August 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/2] Add reset-gpios handling for max98927
by Alejandro Tafalla 29 Aug '21

29 Aug '21

The max98927 codec on some devices (i.e. Xiaomi Mi A2 Lite phone) require hardware-resetting the codec by driving a reset-gpio. This series add support for it through an optional reset-gpios property. Alejandro Tafalla (2): ASoC: max98927: Handle reset gpio when probing i2c dt-bindings: sound: max98927: Add reset-gpios optional property .../devicetree/bindings/sound/max9892x.txt | 3 +++ sound/soc/codecs/max98927.c | 16 ++++++++++++++++ sound/soc/codecs/max98927.h | 1 + 3 files changed, 20 insertions(+) -- 2.32.0

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Question: different install prefixes for libalsa and alsa-plugins?
by GitHub issues - opened 29 Aug '21

29 Aug '21

alsa-project/alsa-lib issue #175 was opened from daravi: Hello, I use Conan to manage my dependencies and wanted to add a recipe for `alsa-plugins`. However it seems that `ALSA_CONFIG_DIR` can only be set to a single path: https://github.com/conan-io/conan-center-index/pull/7098#discussion_r697920… So I was wondering if there is any other way I could have alsa and plugins installed in separate directories? (so that they can each have their own config directory) Issue URL : https://github.com/alsa-project/alsa-lib/issues/175 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: usb-audio: Add lowlatency module option
by Takashi Iwai 29 Aug '21

29 Aug '21

For making user to switch back to the old playback mode, this patch adds a new module option 'lowlatency' to snd-usb-audio driver. When user face a regression due to the recent low-latency playback support, they can test easily by passing lowlatency=0 option without rebuilding the kernel. Fixes: 307cc9baac5c ("ALSA: usb-audio: Reduce latency at playback start, take#2") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 4 ++++ sound/usb/pcm.c | 3 ++- sound/usb/usbaudio.h | 1 + 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index cf8f3953f78f..fd570a42f043 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -68,6 +68,7 @@ static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; +static bool lowlatency = true; static char *quirk_alias[SNDRV_CARDS]; static char *delayed_register[SNDRV_CARDS]; static bool implicit_fb[SNDRV_CARDS]; @@ -93,6 +94,8 @@ MODULE_PARM_DESC(ignore_ctl_error, "Ignore errors from USB controller for mixer interfaces."); module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); +module_param(lowlatency, bool, 0444); +MODULE_PARM_DESC(lowlatency, "Enable low latency playback (default: yes)."); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); module_param_array(delayed_register, charp, NULL, 0444); @@ -623,6 +626,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, chip->setup = device_setup[idx]; chip->generic_implicit_fb = implicit_fb[idx]; chip->autoclock = autoclock; + chip->lowlatency = lowlatency; atomic_set(&chip->active, 1); /* avoid autopm during probing */ atomic_set(&chip->usage_count, 0); atomic_set(&chip->shutdown, 0); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f5cbf61ac366..5dc9266180e3 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -617,7 +617,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) /* check whether early start is needed for playback stream */ subs->early_playback_start = subs->direction == SNDRV_PCM_STREAM_PLAYBACK && - subs->data_endpoint->nominal_queue_size >= subs->buffer_bytes; + (!chip->lowlatency || + (subs->data_endpoint->nominal_queue_size >= subs->buffer_bytes)); if (subs->early_playback_start) ret = start_endpoints(subs); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 94261d19cceb..167834133b9b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -55,6 +55,7 @@ struct snd_usb_audio { bool generic_implicit_fb; /* from the 'implicit_fb' module param */ bool autoclock; /* from the 'autoclock' module param */ + bool lowlatency; /* from the 'lowlatency' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ struct media_device *media_dev; struct media_intf_devnode *ctl_intf_media_devnode; -- 2.26.2

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Static build generates broken symlinks
by GitHub issues - opened 28 Aug '21

28 Aug '21

alsa-project/alsa-plugins issue #34 was opened from daravi: Hello, This snippet seems to generate some symlinks: ``` install-exec-hook: rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_speexrate_*.so $(LN_S) libasound_module_rate_speexrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_speexrate_best.so $(LN_S) libasound_module_rate_speexrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_speexrate_medium.so ``` But it assumes we are doing a shared build. Which results in broken symlinks in a static build. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/34 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH 0/2] ctl/rawmidi/hwdep: fix trivial bugs
by GitHub pull_request - opened 28 Aug '21

