ -299,7 +301,8 @@ static int ipc3_wait_tx_done(struct snd_sof_ipc *ipc, void *reply_data) "ipc tx error for %#x (msg/reply size: %d/%zu): %d\n", hdr->cmd, hdr->size, msg->reply_size, ret); } else { - ipc3_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); + if (sof_debug_check_flag(SOF_DBG_PRINT_IPC_SUCCESS_LOGS)) + ipc3_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); if (msg->reply_size) /* copy the data returned from DSP */ memcpy(reply_data, msg->reply_data, diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70e..afbb37ec841ea 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -37,6 +37,12 @@ #define SOF_DBG_IGNORE_D3_PERSISTENT BIT(7) /* ignore the DSP D3 persistent capability * and always download firmware upon D3 exit */ +#define SOF_DBG_PRINT_DMA_POSITION_UPDATE_LOGS BIT(8) /* print DMA position updates + * in dmesg logs + */ +#define SOF_DBG_PRINT_IPC_SUCCESS_LOGS BIT(9) /* print IPC success + * in dmesg logs + */ /* Flag definitions used for controlling the DSP dump behavior */ #define SOF_DBG_DUMP_REGS BIT(0) -- 2.34.1

2 1

[PATCH] ASoC: SOF: Intel: hda-dai: enhance debug messages
by Pierre-Louis Bossart 14 Jun '22

14 Jun '22

The same message was added twice for dai and link_dma, remove the latter one and add dai name and direction to better understand problematic sequences. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 9823230d2ef4a..cb3d253558fe3 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -257,7 +257,6 @@ static int hda_link_dma_trigger(struct snd_pcm_substream *substream, int cmd) struct hdac_ext_stream *hext_stream = snd_soc_dai_get_dma_data(cpu_dai, substream); int ret; - dev_dbg(cpu_dai->dev, "%s: cmd=%d\n", __func__, cmd); if (!hext_stream) return 0; @@ -408,13 +407,15 @@ static int ipc3_hda_dai_trigger(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w; int ret; + dev_dbg(dai->dev, "%s: cmd=%d dai %s direction %d\n", __func__, cmd, + dai->name, substream->stream); + ret = hda_link_dma_trigger(substream, cmd); if (ret < 0) return ret; w = snd_soc_dai_get_widget(dai, substream->stream); - dev_dbg(dai->dev, "%s: cmd=%d\n", __func__, cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: -- 2.34.1

2 1

[PATCH] ASoC: intel: sof_sdw: add RT711 SDCA card for MTL platform
by Pierre-Louis Bossart 14 Jun '22

14 Jun '22

From: Yong Zhi <yong.zhi(a)intel.com> Enable on-board rt711 based sound card for MTL RVP. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Uday M Bhat <uday.m.bhat(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 8 +++++ .../intel/common/soc-acpi-intel-mtl-match.c | 31 +++++++++++++++++++ 2 files changed, 39 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 77af3a7655c44..d23846572543b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -315,6 +315,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + /* MeteorLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_mtlrvp"), + }, + .driver_data = (void *)(RT711_JD1 | SOF_SDW_TGL_HDMI), + }, {} }; diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index cc594b27e03be..74d3b82f8d35c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -15,6 +15,31 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { + { + .adr = 0x000030025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr mtl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { /* mockup tests need to be first */ @@ -36,6 +61,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt715-rt711-rt1308-mono.tplg", }, + { + .link_mask = BIT(0), + .links = mtl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_sdw_machines); -- 2.34.1

2 1

[PATCH 1/2] ASoC: dt-bindings: fsl, mqs: Add compatible string for i.MX93 platform
by Shengjiu Wang 14 Jun '22

14 Jun '22

Add compatible string "fsl,imx93-mqs" for i.MX93 platform Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,mqs.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt index 40353fc30255..d66284b8bef2 100644 --- a/Documentation/devicetree/bindings/sound/fsl,mqs.txt +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -2,7 +2,7 @@ fsl,mqs audio CODEC Required properties: - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" - "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs", "fsl,imx93-mqs". - clocks : A list of phandles + clock-specifiers, one for each entry in clock-names - clock-names : "mclk" - must required. -- 2.17.1

2 2

[PATCH 0/2] soundwire: Intel: add trigger callback
by Bard Liao 14 Jun '22

14 Jun '22

The .trigger callback will be used in ASoC. Hi Vinod, Could you provied a tag so that Mark can take the tag to his tree? Bard Liao (1): soundwire: Intel: add trigger callback Pierre-Louis Bossart (1): soundwire: intel: uniquify debug message drivers/soundwire/intel.c | 36 ++++++++++++++++++----------- include/linux/soundwire/sdw_intel.h | 1 + 2 files changed, 23 insertions(+), 14 deletions(-) -- 2.17.1

4 11

[PATCH] ALSA: hda/realtek: Apply fixup for Lenovo Yoga Duet 7 properly
by Takashi Iwai 14 Jun '22

14 Jun '22

It turned out that Lenovo shipped two completely different products with the very same PCI SSID, where both require different quirks; namely, Lenovo C940 has already the fixup for its speaker (ALC298_FIXUP_LENOVO_SPK_VOLUME) with the PCI SSID 17aa:3818, while Yoga Duet 7 has also the very same PCI SSID but requires a different quirk, ALC287_FIXUP_YOGA7_14TIL_SPEAKERS. Fortunately, both are with different codecs (C940 with ALC298 and Duet 7 with ALC287), hence we can apply different fixes by checking the codec ID. This patch implements that special fixup function. For easier handling, the internal function for applying a specific fixup entry is exported as __snd_hda_apply_fixup(), so that it can be called from the codec driver. The rest is simply calling it with a different fixup ID depending on the codec ID. Reported-by: Hans de Goede <hdegoede(a)redhat.com> Tested-by: nikitashvets(a)flyium.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/5ca147d1-3a2d-60c6-c491-8aa844183222@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 7 ++++--- sound/pci/hda/hda_local.h | 1 + sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++++- 3 files changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index cd1db943b7e0..7c6b1fe8dfcc 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -819,7 +819,7 @@ static void set_pin_targets(struct hda_codec *codec, snd_hda_set_pin_ctl_cache(codec, cfg->nid, cfg->val); } -static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) +void __snd_hda_apply_fixup(struct hda_codec *codec, int id, int action, int depth) { const char *modelname = codec->fixup_name; @@ -829,7 +829,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) if (++depth > 10) break; if (fix->chained_before) - apply_fixup(codec, fix->chain_id, action, depth + 1); + __snd_hda_apply_fixup(codec, fix->chain_id, action, depth + 1); switch (fix->type) { case HDA_FIXUP_PINS: @@ -870,6 +870,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) id = fix->chain_id; } } +EXPORT_SYMBOL_GPL(__snd_hda_apply_fixup); /** * snd_hda_apply_fixup - Apply the fixup chain with the given action @@ -879,7 +880,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) void snd_hda_apply_fixup(struct hda_codec *codec, int action) { if (codec->fixup_list) - apply_fixup(codec, codec->fixup_id, action, 0); + __snd_hda_apply_fixup(codec, codec->fixup_id, action, 0); } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aca592651870..682dca2057db 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -348,6 +348,7 @@ void snd_hda_apply_verbs(struct hda_codec *codec); void snd_hda_apply_pincfgs(struct hda_codec *codec, const struct hda_pintbl *cfg); void snd_hda_apply_fixup(struct hda_codec *codec, int action); +void __snd_hda_apply_fixup(struct hda_codec *codec, int id, int action, int depth); void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_model_fixup *models, const struct snd_pci_quirk *quirk, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b6ee775ee3f..b937f63d0d09 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7004,6 +7004,7 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC298_FIXUP_LENOVO_C940_DUET7, ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, @@ -7022,6 +7023,23 @@ enum { ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, }; +/* A special fixup for Lenovo C940 and Yoga Duet 7; + * both have the very same PCI SSID, and we need to apply different fixups + * depending on the codec ID + */ +static void alc298_fixup_lenovo_c940_duet7(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + int id; + + if (codec->core.vendor_id == 0x10ec0298) + id = ALC298_FIXUP_LENOVO_SPK_VOLUME; /* C940 */ + else + id = ALC287_FIXUP_YOGA7_14ITL_SPEAKERS; /* Duet 7 */ + __snd_hda_apply_fixup(codec, id, action, 0); +} + static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_GPIO2] = { .type = HDA_FIXUP_FUNC, @@ -8721,6 +8739,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, + [ALC298_FIXUP_LENOVO_C940_DUET7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_lenovo_c940_duet7, + }, [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9274,7 +9296,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), - SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), + SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), -- 2.35.3

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How to add 2 different SND_PCI_QUIRKs for 2 models with the same PCI subsys vend + prod ids?
by Hans de Goede 14 Jun '22

14 Jun '22

Hi All, There is a problem with the sound on the Lenovo Yoga Duet 7 13IML05, which requires a SND_PCI_QUIRK to fix. But there already is an existing quirk for another Lenovo laptop using the same PCI subsys vend + prod ids. For more details see: https://wiki.archlinux.org/title/Lenovo_Yoga_Duet_7_13IML05#Audio So I guess this means that we need to add a way to also include a DMI match for SND_PCI_QUIRKs ? Maybe add a: const struct dmi_systemid *dmi_ids; member to struct snd_pci_quirk and a new SND_PCI_QUIRK_DMI macro to set this? Regards, Hans

2 5

[PATCH AUTOSEL 4.9 01/12] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0d9c4a57301b..f733f6b42b53 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,7 +141,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 4

[PATCH AUTOSEL 4.14 01/14] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0d9c4a57301b..f733f6b42b53 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,7 +141,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 4

[PATCH AUTOSEL 4.19 09/18] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 02c557e1f779..c5b0b56d9c94 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -697,7 +697,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 08/18] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3afa163f7652..dcb01889e177 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -165,13 +165,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 07/18] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index dde015fd70a4..3f75cb3209ff 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3861,6 +3861,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 05/18] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 04f89b751304..deaad703a7db 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -403,9 +403,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 04/18] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 9a3180e71bd8..de311299432b 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,6 +141,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -355,7 +357,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 03/18] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 8995ea45b4ca..86e93904b001 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -351,22 +351,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 02/18] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 3d83c1be1292..9a3180e71bd8 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,7 +141,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 18/23] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b84e12f4f804..489f996d86bc 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -656,6 +656,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 78b5a0f22a41..57da27cd7498 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -382,6 +382,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -519,6 +520,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3184,6 +3186,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3212,6 +3215,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4674,6 +4678,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -4789,6 +4794,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -4939,6 +4945,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5038,6 +5045,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5152,6 +5160,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5248,6 +5257,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5542,6 +5552,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6145,6 +6156,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -9150,6 +9162,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -10592,6 +10605,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 12/23] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 13672928da99..aedfa6b2895b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -791,7 +791,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 11/23] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index fdf64c29f563..4117ab6e9b6f 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 10/23] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d9d59f45833f..ebaee468057b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3854,6 +3854,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 08/23] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 51d7a87ab4c3..8be7d83f0ce9 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 07/23] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0daa3b4c68ca..6c054b357205 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 06/23] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ed22361b35c1..a5a383b92305 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -347,22 +347,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 05/23] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index e9b5f76f27a8..aa32b8c26578 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 04/23] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 2ea4cba3be2a..0daa3b4c68ca 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 02/23] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 78db3bd0b3bc..cd163978792e 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 21/29] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 71a9462e8f6e..e067759561df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -439,6 +439,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -576,6 +577,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3252,6 +3254,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3280,6 +3283,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4849,6 +4853,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -4964,6 +4969,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5114,6 +5120,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5213,6 +5220,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5327,6 +5335,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5428,6 +5437,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5722,6 +5732,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6325,6 +6336,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -9812,6 +9824,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11254,6 +11267,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 14/29] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 51d95437e0fd..10189f44af28 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -800,7 +800,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 13/29] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 7e26231a596a..081b5f189632 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 12/29] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0bd3bbc2aacf..38651022e3d5 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3864,6 +3864,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 09/29] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..fc6a2bc311b4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 08/29] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 06dcfae9dfe7..d41e03193106 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 07/29] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 750e6c923512..38223641bdf6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 06/29] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ed22361b35c1..a5a383b92305 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -347,22 +347,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 05/29] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index e9b5f76f27a8..aa32b8c26578 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 04/29] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index f772628f233e..750e6c923512 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 02/29] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 609aeeb27818..d831959d8ff7 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 27/41] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2b5a610744f6..a2dd23b3a6a3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -438,6 +438,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -575,6 +576,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3242,6 +3244,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3270,6 +3273,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4905,6 +4909,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5020,6 +5025,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5170,6 +5176,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5269,6 +5276,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5383,6 +5391,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5484,6 +5493,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5778,6 +5788,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6380,6 +6391,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -9878,6 +9890,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11328,6 +11341,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 18/41] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f7c800927cb2..08fc1a025b1a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -794,7 +794,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 17/41] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 9632afc2d4d6..ca3b1c00fa78 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 16/41] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ba16bdf9e478..a5a4ae4440cc 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3867,6 +3867,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 12/41] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..fc6a2bc311b4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 11/41] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3cf8a0b4478c..b39c25409c23 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 10/41] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 785caba3f653..c19ad3c24702 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 09/41] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f2087bd38dbc..c2912ad3851b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 08/41] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index d83c1b318c1c..0accdb45ed72 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 07/41] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 80161151b3f2..785caba3f653 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 02/41] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..b436e532993d 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 27/43] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 112ecc256b14..7e7e517d32c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10092,6 +10104,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11542,6 +11555,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 18/43] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0582585236a2..93677b9f3a68 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -328,7 +328,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 17/43] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 9632afc2d4d6..ca3b1c00fa78 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 16/43] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a5584ba962dc..e97eaca7049f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3858,6 +3858,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 12/43] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..fc6a2bc311b4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 11/43] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3cf8a0b4478c..b39c25409c23 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

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[PATCH AUTOSEL 5.17 10/43] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 785caba3f653..c19ad3c24702 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.17 09/43] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f2087bd38dbc..c2912ad3851b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 08/43] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index d83c1b318c1c..0accdb45ed72 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 07/43] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 80161151b3f2..785caba3f653 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 02/43] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..b436e532993d 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

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[PATCH AUTOSEL 5.18 30/47] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 323c74a04268..87240f71892a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10147,6 +10159,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11597,6 +11610,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

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[PATCH AUTOSEL 5.18 21/47] ALSA: hda: MTL: add HD Audio PCI ID and HDMI codec vendor ID
by Sasha Levin 14 Jun '22

14 Jun '22

From: Yong Zhi <yong.zhi(a)intel.com> [ Upstream commit 2e45f2185283a2d927ef2cdbdc246cd65740c8df ] Add HD Audio PCI ID for Intel Meteorlake platform. [ corrected the hex number to lower letters by tiwai ] Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220606204232.144296-1-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1..a77165bd92a9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2525,6 +2525,9 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x51cf), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Meteorlake-P */ + { PCI_DEVICE(0x8086, 0x7e28), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 31fe41795571..6c209cd26c0c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4554,6 +4554,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281f, "Raptorlake-P HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281d, "Meteorlake HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 20/47] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e32c8ded181d..9cfd4f18493f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -333,7 +333,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

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[PATCH AUTOSEL 5.18 19/47] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3f00ead97006..dd53dfd87b04 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

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[PATCH AUTOSEL 5.18 18/47] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2c41d31956aa..f622a6bbd2fb 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3871,6 +3871,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

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[PATCH AUTOSEL 5.18 14/47] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e9c3cb4e2bfc..b9c262a15edf 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.18 13/47] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3cf8a0b4478c..b39c25409c23 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

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[PATCH AUTOSEL 5.18 12/47] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 785caba3f653..c19ad3c24702 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 11/47] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f2087bd38dbc..c2912ad3851b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 10/47] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index d83c1b318c1c..0accdb45ed72 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 09/47] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 80161151b3f2..785caba3f653 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 08/47] ASoC: Intel: cirrus-common: fix incorrect channel mapping
by Sasha Levin 14 Jun '22

14 Jun '22

From: xliu <xiang.liu(a)cirrus.com> [ Upstream commit d69a155555c9d57463b788c400f6b452d976bacd ] The default mapping of ASPRX1 (DAC source) is slot 0. Change the slot mapping of right amplifiers (WR and TR) to slot 1 to receive right channel data. Also update the ACPI instance ID mapping according to HW configuration. Signed-off-by: xliu <xiang.liu(a)cirrus.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220602051922.1232457-1-brent.lu@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_cirrus_common.c | 40 +++++++++++++++++++--- 1 file changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..f4192df962d6 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -54,22 +54,29 @@ static struct snd_soc_dai_link_component cs35l41_components[] = { }, }; +/* + * Mapping between ACPI instance id and speaker position. + * + * Four speakers: + * 0: Tweeter left, 1: Woofer left + * 2: Tweeter right, 3: Woofer right + */ static struct snd_soc_codec_conf cs35l41_codec_conf[] = { { .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "WL", + .name_prefix = "TL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WR", + .name_prefix = "WL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TL", + .name_prefix = "TR", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "TR", + .name_prefix = "WR", }, }; @@ -101,6 +108,21 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) return ret; } +/* + * Channel map: + * + * TL/WL: ASPRX1 on slot 0, ASPRX2 on slot 1 (default) + * TR/WR: ASPRX1 on slot 1, ASPRX2 on slot 0 + */ +static const struct { + unsigned int rx[2]; +} cs35l41_channel_map[] = { + {.rx = {0, 1}}, /* TL */ + {.rx = {0, 1}}, /* WL */ + {.rx = {1, 0}}, /* TR */ + {.rx = {1, 0}}, /* WR */ +}; + static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -134,6 +156,16 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* setup channel map */ + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(cs35l41_channel_map[i].rx), + (unsigned int *)cs35l41_channel_map[i].rx); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } } return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 03/47] ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback
by Sasha Levin 14 Jun '22

