December 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] Don't use 16 bits signed integers for sample rate conversion.
by GitHub issues - opened 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was opened from crocket: Sample rate conversion should be done with the highest resolution possible. Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] [PATCH] ALSA: hda: Modify stream stripe mask only when needed
by Takashi Iwai 15 Dec '19

15 Dec '19

The recent in HD-audio stream management for changing the stripe control seems causing a regression on some platforms. The stripe control is currently used only by HDMI codec, and applying the stripe mask unconditionally may lead to scratchy and static noises as seen on some MacBooks. For addressing the regression, this patch changes the stream management code to apply the stripe mask conditionally only when the codec driver requested. Fixes: 9b6f7e7a296e ("ALSA: hda: program stripe bits for controller") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204477 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_stream.c | 19 ++++++++++++------- sound/pci/hda/patch_hdmi.c | 5 +++++ 3 files changed, 18 insertions(+), 7 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b260c5fd2337..e05b95e83d5a 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -493,6 +493,7 @@ struct hdac_stream { bool prepared:1; bool no_period_wakeup:1; bool locked:1; + bool stripe:1; /* apply stripe control */ /* timestamp */ unsigned long start_wallclk; /* start + minimum wallclk */ diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index d8fe7ff0cd58..f9707fb05efe 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -96,12 +96,14 @@ void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start) 1 << azx_dev->index, 1 << azx_dev->index); /* set stripe control */ - if (azx_dev->substream) - stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream); - else - stripe_ctl = 0; - snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, - stripe_ctl); + if (azx_dev->stripe) { + if (azx_dev->substream) + stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream); + else + stripe_ctl = 0; + snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, + stripe_ctl); + } /* set DMA start and interrupt mask */ snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_DMA_START | SD_INT_MASK); @@ -118,7 +120,10 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_DMA_START | SD_INT_MASK, 0); snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ - snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); + if (azx_dev->stripe) { + snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); + azx_dev->stripe = 0; + } azx_dev->running = false; } EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0032bba8cc9d..ed4e98a1057a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -32,6 +32,7 @@ #include <sound/hda_codec.h> #include "hda_local.h" #include "hda_jack.h" +#include "hda_controller.h" static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); @@ -1249,6 +1250,10 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_pin->cvt_nid = per_cvt->cvt_nid; hinfo->nid = per_cvt->cvt_nid; + /* flip stripe flag for the assigned stream if supported */ + if (get_wcaps(codec, per_cvt->cvt_nid) & AC_WCAP_STRIPE) + azx_stream(get_azx_dev(substream))->stripe = 1; + snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, -- 2.16.4

2 4

[alsa-devel] Code Example for Playing Sound in the Background
by GitHub issues - edited 14 Dec '19

14 Dec '19

alsa-project/alsa-lib issue #17 was edited from leimao: Hello, I wrote a simple C++ player based on alsa-lib which allows me to play `wav` files. I wonder if there is any convenient way to modify my code so that I can play the the `wav` file in the background and the main program is executing something else, say, a `while` loop. _Edited and Updated:_ I tried to use `std::thread` and `std::async`, I was able to play the sound in the background. I looked at the header file of alsa-lib, it seems that alsa-lib natively supports playing sound asynchronously. I wonder how I could do it easily. Thank you very much. Best, Lei Issue URL : https://github.com/alsa-project/alsa-lib/issues/17 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] Code Example for Playing Sound in the Background
by GitHub issues - edited 14 Dec '19

14 Dec '19

alsa-project/alsa-lib issue #17 was edited from leimao: Hello, I wrote a simple C++ player based on alsa-lib which allows me to play `wav` files. I wonder if there is any convenient way to modify my code so that I can play the the `wav` file in the background and the main program is executing something else, say, a `while` loop. Edited and Updated: I tried to use `std::thread` and `std::async`, I was able to play the sound in the background. I looked at the header file of alsa-lib, it seems that alsa-lib natively supports playing sound asynchronously. I wonder how I could do it easily. Thank you very much. Best, Lei Issue URL : https://github.com/alsa-project/alsa-lib/issues/17 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] Code Example for Playing Sound in the Background
by GitHub issues - opened 14 Dec '19

