November 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] pcm: simple: Fix asserts
by Peter Rosin 11 Nov '15

11 Nov '15

From: Peter Rosin <peda(a)axentia.se> Do not error out on the boundaries. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- src/pcm/pcm_simple.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_simple.c b/src/pcm/pcm_simple.c index f943ec0..ce11083 100644 --- a/src/pcm/pcm_simple.c +++ b/src/pcm/pcm_simple.c @@ -173,8 +173,8 @@ int snd_spcm_init(snd_pcm_t *pcm, snd_pcm_sw_params_alloca(&sw_params); assert(pcm); - assert(rate > 5000 && rate < 192000); - assert(channels > 1 && channels < 512); + assert(rate >= 5000 && rate <= 192000); + assert(channels >= 1 && channels <= 512); rrate = rate; err = set_buffer_time(latency, &buffer_time); @@ -233,8 +233,8 @@ int snd_spcm_init_duplex(snd_pcm_t *playback_pcm, assert(playback_pcm); assert(capture_pcm); - assert(rate > 5000 && rate < 192000); - assert(channels > 1 && channels < 512); + assert(rate >= 5000 && rate <= 192000); + assert(channels >= 1 && channels <= 512); pcms[0] = playback_pcm; pcms[1] = capture_pcm; -- 1.7.10.4

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[alsa-devel] Alsa Crackling Noise
by Dinesh Ravindran 11 Nov '15

11 Nov '15

We have been facing a weird issue with Alsa recently. Sometimes the audio plays out with out any noise and sometimes it has lot of cracking noise, really a lot. The weird situation is when another application is printing out some debug statements to a console it sounds good. The behavior is same with a custom developed alsa application and also with aplay. Any thoughts? If we have a really big buffer size of around 2sec it works somehow, but we really want to analyze why alsa exhibits this strange behavior. Is it the alsa itself or is it just a symptom of a bigger issue at system level? All the test we are carrying out it at 8KHz, at higher sampling rate(48k) we do not see this behaviour. These are the scenarios in which it sounds good: 1. If any application is printing to the console while audio is being played out or even just holding the return button on the console helps. 2. If we enable xrun debug with echo 29 > /proc/asound/card0/pcm0p/xrun_debug it sounds good. 3. If we have really big buffer size around 2 seconds. 4. Or sometimes on fresh start of the machine until it enters xrun state once. We have the following setup: Processor: ARM926EJ-S Kernel: 3.14.53 Alsa-lib: 1.0.29 Alsa-Util: 1.0.29 Audio Chip: CONEXANT USB AUDIO > aplay -Dplughw:0,0 8kz_8000.wav #to play file cat /proc/asound/card0/stream0 ---------------------------------------- Playback: Status: Stop Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 1 OUT (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Hw params: -------------- access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 8000 (8000/1) period_size: 1000 buffer_size: 4000 sw params: ------------- tstamp_mode: NONE period_step: 1 avail_min: 1000 start_threshold: 4000 stop_threshold: 4000 silence_threshold: 0 silence_size: 0 boundary: 2097152000 status: -------- state: RUNNING owner_pid : 1852 trigger_time: 3667.436640852 tstamp : 3756.680343977 delay : 3296 avail : 720 avail_max : 1000 ----- hw_ptr : 712720 appl_ptr : 716000

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[alsa-devel] [PATCH v2] ASoC: damp: Reset dapm wcache after freeing damp widgets
by Jyri Sarha 11 Nov '15

