April 2023 - Alsa-devel - mailman.alsa-project.org

USB-Audio: ALC4080 on Gigabyte Z690 AORUS ULTRA
by GitHub pull_request - opened 16 Apr '23

16 Apr '23

alsa-project/alsa-ucm-conf pull request #304 was opened from MichailiK: This PR adds the USB ID of the ALC4080 found in the [Z690 AORUS ULTRA](https://www.gigabyte.com/Motherboard/Z690-AORUS-ULTRA-rev-1x/sp#sp), `0414:a011`. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/304 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/304.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALSA: arecord : silennce recorded as 0x80
by Muni Sekhar 15 Apr '23

15 Apr '23

Hi all, I recorded the silence on the default audio card by running "arecord -d 5 test.wav." Once it exited, I checked the contents by running "hexdump test.wav," and I see that silence is recorded as a sequence of 0x80 instead of 0x00. 0000000 4952 4646 0024 8000 4157 4556 6d66 2074 0000010 0010 0000 0001 0001 1f40 0000 1f40 0000 0000020 0001 0008 6164 6174 0000 8000 8080 8080 0000030 8080 8080 8080 8080 8080 8080 8080 8080 0000040 8080 8080 8080 8080 8080 8080 8080 8080 ... What might be the reason for silence being recorded as 0x80 instead of 0x0? -- Thanks, Sekhar

2 3

arecord: duration incorrect when ALSA chooses different sample rate
by GitHub pull_request - opened 15 Apr '23

15 Apr '23

alsa-project/alsa-utils pull request #200 was opened from iam-TJ: I encountered this issue whilst researching a bug in capture handling in ALSA/PulseAudio/PortAudio/Audacity. When recording for a fixed duration the duration is measured by the quantity of samples, not clock time. In `capture()` a call to `calc_count()` occurs before `set_params()`, but in that latter function the actual sample rate used may differ from the requested rate , but the count is not recalculated. E.g: ``` $ time arecord -D hw:1,0 -d 2 -c 2 -f S16_LE -r 384000 --dump-hw-params /tmp/test.11.wav Recording WAVE '/tmp/test.11.wav' : Signed 16 bit Little Endian, Rate 384000 Hz, Stereo HW Params of device "hw:1,0": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S32_LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [32 64] CHANNELS: 2 RATE: [44100 192000] PERIOD_TIME: (83 11888617) PERIOD_SIZE: [16 524288] PERIOD_BYTES: [128 2097152] PERIODS: [2 32] BUFFER_TIME: (166 23777234) BUFFER_SIZE: [32 1048576] BUFFER_BYTES: [128 4194304] TICK_TIME: ALL -------------------- Warning: rate is not accurate (requested = 384000Hz, got = 192000Hz) please, try the plug plugin real 0m4.009s user 0m0.004s sys 0m0.007s ``` In this example, because the actual sample rate was 1/2 of that requested, count is double what it should be and therefore the recording continues for 4 seconds instead of the requested 2 seconds. Moving the call to `set_params()` to before the count logic solves it: ``` $ time aplay/aplay -C -D hw:1,0 -d 2 -c 2 -f S16_LE -r 384000 --dump-hw-params /tmp/test.11.wav HW Params of device "hw:1,0": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S32_LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [32 64] CHANNELS: 2 RATE: [44100 192000] PERIOD_TIME: (83 11888617) PERIOD_SIZE: [16 524288] PERIOD_BYTES: [128 2097152] PERIODS: [2 32] BUFFER_TIME: (166 23777234) BUFFER_SIZE: [32 1048576] BUFFER_BYTES: [128 4194304] TICK_TIME: ALL -------------------- Warning: rate is not accurate (requested = 384000Hz, got = 192000Hz) please, try the plug plugin Recording WAVE '/tmp/test.11.wav' : Signed 16 bit Little Endian, Rate 192000 Hz, Stereo real 0m2.009s user 0m0.001s sys 0m0.008s ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/200 Patch URL : https://github.com/alsa-project/alsa-utils/pull/200.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