14 Dec '19

alsa-project/alsa-lib issue #17 was opened from leimao: Hello, I wrote a simple C++ player based on alsa-lib which allows me to play `wav` files. I wonder if there is any convenient way to modify my code so that I can play the the `wav` file in the background and the main program is executing something else, say, a `while` loop. I tried to use `std::thread` but I did not succeed. Thank you very much. Best, Lei Issue URL : https://github.com/alsa-project/alsa-lib/issues/17 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH] ALSA: hda: Fix regression by strip mask fix
by Takashi Iwai 14 Dec '19

14 Dec '19

The commit e38e486d66e2 ("ALSA: hda: Modify stream stripe mask only when needed") tried to address the regression by the unconditional application of the stripe mask, but this caused yet another regression for the previously working devices. Namely, the patch clears the azx_dev->stripe flag at snd_hdac_stream_clear(), but this may be called multiple times before restarting the stream, so this ended up with clearance of the flag for the whole time. This patch fixes the regression by moving the azx_dev->stripe flag clearance at the counter-part, the close callback of HDMI codec driver instead. Fixes: e38e486d66e2 ("ALSA: hda: Modify stream stripe mask only when needed") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205855 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204477 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_stream.c | 4 +--- sound/pci/hda/patch_hdmi.c | 2 ++ 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index f9707fb05efe..682ed39f79b0 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -120,10 +120,8 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_DMA_START | SD_INT_MASK, 0); snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ - if (azx_dev->stripe) { + if (azx_dev->stripe) snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); - azx_dev->stripe = 0; - } azx_dev->running = false; } EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78647ee02339..630b1f5c276d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2021,6 +2021,8 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_cvt->assigned = 0; hinfo->nid = 0; + azx_stream(get_azx_dev(substream))->stripe = 0; + mutex_lock(&spec->pcm_lock); snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: control: remove useless assignment in .info callback of PCM chmap element
by Takashi Sakamoto 14 Dec '19

14 Dec '19

Control elements for PCM chmap return information to userspace abount the maximum number of available PCM channels as the number of values in the element. In current implementation the number is once initialized to zero, then assigned to. This is useless and this commit fixes it. Fixes: 2d3391ec0ecc ("ALSA: PCM: channel mapping API implementation") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/pcm_lib.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2236b5e0c1f2..6f0a00fd3ae5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2308,7 +2308,6 @@ static int pcm_chmap_ctl_info(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 0; uinfo->count = info->max_channels; uinfo->value.integer.min = 0; uinfo->value.integer.max = SNDRV_CHMAP_LAST; -- 2.20.1

2 1

[alsa-devel] [PATCH 0/2] Add the broadmobi BM818
by Angus Ainslie (Purism) 14 Dec '19

14 Dec '19

The broadmobi uses slightly different parameters from the option modems so add the paramters and document them. Angus Ainslie (Purism) (2): sound: codecs: gtm601: add Broadmobi bm818 sound profile ASoC: gtm601: add the broadmobi interface .../devicetree/bindings/sound/gtm601.txt | 10 +++++-- sound/soc/codecs/gtm601.c | 29 +++++++++++++++++-- 2 files changed, 35 insertions(+), 4 deletions(-) -- 2.17.1

4 6

[alsa-devel] [PATCH] ALSA: hda: Comment about snd_hdac_bus_update_rirb() and spinlock
by Takashi Iwai 13 Dec '19

13 Dec '19

The call of snd_hdac_bus_update_rirb() needs the bus->reg_lock spinlock protection for concurrency. Comment about it more explicitly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_controller.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 01787081552d..b856184af37f 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -181,6 +181,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_send_cmd); * @bus: HD-audio core bus * * Usually called from interrupt handler. + * The caller needs bus->reg_lock spinlock before calling this. */ void snd_hdac_bus_update_rirb(struct hdac_bus *bus) { -- 2.16.4

1 0