11 Nov '15

If there is anything in damp->path_source_cache or damp->path_sink_cache, it can not be valid after the widgets have been freed. Without this patch a repeated remove and load of a machine driver may cause NULL pointer reference in dapm_wcache_lookup() when a freed widget, not belonging to any list, is haunting in the wcache. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> Reported-by: Felipe Balbi <balbi(a)ti.com> --- The patch fixes the problem reported here: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099431.ht… For v2, followed Lars-Peter's suggestion here: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100335.h… include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 7 +++++++ sound/soc/soc-topology.c | 1 + 3 files changed, 9 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5abba03..ddcbe33 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -398,6 +398,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w); +void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm); /* dapm events */ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ff8bda4..945a792 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2276,6 +2276,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) kfree(w); } +void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm) +{ + dapm->path_sink_cache.widget = NULL; + dapm->path_source_cache.widget = NULL; +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { @@ -2286,6 +2292,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) continue; snd_soc_dapm_free_widget(w); } + snd_soc_dapm_reset_cache(dapm); } static struct snd_soc_dapm_widget *dapm_find_widget( diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 69d01cd..c6ee033 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1805,6 +1805,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, snd_soc_tplg_widget_remove(w); snd_soc_dapm_free_widget(w); } + snd_soc_dapm_reset_cache(dapm); } EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); -- 1.9.1

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[alsa-devel] [PATCH 1/1] ALSA: hda/hdmi - apply Skylake fix-ups to Broxton display codec
by han.lu＠intel.com 11 Nov '15

11 Nov '15

From: "Lu, Han" <han.lu(a)intel.com> Broxton and Skylake have the same behavior on display audio. So this patch applys Skylake fix-ups to Broxton. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f503a88..309274b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -50,8 +50,9 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->core.vendor_id == 0x80862807) #define is_broadwell(codec) ((codec)->core.vendor_id == 0x80862808) #define is_skylake(codec) ((codec)->core.vendor_id == 0x80862809) +#define is_broxton(codec) ((codec)->core.vendor_id == 0x8086280a) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ - || is_skylake(codec)) + || is_skylake(codec) || is_broxton(codec)) #define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) -- 2.5.0

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[alsa-devel] [PATCH] ALSA: ctxfi: constify amixer_rsc_ops structure
by Julia Lawall 11 Nov '15

11 Nov '15

The amixer_rsc_ops structure is never modified, so declare it as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/pci/ctxfi/ctamixer.c | 2 +- sound/pci/ctxfi/ctamixer.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index c7dc38d..465b5c5 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -186,7 +186,7 @@ static int amixer_setup(struct amixer *amixer, struct rsc *input, return 0; } -static struct amixer_rsc_ops amixer_ops = { +static const struct amixer_rsc_ops amixer_ops = { .set_input = amixer_set_input, .set_invalid_squash = amixer_set_invalid_squash, .set_scale = amixer_set_y, diff --git a/sound/pci/ctxfi/ctamixer.h b/sound/pci/ctxfi/ctamixer.h index 72f42f2..2de18aa 100644 --- a/sound/pci/ctxfi/ctamixer.h +++ b/sound/pci/ctxfi/ctamixer.h @@ -58,7 +58,7 @@ struct amixer { unsigned char idx[8]; struct rsc *input; /* pointer to a resource acting as source */ struct sum *sum; /* Put amixer output to this summation node */ - struct amixer_rsc_ops *ops; /* AMixer specific operations */ + const struct amixer_rsc_ops *ops; /* AMixer specific operations */ }; struct amixer_rsc_ops {

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[alsa-devel] [PATCH] ASoC: core: Fix warning while make xmldocs caused by soc-ops.c
by Masanari Iida 11 Nov '15

11 Nov '15

This patch fix following warning while "make xmldocs". .//sound/soc/soc-ops.c:415: warning: No description found for parameter 'ucontrol' .//sound/soc/soc-ops.c:415: warning: Excess function parameter 'uinfo' description in 'snd_soc_put_volsw_sx' Signed-off-by: Masanari Iida <standby24x7(a)gmail.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 ecd38e5..2f67ba6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * -- 2.6.2.402.g2635c2b

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[alsa-devel] [PATCH MC Next Gen v3 0/6] Add mixer and control interface media entities
by Shuah Khan 10 Nov '15