ALSA: struct snd_pcm_hardware.formats
by Muni Sekhar 14 Apr '23

14 Apr '23

Hi all, struct snd_pcm_hardware.formats (SNDRV_PCM_FMTBIT_S8, SNDRV_PCM_FMTBIT_S16_LE, SNDRV_PCM_FMTBIT_U16_LE, SNDRV_PCM_FMTBIT_S16_BE, etc.), are these sound card hardware capabilities or something that needs to be taken care of on the software (driver or userspace) side? -- Thanks, Sekhar

1 0

[PATCH 1/2] ALSA: pcm: rewrite snd_pcm_playback_silence()
by Oswald Buddenhagen 14 Apr '23

14 Apr '23

The auto-silencer supports two modes: "thresholded" to fill up "just enough", and "top-up" to fill up "as much as possible". The two modes used rather distinct code paths, which this patch unifies. The only remaining distinction is how much we actually want to fill. This fixes a bug in thresholded mode, where we failed to use new_hw_ptr, resulting in under-fill. Top-up mode is now more well-behaved and much easier to understand in corner cases. This also updates comments in the proximity of silencing-related data structures. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- .../kernel-api/writing-an-alsa-driver.rst | 14 +-- include/sound/pcm.h | 14 +-- include/uapi/sound/asound.h | 9 +- sound/core/pcm_lib.c | 86 ++++++++----------- sound/core/pcm_local.h | 3 +- sound/core/pcm_native.c | 6 +- 6 files changed, 62 insertions(+), 70 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index a368529e8ed3..f2834a016473 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -1577,14 +1577,16 @@ are the contents of this file: unsigned int period_step; unsigned int sleep_min; /* min ticks to sleep */ snd_pcm_uframes_t start_threshold; - snd_pcm_uframes_t stop_threshold; - snd_pcm_uframes_t silence_threshold; /* Silence filling happens when - noise is nearest than this */ - snd_pcm_uframes_t silence_size; /* Silence filling size */ + // The following two thresholds alleviate playback buffer underruns; when + // hw_avail drops below the threshold, the respective action is triggered: + snd_pcm_uframes_t stop_threshold; /* stop playback */ + snd_pcm_uframes_t silence_threshold; /* pre-fill buffer with silence */ + snd_pcm_uframes_t silence_size; /* msx size of silence pre-fill */ snd_pcm_uframes_t boundary; /* pointers wrap point */ - snd_pcm_uframes_t silenced_start; - snd_pcm_uframes_t silenced_size; + // internal data of auto-silencer + snd_pcm_uframes_t silence_start; /* starting pointer to silence area */ + snd_pcm_uframes_t silence_filled; /* size filled with silence */ snd_pcm_sync_id_t sync; /* hardware synchronization ID */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 27040b472a4f..f20400bb7032 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -378,18 +378,18 @@ struct snd_pcm_runtime { unsigned int rate_den; unsigned int no_period_wakeup: 1; - /* -- SW params -- */ - int tstamp_mode; /* mmap timestamp is updated */ + /* -- SW params; see struct snd_pcm_sw_params for comments -- */ + int tstamp_mode; unsigned int period_step; snd_pcm_uframes_t start_threshold; snd_pcm_uframes_t stop_threshold; - snd_pcm_uframes_t silence_threshold; /* Silence filling happens when - noise is nearest than this */ - snd_pcm_uframes_t silence_size; /* Silence filling size */ - snd_pcm_uframes_t boundary; /* pointers wrap point */ + snd_pcm_uframes_t silence_threshold; + snd_pcm_uframes_t silence_size; + snd_pcm_uframes_t boundary; + // internal data of auto-silencer snd_pcm_uframes_t silence_start; /* starting pointer to silence area */ - snd_pcm_uframes_t silence_filled; /* size filled with silence */ + snd_pcm_uframes_t silence_filled; /* already filled part of silence area */ union snd_pcm_sync_id sync; /* hardware synchronization ID */ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index de6810e94abe..50aa1d98873f 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -429,9 +429,12 @@ struct snd_pcm_sw_params { snd_pcm_uframes_t avail_min; /* min avail frames for wakeup */ snd_pcm_uframes_t xfer_align; /* obsolete: xfer size need to be a multiple */ snd_pcm_uframes_t start_threshold; /* min hw_avail frames for automatic start */ - snd_pcm_uframes_t stop_threshold; /* min avail frames for automatic stop */ - snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */ - snd_pcm_uframes_t silence_size; /* silence block size */ + // The following two thresholds alleviate playback buffer underruns; when + // hw_avail drops below the threshold, the respective action is triggered: + snd_pcm_uframes_t stop_threshold; /* stop playback */ + snd_pcm_uframes_t silence_threshold; /* pre-fill buffer with silence */ + snd_pcm_uframes_t silence_size; /* max size of silence pre-fill; when >= boundary, + * fill played area with silence immediately */ snd_pcm_uframes_t boundary; /* pointers wrap point */ unsigned int proto; /* protocol version */ unsigned int tstamp_type; /* timestamp type (req. proto >= 2.0.12) */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 02fd65993e7e..b074c5b926db 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -42,70 +42,58 @@ static int fill_silence_frames(struct snd_pcm_substream *substream, * * when runtime->silence_size >= runtime->boundary - fill processed area with silence immediately */ -void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr) +void snd_pcm_playback_silence(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); + snd_pcm_sframes_t hw_avail, added, noise_dist; snd_pcm_uframes_t frames, ofs, transfer; int err; + // This will "legitimately" turn negative on underrun, and will be mangled + // into a huge number by the boundary crossing handling. The initial state + // might also be not quite sane. The code below MUST account for these cases. + hw_avail = appl_ptr - runtime->status->hw_ptr; + if (hw_avail < 0) + hw_avail += runtime->boundary; + else if ((snd_pcm_uframes_t) hw_avail >= runtime->boundary) + hw_avail -= runtime->boundary; + + added = appl_ptr - runtime->silence_start; + if (added) { + if (added < 0) + added += runtime->boundary; + if ((snd_pcm_uframes_t)added < runtime->silence_filled) + runtime->silence_filled -= added; + else + runtime->silence_filled = 0; + runtime->silence_start = appl_ptr; + } + + noise_dist = hw_avail + runtime->silence_filled; if (runtime->silence_size < runtime->boundary) { - snd_pcm_sframes_t noise_dist, n; - snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); - if (runtime->silence_start != appl_ptr) { - n = appl_ptr - runtime->silence_start; - if (n < 0) - n += runtime->boundary; - if ((snd_pcm_uframes_t)n < runtime->silence_filled) - runtime->silence_filled -= n; - else - runtime->silence_filled = 0; - runtime->silence_start = appl_ptr; - } - if (runtime->silence_filled >= runtime->buffer_size) - return; - noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled; - if (noise_dist >= (snd_pcm_sframes_t) runtime->silence_threshold) - return; frames = runtime->silence_threshold - noise_dist; + if ((snd_pcm_sframes_t) frames <= 0) + return; if (frames > runtime->silence_size) frames = runtime->silence_size; } else { - if (new_hw_ptr == ULONG_MAX) { /* initialization */ - snd_pcm_sframes_t avail = snd_pcm_playback_hw_avail(runtime); - if (avail > runtime->buffer_size) - avail = runtime->buffer_size; - runtime->silence_filled = avail > 0 ? avail : 0; - runtime->silence_start = (runtime->status->hw_ptr + - runtime->silence_filled) % - runtime->boundary; - } else { - ofs = runtime->status->hw_ptr; - frames = new_hw_ptr - ofs; - if ((snd_pcm_sframes_t)frames < 0) - frames += runtime->boundary; - runtime->silence_filled -= frames; - if ((snd_pcm_sframes_t)runtime->silence_filled < 0) { - runtime->silence_filled = 0; - runtime->silence_start = new_hw_ptr; - } else { - runtime->silence_start = ofs; - } - } - frames = runtime->buffer_size - runtime->silence_filled; + frames = runtime->buffer_size - noise_dist; + if ((snd_pcm_sframes_t) frames <= 0) + return; } + if (snd_BUG_ON(frames > runtime->buffer_size)) return; - if (frames == 0) - return; - ofs = runtime->silence_start % runtime->buffer_size; - while (frames > 0) { + ofs = (runtime->silence_start + runtime->silence_filled) % runtime->buffer_size; + do { transfer = ofs + frames > runtime->buffer_size ? runtime->buffer_size - ofs : frames; err = fill_silence_frames(substream, ofs, transfer); snd_BUG_ON(err < 0); runtime->silence_filled += transfer; frames -= transfer; ofs = 0; - } + } while (frames > 0); snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); } @@ -439,10 +427,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, return 0; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream, new_hw_ptr); - if (in_interrupt) { delta = new_hw_ptr - runtime->hw_ptr_interrupt; if (delta < 0) @@ -460,6 +444,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_wrap += runtime->boundary; } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + runtime->silence_size > 0) + snd_pcm_playback_silence(substream); + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return snd_pcm_update_state(substream, runtime); diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index ecb21697ae3a..42fe3a4e9154 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -29,8 +29,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); -void snd_pcm_playback_silence(struct snd_pcm_substream *substream, - snd_pcm_uframes_t new_hw_ptr); +void snd_pcm_playback_silence(struct snd_pcm_substream *substream); static inline snd_pcm_uframes_t snd_pcm_avail(struct snd_pcm_substream *substream) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 331380c2438b..0e3e7997dc58 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -958,7 +958,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (snd_pcm_running(substream)) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); @@ -1455,7 +1455,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, __snd_pcm_set_state(runtime, state); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } @@ -1916,7 +1916,7 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *substream, runtime->control->appl_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); snd_pcm_stream_unlock_irq(substream); } -- 2.40.0.152.g15d061e6df