14 Jun '22

From: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> [ Upstream commit ef8d89b83bf453ea9cc3c4873a84b50ff334f797 ] Replace page protection permissions from noncashed to writecombine, in lpass codec DMA path mmp callabck, to support 64 bit chromeOS. Avoid SIGBUS error in userspace caused by noncached permissions in 64 bit chromeOS. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Link: https://lore.kernel.org/r/1653660608-27245-1-git-send-email-quic_srivasam@q… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 74d62f377dfd..ae2a7837e5cc 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -898,7 +898,7 @@ static int lpass_platform_cdc_dma_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; unsigned long size, offset; - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot); size = vma->vm_end - vma->vm_start; offset = vma->vm_pgoff << PAGE_SHIFT; return io_remap_pfn_range(vma, vma->vm_start, -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 02/47] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..b436e532993d 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

1 0

[PATCH] MAINTAINERS: update ASoC/Intel/SOF maintainers
by Pierre-Louis Bossart 13 Jun '22

13 Jun '22

Keyon Jie was a key contributor to the Intel ASoC and SOF Intel drivers, but he's moved on to a different role within Intel. We wish him all the best in his new endeavors. Bard Liao, Kai Vehmanen, Ranjani Sridharan and Peter Ujfalusi have been involved in the Intel multi-maintainer team, it's time to update the MAINTAINERS entry to reflect their contributions and clarify their role. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- MAINTAINERS | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a6d3bd9d2a8d0..440f3d7c93b9e 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -9803,7 +9803,10 @@ INTEL ASoC DRIVERS M: Cezary Rojewski <cezary.rojewski(a)intel.com> M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> M: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> -M: Jie Yang <yang.jie(a)linux.intel.com> +M: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> +M: Bard Liao <yung-chuan.liao(a)linux.intel.com> +M: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> +M: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/intel/ @@ -18670,8 +18673,10 @@ F: sound/soc/ SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> M: Liam Girdwood <lgirdwood(a)gmail.com> +M: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> +M: Bard Liao <yung-chuan.liao(a)linux.intel.com> M: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> -M: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> +R: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> M: Daniel Baluta <daniel.baluta(a)nxp.com> L: sound-open-firmware(a)alsa-project.org (moderated for non-subscribers) S: Supported -- 2.34.1

2 1

[PATCH] ASoC: Intel: bytcr_wm5102: Fix GPIO related probe-ordering problem
by Hans de Goede 13 Jun '22

13 Jun '22

The "wlf,spkvdd-ena" GPIO needed by the bytcr_wm5102 driver is made available through a gpio-lookup table. This gpio-lookup table is registered by drivers/mfd/arizona-spi.c, which may get probed after the bytcr_wm5102 driver. If the gpio-lookup table has not registered yet then the gpiod_get() will return -ENOENT. Treat -ENOENT as -EPROBE_DEFER to still keep things working in this case. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_wm5102.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 00384c6fbcaa..330c0ace1638 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -421,8 +421,17 @@ static int snd_byt_wm5102_mc_probe(struct platform_device *pdev) priv->spkvdd_en_gpio = gpiod_get(codec_dev, "wlf,spkvdd-ena", GPIOD_OUT_LOW); put_device(codec_dev); - if (IS_ERR(priv->spkvdd_en_gpio)) - return dev_err_probe(dev, PTR_ERR(priv->spkvdd_en_gpio), "getting spkvdd-GPIO\n"); + if (IS_ERR(priv->spkvdd_en_gpio)) { + ret = PTR_ERR(priv->spkvdd_en_gpio); + /* + * The spkvdd gpio-lookup is registered by: drivers/mfd/arizona-spi.c, + * so -ENOENT means that arizona-spi hasn't probed yet. + */ + if (ret == -ENOENT) + ret = -EPROBE_DEFER; + + return dev_err_probe(dev, ret, "getting spkvdd-GPIO\n"); + } /* override platform name, if required */ byt_wm5102_card.dev = dev; -- 2.36.0

3 2

[PATCH 00/23] ASoC: SOF: IPC4: Add topology, control and PCM ops
by Ranjani Sridharan 13 Jun '22

13 Jun '22

This set of patches includes changes to add the topology, control and PCM ops for IPC4. It also includes a couple of patches to set the IPC4 BE DAI trigger ops for SSP/DMIC/HDA type DAI's. Bard Liao (1): ASoC: SOF: IPC4: add sdw blob Ranjani Sridharan (22): ASoC: SOF: Add topology tokens for IPC4 ASoC: SOF: IPC4: Introduce topology ops ASoC: SOF: ipc4-topology: Add support for parsing AIF_IN/AIF_OUT widgets ASoC: SOF: ipc4-topology: Add support for parsing DAI_IN/DAI_OUT widgets ASoC: SOF: ipc4-topology: Add prepare op for AIF type widgets ASoC: SOF: ipc4-topology: Add prepare op for DAI type widgets ASoC: SOF: ipc4-topology: Add support for parsing and preparing pga widgets ASoC: SOF: ipc4-topology: Add support for parsing mixer widgets ASoC: SOF: ipc4-topology: Add control_setup op ASoC: SOF: ipc4-topology: Add control IO ops ASoC: SOF: IPC4: Add pcm ops ASoC: SOF: ipc4-topology: Add widget_setup/widget_free ops ASoC: SOF: ipc4-topology: Add route_setup/route_free ops ASoC: SOF: ipc4-topology: Add the dai_config op ASoC: SOF: ipc4-pcm: Expose sof_ipc4_set_pipeline_state() ASoC: SOF: IPC4: set the BE DAI ops ASoC: SOF: Add ops_free ASoC: SOF: Intel: hda: init NHLT for IPC4 ASoC: SOF: Add two new structures for topology manifest data ASoC: SOF: Add a new IPC op for parsing topology manifest ASoC: SOF: ipc4-topology: Add support for SSP/DMIC DAI's AsoC: SOF: ipc4-topology: Add dai_get_clk op include/sound/sof.h | 1 + include/uapi/sound/sof/abi.h | 2 + include/uapi/sound/sof/header.h | 30 + include/uapi/sound/sof/tokens.h | 40 + sound/soc/sof/Makefile | 2 +- sound/soc/sof/core.c | 7 +- sound/soc/sof/intel/hda-dai.c | 201 +++- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/ipc3-topology.c | 48 + sound/soc/sof/ipc4-control.c | 216 ++++ sound/soc/sof/ipc4-pcm.c | 230 +++++ sound/soc/sof/ipc4-priv.h | 7 + sound/soc/sof/ipc4-topology.c | 1702 +++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.h | 234 +++++ sound/soc/sof/ipc4.c | 2 + sound/soc/sof/ops.h | 6 + sound/soc/sof/sof-audio.h | 11 + sound/soc/sof/topology.c | 114 +-- 22 files changed, 2794 insertions(+), 64 deletions(-) create mode 100644 sound/soc/sof/ipc4-control.c create mode 100644 sound/soc/sof/ipc4-pcm.c create mode 100644 sound/soc/sof/ipc4-topology.c create mode 100644 sound/soc/sof/ipc4-topology.h -- 2.25.1

2 24

[PATCH] ASoC: hdmi-codec: Update to modern DAI terminology
by Mark Brown 13 Jun '22

13 Jun '22

As part of retiring the old defines used to specify DAI formats update the hdmi_codec driver to use the modern names, including the variables in the struct hdmi_codec_daifmt exported to the DRM drivers. In updating this I did note that the only use of this information in DRM drivers is to reject clock provider settings, thinking about what this hardware is doing I rather suspect that there might not be any hardware out there which needs the configuration so it may be worth considering just having hdmi-codec support only clock consumer. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/gpu/drm/bridge/sii902x.c | 5 +++-- .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 2 +- drivers/gpu/drm/exynos/exynos_hdmi.c | 8 ++++---- drivers/gpu/drm/i2c/tda998x_drv.c | 6 +++--- drivers/gpu/drm/sti/sti_hdmi.c | 8 ++++---- include/sound/hdmi-codec.h | 4 ++-- sound/soc/codecs/hdmi-codec.c | 18 +++++++++--------- 7 files changed, 26 insertions(+), 25 deletions(-) diff --git a/drivers/gpu/drm/bridge/sii902x.c b/drivers/gpu/drm/bridge/sii902x.c index 65549fbfdc87..be9736f67542 100644 --- a/drivers/gpu/drm/bridge/sii902x.c +++ b/drivers/gpu/drm/bridge/sii902x.c @@ -549,8 +549,9 @@ static int sii902x_audio_hw_params(struct device *dev, void *data, unsigned long mclk_rate; int i, ret; - if (daifmt->bit_clk_master || daifmt->frame_clk_master) { - dev_dbg(dev, "%s: I2S master mode not supported\n", __func__); + if (daifmt->bit_clk_provider || daifmt->frame_clk_provider) { + dev_dbg(dev, "%s: I2S clock provider mode not supported\n", + __func__); return -EINVAL; } diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index f50b47ac11a8..a2f0860b20bb 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -45,7 +45,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, u8 inputclkfs = 0; /* it cares I2S only */ - if (fmt->bit_clk_master | fmt->frame_clk_master) { + if (fmt->bit_clk_provider | fmt->frame_clk_provider) { dev_err(dev, "unsupported clock settings\n"); return -EINVAL; } diff --git a/drivers/gpu/drm/exynos/exynos_hdmi.c b/drivers/gpu/drm/exynos/exynos_hdmi.c index 7655142a4651..10b0036f8a2e 100644 --- a/drivers/gpu/drm/exynos/exynos_hdmi.c +++ b/drivers/gpu/drm/exynos/exynos_hdmi.c @@ -1594,12 +1594,12 @@ static int hdmi_audio_hw_params(struct device *dev, void *data, struct hdmi_context *hdata = dev_get_drvdata(dev); if (daifmt->fmt != HDMI_I2S || daifmt->bit_clk_inv || - daifmt->frame_clk_inv || daifmt->bit_clk_master || - daifmt->frame_clk_master) { + daifmt->frame_clk_inv || daifmt->bit_clk_provider || + daifmt->frame_clk_provider) { dev_err(dev, "%s: Bad flags %d %d %d %d\n", __func__, daifmt->bit_clk_inv, daifmt->frame_clk_inv, - daifmt->bit_clk_master, - daifmt->frame_clk_master); + daifmt->bit_clk_provider, + daifmt->frame_clk_provider); return -EINVAL; } diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c index b7ec6c374fbd..c4fadaecbb2d 100644 --- a/drivers/gpu/drm/i2c/tda998x_drv.c +++ b/drivers/gpu/drm/i2c/tda998x_drv.c @@ -1095,11 +1095,11 @@ static int tda998x_audio_hw_params(struct device *dev, void *data, if (!spdif && (daifmt->bit_clk_inv || daifmt->frame_clk_inv || - daifmt->bit_clk_master || daifmt->frame_clk_master)) { + daifmt->bit_clk_provider || daifmt->frame_clk_provider)) { dev_err(dev, "%s: Bad flags %d %d %d %d\n", __func__, daifmt->bit_clk_inv, daifmt->frame_clk_inv, - daifmt->bit_clk_master, - daifmt->frame_clk_master); + daifmt->bit_clk_provider, + daifmt->frame_clk_provider); return -EINVAL; } diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c index b3fbee7eac11..622d0dfe7bb9 100644 --- a/drivers/gpu/drm/sti/sti_hdmi.c +++ b/drivers/gpu/drm/sti/sti_hdmi.c @@ -1175,12 +1175,12 @@ static int hdmi_audio_hw_params(struct device *dev, DRM_DEBUG_DRIVER("\n"); if ((daifmt->fmt != HDMI_I2S) || daifmt->bit_clk_inv || - daifmt->frame_clk_inv || daifmt->bit_clk_master || - daifmt->frame_clk_master) { + daifmt->frame_clk_inv || daifmt->bit_clk_providre || + daifmt->frame_clk_provider) { dev_err(dev, "%s: Bad flags %d %d %d %d\n", __func__, daifmt->bit_clk_inv, daifmt->frame_clk_inv, - daifmt->bit_clk_master, - daifmt->frame_clk_master); + daifmt->bit_clk_provider, + daifmt->frame_clk_provider); return -EINVAL; } diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 4fc733c8c570..48ad33aba393 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -32,8 +32,8 @@ struct hdmi_codec_daifmt { } fmt; unsigned int bit_clk_inv:1; unsigned int frame_clk_inv:1; - unsigned int bit_clk_master:1; - unsigned int frame_clk_master:1; + unsigned int bit_clk_provider:1; + unsigned int frame_clk_provider:1; /* bit_fmt could be standard PCM format or * IEC958 encoded format. ALSA IEC958 plugin will pass * IEC958_SUBFRAME format to the underneath driver. diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b773466619b2..7d1e351f863a 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -606,18 +606,18 @@ static int hdmi_codec_i2s_set_fmt(struct snd_soc_dai *dai, /* Reset daifmt */ memset(cf, 0, sizeof(*cf)); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - cf->bit_clk_master = 1; - cf->frame_clk_master = 1; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + cf->bit_clk_provider = 1; + cf->frame_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBS_CFM: - cf->frame_clk_master = 1; + case SND_SOC_DAIFMT_CBC_CFP: + cf->frame_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBM_CFS: - cf->bit_clk_master = 1; + case SND_SOC_DAIFMT_CBP_CFC: + cf->bit_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.30.2

1 1

[PATCH] ASoC: wcd9335: Remove RX channel from old list before adding it to a new one
by Yassine Oudjana 13 Jun '22

13 Jun '22

From: Yassine Oudjana <y.oudjana(a)protonmail.com> Currently in slim_rx_mux_put, an RX channel gets added to a new list even if it is already in one. This can mess up links and make either it, the new list head, or both, get linked to the wrong entries. This can cause an entry to link to itself which in turn ends up making list_for_each_entry in other functions loop infinitely. To avoid issues, always remove the RX channel from any list it's in before adding it to a new list. Signed-off-by: Yassine Oudjana <y.oudjana(a)protonmail.com> --- sound/soc/codecs/wcd9335.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 617a36a89dfe..597420679505 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1289,9 +1289,12 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; + /* Remove channel from any list it's in before adding it to a new one */ + list_del_init(&wcd->rx_chs[port_id].list); + switch (wcd->rx_port_value[port_id]) { case 0: - list_del_init(&wcd->rx_chs[port_id].list); + /* Channel already removed from lists. Nothing to do here */ break; case 1: list_add_tail(&wcd->rx_chs[port_id].list, -- 2.36.1

2 1

[PATCH] ASoC: wcd9335: Fix spurious event generation
by Mark Brown 13 Jun '22

13 Jun '22

The slimbus mux put operation unconditionally reports a change in value which means that spurious events are generated. Fix this by exiting early in that case. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd9335.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 617a36a89dfe..d25594a708e0 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1287,6 +1287,9 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, struct snd_soc_dapm_update *update = NULL; u32 port_id = w->shift; + if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0]) + return 0; + wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; switch (wcd->rx_port_value[port_id]) { -- 2.30.2

1 1

[PATCH] ASoC: wcd938x: Fix event generation for some controls
by Mark Brown 13 Jun '22

13 Jun '22

Currently wcd938x_*_put() unconditionally report that the value of the control changed, resulting in spurious events being generated. Return 0 in that case instead as we should. There is still an issue in the compander control which is a bit more complex. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd938x.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index c1b61b997f69..d956c58298c3 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -2519,6 +2519,9 @@ static int wcd938x_tx_mode_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int path = e->shift_l; + if (wcd938x->tx_mode[path] == ucontrol->value.enumerated.item[0]) + return 0; + wcd938x->tx_mode[path] = ucontrol->value.enumerated.item[0]; return 1; @@ -2540,6 +2543,10 @@ static int wcd938x_rx_hph_mode_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + int change; + + if (wcd938x->hph_mode == ucontrol->value.enumerated.item[0]) + return 0; wcd938x->hph_mode = ucontrol->value.enumerated.item[0]; @@ -2632,6 +2639,9 @@ static int wcd938x_ldoh_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->ldoh == ucontrol->value.integer.value[0]) + return 0; + wcd938x->ldoh = ucontrol->value.integer.value[0]; return 1; @@ -2654,6 +2664,9 @@ static int wcd938x_bcs_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->bcs_dis == ucontrol->value.integer.value[0]) + return 0; + wcd938x->bcs_dis = ucontrol->value.integer.value[0]; return 1; -- 2.30.2