28 Aug '21

alsa-project/alsa-gobject pull request #62 was opened from takaswie: Current implementation of libalsactl0, libalsarawmidi0, and libalsahwdep0 includes trivial bugs such as usage of uninitialized local variable. This patchset fixes the bugs. ``` Takashi Sakamoto (2): ctl/rawmidi/hwdep: fix compiler warning due to uninitialized variables seq: use safer way to copy strings src/ctl/query.c | 4 ++-- src/hwdep/query.c | 4 ++-- src/rawmidi/query.c | 4 ++-- src/seq/client-info.c | 2 +- src/seq/port-info.c | 2 +- src/seq/query.c | 2 +- src/seq/queue-info.c | 2 +- 7 files changed, 10 insertions(+), 10 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/62 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/62.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH 1/2] ALSA: hda/cs8409: Ensure Type Detection is only run on startup when necessary
by Vitaly Rodionov 27 Aug '21

27 Aug '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Type Detection should only be run after init and when the controls have been built. There is no need to run it multiple times. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409.c | 34 ++++++++++++++++++++++++++++------ sound/pci/hda/patch_cs8409.h | 2 ++ 2 files changed, 30 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 9db16b6292f4..805441b7bf86 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -784,6 +784,8 @@ static void cs42l42_suspend(struct sub_codec *cs42l42) cs8409_i2c_write(cs42l42, 0x1102, 0x9C); cs42l42->suspended = 1; cs42l42->last_page = 0; + cs42l42->hp_jack_in = 0; + cs42l42->mic_jack_in = 0; /* Put CS42L42 into Reset */ gpio_data = snd_hda_codec_read(codec, CS8409_PIN_AFG, 0, AC_VERB_GET_GPIO_DATA, 0); @@ -852,6 +854,8 @@ static int cs8409_cs42l42_suspend(struct hda_codec *codec) struct cs8409_spec *spec = codec->spec; int i; + spec->init_done = 0; + cs8409_enable_ur(codec, 0); for (i = 0; i < spec->num_scodecs; i++) @@ -1025,14 +1029,21 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, break; case HDA_FIXUP_ACT_INIT: cs8409_cs42l42_hw_init(codec); - fallthrough; + spec->init_done = 1; + if (spec->init_done && spec->build_ctrl_done + && !spec->scodecs[CS8409_CODEC0]->hp_jack_in) + cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); + break; case HDA_FIXUP_ACT_BUILD: + spec->build_ctrl_done = 1; /* Run jack auto detect first time on boot * after controls have been added, to check if jack has * been already plugged in. * Run immediately after init. */ - cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); + if (spec->init_done && spec->build_ctrl_done + && !spec->scodecs[CS8409_CODEC0]->hp_jack_in) + cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); break; default: break; @@ -1223,16 +1234,27 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac break; case HDA_FIXUP_ACT_INIT: dolphin_hw_init(codec); - fallthrough; + spec->init_done = 1; + if (spec->init_done && spec->build_ctrl_done) { + for (i = 0; i < spec->num_scodecs; i++) { + if (!spec->scodecs[i]->hp_jack_in) + cs42l42_run_jack_detect(spec->scodecs[i]); + } + } + break; case HDA_FIXUP_ACT_BUILD: + spec->build_ctrl_done = 1; /* Run jack auto detect first time on boot * after controls have been added, to check if jack has * been already plugged in. * Run immediately after init. */ - for (i = 0; i < spec->num_scodecs; i++) - cs42l42_run_jack_detect(spec->scodecs[i]); - + if (spec->init_done && spec->build_ctrl_done) { + for (i = 0; i < spec->num_scodecs; i++) { + if (!spec->scodecs[i]->hp_jack_in) + cs42l42_run_jack_detect(spec->scodecs[i]); + } + } break; default: break; diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index 207315ad5bf6..ade2b838590c 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -333,6 +333,8 @@ struct cs8409_spec { unsigned int playback_started:1; unsigned int capture_started:1; + unsigned int init_done:1; + unsigned int build_ctrl_done:1; /* verb exec op override */ int (*exec_verb)(struct hdac_device *dev, unsigned int cmd, unsigned int flags, -- 2.25.1

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[PATCH v2] ALSA: pcm: fix divide error in snd_pcm_lib_ioctl
by Zubin Mithra 27 Aug '21

27 Aug '21

Syzkaller reported a divide error in snd_pcm_lib_ioctl. fifo_size is of type snd_pcm_uframes_t(unsigned long). If frame_size is 0x100000000, the error occurs. Fixes: a9960e6a293e ("ALSA: pcm: fix fifo_size frame calculation") Signed-off-by: Zubin Mithra <zsm(a)chromium.org> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d5883432085..a144a3f68e9e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1746,7 +1746,7 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, channels = params_channels(params); frame_size = snd_pcm_format_size(format, channels); if (frame_size > 0) - params->fifo_size /= (unsigned)frame_size; + params->fifo_size /= frame_size; } return 0; } -- 2.33.0.259.gc128427fd7-goog