3 28

[PATCH] ALSA: hda/hdmi: disable KAE for Intel DG2
by Kai Vehmanen 14 Apr '23

14 Apr '23

Use of keep-alive (KAE) has resulted in loss of audio on some A750/770 cards as the transition from keep-alive to stream playback is not working as expected. As there is limited benefit of the new KAE mode on discrete cards, revert back to older silent-stream implementation on these systems. Cc: stable(a)vger.kernel.org Fixes: 15175a4f2bbb ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2") Link: https://gitlab.freedesktop.org/drm/intel/-/issues/8307 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4ffa3a59f419..5c6980394dce 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4604,7 +4604,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862818, "Raptorlake HDMI", patch_i915_tgl_hdmi), -HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), 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), base-commit: be6247640eea9d9b0ff15607fab7a12f40974985 -- 2.40.0

2 1

[PATCH v2] ASoC: amd: yc: Add ThinkBook 14 G5+ ARP to quirks list for acp6x
by Baishan Jiang 13 Apr '23

13 Apr '23

ThinkBook 14 G5+ ARP uses Ryzen 7735H processor, and has the same microphone problem as ThinkBook 14 G4+ ARA. Adding 21HY to acp6x quirks table enables microphone for ThinkBook 14 G5+ ARP. Signed-off-by: Baishan Jiang <bjiang400(a)outlook.com> --- V1 -> V2: Fixed signed-off-by format --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 0acdf0156f07..62d8a8e060d2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -178,6 +178,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21EN"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21HY"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.40.0

4 3

[PATCH] ASoC: da7218: Use devm_clk_get_optional()
by Christophe JAILLET 13 Apr '23

13 Apr '23

Use devm_clk_get_optional() instead of hand writing it. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/da7218.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 91372909d184..d9c28e701613 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2893,14 +2893,10 @@ static int da7218_probe(struct snd_soc_component *component) da7218_handle_pdata(component); /* Check if MCLK provided, if not the clock is NULL */ - da7218->mclk = devm_clk_get(component->dev, "mclk"); + da7218->mclk = devm_clk_get_optional(component->dev, "mclk"); if (IS_ERR(da7218->mclk)) { - if (PTR_ERR(da7218->mclk) != -ENOENT) { - ret = PTR_ERR(da7218->mclk); - goto err_disable_reg; - } else { - da7218->mclk = NULL; - } + ret = PTR_ERR(da7218->mclk); + goto err_disable_reg; } /* Default PC to free-running */ -- 2.34.1