2 3

There is headset Mic issue on HP ALC897 Platform
by Kailang 13 Jun '22

13 Jun '22

Hi Takashi, HP ALC897 platform. It need to fill 0x19 verb table to enable Headset MIC. Attach was the supporting patch. BR, Kailang

2 1

[PATCH 0/2] Update external mclk name
by Srinivasa Rao Mandadapu 13 Jun '22

13 Jun '22

This patch set is to update external mclk name. Srinivasa Rao Mandadapu (2): ASoC: dt-bindings: lpass-cpu: Update external mclck0 name ASoC: qcom: lpass-sc7280: Update external mclk0 name Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.7.4

3 5

[PATCH V0 0/1] asoc: msm: use hashtable to check kcontrol
by Raghu Bankapur 13 Jun '22

13 Jun '22

use hashtabe instead of linear list to check kcontrol before adding them for improving early audio KPI. With this changes we see 600ms improvement in start of audio Raghu Bankapur (1): asoc: msm: use hashtable to check kcontrol include/sound/control.h | 4 ++ include/sound/core.h | 12 +++++- sound/core/control.c | 92 +++++++++++++++++++++++++++++++++-------- sound/core/init.c | 3 ++ sound/soc/Kconfig | 9 ++++ 5 files changed, 101 insertions(+), 19 deletions(-) -- 2.17.1

5 10

[PATCH 00/56] Specify clock provider directly to CPU DAIs
by Charles Keepax 13 Jun '22

13 Jun '22

Currently the set_fmt callback always passes clock provider/consumer with respect to the CODEC. This made sense when the framework was directly broken down into platforms and CODECs. However, as things are now broken down into components which can be connected as either the CPU or CODEC side of a DAI link it simplifies things if each side of the link is just told if it is provider or consumer of the clocks. Making this change allows us to remove one of the last parts of the ASoC core that needs to know if a driver is a CODEC driver, where it flips the clock format specifier if a CODEC driver is used on the CPU side of a DAI link, as well as just being conceptually more consistent with componentisation. The basic idea of this patch chain is to change the set_fmt callback from specifying if the CODEC is provider/consumer into directly specifying if the component is provider/consumer. To do this we add some new defines, and then to preserve bisectability, the migration is done by adding a new callback, converting over all existing CPU side drivers, converting the core, and then finally reverting back to the old callback. Converting the platform drivers makes sense as the existing defines are from the perspective of the CODEC and there are more CODEC drivers than platform drivers. Obviously a fair amount of this patch chain I was only able to build test, so any testing that can be done would be greatly appreciated. Thanks, Charles Charles Keepax (56): ASoC: core: Add set_fmt_new callback that directly specifies provider ASoC: amd: vangogh: Update to use set_fmt_new callback ASoC: atmel: Update to use set_fmt_new callback ASoC: au1x: Update to use set_fmt_new callback ASoC: bcm: Update to use set_fmt_new callback ASoC: ep93xx: Update to use set_fmt_new callback ASoC: dwc: Update to use set_fmt_new callback ASoC: fsl: Update to use set_fmt_new callback ASoC: hisilicon: Update to use set_fmt_new callback ASoC: img: Update to use set_fmt_new callback ASoC: Intel: Update to use set_fmt_new callback ASoC: js4740-i2s: Update to use set_fmt_new callback ASoC: mediatek: Update to use set_fmt_new callback ASoC: meson: Update to use set_fmt_new callback ASoC: mxs-saif: Update to use set_fmt_new callback ASoC: pxa: Update to use set_fmt_new callback ASoC: qcom: Update to use set_fmt_new callback ASoC: rockchip: Update to use set_fmt_new callback ASoC: samsung: Update to use set_fmt_new callback ASoC: sh: Update to use set_fmt_new callback ASoC: stm: Update to use set_fmt_new callback ASoC: sunxi: Update to use set_fmt_new callback ASoC: tegra: Update to use set_fmt_new callback ASoC: test-component: Update to use set_fmt_new callback ASoC: ti: Update to use set_fmt_new callback ASoC: ux500: Update to use set_fmt_new callback ASoC: xtensa: Update to use set_fmt_new callback ASoC: core: Always send the CPU DAI a direct clock specifier ASoC: amd: vangogh: Rename set_fmt_new back to set_fmt ASoC: atmel: Rename set_fmt_new back to set_fmt ASoC: au1x: Rename set_fmt_new back to set_fmt ASoC: bcm: Rename set_fmt_new back to set_fmt ASoC: ep93xx: Rename set_fmt_new back to set_fmt ASoC: dwc: Rename set_fmt_new back to set_fmt ASoC: fsl: Rename set_fmt_new back to set_fmt ASoC: hisilicon: Rename set_fmt_new back to set_fmt ASoC: img: Rename set_fmt_new back to set_fmt ASoC: Intel: Rename set_fmt_new back to set_fmt ASoC: jz4740-i2s: Rename set_fmt_new back to set_fmt ASoC: mediatek: Rename set_fmt_new back to set_fmt ASoC: meson: Rename set_fmt_new back to set_fmt ASoC: mxs-saif: Rename set_fmt_new back to set_fmt ASoC: pxa: Rename set_fmt_new back to set_fmt ASoC: qcom: Rename set_fmt_new back to set_fmt ASoC: rockchip: Rename set_fmt_new back to set_fmt ASoC: samsung: Rename set_fmt_new back to set_fmt ASoC: sh: Rename set_fmt_new back to set_fmt ASoC: stm: Rename set_fmt_new back to set_fmt ASoC: sunxi: Rename set_fmt_new back to set_fmt ASoC: tegra: Rename set_fmt_new back to set_fmt ASoC: test-component: Rename set_fmt_new back to set_fmt ASoC: ti: Rename set_fmt_new back to set_fmt ASoC: ux500: Rename set_fmt_new back to set_fmt ASoC: xtensa: Rename set_fmt_new back to set_fmt ASoC: soc-dai: Remove set_fmt_new callback ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local include/sound/soc-component.h | 5 --- include/sound/soc-dai.h | 6 ++++ sound/soc/amd/vangogh/acp5x-i2s.c | 4 +-- sound/soc/atmel/atmel-i2s.c | 4 +-- sound/soc/atmel/atmel_ssc_dai.c | 18 +++++----- sound/soc/atmel/mchp-i2s-mcc.c | 8 ++--- sound/soc/atmel/mchp-pdmc.c | 4 +-- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-i2s.c | 4 +-- sound/soc/bcm/bcm2835-i2s.c | 20 +++++------ sound/soc/bcm/cygnus-ssp.c | 4 +-- sound/soc/cirrus/ep93xx-i2s.c | 4 +-- sound/soc/dwc/dwc-i2s.c | 8 ++--- sound/soc/fsl/fsl_audmix.c | 6 ++-- sound/soc/fsl/fsl_esai.c | 8 ++--- sound/soc/fsl/fsl_mqs.c | 2 +- sound/soc/fsl/fsl_sai.c | 8 ++--- sound/soc/fsl/fsl_ssi.c | 22 ++++++------ sound/soc/fsl/imx-audmix.c | 4 +-- sound/soc/fsl/imx-card.c | 2 +- sound/soc/generic/simple-card-utils.c | 7 +++- sound/soc/generic/test-component.c | 18 +++++----- sound/soc/hisilicon/hi6210-i2s.c | 18 +++++----- sound/soc/img/img-i2s-in.c | 4 +-- sound/soc/img/img-i2s-out.c | 6 ++-- sound/soc/intel/atom/sst-atom-controls.c | 4 +-- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcht_nocodec.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/keembay/kmb_platform.c | 4 +-- sound/soc/jz4740/jz4740-i2s.c | 10 +++--- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 6 ++-- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 6 ++-- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/axg-tdm-interface.c | 14 ++++---- sound/soc/mxs/mxs-saif.c | 4 +-- sound/soc/pxa/magician.c | 8 ++--- sound/soc/pxa/mmp-sspa.c | 6 ++-- sound/soc/pxa/pxa-ssp.c | 22 ++++++------ sound/soc/pxa/pxa2xx-i2s.c | 6 ++-- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/qdsp6/audioreach.c | 4 +-- sound/soc/qcom/qdsp6/q6afe.c | 6 ++-- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sdm845.c | 6 ++-- sound/soc/qcom/sm8250.c | 4 +-- sound/soc/rockchip/rockchip_i2s.c | 6 ++-- sound/soc/rockchip/rockchip_i2s_tdm.c | 6 ++-- sound/soc/samsung/i2s.c | 6 ++-- sound/soc/samsung/pcm.c | 4 +-- sound/soc/samsung/s3c-i2s-v2.c | 6 ++-- sound/soc/samsung/s3c24xx-i2s.c | 6 ++-- sound/soc/sh/fsi.c | 6 ++-- sound/soc/sh/rcar/core.c | 4 +-- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/sh/ssi.c | 10 +++--- sound/soc/soc-core.c | 14 ++------ sound/soc/soc-dai.c | 3 +- sound/soc/stm/stm32_i2s.c | 8 ++--- sound/soc/stm/stm32_sai_sub.c | 8 ++--- sound/soc/sunxi/sun4i-i2s.c | 18 +++++----- sound/soc/sunxi/sun8i-codec.c | 6 ++-- sound/soc/tegra/tegra20_i2s.c | 6 ++-- sound/soc/tegra/tegra210_i2s.c | 6 ++-- sound/soc/tegra/tegra30_i2s.c | 6 ++-- sound/soc/ti/davinci-i2s.c | 32 ++++++++--------- sound/soc/ti/davinci-mcasp.c | 10 +++--- sound/soc/ti/omap-mcbsp.c | 12 +++---- sound/soc/ux500/ux500_msp_dai.c | 36 ++++++++++---------- sound/soc/xtensa/xtfpga-i2s.c | 4 +-- 77 files changed, 279 insertions(+), 283 deletions(-) -- 2.30.2

8 72

[PATCH 00/17] ASoC: Intel: haswell and broadwell boards update
by Cezary Rojewski 13 Jun '22

13 Jun '22

A number of patches improving overall quality and readability of haswell.c and broadwell.c source files found in sound/soc/intel/boards. Both files are first renamed and only then actual changes are being incrementally added. The respective names are: hsw_rt5640 and bdw_rt286 to match the pattern found in more recent boards. Most patches bring no functional change - the more impactful patches at are placed the end: Refactor of suspend/resume flow for the bdw_rt286 board by dropping dev->remove() in favour of card->remove() and adjust jack handling to reduce code size slightly by implementing card_set_jack(). The last patch is removing of FE DAI ops. Given the existence of platform FE DAI capabilities (either static declaration or through topology file), this code is redundant. Cezary Rojewski (17): ASoC: Intel: Rename haswell source file to hsw_rt5640 ASoC: Intel: hsw_rt5640: Reword prefixes of all driver members ASoC: Intel: hsw_rt5640: Reword driver name ASoC: Intel: hsw_rt5640: Update code indentation ASoC: Intel: hsw_rt5640: Update file comments ASoC: Intel: hsw_rt5640: Improve probe() function quality ASoC: Intel: hsw_rt5640: Improve hw_params() debug-ability ASoC: Intel: Rename broadwell source file to bdw_rt286 ASoC: Intel: bdw_rt286: Reword prefixes of all driver members ASoC: Intel: bdw_rt286: Reword driver name ASoC: Intel: bdw_rt286: Update code indentation ASoC: Intel: bdw_rt286: Update file comments ASoC: Intel: bdw_rt286: Improve probe() function quality ASoC: Intel: bdw_rt286: Improve hw_params() debug-ability ASoC: Intel: bdw_rt286: Improve codec_init() quality ASoC: Intel: bdw_rt286: Refactor suspend/resume ASoC: Intel: bdw_rt286: Remove FE DAI ops sound/soc/intel/boards/Kconfig | 4 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bdw_rt286.c | 257 +++++++++++++ sound/soc/intel/boards/broadwell.c | 338 ------------------ sound/soc/intel/boards/haswell.c | 202 ----------- sound/soc/intel/boards/hsw_rt5640.c | 176 +++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 6 +- 7 files changed, 440 insertions(+), 547 deletions(-) create mode 100644 sound/soc/intel/boards/bdw_rt286.c delete mode 100644 sound/soc/intel/boards/broadwell.c delete mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/hsw_rt5640.c -- 2.25.1

2 23

[PATCH 00/23] Update Documentation/ cross-references
by Mauro Carvalho Chehab 13 Jun '22

13 Jun '22

Hi John, There were a number of DT binding conversions and other docs change that were not updated. Address them, in order to keep the cross-references on a sane state. Patch series is against v5.19-rc1 (and applies cleanly on the top of today's -next). Mauro Carvalho Chehab (23): dt-bindings: mfd: bd9571mwv: update rohm,bd9571mwv.yaml reference dt-bindings: interrupt-controller: update brcm,l2-intc.yaml reference dt-bindings: arm: update vexpress-config.yaml references dt-bindings: reset: update st,stih407-powerdown.yaml references dt-bindings: mfd: rk808: update rockchip,rk808.yaml reference dt-bindings: mmc: exynos-dw-mshc: update samsung,pinctrl.yaml reference docs: netdev: update maintainer-netdev.rst reference docs: filesystems: update netfs-api.rst reference Documentation: update watch_queue.rst references Documentation: KVM: update s390-pv.rst reference Documentation: KVM: update amd-memory-encryption.rst references Documentation: KVM: update msr.rst reference Documentation: KVM: update s390-diag.rst reference MAINTAINERS: update arm,hdlcd.yaml reference MAINTAINERS: update arm,komeda.yaml reference MAINTAINERS: update arm,malidp.yaml reference MAINTAINERS: update cortina,gemini-ethernet.yaml reference MAINTAINERS: update dongwoon,dw9807-vcm.yaml reference MAINTAINERS: update maxim,max77693.yaml reference MAINTAINERS: update snps,axs10x-reset.yaml reference objtool: update objtool.txt references ASoC: wm8731: update wlf,wm8731.yaml reference arch: m68k: q40: README: drop references to IDE driver .../ABI/testing/sysfs-driver-bd9571mwv-regulator | 2 +- Documentation/admin-guide/kernel-parameters.txt | 2 +- .../bindings/cpufreq/brcm,stb-avs-cpu-freq.txt | 2 +- .../devicetree/bindings/hwmon/vexpress.txt | 2 +- .../devicetree/bindings/mmc/exynos-dw-mshc.txt | 2 +- .../devicetree/bindings/phy/phy-stih407-usb.txt | 2 +- .../devicetree/bindings/pinctrl/pinctrl-rk805.txt | 2 +- .../devicetree/bindings/regulator/vexpress.txt | 2 +- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 2 +- Documentation/devicetree/bindings/usb/dwc3-st.txt | 2 +- Documentation/devicetree/bindings/usb/ehci-st.txt | 2 +- Documentation/devicetree/bindings/usb/ohci-st.txt | 2 +- Documentation/security/keys/core.rst | 2 +- Documentation/security/secrets/coco.rst | 2 +- .../translations/it_IT/networking/netdev-FAQ.rst | 2 +- Documentation/virt/kvm/api.rst | 4 ++-- Documentation/virt/kvm/s390/s390-pv-boot.rst | 2 +- Documentation/virt/kvm/x86/hypercalls.rst | 2 +- Documentation/x86/orc-unwinder.rst | 2 +- MAINTAINERS | 14 +++++++------- arch/m68k/q40/README | 4 +--- include/linux/fscache.h | 2 +- include/linux/objtool.h | 2 +- include/linux/watch_queue.h | 2 +- init/Kconfig | 2 +- kernel/watch_queue.c | 2 +- lib/Kconfig.debug | 2 +- tools/include/linux/objtool.h | 2 +- tools/objtool/check.c | 2 +- 29 files changed, 36 insertions(+), 38 deletions(-) -- 2.36.1

4 5

ucm2: add ALC4080 ID for Asus Z690-I Gaming Wifi
by GitHub pull_request - opened 13 Jun '22

13 Jun '22

alsa-project/alsa-ucm-conf pull request #169 was opened from brenmous: ``` inxi -M Machine: Type: Desktop System: ASUS product: N/A v: N/A serial: <superuser required> Mobo: ASUSTeK model: ROG STRIX Z690-I GAMING WIFI v: Rev 1.xx serial: <superuser required> UEFI: American Megatrends v: 1403 date: 03/30/2022 ``` ``` aplay -l **** List of PLAYBACK Hardware Devices **** card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [27GL850] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 11: HDMI 5 [HDMI 5] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 0/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 3: USB Audio [USB Audio #3] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` Card 3 is ALC4080 ``` udevadm info -a -p /sys/class/sound/card3 | grep idVendor -m1 ATTRS{idVendor}=="0b05" udevadm info -a -p /sys/class/sound/card3 | grep idProduct -m1 ATTRS{idProduct}=="1a20" ``` Rear panel line out and front panel headphone now work. Jack detection also working properly. Haven't tested any of the other inputs/outputs as I don't have suitable devices. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/169 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/169.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] soc: fix typos in comment
by Ren Yu 13 Jun '22