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[PATCH] ALSA: usb-audio: Work around for XRUN with low latency playback
by Takashi Iwai 27 Aug '21

27 Aug '21

The recent change for low latency playback works in most of test cases but it turned out still to hit errors on some use cases, most notably with JACK with small buffer sizes. This is because USB-audio driver fills up and submits full URBs at the beginning, while the URBs would return immediately and try to fill more -- that can easily trigger XRUN. It was more or less expected, but in the small buffer size, the problem became pretty obvious. Fixing this behavior properly would require the change of the fundamental driver design, so it's no trivial task, unfortunately. Instead, here we work around the problem just by switching back to the old method when the given configuration is too fragile with the low latency stream handling. As a threshold, we calculate the total buffer bytes in all plus one URBs, and check whether it's beyond the PCM buffer bytes. The one extra URB is needed because XRUN happens at the next submission after the first round. Fixes: 307cc9baac5c ("ALSA: usb-audio: Reduce latency at playback start, take#2") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.h | 2 ++ sound/usb/endpoint.c | 4 ++++ sound/usb/pcm.c | 13 +++++++++++-- 3 files changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 6c0a052a28f9..5b19901f305a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -94,6 +94,7 @@ struct snd_usb_endpoint { struct list_head ready_playback_urbs; /* playback URB FIFO for implicit fb */ unsigned int nurbs; /* # urbs */ + unsigned int nominal_queue_size; /* total buffer sizes in URBs */ unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ char *syncbuf; /* sync buffer for all sync URBs */ @@ -187,6 +188,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + bool early_playback_start; /* early start needed for playback? */ struct media_ctl *media_ctl; }; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8b1bec51c806..bf26c04cf471 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1126,6 +1126,10 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep) INIT_LIST_HEAD(&u->ready_list); } + /* total buffer bytes of all URBs plus the next queue; + * referred in pcm.c + */ + ep->nominal_queue_size = maxsize * urb_packs * (ep->nurbs + 1); return 0; out_of_memory: diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4e5031a68064..f5cbf61ac366 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -614,6 +614,14 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->period_elapsed_pending = 0; runtime->delay = 0; + /* check whether early start is needed for playback stream */ + subs->early_playback_start = + subs->direction == SNDRV_PCM_STREAM_PLAYBACK && + subs->data_endpoint->nominal_queue_size >= subs->buffer_bytes; + + if (subs->early_playback_start) + ret = start_endpoints(subs); + unlock: snd_usb_unlock_shutdown(chip); return ret; @@ -1394,7 +1402,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, subs->trigger_tstamp_pending_update = false; } - if (period_elapsed && !subs->running) { + if (period_elapsed && !subs->running && !subs->early_playback_start) { subs->period_elapsed_pending = 1; period_elapsed = 0; } @@ -1448,7 +1456,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea prepare_playback_urb, retire_playback_urb, subs); - if (cmd == SNDRV_PCM_TRIGGER_START) { + if (!subs->early_playback_start && + cmd == SNDRV_PCM_TRIGGER_START) { err = start_endpoints(subs); if (err < 0) { snd_usb_endpoint_set_callback(subs->data_endpoint, -- 2.26.2

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Adjust appl_ptr for multi plugin on rewind/forward operations
by GitHub pull_request - edited 27 Aug '21

27 Aug '21

alsa-project/alsa-lib pull request #161 was edited from sava-j: * When doing rewind or forward operations, application pointer needs to be properly adjusted, in the same way when doing commit/write operations. * Introduce helper functions for manipulating the application pointer. Request URL : https://github.com/alsa-project/alsa-lib/pull/161 Patch URL : https://github.com/alsa-project/alsa-lib/pull/161.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: pcm: fix divide error in snd_pcm_lib_ioctl
by Zubin Mithra 27 Aug '21

27 Aug '21

Syzkaller reported a divide error in snd_pcm_lib_ioctl. fifo_size is of type snd_pcm_uframes_t(unsigned long). If frame_size is 0x100000000, the error occurs. Fixes: a9960e6a293e ("ALSA: pcm: fix fifo_size frame calculation") Signed-off-by: Zubin Mithra <zsm(a)chromium.org> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d5883432085..e41b4e01aa37 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1746,7 +1746,7 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, channels = params_channels(params); frame_size = snd_pcm_format_size(format, channels); if (frame_size > 0) - params->fifo_size /= (unsigned)frame_size; + params->fifo_size /= (unsigned long)frame_size; } return 0; } -- 2.33.0.259.gc128427fd7-goog

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