2 1

[PATCH] ASoC: da7219: Improve the relability of AAD IRQ process
by David Rau 13 Apr '23

13 Apr '23

- Configure the default gound switch delay time before enabling IRQ to avoid the unexpected delay time is set up - Apply DA7219 AAD own work queue to handle AAD events - Replace msleep with queue_delayed_work to have better relability This commit improves the control of ground switches in AAD IRQ Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- sound/soc/codecs/da7219-aad.c | 60 +++++++++++++++++------------------ sound/soc/codecs/da7219-aad.h | 5 ++- 2 files changed, 31 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index e3d398b8f54e..993a0d00bc48 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -342,36 +342,17 @@ static void da7219_aad_hptest_work(struct work_struct *work) static void da7219_aad_jack_det_work(struct work_struct *work) { struct da7219_aad_priv *da7219_aad = - container_of(work, struct da7219_aad_priv, jack_det_work); + container_of(work, struct da7219_aad_priv, jack_det_work.work); struct snd_soc_component *component = da7219_aad->component; - u8 srm_st; - mutex_lock(&da7219_aad->jack_det_mutex); - - srm_st = snd_soc_component_read(component, DA7219_PLL_SRM_STS) & DA7219_PLL_SRM_STS_MCLK; - msleep(da7219_aad->gnd_switch_delay * ((srm_st == 0x0) ? 2 : 1) - 4); /* Enable ground switch */ snd_soc_component_update_bits(component, 0xFB, 0x01, 0x01); - - mutex_unlock(&da7219_aad->jack_det_mutex); } - /* * IRQ */ -static irqreturn_t da7219_aad_pre_irq_thread(int irq, void *data) -{ - - struct da7219_aad_priv *da7219_aad = data; - - if (!da7219_aad->jack_inserted) - schedule_work(&da7219_aad->jack_det_work); - - return IRQ_WAKE_THREAD; -} - static irqreturn_t da7219_aad_irq_thread(int irq, void *data) { struct da7219_aad_priv *da7219_aad = data; @@ -392,6 +373,18 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) /* Read status register for jack insertion & type status */ statusa = snd_soc_component_read(component, DA7219_ACCDET_STATUS_A); + if (events[DA7219_AAD_IRQ_REG_A] & DA7219_E_JACK_INSERTED_MASK) { + u8 srm_st; + int delay = 0; + + srm_st = snd_soc_component_read(component, + DA7219_PLL_SRM_STS) & DA7219_PLL_SRM_STS_MCLK; + delay = (da7219_aad->gnd_switch_delay * ((srm_st == 0x0) ? 2 : 1) - 2); + queue_delayed_work(da7219_aad->aad_wq, + &da7219_aad->jack_det_work, + msecs_to_jiffies(delay)); + } + /* Clear events */ regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, events, DA7219_AAD_IRQ_REG_MAX); @@ -400,9 +393,6 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B], statusa); - if (!da7219_aad->jack_inserted) - cancel_work_sync(&da7219_aad->jack_det_work); - if (statusa & DA7219_JACK_INSERTION_STS_MASK) { /* Jack Insertion */ if (events[DA7219_AAD_IRQ_REG_A] & @@ -430,9 +420,9 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) if (statusa & DA7219_JACK_TYPE_STS_MASK) { report |= SND_JACK_HEADSET; mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; - schedule_work(&da7219_aad->btn_det_work); + queue_work(da7219_aad->aad_wq, &da7219_aad->btn_det_work); } else { - schedule_work(&da7219_aad->hptest_work); + queue_work(da7219_aad->aad_wq, &da7219_aad->hptest_work); } } @@ -465,6 +455,7 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) da7219_aad->jack_inserted = false; /* Cancel any pending work */ + cancel_delayed_work_sync(&da7219_aad->jack_det_work); cancel_work_sync(&da7219_aad->btn_det_work); cancel_work_sync(&da7219_aad->hptest_work); @@ -964,13 +955,19 @@ int da7219_aad_init(struct snd_soc_component *component) snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1, DA7219_BUTTON_CONFIG_MASK, 0); + da7219_aad_handle_gnd_switch_time(component); + + da7219_aad->aad_wq = create_singlethread_workqueue("da7219-aad"); + if (!da7219_aad->aad_wq) { + dev_err(component->dev, "Failed to create aad workqueue\n"); + return -ENOMEM; + } + + INIT_DELAYED_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work); INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work); INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); - INIT_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work); - - mutex_init(&da7219_aad->jack_det_mutex); - ret = request_threaded_irq(da7219_aad->irq, da7219_aad_pre_irq_thread, + ret = request_threaded_irq(da7219_aad->irq, NULL, da7219_aad_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "da7219-aad", da7219_aad); @@ -984,8 +981,6 @@ int da7219_aad_init(struct snd_soc_component *component) regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, &mask, DA7219_AAD_IRQ_REG_MAX); - da7219_aad_handle_gnd_switch_time(component); - return 0; } @@ -1002,8 +997,10 @@ void da7219_aad_exit(struct snd_soc_component *component) free_irq(da7219_aad->irq, da7219_aad); + cancel_delayed_work_sync(&da7219_aad->jack_det_work); cancel_work_sync(&da7219_aad->btn_det_work); cancel_work_sync(&da7219_aad->hptest_work); + destroy_workqueue(da7219_aad->aad_wq); } /* @@ -1031,4 +1028,5 @@ int da7219_aad_probe(struct i2c_client *i2c) MODULE_DESCRIPTION("ASoC DA7219 AAD Driver"); MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com>"); +MODULE_AUTHOR("David Rau <David.Rau.opensource(a)dm.renesas.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h index be87ee47edde..fbfbf3e67918 100644 --- a/sound/soc/codecs/da7219-aad.h +++ b/sound/soc/codecs/da7219-aad.h @@ -197,9 +197,8 @@ struct da7219_aad_priv { struct work_struct btn_det_work; struct work_struct hptest_work; - struct work_struct jack_det_work; - - struct mutex jack_det_mutex; + struct delayed_work jack_det_work; + struct workqueue_struct *aad_wq; struct snd_soc_jack *jack; bool micbias_resume_enable; -- 2.17.1