13 Jun '22

change 'slection' to 'selection' Signed-off-by: Ren Yu <renyu(a)nfschina.com> --- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/rt5631.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index bd0078e4499b..2eb43e93778e 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1201,7 +1201,7 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, case 12288000: case 26000000: case 19200000: - /* pll clk slection */ + /* pll clk selection */ pll_clk = 0; break; case 48000: diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h index 8a6b99a48c7c..0caca2e1d31b 100644 --- a/sound/soc/codecs/rt5631.h +++ b/sound/soc/codecs/rt5631.h @@ -373,7 +373,7 @@ */ #define RT5631_SDP_DAC_DATA_L_R_SWAP (0x1 << 4) -/* Data Length Slection */ +/* Data Length Selection */ #define RT5631_SDP_I2S_DL_MASK (0x3 << 2) #define RT5631_SDP_I2S_DL_16 (0x0 << 2) #define RT5631_SDP_I2S_DL_20 (0x1 << 2) -- 2.11.0

1 0

[PATCH v7 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 13 Jun '22

13 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v6: - The Makefile and Kconfig updates are moved into the driver patches so that can keep independence of each patch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3009 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 14 files changed, 5956 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.18.0

2 12

[PATCH] soc: There are several spelling mistakes. Fix them
by Ren Yu 13 Jun '22

13 Jun '22

Signed-off-by: Ren Yu <renyu(a)nfschina.com> --- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/rt5631.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index bd0078e4499b..2eb43e93778e 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1201,7 +1201,7 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, case 12288000: case 26000000: case 19200000: - /* pll clk slection */ + /* pll clk selection */ pll_clk = 0; break; case 48000: diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h index 8a6b99a48c7c..0caca2e1d31b 100644 --- a/sound/soc/codecs/rt5631.h +++ b/sound/soc/codecs/rt5631.h @@ -373,7 +373,7 @@ */ #define RT5631_SDP_DAC_DATA_L_R_SWAP (0x1 << 4) -/* Data Length Slection */ +/* Data Length Selection */ #define RT5631_SDP_I2S_DL_MASK (0x3 << 2) #define RT5631_SDP_I2S_DL_16 (0x0 << 2) #define RT5631_SDP_I2S_DL_20 (0x1 << 2) -- 2.11.0

1 0

[PATCH] ALSA: usb-audio: US16x08: Move overflow check before array access
by Daniil Dementev 13 Jun '22

13 Jun '22

Buffer overflow could occur in the loop "while", due to accessing an array element before checking the index. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Daniil Dementev <d.dementev(a)ispras.ru> Reviewed-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> --- sound/usb/mixer_us16x08.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index b7b6f3834ed5..6eb7d93b358d 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -637,10 +637,10 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) } } else { /* skip channels with no compressor active */ - while (!store->comp_store->val[ + while (store->comp_index <= SND_US16X08_MAX_CHANNELS + && !store->comp_store->val[ COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] - [store->comp_index - 1] - && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + [store->comp_index - 1]) { store->comp_index++; } ret = store->comp_index++; -- 2.35.1.windows.2

2 1

Asus Z690-E: No Profiles Available in alsa-lib 1.2.7-2
by GitHub issues - opened 11 Jun '22

11 Jun '22

alsa-project/alsa-ucm-conf issue #168 was opened from BrothersRM: Upgrading alsa-lib from 1.2.6.1-1 to 1.2.7-2 seems to have broken support for my motherboard's sound card. Downgrading the package did resolve the issue. Device is listed in `lsusb -v` as : `Bus 001 Device 002: ID 0b05:1996 ASUSTek Computer, Inc. USB Audio` My hardware configuration uses speakers attached to the front, surround and subwoofer ports on my motherboard. Some interesting things I noticed while it was broken: - Plugging headphones into the front-panel headphone port allowed me to turn on the rear speakers, though they didn't seem to use the correct profile. Max volume was far lower - Though the device was listed as unavailable, I could manually enable the rear speakers using pavucontrol when front headphones weren't plugged in - Trying to run speaker-test on the device failed with 'no valid configuration' Please let me know if I can provide more information / do any testing. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/168 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

1.2.7 aplay doesn't compile
by GitHub issues - opened 11 Jun '22

11 Jun '22

alsa-project/alsa-utils issue #153 was opened from LoneFox78: I'm getting this error: ``` Making all in aplay make[1]: Entering directory '/mnt/pkg/pkgmk_work/alsa-utils/src/alsa-utils-1.2.7/aplay' gcc -DHAVE_CONFIG_H -I. -I../include -I../include -s -Wl,-s -Wall -O2 -march=native -ftree-vectorize -pipe -MT aplay.o -MD -MP -MF .deps/aplay.Tpo -c -o aplay.o aplay.c aplay.c:44:10: fatal error: alsa/use-case.h: No such file or directory 44 | #include <alsa/use-case.h> | ^~~~~~~~~~~~~~~~~ compilation terminated. make[1]: *** [Makefile:495: aplay.o] Error 1 make[1]: Leaving directory '/mnt/pkg/pkgmk_work/alsa-utils/src/alsa-utils-1.2.7/aplay' make: *** [Makefile:461: all-recursive] Error 1 ``` The configure script already looks for that header, but I guess someone forgot to add the corresponding <code>#ifdef</code>s in this file. Issue URL : https://github.com/alsa-project/alsa-utils/issues/153 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: pcm: add 24KHz sample rate definition
by Patrick Lai 11 Jun '22

11 Jun '22

Add 24KHz sample rate definition as some hardware can support 24KHz sample rate PCM data Signed-off-by: Patrick Lai <quic_plai(a)quicinc.com> --- include/sound/pcm.h | 24 +++++++++++++----------- sound/core/pcm_native.c | 4 ++-- 2 files changed, 15 insertions(+), 13 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6b99310b5b88..a761a4459091 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -111,23 +111,25 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_11025 (1<<2) /* 11025Hz */ #define SNDRV_PCM_RATE_16000 (1<<3) /* 16000Hz */ #define SNDRV_PCM_RATE_22050 (1<<4) /* 22050Hz */ -#define SNDRV_PCM_RATE_32000 (1<<5) /* 32000Hz */ -#define SNDRV_PCM_RATE_44100 (1<<6) /* 44100Hz */ -#define SNDRV_PCM_RATE_48000 (1<<7) /* 48000Hz */ -#define SNDRV_PCM_RATE_64000 (1<<8) /* 64000Hz */ -#define SNDRV_PCM_RATE_88200 (1<<9) /* 88200Hz */ -#define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ -#define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ -#define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ -#define SNDRV_PCM_RATE_352800 (1<<13) /* 352800Hz */ -#define SNDRV_PCM_RATE_384000 (1<<14) /* 384000Hz */ +#define SNDDV_PCM_RATE_24000 (1<<5) /* 24000Hz */ +#define SNDRV_PCM_RATE_32000 (1<<6) /* 32000Hz */ +#define SNDRV_PCM_RATE_44100 (1<<7) /* 44100Hz */ +#define SNDRV_PCM_RATE_48000 (1<<8) /* 48000Hz */ +#define SNDRV_PCM_RATE_64000 (1<<9) /* 64000Hz */ +#define SNDRV_PCM_RATE_88200 (1<<10) /* 88200Hz */ +#define SNDRV_PCM_RATE_96000 (1<<11) /* 96000Hz */ +#define SNDRV_PCM_RATE_176400 (1<<12) /* 176400Hz */ +#define SNDRV_PCM_RATE_192000 (1<<13) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1<<14) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1<<15) /* 384000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ #define SNDRV_PCM_RATE_8000_44100 (SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_11025|\ SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_22050|\ - SNDRV_PCM_RATE_32000|SNDRV_PCM_RATE_44100) + SNDRV_PCM_RATE_24000|SNDRV_PCM_RATE_32000|\ + SNDRV_PCM_RATE_44100) #define SNDRV_PCM_RATE_8000_48000 (SNDRV_PCM_RATE_8000_44100|SNDRV_PCM_RATE_48000) #define SNDRV_PCM_RATE_8000_96000 (SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_64000|\ SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4adaee62ef33..72e3eab09518 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2442,12 +2442,12 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 13 #error "Change this table" #endif static const unsigned int rates[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 5512, 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000 }; -- 2.36.1

2 1

[PATCH v3 0/2] [PATCH v3 0/2] Add pinctrl support adsp bypass platforms
by Srinivasa Rao Mandadapu 11 Jun '22

11 Jun '22

This patch set is to make clock voting optinal for adsp bypass sc7280 platforms. Changes Since V2: -- Remove redundant lpi pincontrol variant data structure and compatible entry. -- Add adsp bypass mode boolean param check. -- Remove compatible name in dt bindings. -- Update dt binding bypass mode boolean param. Changes Since V1: -- Update commit message. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add boolean param for ADSP bypass platforms pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets .../bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 9 ++++++++- drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 3 +++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 2 +- 3 files changed, 12 insertions(+), 2 deletions(-) -- 2.7.4

2 5

[PATCH] ASoC: samsung: i2s: Check before clk_unregister() not needed
by Yihao Han 10 Jun '22

10 Jun '22

clk_unregister() already checks the clk ptr using !clk || WARN_ON_ONCE(IS_ERR(clk)) so there is no need to check it again before calling it. Signed-off-by: Yihao Han <hanyihao(a)vivo.com> --- sound/soc/samsung/i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 70c827162be4..84e21f7fc179 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1247,8 +1247,7 @@ static void i2s_unregister_clocks(struct samsung_i2s_priv *priv) int i; for (i = 0; i < priv->clk_data.clk_num; i++) { - if (!IS_ERR(priv->clk_table[i])) - clk_unregister(priv->clk_table[i]); + clk_unregister(priv->clk_table[i]); } } -- 2.17.1

3 3

[PATCH] ASoC: qdsp6: q6apm-dai: unprepare stream if its already prepared
by Srinivas Kandagatla 10 Jun '22

10 Jun '22

prepare callback can be called multiple times, so unprepare the stream if its already prepared. Without this DSP is not happy to setting the params on a already prepared graph. Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support") Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 19c4a90ec1ea..ee59ef36b85a 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -147,6 +147,12 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, cfg.num_channels = runtime->channels; cfg.bit_width = prtd->bits_per_sample; + if (prtd->state) { + /* clear the previous setup if any */ + q6apm_graph_stop(prtd->graph); + q6apm_unmap_memory_regions(prtd->graph, substream->stream); + } + prtd->pcm_count = snd_pcm_lib_period_bytes(substream); prtd->pos = 0; /* rate and channels are sent to audio driver */ -- 2.25.1

2 1

[PATCH] ASoC: Remove unused hw_write_t type
by Amadeusz Sławiński 10 Jun '22

10 Jun '22

Commit 81da8a0b7975 ("ASoC: remove codec hw_write/control_data") removed use of hw_write_t in struct snd_soc_codec, but it left type definition. Fully clean it up. Fixes: 81da8a0b7975 ("ASoC: remove codec hw_write/control_data") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f20f5f890794..b276dcb5d4e8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -408,8 +408,6 @@ struct snd_soc_jack_pin; struct snd_soc_jack_gpio; -typedef int (*hw_write_t)(void *,const char* ,int); - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, -- 2.25.1

2 1

[PATCH] ASoC: Intel: avs: Fix parsing UUIDs in topology
by Amadeusz Sławiński 10 Jun '22

10 Jun '22

Use correct type for parsing UUIDs, this eliminates warning present, when compiling with W=1. Fixes: 34ae2cd53673 ("ASoC: Intel: avs: Add topology parsing infrastructure") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/avs/topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 0d11cc8aab0b..6a06fe387d13 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -128,10 +128,10 @@ struct avs_tplg_token_parser { static int avs_parse_uuid_token(struct snd_soc_component *comp, void *elem, void *object, u32 offset) { - struct snd_soc_tplg_vendor_value_elem *tuple = elem; + struct snd_soc_tplg_vendor_uuid_elem *tuple = elem; guid_t *val = (guid_t *)((u8 *)object + offset); - guid_copy((guid_t *)val, (const guid_t *)&tuple->value); + guid_copy((guid_t *)val, (const guid_t *)&tuple->uuid); return 0; } -- 2.25.1

2 1

[PATCH 0/6] ASoC: SOF: Add support ctx_save with IPC4
by Peter Ujfalusi 10 Jun '22

10 Jun '22

Hi, The context save functionality with IPC4 is triggered by sending a message to the firmware about the pending power down of the primary core by the host. In order to have this functionality implemented in a clean way we need to introduce a new IPC level PM ops for core state management and use that instead of open coding IPC messages here and there. The first patch updates the ctx store/ctx_restore documentation to clarify that they are optional. Regards, Peter --- Peter Ujfalusi (6): ASoC: SOF: make ctx_store and ctx_restore as optional ASoC: SOF: sof_ipc_pm_ops: Add support for DSP core power management ASoC: SOF: ipc3: Add set_core_state pm_ops implementation ASoC: SOF: ipc4: Add set_core_state pm_ops implementation ASoC: SOF: Intel: Switch to use the generic pm_ops.set_core_state ASoC: SOF: ipc4: implement pm ctx_save callback include/sound/sof/ipc4/header.h | 8 +++++++ sound/soc/sof/intel/hda-dsp.c | 15 +++++------- sound/soc/sof/intel/tgl.c | 30 ++++++++---------------- sound/soc/sof/ipc3.c | 18 +++++++++++++++ sound/soc/sof/ipc4.c | 41 +++++++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 6 +++-- 6 files changed, 87 insertions(+), 31 deletions(-) -- 2.36.1

2 7

[PATCH 0/3] ASoC: SOF: ipc3-dtrace: Handle race during initialization
by Peter Ujfalusi 10 Jun '22

10 Jun '22

Hi, This series handles the race which can result missing the first position update after the trace is enabled. In short: the firmware might send the position update (if we have enough trace data generated) after the dma-trace is enabled by the TRACE_DMA_PARAMS_EXT message. Depending on scheduling, load, preemption on Linux side we have seen that occasionally this first position update got missed and we missed reading it out. A new state and more strict handling of host_offset can overcome this issue, making the dtrace more reliable. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: ipc3-dtrace: Introduce SOF_DTRACE_INITIALIZING state ASoC: SOF: ipc3-dtrace: Add helper function to update the sdev->host_offset ASoC: SOF: ipc3-dtrace: Return from dtrace_read if there is no new data available sound/soc/sof/ipc3-dtrace.c | 47 ++++++++++++++++++++++++++++++------- 1 file changed, 38 insertions(+), 9 deletions(-) -- 2.36.1

2 4

[PATCH 0/2] ASoC: codecs: qualcomm move gain to S8_TLV
by Srinivas Kandagatla 10 Jun '22

10 Jun '22

Move all the digital and IIR gains form using SX_TLV to S8_TLV, these gains are actually 8 bit gains with 7th signed bit and ranges from -84dB to +40dB Tested on DB410c with Headset playback Thanks, Srini Srinivas Kandagatla (2): ASoC: codecs: msm8916-wcd-digital: move gains from SX_TLV to S8_TLV ASoC: codecs: wcd9335: move gains from SX_TLV to S8_TLV sound/soc/codecs/msm8916-wcd-digital.c | 46 +++++++-------- sound/soc/codecs/wcd9335.c | 81 ++++++++++++-------------- 2 files changed, 59 insertions(+), 68 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: SOF: debug: Clarify the IPC timeout handling path
by Peter Ujfalusi 10 Jun '22

10 Jun '22

The dmesg log message of "Firmware exception" causes lots of confusion as the snd_sof_handle_fw_exception() is only called in case of an IPC tx timeout, where such a message does not make much sense. To not limit the snd_sof_handle_fw_exception() handler to just one error case, add a parameter to allow the caller to specify a meaningful message to be printed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Yaochun Hung <yc.hung(a)mediatek.com> --- sound/soc/sof/debug.c | 5 ++--- sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/sof-priv.h | 2 +- 3 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index cf1271eb29b2..c5d797e97c02 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -428,7 +428,7 @@ static void snd_sof_ipc_dump(struct snd_sof_dev *sdev) } } -void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev) +void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg) { if (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT) || sof_debug_check_flag(SOF_DBG_RETAIN_CTX)) { @@ -441,8 +441,7 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev) /* dump vital information to the logs */ snd_sof_ipc_dump(sdev); - snd_sof_dsp_dbg_dump(sdev, "Firmware exception", - SOF_DBG_DUMP_REGS | SOF_DBG_DUMP_MBOX); + snd_sof_dsp_dbg_dump(sdev, msg, SOF_DBG_DUMP_REGS | SOF_DBG_DUMP_MBOX); sof_fw_trace_fw_crashed(sdev); } EXPORT_SYMBOL(snd_sof_handle_fw_exception); diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index dff5feaad370..ef8019e009b7 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -290,7 +290,7 @@ static int ipc3_wait_tx_done(struct snd_sof_ipc *ipc, void *reply_data) dev_err(sdev->dev, "ipc tx timed out for %#x (msg/reply size: %d/%zu)\n", hdr->cmd, hdr->size, msg->reply_size); - snd_sof_handle_fw_exception(ipc->sdev); + snd_sof_handle_fw_exception(ipc->sdev, "IPC timeout"); ret = -ETIMEDOUT; } else { ret = msg->reply_error; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70..32c152528f1d 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -655,7 +655,7 @@ void sof_print_oops_and_stack(struct snd_sof_dev *sdev, const char *level, u32 panic_code, u32 tracep_code, void *oops, struct sof_ipc_panic_info *panic_info, void *stack, size_t stack_words); -void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev); +void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg); int snd_sof_dbg_memory_info_init(struct snd_sof_dev *sdev); int snd_sof_debugfs_add_region_item_iomem(struct snd_sof_dev *sdev, enum snd_sof_fw_blk_type blk_type, u32 offset, size_t size, -- 2.36.1

2 1

[PATCH] ASoC: SOF: ipc3-dtrace: use pm_runtime_resume_and_get()
by Peter Ujfalusi 10 Jun '22

10 Jun '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Use pm_runtime_resume_and_get() to replace the pm_runtime_get_sync() and pm_runtime_put_noidle() pattern. No functional changes. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc3-dtrace.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index b4e1343f9138..45bf9c5dc412 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -157,9 +157,8 @@ static int ipc3_trace_update_filter(struct snd_sof_dev *sdev, int num_elems, msg->elem_cnt = num_elems; memcpy(&msg->elems[0], elems, num_elems * sizeof(*elems)); - ret = pm_runtime_get_sync(sdev->dev); + ret = pm_runtime_resume_and_get(sdev->dev); if (ret < 0 && ret != -EACCES) { - pm_runtime_put_noidle(sdev->dev); dev_err(sdev->dev, "enabling device failed: %d\n", ret); goto error; } -- 2.36.1

2 1

[RFC PATCH v2 0/5] Apple Macs machine/platform ASoC driver
by Martin Povišer 10 Jun '22

10 Jun '22

Hi all, This is again RFC with a machine-level ASoC driver for recent Apple Macs with the M1 line of chips. This time I attached the platform driver too for good measure. What I am interested in the most is checking the overall approach, especially on two points (both in some ways already discussed in previous RFC [0]): - The way the platform/machine driver handles the fact that multiple I2S ports (now backend DAIs) can be driven by/connected to the same SERDES unit (now in effect a frontend DAI). After previous discussion I have transitioned to DPCM to model this. I took the opportunity of dynamic backend/frontend routing to support speakers/headphones runtime switching. More on this in comments at top of the machine and platform driver. - The way the machine driver deactivates some of the controls where suitable, and limits volume on others. I added a new ASoC card method to that end. Kind regards, Martin [0] https://lore.kernel.org/linux-devicetree/20220331000449.41062-1-povik+lin@c… Martin Povišer (5): dt-bindings: sound: Add Apple MCA I2S transceiver dt-bindings: sound: Add Apple Macs sound peripherals ASoC: apple: Add MCA platform driver for Apple SoCs ASoC: Introduce 'fixup_controls' card method ASoC: apple: Add macaudio machine driver .../bindings/sound/apple,macaudio.yaml | 157 +++ .../devicetree/bindings/sound/apple,mca.yaml | 102 ++ include/sound/soc-card.h | 1 + include/sound/soc.h | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/apple/Kconfig | 25 + sound/soc/apple/Makefile | 5 + sound/soc/apple/macaudio.c | 1004 +++++++++++++++ sound/soc/apple/mca.c | 1122 +++++++++++++++++ sound/soc/soc-card.c | 6 + sound/soc/soc-core.c | 1 + 12 files changed, 2426 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/apple,macaudio.yaml create mode 100644 Documentation/devicetree/bindings/sound/apple,mca.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/macaudio.c create mode 100644 sound/soc/apple/mca.c -- 2.33.0

4 24

[PATCH] ASoC: SOF: ipc3-topology: Move and correct size checks in sof_ipc3_control_load_bytes()
by Peter Ujfalusi 10 Jun '22

10 Jun '22

Move the size checks prior to allocating memory as these checks do not need the data to be allocated and in case of an error we would not need to free the allocation. The max size must not be less than the size of struct sof_ipc_ctrl_data + struct sof_abi_hdr as the ABI header needs to be present under all circumstances. The check was incorrectly used or between the two size checks. Fixes: b5cee8feb1d4 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(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> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 043554d7cb4a..10740c55294d 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1577,24 +1577,23 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ struct sof_ipc_ctrl_data *cdata; int ret; - scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); - if (!scontrol->ipc_control_data) - return -ENOMEM; - - if (scontrol->max_size < sizeof(*cdata) || - scontrol->max_size < sizeof(struct sof_abi_hdr)) { - ret = -EINVAL; - goto err; + if (scontrol->max_size < (sizeof(*cdata) + sizeof(struct sof_abi_hdr))) { + dev_err(sdev->dev, "%s: insufficient size for a bytes control: %zu.\n", + __func__, scontrol->max_size); + return -EINVAL; } - /* init the get/put bytes data */ if (scontrol->priv_size > scontrol->max_size - sizeof(*cdata)) { - dev_err(sdev->dev, "err: bytes data size %zu exceeds max %zu.\n", + dev_err(sdev->dev, + "%s: bytes data size %zu exceeds max %zu.\n", __func__, scontrol->priv_size, scontrol->max_size - sizeof(*cdata)); - ret = -EINVAL; - goto err; + return -EINVAL; } + scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); + if (!scontrol->ipc_control_data) + return -ENOMEM; + scontrol->size = sizeof(struct sof_ipc_ctrl_data) + scontrol->priv_size; cdata = scontrol->ipc_control_data; -- 2.36.1

2 1

[PATCH 1/3] ASoC: wm8940: Remove warning when no plat data
by Lukasz Majewski 10 Jun '22

10 Jun '22

The lack of platform data in the contemporary Linux shall not be the reason to display warnings to the kernel logs. Signed-off-by: Lukasz Majewski <lukma(a)denx.de> --- sound/soc/codecs/wm8940.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 440d048ef0c0..7cea54720436 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -709,9 +709,7 @@ static int wm8940_probe(struct snd_soc_component *component) if (ret < 0) return ret; - if (!pdata) - dev_warn(component->dev, "No platform data supplied\n"); - else { + if (pdata) { reg = snd_soc_component_read(component, WM8940_OUTPUTCTL); ret = snd_soc_component_write(component, WM8940_OUTPUTCTL, reg | pdata->vroi); if (ret < 0) -- 2.20.1

2 13

snd-usb-audio: probe of 1-1:1.0 failed with error -16
by Angelo Capone 10 Jun '22

10 Jun '22

Hi, my linux is not able to recognize usb headset, below you canfind dmesg output, while attached the alsa-info output. Can you help me? Thanks in advance. [10996.537749] usb 1-1: new full-speed USB device number 11 using xhci_hcd [10996.698320] usb 1-1: New USB device found, idVendor=0b0e, idProduct=0300, bcdDevice= 3.08 [10996.698337] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [10996.698346] usb 1-1: Product: Jabra EVOLVE 20 MS [10996.698354] usb 1-1: Manufacturer: GN Netcom A/S [10996.698362] usb 1-1: SerialNumber: 000D06866CEE0A [10996.699759] snd-usb-audio 1-1:1.0: cannot find the slot for index 0 (range 0-0), error: -16 [10996.699788] usb 1-1: cannot create card instance 0 [10996.699813] snd-usb-audio: probe of 1-1:1.0 failed with error -16 [10996.706056] input: GN Netcom A/S Jabra EVOLVE 20 MS as /devices/pci0000:00/0000:00:14.0/usb1/1-1/1-1:1.3/0003:0B0E:0300.0009/input/input38 [10996.766432] jabra 0003:0B0E:0300.0009: input,hiddev0,hidraw3: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE 20 MS] on usb-0000:00:14.0-1/input3 -- Angelo

1 0

[PATCH v4 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 10 Jun '22

10 Jun '22

This patch set is to add software clock gating requirement check Changes Since V3: -- Remove redundant check before reset control call -- Reorganiaze patches. -- Fix typos. Changes Since V2: -- Fix if check before reset control call Changes Since V1: -- Use boolean flag for bool variable initialization instead of hard coding. Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Enable software clock gating requirement flag drivers/soundwire/qcom.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) -- 2.7.4

2 4

[PATCH v2 00/14] ASoC: Intel: avs: Topology and path management
by Cezary Rojewski 10 Jun '22

10 Jun '22

A continuation of avs-driver initial series [1]. This chapter covers path management and topology parsing part which was ealier path of the main series. The two patches that represented these two subjects in the initial series, have been split into many to allow for easier review and discussion. AVS topology is split into two major parts: dictionaries - found within ASoC topology manifest - and path templates. Dictionaries job is to reduce the total amount of memory occupied by topology elements. Rather than having every pipeline and module carry its own information, each refers to specific entry in specific dictionary by provided (from topology file) indexes. In consequence, most struct avs_tplg_xxx are made out of pointers. Path templates are similar to path descriptions found in skylake-driver and they describe how given path shall look like in runtime - number of modules and pipelines that shape it and how they are laid out. A single path template is tied either to FE or BE and thus at most to a single, user-visible endpoint when speaking of FE. Path is a software representation of its ADSP firmware equivalent. It's a logical container for pipelines which are themselves containers - this time for modules i.e. processing units. Depending on the number of audio formats supported, each path template may carry one or more descriptions of given path. During runtime, when audio format is known, description matching said format is selected and used when instantiating path on ADSP firmware side through IPCs. Changes v1 -> v2: - fixed 'set but not used' warning in patch 10/14 as mentioned by ikp builder Changes RFC -> v1: - Plenty of rewordings and spelling fixes as requested by Pierre - any sysfs-functionality-related left in RFC dropped, will be part of separate subject in the future - separated parsing of formatted strings into: "Support link_mask formatted string" patch, increasing series size to 14 patches [1]: https://lore.kernel.org/alsa-devel/20220207122108.3780926-1-cezary.rojewski… Cezary Rojewski (14): ASoC: Intel: avs: Declare vendor tokens ASoC: Intel: avs: Add topology parsing infrastructure ASoC: Intel: avs: Parse module-extension tuples ASoC: Intel: avs: Parse pplcfg and binding tuples ASoC: Intel: avs: Parse pipeline and module tuples ASoC: Intel: avs: Parse path and path templates tuples ASoC: Intel: avs: Add topology loading operations ASoC: Intel: avs: Support link_mask formatted string ASoC: Intel: avs: Declare path and its components ASoC: Intel: avs: Path creation and freeing ASoC: Intel: avs: Path state management ASoC: Intel: avs: Arm paths after creating them ASoC: Intel: avs: Prepare modules before bindings them ASoC: Intel: avs: Configure modules according to their type include/uapi/sound/intel/avs/tokens.h | 126 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 3 +- sound/soc/intel/avs/avs.h | 23 + sound/soc/intel/avs/path.c | 1005 ++++++++++++++++ sound/soc/intel/avs/path.h | 72 ++ sound/soc/intel/avs/topology.c | 1600 +++++++++++++++++++++++++ sound/soc/intel/avs/topology.h | 194 +++ 8 files changed, 3024 insertions(+), 1 deletion(-) create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/intel/avs/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h base-commit: bc55cfd5718c7c23e5524582e9fa70b4d10f2433 -- 2.25.1

3 19

[PATCH] ALSA: hda/realtek: Add mute LED quirk for HP Omen laptop
by contact＠sohamsen.me 10 Jun '22

10 Jun '22

From: Soham Sen <contact(a)sohamsen.me> The HP Omen 15 laptop needs a quirk to toggle the mute LED. It already is implemented for a different variant of the HP Omen laptop so a fixup entry is needed for this variant. Signed-off-by: Soham Sen <contact(a)sohamsen.me> --- 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 f3ad454b3..495b919bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9010,6 +9010,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), -- 2.36.1

2 1

[PATCH v3] ALSA: control: Use xarray for faster lookups
by Takashi Iwai 10 Jun '22

10 Jun '22

The control elements are managed in a single linked list and we traverse the whole list for matching each numid or ctl id per every inquiry of a control element. This is OK-ish for a small number of elements but obviously it doesn't scale. Especially the matching with the ctl id takes time because it checks each field of the snd_ctl_id element, e.g. the name string is matched with strcmp(). This patch adds the hash tables with Xarray for improving the lookup speed of a control element. There are two xarray tables added to the card; one for numid and another for ctl id. For the numid, we use the numid as the index, while for the ctl id, we calculate a hash key. The lookup is done via a single xa_load() execution. As long as the given control element is found on the Xarray table, that's fine, we can give back a quick lookup result. The problem is when no entry hits on the table, and for this case, we have a slight optimization. Namely, the driver checks whether we had a collision on Xarray table, and do a fallback search (linear lookup of the full entries) only if a hash key collision happened beforehand. So, in theory, the inquiry for a non-existing element might take still time even with this patch in a worst case, but this must be pretty rare. The feature is enabled via CONFIG_SND_CTL_FAST_LOOKUP, which is turned on as default. For simplicity, the option can be turned off only when CONFIG_EXPERT is set ("You are expert? Then you manage 1000 knobs"). Link: https://lore.kernel.org/r/20211028130027.18764-1-tiwai@suse.de Link: https://lore.kernel.org/r/20220609180504.775-1-tiwai@suse.de Link: https://lore.kernel.org/all/cover.1653813866.git.quic_rbankapu@quicinc.com/ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: optimize with collision detection, cleanups v2->v3: Fix missing dependency on CONFIG_XARRAY_MULTI include/sound/core.h | 6 ++ sound/core/Kconfig | 10 +++ sound/core/control.c | 180 +++++++++++++++++++++++++++++++++++-------- sound/core/init.c | 4 + 4 files changed, 168 insertions(+), 32 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..dd28de2343b8 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/xarray.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -103,6 +104,11 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + struct xarray ctl_numids; /* hash table for numids */ + struct xarray ctl_hash; /* hash table for ctl id matching */ + bool ctl_hash_collision; /* ctl_hash collision seen? */ +#endif struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/Kconfig b/sound/core/Kconfig index dd7b40734723..25b2434e4556 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -154,6 +154,16 @@ config SND_VERBOSE_PRINTK You don't need this unless you're debugging ALSA. +config SND_CTL_FAST_LOOKUP + bool "Fast lookup of control elements" if EXPERT + default y + select XARRAY_MULTI + help + This option enables the faster lookup of control elements. + It will consume more memory because of the additional Xarray. + If you want to choose the memory footprint over the performance + inevitably, turn this off. + config SND_DEBUG bool "Debug" help diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..6a8fd9933f06 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -364,6 +364,93 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } +/* check whether the given id is contained in the given kctl */ +static bool elem_id_matches(const struct snd_kcontrol *kctl, + const struct snd_ctl_elem_id *id) +{ + return kctl->id.iface == id->iface && + kctl->id.device == id->device && + kctl->id.subdevice == id->subdevice && + !strncmp(kctl->id.name, id->name, sizeof(kctl->id.name)) && + kctl->id.index <= id->index && + kctl->id.index + kctl->count > id->index; +} + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP +/* Compute a hash key for the corresponding ctl id + * It's for the name lookup, hence the numid is excluded. + * The hash key is bound in LONG_MAX to be used for Xarray key. + */ +#define MULTIPLIER 37 +static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) +{ + unsigned long h; + const unsigned char *p; + + h = id->iface; + h = MULTIPLIER * h + id->device; + h = MULTIPLIER * h + id->subdevice; + for (p = id->name; *p; p++) + h = MULTIPLIER * h + *p; + h = MULTIPLIER * h + id->index; + h &= LONG_MAX; + return h; +} + +/* add hash entries to numid and ctl xarray tables */ +static void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + int i; + + xa_store_range(&card->ctl_numids, kcontrol->id.numid, + kcontrol->id.numid + kcontrol->count - 1, + kcontrol, GFP_KERNEL); + + for (i = 0; i < kcontrol->count; i++) { + id.index = kcontrol->id.index + i; + if (xa_insert(&card->ctl_hash, get_ctl_id_hash(&id), + kcontrol, GFP_KERNEL)) { + /* skip hash for this entry, noting we had collision */ + card->ctl_hash_collision = true; + dev_dbg(card->dev, "ctl_hash collision %d:%s:%d\n", + id.iface, id.name, id.index); + } + } +} + +/* remove hash entries that have been added */ +static void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + struct snd_kcontrol *matched; + unsigned long h; + int i; + + for (i = 0; i < kcontrol->count; i++) { + xa_erase(&card->ctl_numids, id.numid); + h = get_ctl_id_hash(&id); + matched = xa_load(&card->ctl_hash, h); + if (matched && (matched == kcontrol || + elem_id_matches(matched, &id))) + xa_erase(&card->ctl_hash, h); + id.index++; + id.numid++; + } +} +#else /* CONFIG_SND_CTL_FAST_LOOKUP */ +static inline void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +static inline void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +#endif /* CONFIG_SND_CTL_FAST_LOOKUP */ + enum snd_ctl_add_mode { CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, }; @@ -408,6 +495,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + add_hash_entries(card, kcontrol); + for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); @@ -479,6 +568,26 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_replace); +static int __snd_ctl_remove(struct snd_card *card, + struct snd_kcontrol *kcontrol, + bool remove_hash) +{ + unsigned int idx; + + if (snd_BUG_ON(!card || !kcontrol)) + return -EINVAL; + list_del(&kcontrol->list); + + if (remove_hash) + remove_hash_entries(card, kcontrol); + + card->controls_count -= kcontrol->count; + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); + snd_ctl_free_one(kcontrol); + return 0; +} + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance @@ -492,16 +601,7 @@ EXPORT_SYMBOL(snd_ctl_replace); */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - unsigned int idx; - - if (snd_BUG_ON(!card || !kcontrol)) - return -EINVAL; - list_del(&kcontrol->list); - card->controls_count -= kcontrol->count; - for (idx = 0; idx < kcontrol->count; idx++) - snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); - snd_ctl_free_one(kcontrol); - return 0; + return __snd_ctl_remove(card, kcontrol, true); } EXPORT_SYMBOL(snd_ctl_remove); @@ -642,14 +742,30 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return -ENOENT; } + remove_hash_entries(card, kctl); kctl->id = *dst_id; kctl->id.numid = card->last_numid + 1; card->last_numid += kctl->count; + add_hash_entries(card, kctl); up_write(&card->controls_rwsem); return 0; } EXPORT_SYMBOL(snd_ctl_rename_id); +#ifndef CONFIG_SND_CTL_FAST_LOOKUP +static struct snd_kcontrol * +snd_ctl_find_numid_slow(struct snd_card *card, unsigned int numid) +{ + struct snd_kcontrol *kctl; + + list_for_each_entry(kctl, &card->controls, list) { + if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) + return kctl; + } + return NULL; +} +#endif /* !CONFIG_SND_CTL_FAST_LOOKUP */ + /** * snd_ctl_find_numid - find the control instance with the given number-id * @card: the card instance @@ -665,15 +781,13 @@ EXPORT_SYMBOL(snd_ctl_rename_id); */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - struct snd_kcontrol *kctl; - if (snd_BUG_ON(!card || !numid)) return NULL; - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) - return kctl; - } - return NULL; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + return xa_load(&card->ctl_numids, numid); +#else + return snd_ctl_find_numid_slow(card, numid); +#endif } EXPORT_SYMBOL(snd_ctl_find_numid); @@ -699,21 +813,18 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, return NULL; if (id->numid != 0) return snd_ctl_find_numid(card, id->numid); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.iface != id->iface) - continue; - if (kctl->id.device != id->device) - continue; - if (kctl->id.subdevice != id->subdevice) - continue; - if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name))) - continue; - if (kctl->id.index > id->index) - continue; - if (kctl->id.index + kctl->count <= id->index) - continue; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); + if (kctl && elem_id_matches(kctl, id)) return kctl; - } + if (!card->ctl_hash_collision) + return NULL; /* we can rely on only hash table */ +#endif + /* no matching in hash table - try all as the last resort */ + list_for_each_entry(kctl, &card->controls, list) + if (elem_id_matches(kctl, id)) + return kctl; + return NULL; } EXPORT_SYMBOL(snd_ctl_find_id); @@ -2195,8 +2306,13 @@ static int snd_ctl_dev_free(struct snd_device *device) down_write(&card->controls_rwsem); while (!list_empty(&card->controls)) { control = snd_kcontrol(card->controls.next); - snd_ctl_remove(card, control); + __snd_ctl_remove(card, control, false); } + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_destroy(&card->ctl_numids); + xa_destroy(&card->ctl_hash); +#endif up_write(&card->controls_rwsem); put_device(&card->ctl_dev); return 0; diff --git a/sound/core/init.c b/sound/core/init.c index 726a8353201f..1870aee7b64f 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -310,6 +310,10 @@ static int snd_card_init(struct snd_card *card, struct device *parent, rwlock_init(&card->ctl_files_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_init(&card->ctl_numids); + xa_init(&card->ctl_hash); +#endif spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); mutex_init(&card->memory_mutex); -- 2.35.3