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[PATCH] ASoC: SOF: topology: Move a variable assignment behind condition checks in sof_dai_load()
by Markus Elfring 13 Apr '23

13 Apr '23

Date: Thu, 13 Apr 2023 13:56:44 +0200 The address of a data structure member was determined before a corresponding null pointer check in the implementation of the function “sof_dai_load”. Thus avoid the risk for undefined behaviour by moving the assignment for the local variable “private” behind some condition checks. This issue was detected by using the Coccinelle software. Fixes: c5232c0171428f005a3204e1c264231fb5999b28 ("ASoC: SOF: topology: parse and store d0i3_compatible flag") Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/sof/topology.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index d3d536b0a8f5..3fffe3826160 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1680,7 +1680,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); const struct sof_ipc_pcm_ops *ipc_pcm_ops = sof_ipc_get_ops(sdev, pcm); struct snd_soc_tplg_stream_caps *caps; - struct snd_soc_tplg_private *private = &pcm->priv; + struct snd_soc_tplg_private *private; struct snd_sof_pcm *spcm; int stream; int ret; @@ -1716,6 +1716,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, dai_drv->dobj.private = spcm; list_add(&spcm->list, &sdev->pcm_list); + private = &pcm->priv; ret = sof_parse_tokens(scomp, spcm, stream_tokens, ARRAY_SIZE(stream_tokens), private->array, le32_to_cpu(private->size)); -- 2.40.0

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