1 0

[PATCH v1 14/16] soundwire: Use acpi_dev_for_each_child()
by Rafael J. Wysocki 09 Jun '22

09 Jun '22

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Instead of walking the list of children of an ACPI device directly, use acpi_dev_for_each_child() to carry out an action for all of the given ACPI device's children. This will help to eliminate the children list head from struct acpi_device as it is redundant and it is used in questionable ways in some places (in particular, locking is needed for walking the list pointed to it safely, but it is often missing). Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- drivers/soundwire/slave.c | 115 ++++++++++++++++++++++++++-------------------- 1 file changed, 67 insertions(+), 48 deletions(-) Index: linux-pm/drivers/soundwire/slave.c =================================================================== --- linux-pm.orig/drivers/soundwire/slave.c +++ linux-pm/drivers/soundwire/slave.c @@ -127,6 +127,71 @@ static bool find_slave(struct sdw_bus *b return true; } +struct sdw_acpi_child_walk_data { + struct sdw_bus *bus; + struct acpi_device *adev; + struct sdw_slave_id id; + bool ignore_unique_id; +}; + +static int sdw_acpi_check_duplicate(struct acpi_device *adev, void *data) +{ + struct sdw_acpi_child_walk_data *cwd = data; + struct sdw_bus *bus = cwd->bus; + struct sdw_slave_id id; + + if (adev == cwd->adev) + return 0; + + if (!find_slave(bus, adev, &id)) + return 0; + + if (cwd->id.sdw_version != id.sdw_version || cwd->id.mfg_id != id.mfg_id || + cwd->id.part_id != id.part_id || cwd->id.class_id != id.class_id) + return 0; + + if (cwd->id.unique_id != id.unique_id) { + dev_dbg(bus->dev, + "Valid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", + cwd->id.unique_id, id.unique_id, cwd->id.mfg_id, + cwd->id.part_id); + cwd->ignore_unique_id = false; + return 0; + } + + dev_err(bus->dev, + "Invalid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", + cwd->id.unique_id, id.unique_id, cwd->id.mfg_id, cwd->id.part_id); + return -ENODEV; +} + +static int sdw_acpi_find_one(struct acpi_device *adev, void *data) +{ + struct sdw_bus *bus = data; + struct sdw_acpi_child_walk_data cwd = { + .bus = bus, + .adev = adev, + .ignore_unique_id = true, + }; + int ret; + + if (!find_slave(bus, adev, &cwd.id)) + return 0; + + /* Brute-force O(N^2) search for duplicates. */ + ret = acpi_dev_for_each_child(ACPI_COMPANION(bus->dev), + sdw_acpi_check_duplicate, &cwd); + if (ret) + return ret; + + if (cwd.ignore_unique_id) + cwd.id.unique_id = SDW_IGNORED_UNIQUE_ID; + + /* Ignore errors and continue. */ + sdw_slave_add(bus, &cwd.id, acpi_fwnode_handle(adev)); + return 0; +} + /* * sdw_acpi_find_slaves() - Find Slave devices in Master ACPI node * @bus: SDW bus instance @@ -135,8 +200,7 @@ static bool find_slave(struct sdw_bus *b */ int sdw_acpi_find_slaves(struct sdw_bus *bus) { - struct acpi_device *adev, *parent; - struct acpi_device *adev2, *parent2; + struct acpi_device *parent; parent = ACPI_COMPANION(bus->dev); if (!parent) { @@ -144,52 +208,7 @@ int sdw_acpi_find_slaves(struct sdw_bus return -ENODEV; } - list_for_each_entry(adev, &parent->children, node) { - struct sdw_slave_id id; - struct sdw_slave_id id2; - bool ignore_unique_id = true; - - if (!find_slave(bus, adev, &id)) - continue; - - /* brute-force O(N^2) search for duplicates */ - parent2 = parent; - list_for_each_entry(adev2, &parent2->children, node) { - - if (adev == adev2) - continue; - - if (!find_slave(bus, adev2, &id2)) - continue; - - if (id.sdw_version != id2.sdw_version || - id.mfg_id != id2.mfg_id || - id.part_id != id2.part_id || - id.class_id != id2.class_id) - continue; - - if (id.unique_id != id2.unique_id) { - dev_dbg(bus->dev, - "Valid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", - id.unique_id, id2.unique_id, id.mfg_id, id.part_id); - ignore_unique_id = false; - } else { - dev_err(bus->dev, - "Invalid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", - id.unique_id, id2.unique_id, id.mfg_id, id.part_id); - return -ENODEV; - } - } - - if (ignore_unique_id) - id.unique_id = SDW_IGNORED_UNIQUE_ID; - - /* - * don't error check for sdw_slave_add as we want to continue - * adding Slaves - */ - sdw_slave_add(bus, &id, acpi_fwnode_handle(adev)); - } + acpi_dev_for_each_child(parent, sdw_acpi_find_one, bus); return 0; }

3 5

[PATCH v2] ALSA: control: Use xarray for faster lookups
by Takashi Iwai 09 Jun '22

09 Jun '22

The control elements are managed in a single linked list and we traverse the whole list for matching each numid or ctl id per every inquiry of a control element. This is OK-ish for a small number of elements but obviously it doesn't scale. Especially the matching with the ctl id takes time because it checks each field of the snd_ctl_id element, e.g. the name string is matched with strcmp(). This patch adds the hash tables with Xarray for improving the lookup speed of a control element. There are two xarray tables added to the card; one for numid and another for ctl id. For the numid, we use the numid as the index, while for the ctl id, we calculate a hash key. The lookup is done via a single xa_load() execution. As long as the given control element is found on the Xarray table, that's fine, we can give back a quick lookup result. The problem is when no entry hits on the table, and for this case, we have a slight optimization. Namely, the driver checks whether we had a collision on Xarray table, and do a fallback search (linear lookup of the full entries) only if a hash key collision happened beforehand. So, in theory, the inquiry for a non-existing element might take still time even with this patch in a worst case, but this must be pretty rare. The feature is enabled via CONFIG_SND_CTL_FAST_LOOKUP, which is turned on as default. For simplicity, the option can be turned off only when CONFIG_EXPERT is set ("You are expert? Then you manage 1000 knobs"). Link: https://lore.kernel.org/r/20211028130027.18764-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- So this is a revised patch for the previous RFC (see the link above), as there seems real demands for the fast lookup of control elements. A minor improvement from the previous version is the avoidance of fallback in the case of hash non-collision. include/sound/core.h | 6 ++ sound/core/Kconfig | 9 +++ sound/core/control.c | 180 +++++++++++++++++++++++++++++++++++-------- sound/core/init.c | 4 + 4 files changed, 167 insertions(+), 32 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..dd28de2343b8 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/xarray.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -103,6 +104,11 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + struct xarray ctl_numids; /* hash table for numids */ + struct xarray ctl_hash; /* hash table for ctl id matching */ + bool ctl_hash_collision; /* ctl_hash collision seen? */ +#endif struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/Kconfig b/sound/core/Kconfig index dd7b40734723..fa0f6faf821a 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -154,6 +154,15 @@ config SND_VERBOSE_PRINTK You don't need this unless you're debugging ALSA. +config SND_CTL_FAST_LOOKUP + bool "Fast lookup of control elements" if EXPERT + default y + help + This option enables the faster lookup of control elements. + It will consume more memory because of the additional Xarray. + If you want to choose the memory footprint over the performance + inevitably, turn this off. + config SND_DEBUG bool "Debug" help diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..6a8fd9933f06 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -364,6 +364,93 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } +/* check whether the given id is contained in the given kctl */ +static bool elem_id_matches(const struct snd_kcontrol *kctl, + const struct snd_ctl_elem_id *id) +{ + return kctl->id.iface == id->iface && + kctl->id.device == id->device && + kctl->id.subdevice == id->subdevice && + !strncmp(kctl->id.name, id->name, sizeof(kctl->id.name)) && + kctl->id.index <= id->index && + kctl->id.index + kctl->count > id->index; +} + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP +/* Compute a hash key for the corresponding ctl id + * It's for the name lookup, hence the numid is excluded. + * The hash key is bound in LONG_MAX to be used for Xarray key. + */ +#define MULTIPLIER 37 +static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) +{ + unsigned long h; + const unsigned char *p; + + h = id->iface; + h = MULTIPLIER * h + id->device; + h = MULTIPLIER * h + id->subdevice; + for (p = id->name; *p; p++) + h = MULTIPLIER * h + *p; + h = MULTIPLIER * h + id->index; + h &= LONG_MAX; + return h; +} + +/* add hash entries to numid and ctl xarray tables */ +static void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + int i; + + xa_store_range(&card->ctl_numids, kcontrol->id.numid, + kcontrol->id.numid + kcontrol->count - 1, + kcontrol, GFP_KERNEL); + + for (i = 0; i < kcontrol->count; i++) { + id.index = kcontrol->id.index + i; + if (xa_insert(&card->ctl_hash, get_ctl_id_hash(&id), + kcontrol, GFP_KERNEL)) { + /* skip hash for this entry, noting we had collision */ + card->ctl_hash_collision = true; + dev_dbg(card->dev, "ctl_hash collision %d:%s:%d\n", + id.iface, id.name, id.index); + } + } +} + +/* remove hash entries that have been added */ +static void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + struct snd_kcontrol *matched; + unsigned long h; + int i; + + for (i = 0; i < kcontrol->count; i++) { + xa_erase(&card->ctl_numids, id.numid); + h = get_ctl_id_hash(&id); + matched = xa_load(&card->ctl_hash, h); + if (matched && (matched == kcontrol || + elem_id_matches(matched, &id))) + xa_erase(&card->ctl_hash, h); + id.index++; + id.numid++; + } +} +#else /* CONFIG_SND_CTL_FAST_LOOKUP */ +static inline void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +static inline void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +#endif /* CONFIG_SND_CTL_FAST_LOOKUP */ + enum snd_ctl_add_mode { CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, }; @@ -408,6 +495,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + add_hash_entries(card, kcontrol); + for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); @@ -479,6 +568,26 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_replace); +static int __snd_ctl_remove(struct snd_card *card, + struct snd_kcontrol *kcontrol, + bool remove_hash) +{ + unsigned int idx; + + if (snd_BUG_ON(!card || !kcontrol)) + return -EINVAL; + list_del(&kcontrol->list); + + if (remove_hash) + remove_hash_entries(card, kcontrol); + + card->controls_count -= kcontrol->count; + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); + snd_ctl_free_one(kcontrol); + return 0; +} + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance @@ -492,16 +601,7 @@ EXPORT_SYMBOL(snd_ctl_replace); */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - unsigned int idx; - - if (snd_BUG_ON(!card || !kcontrol)) - return -EINVAL; - list_del(&kcontrol->list); - card->controls_count -= kcontrol->count; - for (idx = 0; idx < kcontrol->count; idx++) - snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); - snd_ctl_free_one(kcontrol); - return 0; + return __snd_ctl_remove(card, kcontrol, true); } EXPORT_SYMBOL(snd_ctl_remove); @@ -642,14 +742,30 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return -ENOENT; } + remove_hash_entries(card, kctl); kctl->id = *dst_id; kctl->id.numid = card->last_numid + 1; card->last_numid += kctl->count; + add_hash_entries(card, kctl); up_write(&card->controls_rwsem); return 0; } EXPORT_SYMBOL(snd_ctl_rename_id); +#ifndef CONFIG_SND_CTL_FAST_LOOKUP +static struct snd_kcontrol * +snd_ctl_find_numid_slow(struct snd_card *card, unsigned int numid) +{ + struct snd_kcontrol *kctl; + + list_for_each_entry(kctl, &card->controls, list) { + if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) + return kctl; + } + return NULL; +} +#endif /* !CONFIG_SND_CTL_FAST_LOOKUP */ + /** * snd_ctl_find_numid - find the control instance with the given number-id * @card: the card instance @@ -665,15 +781,13 @@ EXPORT_SYMBOL(snd_ctl_rename_id); */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - struct snd_kcontrol *kctl; - if (snd_BUG_ON(!card || !numid)) return NULL; - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) - return kctl; - } - return NULL; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + return xa_load(&card->ctl_numids, numid); +#else + return snd_ctl_find_numid_slow(card, numid); +#endif } EXPORT_SYMBOL(snd_ctl_find_numid); @@ -699,21 +813,18 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, return NULL; if (id->numid != 0) return snd_ctl_find_numid(card, id->numid); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.iface != id->iface) - continue; - if (kctl->id.device != id->device) - continue; - if (kctl->id.subdevice != id->subdevice) - continue; - if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name))) - continue; - if (kctl->id.index > id->index) - continue; - if (kctl->id.index + kctl->count <= id->index) - continue; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); + if (kctl && elem_id_matches(kctl, id)) return kctl; - } + if (!card->ctl_hash_collision) + return NULL; /* we can rely on only hash table */ +#endif + /* no matching in hash table - try all as the last resort */ + list_for_each_entry(kctl, &card->controls, list) + if (elem_id_matches(kctl, id)) + return kctl; + return NULL; } EXPORT_SYMBOL(snd_ctl_find_id); @@ -2195,8 +2306,13 @@ static int snd_ctl_dev_free(struct snd_device *device) down_write(&card->controls_rwsem); while (!list_empty(&card->controls)) { control = snd_kcontrol(card->controls.next); - snd_ctl_remove(card, control); + __snd_ctl_remove(card, control, false); } + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_destroy(&card->ctl_numids); + xa_destroy(&card->ctl_hash); +#endif up_write(&card->controls_rwsem); put_device(&card->ctl_dev); return 0; diff --git a/sound/core/init.c b/sound/core/init.c index 726a8353201f..1870aee7b64f 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -310,6 +310,10 @@ static int snd_card_init(struct snd_card *card, struct device *parent, rwlock_init(&card->ctl_files_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_init(&card->ctl_numids); + xa_init(&card->ctl_hash); +#endif spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); mutex_init(&card->memory_mutex); -- 2.35.3

1 0

Re: [RFC PATCH v2 5/5] ASoC: apple: Add macaudio machine driver
by Martin Povišer 09 Jun '22

09 Jun '22

> On 9. 6. 2022, at 17:50, Mark Brown <broonie(a)kernel.org> wrote: > > On Thu, Jun 09, 2022 at 05:24:49PM +0200, Martin Povišer wrote: >>> On 9. 6. 2022, at 17:03, Mark Brown <broonie(a)kernel.org> wrote: > > Why is this off list? By accident, added the CC list back with this reply (hopefully it still attaches to the thread when people receive it). >>> That's basically no userspaces at this point TBH. I'm not convinced >>> it's a good idea to be adding custom code for that use case. >> >> FWIW I know of at least one user of the WIP audio support on Macs who >> would welcome this feature. My preference is to keep it in, but in >> the end I guess it’s your call. > > I'd rather not have this open coded in individual drivers, we already > have an unfortunate abundance of jack detection interfaces. If we're > going to add anything I'd rather it were in core code and TBH I'm > struggling to be enthusiastic. Noted. > Can you say anything more about the use case? I can restate: The alleged use case is running userspace without sound server, but having playback switch transparently between speakers and headphones even mid-stream based on jack detection. >>>> I looked at the existing DAPM integration but I couldn’t figure out >>>> how to switch the demux with it. > >>> Yes, it won't do that. If you can't stream the same audio to both then >>> you'd need something else. > >> I don’t understand what’s meant by streaming the same audio here. > > Playing one audio stream from the host which appears on both speakers > and headphones - I don't know what the mixing and muxing capabilities of > the hardware are. > >> Taking a guess: The existing DAPM integration can enable the headphones >> path based on jack being plugged in, but it can’t disable the speakers >> path like the demux does? > > No, that works perfectly fine - you can enable or disable pins depending > on the jack state. Ah, I peeked into soc-jack.c. What about this then: If I understand what pins represent, they would be at the remote end of the DAPM paths. So if for the speakers I add something like Headphones Codec Out —> Jack pin +--> Always-on pin | Speaker Amp Out -> Mux | +--> Jack inverted pin and let userspace control the mux, it would in effect support the same use cases as what I attempted in the code so far. Sounds somewhat right?

2 1

[PATCH 0/3] ASoC: SOF: Intel: Fix IMR boot race caused by early core booting
by Peter Ujfalusi 09 Jun '22

09 Jun '22

Hi, After the revisited IMR sequence patch 2a68ff846164 ("ASoC: SOF: Intel: hda: Revisit IMR boot sequence") we have started to see random boot issues manifesting in ghost reply messages from the DSP for a never sent message. It turned out that an earlier commit: d416519982cb ("ASoC: SOF: hda: don't use the core op for power up/power down") replaced the core power up step with a full blown core power up + enable call. The result was that we have booted the core before we would place the message to the ROM code and based on luck and timing it got received by the (IMR) booted firmware or in between ROM and firmware boot. Mark, can you queue this series for 5.19 as the 2a68ff846164 is in rc1 to make sure that we fix this random issue? Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: hda-dsp: Expose hda_dsp_core_power_up() ASoC: SOF: Intel: hda-loader: Make sure that the fw load sequence is followed ASoC: SOF: Intel: hda-loader: Clarify the cl_dsp_init() flow sound/soc/sof/intel/hda-dsp.c | 10 +++++++++- sound/soc/sof/intel/hda-loader.c | 10 +++++----- sound/soc/sof/intel/hda.h | 1 + 3 files changed, 15 insertions(+), 6 deletions(-) -- 2.36.1

2 4

[PATCH V2] ASoC: imx-audmux: remove unnecessary check of clk_disable_unprepare/clk_prepare_enable
by cgel.zte＠gmail.com 09 Jun '22

09 Jun '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Because clk_disable_unprepare/clk_prepare_enable already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- v1->v2: remove the check of audmux_clk before "clk_prepare_enable" sound/soc/fsl/imx-audmux.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index dfa05d40b276..3ba82adace42 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -62,17 +62,14 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, uintptr_t port = (uintptr_t)file->private_data; u32 pdcr, ptcr; - if (audmux_clk) { - ret = clk_prepare_enable(audmux_clk); - if (ret) - return ret; - } + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) @@ -209,17 +206,14 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, if (!audmux_base) return -ENOSYS; - if (audmux_clk) { - ret = clk_prepare_enable(audmux_clk); - if (ret) - return ret; - } + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); return 0; } -- 2.25.1

3 2

Audio mem2mem devices aka asymmetric sample rate converters
by Sascha Hauer 09 Jun '22

09 Jun '22

Hi All, I am currently looking into getting the asymmetric sample rate converters (ASRC) found on some i.MX SoCs to do something useful. The ASRC units are completely independent units, i.e. independent of the rest of the audio subsystem. They can read from memory using the SDMA engine, convert sample rates and/or audio formats and write back to memory also using the SDMA engine. The ASRC on the i.MX8MN has four contexts to convert up to four streams simultanously. I am not aware of any other non i.MX SoCs having such a unit, but I am pretty sure they exist on other SoCs as well. There are likely two usecases for such a unit. First would be to offload sample rate and format conversions to hardware. The other would be to synchronize different audio sources/sinks with different master clocks to each other when the master clocks drift away. How would such units be integrated into ASoC? I can think of two ways. First would be to create an separate audio card from them which records on one end and plays back with a different sample rate / format on the other end, in the v4l2 world that would be a classical mem2mem device. Is Alsa/ASoc prepared for something like this? Would it be feasible to go into such a direction? I haven't found any examples for this in the tree. The other way is to attach the ASRC to an existing audio card. That is done with the existing in-tree sound/soc/fsl/fsl_asrc.c and sound/soc/fsl/fsl_easrc.c drivers. This approach feels somehow limited as it's not possible to just do conversions without playing/recording something. OTOH userspace is unaffected which might be an advantage. What nags me with that approach is that it's currently not integrated into the simple-audio-card or audio-graph-card bindings. Currently the driver can only be used in conjunction with the fsl,imx-audio-* card driver. It seems backward to integrate such a generic ASRC unit into a special purpose audio card driver. The ASoC core is fully unaware of the ASRC with this approach currently which also doesn't look very appealing. OTOH I don't know if ASoC could handle this. Can ASoC handle for example a chain of DAIs when there are different sample rates and formats in that chain? Currently I don't really know how to proceed. It would be great if you could share some thoughts to this topic. Thanks, Sascha -- Pengutronix e.K. | | Steuerwalder Str. 21 | http://www.pengutronix.de/ | 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

2 3

[PATCH 0/4] ALSA: input validation for control element writes
by Takashi Iwai 09 Jun '22

09 Jun '22

Hi, this is a patch set to add a new feature to ALSA control interface, a validation for the write to control elements. So far we rely fully on the driver implementation to deal with the user inputs with invalid values (e.g. a value outside the defined range). This patch set allows ALSA core to verify the input values beforehand and returns the error immediately if the validation fails. The feature is enabled with a new Kconfig for now, as it brings a slight performance overhead, although this could be turned on as default in most cases, IMO. The patch set contains a few preliminary cleanup patches. The essential change is only the last one. Takashi === Takashi Iwai (4): ASoC: topology: Drop superfluous check of CONFIG_SND_CTL_VALIDATION ALSA: control: Rename CONFIG_SND_CTL_VALIDATION to CONFIG_SND_CTL_DEBUG ALSA: control: Drop superfluous ifdef CONFIG_SND_CTL_DEBUG ALSA: control: Add input validation include/sound/control.h | 2 +- sound/core/Kconfig | 27 ++++++++++--- sound/core/control.c | 87 +++++++++++++++++++++++----------------- sound/soc/soc-topology.c | 2 +- 4 files changed, 73 insertions(+), 45 deletions(-) -- 2.35.3

1 4

[PATCH v3 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 09 Jun '22

09 Jun '22

This patch set is to add software clock gating requirement check Changes Since V1: -- Use boolean flag for bool variable initialization instead of hard coding. Changes Since V1: -- Fix if check before reset control call Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Add software clock gating requirement check drivers/soundwire/qcom.c | 28 +++++++++++++++++++++------- 1 file changed, 21 insertions(+), 7 deletions(-) -- 2.7.4

3 6

[PATCH v6 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 09 Jun '22

09 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3009 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 14 files changed, 5956 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.25.1

2 10

About Cleanup ASoC
by Kuninori Morimoto 09 Jun '22

09 Jun '22

Hi ASoC Cc Mark, Pierre-Louis, Cezary I have very interesting to clean up ASoC. I think it is still very complex, thus, non flexible. I'm thinking that we want to cleanup... - Component with multi Card connection - fixed playback vs capture - fixed CPU vs Codec - DPCM connection vs normal connection About first topic, I guess the biggest reason why we have limitation is its connections. Text based image is a little bit difficult, but if my understanding was correct, current connections are... card <-> rtd <-> rtd <-> rtd <-> ... | | +--------|-----+---------------+-----------+ (A) | | | | +-> component -> component -> component -> ... | (CPU) (Codec) (Platform) | | | | | +-> dai -> dai -> dai -> dai -> ... Card has "component list", and Component has pointer to Card (A). This makes the limitation which blocking multiple Card connections, I think. One note here is that "Platform" doesn't have "DAI". So if rtd have something other connector list, let's say "ep" list here for now, instead of "dai", and remove Card <-> Component connection, I think the connection can be simple, and the issue can be gone, but what do you think ? card <-> rtd <-> rtd <-> rtd <-> ... | | dai dai dai dai | | | | | +--> ep --> ep --> ep -> ep -> ep -> ... | | | | | component component component (CPU) (Codec) (Platform) Here, ep is like this struct rtd_entry_point { struct list_head list; // list for rtd struct struct snd_soc_pcm_runtime *rtd; struct snd_soc_component *component; struct snd_soc_dai *dai; }; Current for_each_xxx() can be like this for_each_rtd_components() = for_each_rtd_ep() + ep_to_component() for_each_rtd_dais() = for_each_rtd_ep() + ep_to_dai() for_each_card_components() = for_each_card_rtds() + for_each_rtd_components() I guess Component can have multiple Card connection on this idea, but what do you think ? Maybe I'm misunderstanding and/or missing something, because ASoC is complex enough :) Please let me know if I was misunderstanding or missing. Thank you for your help !! Best regards --- Kuninori Morimoto

5 23

[syzbot] general protection fault in snd_virmidi_output_trigger
by syzbot 09 Jun '22

09 Jun '22

Hello, syzbot found the following issue on: HEAD commit: 952923ddc011 Merge tag 'pull-18-rc1-work.namei' of git://g.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=10b95d0bf00000 kernel config: https://syzkaller.appspot.com/x/.config?x=3096247591885bfa dashboard link: https://syzkaller.appspot.com/bug?extid=8e407e6b1dba0a275884 compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+8e407e6b1dba0a275884(a)syzkaller.appspotmail.com general protection fault, probably for non-canonical address 0xdffffc0000000003: 0000 [#1] PREEMPT SMP KASAN KASAN: null-ptr-deref in range [0x0000000000000018-0x000000000000001f] CPU: 1 PID: 18021 Comm: syz-executor.1 Not tainted 5.18.0-syzkaller-13842-g952923ddc011 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:snd_virmidi_output_trigger+0x82/0x130 sound/core/seq/seq_virmidi.c:169 Code: 8b ad 30 01 00 00 31 ff 89 de e8 d9 8b 4f fa 85 db 48 b8 00 00 00 00 00 fc ff df 41 0f 95 c4 48 8d 7d 1c 48 89 fa 48 c1 ea 03 <0f> b6 04 02 48 89 fa 83 e2 07 38 d0 7f 04 84 c0 75 60 31 ff 44 88 RSP: 0018:ffffc9000c6e7c48 EFLAGS: 00010207 RAX: dffffc0000000000 RBX: 0000000000000001 RCX: ffffffff872af747 RDX: 0000000000000003 RSI: 0000000000000000 RDI: 000000000000001c RBP: 0000000000000000 R08: 0000000000000005 R09: 0000000000000000 R10: 0000000000000001 R11: 0000000000000001 R12: ffff888078d6c801 R13: 000000007f012000 R14: ffff88814ab3c400 R15: 0000000000001000 FS: 00007fc5b5868700(0000) GS:ffff8880b9b00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000020fec000 CR3: 0000000046903000 CR4: 00000000003506e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: <TASK> snd_rawmidi_output_trigger sound/core/rawmidi.c:186 [inline] snd_rawmidi_output_trigger sound/core/rawmidi.c:182 [inline] snd_rawmidi_kernel_write1+0x548/0x870 sound/core/rawmidi.c:1502 snd_rawmidi_write+0x273/0xb90 sound/core/rawmidi.c:1555 vfs_write+0x269/0xac0 fs/read_write.c:589 ksys_write+0x1e8/0x250 fs/read_write.c:644 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 RIP: 0033:0x7fc5b4689109 Code: ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 40 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007fc5b5868168 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007fc5b479bf60 RCX: 00007fc5b4689109 RDX: 00000000fffffd2c RSI: 0000000020000000 RDI: 0000000000000004 RBP: 00007fc5b46e308d R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000000 R13: 00007fc5b4ccfb1f R14: 00007fc5b5868300 R15: 0000000000022000 </TASK> Modules linked in: ---[ end trace 0000000000000000 ]--- RIP: 0010:snd_virmidi_output_trigger+0x82/0x130 sound/core/seq/seq_virmidi.c:169 Code: 8b ad 30 01 00 00 31 ff 89 de e8 d9 8b 4f fa 85 db 48 b8 00 00 00 00 00 fc ff df 41 0f 95 c4 48 8d 7d 1c 48 89 fa 48 c1 ea 03 <0f> b6 04 02 48 89 fa 83 e2 07 38 d0 7f 04 84 c0 75 60 31 ff 44 88 RSP: 0018:ffffc9000c6e7c48 EFLAGS: 00010207 RAX: dffffc0000000000 RBX: 0000000000000001 RCX: ffffffff872af747 RDX: 0000000000000003 RSI: 0000000000000000 RDI: 000000000000001c RBP: 0000000000000000 R08: 0000000000000005 R09: 0000000000000000 R10: 0000000000000001 R11: 0000000000000001 R12: ffff888078d6c801 R13: 000000007f012000 R14: ffff88814ab3c400 R15: 0000000000001000 FS: 00007fc5b5868700(0000) GS:ffff8880b9a00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000001b2bc27000 CR3: 0000000046903000 CR4: 00000000003506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 ---------------- Code disassembly (best guess): 0: 8b ad 30 01 00 00 mov 0x130(%rbp),%ebp 6: 31 ff xor %edi,%edi 8: 89 de mov %ebx,%esi a: e8 d9 8b 4f fa callq 0xfa4f8be8 f: 85 db test %ebx,%ebx 11: 48 b8 00 00 00 00 00 movabs $0xdffffc0000000000,%rax 18: fc ff df 1b: 41 0f 95 c4 setne %r12b 1f: 48 8d 7d 1c lea 0x1c(%rbp),%rdi 23: 48 89 fa mov %rdi,%rdx 26: 48 c1 ea 03 shr $0x3,%rdx * 2a: 0f b6 04 02 movzbl (%rdx,%rax,1),%eax <-- trapping instruction 2e: 48 89 fa mov %rdi,%rdx 31: 83 e2 07 and $0x7,%edx 34: 38 d0 cmp %dl,%al 36: 7f 04 jg 0x3c 38: 84 c0 test %al,%al 3a: 75 60 jne 0x9c 3c: 31 ff xor %edi,%edi 3e: 44 rex.R 3f: 88 .byte 0x88 --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[PATCH v2 1/2] ASoC: dt-bindings: nxp,tfa989x: Add tfa9890 support
by Alexander Martinz 08 Jun '22

08 Jun '22

Document TFA9890 binding for tfa989x. Signed-off-by: Alexander Martinz <amartinz(a)shiftphones.com> Reviewed-by: Stephan Gerhold <stephan(a)gerhold.net> Acked-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml index b9b1dba40856..7f2e68ff6d34 100644 --- a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -15,6 +15,7 @@ allOf: properties: compatible: enum: + - nxp,tfa9890 - nxp,tfa9895 - nxp,tfa9897 -- 2.36.1

2 2

no input from razer headset microphone when using RAZER BASE STATION V2 CHROMA
by GitHub issues - opened 08 Jun '22

08 Jun '22

alsa-project/alsa-lib issue #236 was opened from fiercebrake: Greetings everyone! So I currently have a razer "blacksharck v2 pro" headset that i connect to a razer "base station v2 chroma" The razer base station is properly detected/listed by (arch) linux. Problem is though the card and mic are detected and listed, the mic has no actual input at all. For the mic to start showing/taking any input I need to manually unplug and replug the headset from the base station. The jack in the base station is a 4-pin 3.5mm (TRRS) jack This article/fix below seems to alleviate the problem, say 50% of the times. So I am wondering if there is anything else I can do via ALSA to completely fix the problem "https://wiki.archlinux.org/title/Advanced_Linux_Sound_Architecture#Correctl…" Issue URL : https://github.com/alsa-project/alsa-lib/issues/236 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda/realtek: Add quirk for HP Dev One
by Tim Crawford 08 Jun '22

08 Jun '22

From: Jeremy Soller <jeremy(a)system76.com> Enables the audio mute LEDs and limits the mic boost to avoid picking up noise. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- 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 7170e086f166..b0f954118e72 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9071,6 +9071,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.35.3

2 1

[PATCH][next] ASoC: mediatek: mt8186: Fix a handful of spelling mistakes
by Colin Ian King 08 Jun '22

08 Jun '22

There are several spelling mistakes in dev_err messages. Fix them. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 8 ++++---- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c index 5ba28095b7da..255ffba637d3 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -137,25 +137,25 @@ static int mt8186_afe_gpio_adda_dl(struct device *dev, bool enable) if (enable) { ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MOSI_ON); if (ret) { - dev_err(dev, "%s(), MOSI CLK ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI CLK ON select fail!\n", __func__); return ret; } ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MOSI_ON); if (ret) { - dev_err(dev, "%s(), MOSI DAT ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI DAT ON select fail!\n", __func__); return ret; } } else { ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MOSI_OFF); if (ret) { - dev_err(dev, "%s(), MOSI DAT OFF slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI DAT OFF select fail!\n", __func__); return ret; } ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MOSI_OFF); if (ret) { - dev_err(dev, "%s(), MOSI CLK ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI CLK ON select fail!\n", __func__); return ret; } } diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index c66861fd197d..db71b032770d 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -341,7 +341,7 @@ static int mtk_adda_mtkaif_cfg_event(struct snd_soc_dapm_widget *w, if (afe_priv->mtkaif_chosen_phase[0] < 0 || afe_priv->mtkaif_chosen_phase[1] < 0) { dev_err(afe->dev, - "%s(), skip dealy setting mtkaif_chosen_phase[0/1]:%d/%d\n", + "%s(), skip delay setting mtkaif_chosen_phase[0/1]:%d/%d\n", __func__, afe_priv->mtkaif_chosen_phase[0], afe_priv->mtkaif_chosen_phase[1]); -- 2.35.3

2 1

[PATCH 0/7] ASoC: harden SoundWire codec/machine drivers used on Intel platforms
by Pierre-Louis Bossart 08 Jun '22

08 Jun '22

While testing fixes for SoundWire race conditions initially reported in [1], I found additional issues in codec drivers. When the order in which drivers are probed is changed, multiple errors are reported, from unbalanced pm_runtime_enable() calls to invalid mutex lock magic numbers, workqueues not initialized and missing resume sequences. In 'nominal' usages, there is no change of functionality, this is just a first step to test random device/driver bind/unbind sequences. Important note: these changes only touch Intel-based platforms, I don't have any background and ability to test on Qualcomm-based devices. [1] https://lore.kernel.org/alsa-devel/d0559e97-c4a0-b817-428c-d3e305390270@lin… Pierre-Louis Bossart (7): ASoC: Realtek/Maxim SoundWire codecs: disable pm_runtime on remove ASoC: rt711-sdca-sdw: fix calibrate mutex initialization ASoC: Intel: sof_sdw: handle errors on card registration ASoC: rt711: fix calibrate mutex initialization ASoC: rt7*-sdw: harden jack_detect_handler ASoC: codecs: rt700/rt711/rt711-sdca: initialize workqueues in probe ASoC: codecs: rt700/rt711/rt711-sdca: resume bus/codec in .set_jack_detect sound/soc/codecs/max98373-sdw.c | 12 +++++++- sound/soc/codecs/rt1308-sdw.c | 11 +++++++ sound/soc/codecs/rt1316-sdw.c | 11 +++++++ sound/soc/codecs/rt5682-sdw.c | 5 ++- sound/soc/codecs/rt700-sdw.c | 6 +++- sound/soc/codecs/rt700.c | 30 +++++++++++------- sound/soc/codecs/rt711-sdca-sdw.c | 9 +++++- sound/soc/codecs/rt711-sdca.c | 40 ++++++++++++------------ sound/soc/codecs/rt711-sdw.c | 9 +++++- sound/soc/codecs/rt711.c | 40 ++++++++++++------------ sound/soc/codecs/rt715-sdca-sdw.c | 12 ++++++++ sound/soc/codecs/rt715-sdw.c | 12 ++++++++ sound/soc/intel/boards/sof_sdw.c | 51 ++++++++++++++++++------------- 13 files changed, 169 insertions(+), 79 deletions(-) -- 2.34.1

2 8

[PATCH v2 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 08 Jun '22

08 Jun '22

This patch set is to add software clock gating requirement check Changes Since V1: -- Fix if check before reset control call Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Add software clock gating requirement check drivers/soundwire/qcom.c | 28 +++++++++++++++++++++------- 1 file changed, 21 insertions(+), 7 deletions(-) -- 2.7.4

2 4

[PATCH] ASoC: ops: Fix off by one in range control validation
by Mark Brown 08 Jun '22

08 Jun '22

We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 93e72a016b4d..e11109dcaa50 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -542,7 +542,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -563,7 +563,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.30.2

1 1

[PATCH 00/20] ASoC: DAI clocking terminology modernisation
by Mark Brown 08 Jun '22

08 Jun '22

Update the last batch of CODEC drivers without specific maintainers to use the new defines for DAI clocking. Mark Brown (20): ASoC: cx2072x: Use modern ASoC DAI format terminology ASoC: max98090: Use modern ASoC DAI format terminology ASoC: rk3328: Use modern ASoC DAI format terminology ASoC: sta32x: Use modern ASoC DAI format terminology ASoC: sta350: Use modern ASoC DAI format terminology ASoC: sti-sas: Use modern ASoC DAI format terminology ASoC: tas2552: Use modern ASoC DAI format terminology ASoC: tas2770: Use modern ASoC DAI format terminology ASoC: tas5086: Use modern ASoC DAI format terminology ASoC: tas5720: Use modern ASoC DAI format terminology ASoC: tas6424: Use modern ASoC DAI format terminology ASoC: uda1334: Use modern ASoC DAI format terminology ASoC: tlv320adc3xxx: Use modern ASoC DAI format terminology ASoC: tlv320adcx140: Use modern ASoC DAI format terminology ASoC: tlv320aic23: Use modern ASoC DAI format terminology ASoC: tlv320aic26: Use modern ASoC DAI format terminology ASoC: tlv320aic31xx: Use modern ASoC DAI format terminology ASoC: tlv320aic32x4: Use modern ASoC DAI format terminology ASoC: tlv320aic33: Use modern ASoC DAI format terminology ASoC: tlv320dac3x: Use modern ASoC DAI format terminology sound/soc/codecs/cx2072x.c | 17 +++++++---------- sound/soc/codecs/max98090.c | 12 +++++------- sound/soc/codecs/rk3328_codec.c | 6 +++--- sound/soc/codecs/sta32x.c | 4 ++-- sound/soc/codecs/sta350.c | 4 ++-- sound/soc/codecs/sti-sas.c | 6 +++--- sound/soc/codecs/tas2552.c | 10 +++++----- sound/soc/codecs/tas2770.c | 6 +++--- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tas5720.c | 4 ++-- sound/soc/codecs/tas6424.c | 6 +++--- sound/soc/codecs/tlv320adc3xxx.c | 3 +-- sound/soc/codecs/tlv320adcx140.c | 10 ++++------ sound/soc/codecs/tlv320aic23.c | 7 +++---- sound/soc/codecs/tlv320aic26.c | 13 ++++++------- sound/soc/codecs/tlv320aic31xx.c | 19 +++++++++---------- sound/soc/codecs/tlv320aic32x4.c | 9 ++++----- sound/soc/codecs/tlv320aic3x.c | 11 +++++------ sound/soc/codecs/tlv320dac33.c | 12 +++++------- sound/soc/codecs/uda1334.c | 2 +- 20 files changed, 74 insertions(+), 89 deletions(-) -- 2.30.2

2 23

[PATCH v4] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 08 Jun '22

08 Jun '22

Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..cd1281691767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11545,6 +11558,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.20.1

2 1

[PATCH v3] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 08 Jun '22

08 Jun '22

Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..cd1281691767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11545,6 +11558,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.20.1

3 4

[PATCH] [PATCH v2]ALSA: hda/realtek - Add HW8326 support
by huangwenhui 08 Jun '22

08 Jun '22

Added the support of new Huawei codec HW8326.The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..cd1281691767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11545,6 +11558,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.20.1

2 1

Combined wait for pcm_wait and ctl_wait
by Pavel Hofman 08 Jun '22

08 Jun '22

Hi, Please is there any way to wait for nonblocking read/write at min_avail (like snd_pcm_wait) AND nonblocking subscribed ctl event (like snd_ctl_wait) in a single combined wait? The goal is to react as fast as possible to ctls in aloop and audio gadget devices which notify about stream stop and start (at new rate/format/channel count) on the other side. I guess some wizardry with file descriptors and polling multiple fds could do. Thanks a lot for any help or hints. Best regards, Pavel.

2 2

[PATCH] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 08 Jun '22

08 Jun '22

Added the support of new Huawei codec HW8326. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..cd1281691767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11545,6 +11558,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.20.1

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[PATCH 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 08 Jun '22

08 Jun '22

This patch set is to add software clock gating requirement check Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Add software clock gating requirement check drivers/soundwire/qcom.c | 28 +++++++++++++++++++++------- 1 file changed, 21 insertions(+), 7 deletions(-) -- 2.7.4

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[PATCH 0/6] ASoC: ak4613: add TDM256 test support
by Kuninori Morimoto 08 Jun '22

08 Jun '22

Renesas is the only user of ak4613 driver on upstream for now. It has STEREO/TDM512/TDM256/TDM128 mode, but STEREO only is used, because of Renesas board connection. I noticed that I can test first 2ch out of TDM256 mode 8ch Playback even in such a situation. [PATCH 1/6] - [PATCH 5/6] are prepare for TDM mode support. [PATCH 6/6] uses "ifdef style" to enable TDM because I can't test full TDM256 and/or other TDM mode. "ifdef style" has no effect to current supported STEREO mode. Kuninori Morimoto (6): 1) ASoC: ak4613: add missing mutex_lock() 2) ASoC: ak4613: tidyup ak4613_interface 3) ASoC: ak4613: return error if it was setup as clock provider 4) ASoC: ak4613: priv has ctrl1 instead of iface 5) ASoC: ak4613: rename constraint to constraint_rates 6) ASoC: ak4613: add TDM256 support sound/soc/codecs/ak4613.c | 367 +++++++++++++++++++++++++++++++------- 1 file changed, 300 insertions(+), 67 deletions(-) -- 2.25.1

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[PATCH] ASoC: samsung: Fix error handling in aries_audio_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_get_child_by_name() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. This function is missing of_node_put(cpu) in the error path. Fix this by goto out label. of_node_put() will check NULL pointer. Fixes: format:7a3a7671fa6c ("ASoC: samsung: Add driver for Aries boards") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/samsung/aries_wm8994.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index bb0cf4244e00..edee02d7f100 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -628,8 +628,10 @@ static int aries_audio_probe(struct platform_device *pdev) return -EINVAL; codec = of_get_child_by_name(dev->of_node, "codec"); - if (!codec) - return -EINVAL; + if (!codec) { + ret = -EINVAL; + goto out; + } for_each_card_prelinks(card, i, dai_link) { dai_link->codecs->of_node = of_parse_phandle(codec, -- 2.25.1

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[PATCH] ASoC: topology: Avoid card NULL deref in snd_soc_tplg_component_remove()
by Dean Gehnert 07 Jun '22

07 Jun '22

Don't deference card in comp->card->snd_card before checking for NULL card. During the unloading of ASoC kernel modules, there is a kernel oops in snd_soc_tplg_component_remove() that happens because comp->card is set to NULL in soc_cleanup_component(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: stable(a)vger.kernel.org Fixes: 7e567b5ae063 ("ASoC: topology: Add missing rwsem around snd_ctl_remove() calls") Signed-off-by: Dean Gehnert <deang(a)tpi.com> --- sound/soc/soc-topology.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 3f9d314fba16..cf0efe1147c2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2613,15 +2613,18 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); /* remove dynamic controls from the component driver */ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) { - struct snd_card *card = comp->card->snd_card; + struct snd_card *card; struct snd_soc_dobj *dobj, *next_dobj; int pass; /* process the header types from end to start */ for (pass = SOC_TPLG_PASS_END; pass >= SOC_TPLG_PASS_START; pass--) { + card = (comp->card) ? comp->card->snd_card : NULL; + /* remove mixer controls */ - down_write(&card->controls_rwsem); + if (card) + down_write(&card->controls_rwsem); list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, list) { @@ -2660,7 +2663,8 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) break; } } - up_write(&card->controls_rwsem); + if (card) + up_write(&card->controls_rwsem); } /* let caller know if FW can be freed when no objects are left */ -- 2.17.1

2 2

[PATCH AUTOSEL 5.15 19/51] soundwire: qcom: adjust autoenumeration timeout
by Sasha Levin 07 Jun '22

07 Jun '22

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 74da272400b46f2e898f115d1b1cd60828766919 ] Currently timeout for autoenumeration during probe and bus reset is set to 2 secs which is really a big value. This can have an adverse effect on boot time if the slave device is not ready/reset. This was the case with wcd938x which was not reset yet but we spent 2 secs waiting in the soundwire controller probe. Reduce this time to 1/10 of Hz which should be good enough time to finish autoenumeration if any slaves are available on the bus. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20220506084705.18525-1-srinivas.kandagatla@linaro… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 0ef79d60e88e..f5955826b152 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -97,7 +97,7 @@ #define SWRM_SPECIAL_CMD_ID 0xF #define MAX_FREQ_NUM 1 -#define TIMEOUT_MS (2 * HZ) +#define TIMEOUT_MS 100 #define QCOM_SWRM_MAX_RD_LEN 0x1 #define QCOM_SDW_MAX_PORTS 14 #define DEFAULT_CLK_FREQ 9600000 -- 2.35.1

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[PATCH AUTOSEL 5.17 36/60] ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM
by Sasha Levin 07 Jun '22

07 Jun '22

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 832296804bc7171730884e78c761c29f6d258e13 ] The pin "Platform Clock" was only used by the Intel Byt CR platform. In the others, the error log will be informed. The patch will set the flag to avoid the pin "Platform Clock" manipulated by the other platforms. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20220516103055.20003-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e7a82565b905..f078463346e8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2097,12 +2097,14 @@ EXPORT_SYMBOL_GPL(rt5640_sel_asrc_clk_src); void rt5640_enable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); /* OVCD is unreliable when used with RCCLK as sysclk-source */ - snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_sync_unlocked(dapm); snd_soc_dapm_mutex_unlock(dapm); } @@ -2111,9 +2113,11 @@ EXPORT_SYMBOL_GPL(rt5640_enable_micbias1_for_ovcd); void rt5640_disable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); - snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); snd_soc_dapm_disable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_sync_unlocked(dapm); @@ -2538,6 +2542,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, rt5640->jd_gpio_irq_requested = true; } + if (jack_data && jack_data->use_platform_clock) + rt5640->use_platform_clock = jack_data->use_platform_clock; + ret = request_irq(rt5640->irq, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 9e49b9a0ccaa..505c93514051 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2155,11 +2155,13 @@ struct rt5640_priv { bool jd_inverted; unsigned int ovcd_th; unsigned int ovcd_sf; + bool use_platform_clock; }; struct rt5640_set_jack_data { int codec_irq_override; struct gpio_desc *jd_gpio; + bool use_platform_clock; }; int rt5640_dmic_enable(struct snd_soc_component *component, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 2ace32c03ec9..cc5703ecf523 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1179,12 +1179,14 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); + struct rt5640_set_jack_data *jack_data = &priv->jack_data; struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map = NULL; int num_routes = 0; int ret; card->dapm.idle_bias_off = true; + jack_data->use_platform_clock = true; /* Start with RC clk for jack-detect (we disable MCLK below) */ if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) -- 2.35.1

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[PATCH AUTOSEL 5.17 20/60] soundwire: qcom: adjust autoenumeration timeout
by Sasha Levin 07 Jun '22

07 Jun '22

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 74da272400b46f2e898f115d1b1cd60828766919 ] Currently timeout for autoenumeration during probe and bus reset is set to 2 secs which is really a big value. This can have an adverse effect on boot time if the slave device is not ready/reset. This was the case with wcd938x which was not reset yet but we spent 2 secs waiting in the soundwire controller probe. Reduce this time to 1/10 of Hz which should be good enough time to finish autoenumeration if any slaves are available on the bus. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20220506084705.18525-1-srinivas.kandagatla@linaro… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 54813417ef8e..249f1b69ec94 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -99,7 +99,7 @@ #define SWRM_SPECIAL_CMD_ID 0xF #define MAX_FREQ_NUM 1 -#define TIMEOUT_MS (2 * HZ) +#define TIMEOUT_MS 100 #define QCOM_SWRM_MAX_RD_LEN 0x1 #define QCOM_SDW_MAX_PORTS 14 #define DEFAULT_CLK_FREQ 9600000 -- 2.35.1

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