10 Nov '15

This patch v3 series adds the following: A fix to media_stream_delete() to remove intf devnode. Add support for creating MEDIA_ENT_F_AUDIO_MIXER entity for each mixer and a MEDIA_INTF_T_ALSA_CONTROL control interface entity that links to mixer entities. MEDIA_INTF_T_ALSA_CONTROL entity corresponds to the control device for the card. Change au0828_create_media_graph() to create pad link between MEDIA_ENT_F_AUDIO_MIXER entity and decoder's AU8522_PAD_AUDIO_OUT. With mixer entity now linked to decoder, change to link MEDIA_ENT_F_AUDIO_CAPTURE to mixer's source pad In addition, this patch series consists of fixes to compile warnings found by kbuild bot, and an additional fix in media stream cleanup path. Changes since v2: 1. Addressed Takashi's comments on the following ALSA patch: sound/usb: Create media mixer function and control interface entities 2. Include patches to fix compile warnings found by kbuild bot, and an additional fix in media stream cleanup path. 3. No changes to patches 1 and 3. 4. Changed patches: 2 5. New patches: 4,5, and 6 Changes since v1: 1. Included the fix to media_stream_delete() which was by mistake not included in the patch v1. 2. Fixed mixer to decoder link and mixer to capture node links based on Mauro Chehab's review comments on the media graph. This patch series is dependent on an earlier patch series-es: Patch series v2: https://www.mail-archive.com/linux-media@vger.kernel.org/msg93084.html Patch series v1: Update ALSA, and au0828 drivers to use Managed Media Controller API: https://www.mail-archive.com/linux-media@vger.kernel.org/msg92752.html Please find the media graphs for ALSA work with mixer and control interface nodes: (these are generated using mc_next_gen_test tool. https://drive.google.com/folderview?id=0B0NIL0BQg-AlLWE3SzAxazBJWm8&usp=sha… v4l2-utils patch adds mixer node support mc_next_gen_test tool: https://www.mail-archive.com/linux-media@vger.kernel.org/msg93086.html Shuah Khan (6): sound/usb: Fix media_stream_delete() to remove intf devnode sound/usb: Create media mixer function and control interface entities media: au0828 create link between ALSA Mixer and decoder sound/usb: Fix media_stream_init() and media_stream_delete() error paths media: Fix compile warning when CONFIG_MEDIA_CONTROLLER_DVB is disabled media: au0828-core fix ignoring return value errors drivers/media/dvb-core/dvb_frontend.c | 2 +- drivers/media/usb/au0828/au0828-core.c | 38 +++++++++--- drivers/media/usb/au0828/au0828.h | 1 + sound/usb/card.c | 5 ++ sound/usb/media.c | 102 ++++++++++++++++++++++++++++++++- sound/usb/media.h | 20 +++++++ sound/usb/mixer.h | 1 + sound/usb/usbaudio.h | 1 + 8 files changed, 158 insertions(+), 12 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: rsnd: make sure SRC In Rate feature enablement
by Kuninori Morimoto 10 Nov '15

10 Nov '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SRC In Rate convert feature cannot be used if data path is using DVC. This patch judges it, and not allowed to use it in such case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index c103aa7..e66f59e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -498,6 +498,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, struct snd_soc_pcm_runtime *rtd) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); struct rsnd_src *src = rsnd_mod_to_src(mod); int ret; @@ -512,6 +513,12 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, return 0; /* + * SRC In doesn't work if DVC was enabled + */ + if (dvc && !rsnd_io_is_play(io)) + return 0; + + /* * enable sync convert */ ret = rsnd_kctrl_new_s(mod, io, rtd, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5645: Use the mod_delayed_work instead of the queue_delayed_work and cancel_delayed_work_sync
by Oder Chiou 10 Nov '15

10 Nov '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 672fafd..4e81181 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -572,14 +572,12 @@ static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol, struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); int ret; - cancel_delayed_work_sync(&rt5645->rcclock_work); - regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PU); ret = snd_soc_put_volsw(kcontrol, ucontrol); - queue_delayed_work(system_power_efficient_wq, &rt5645->rcclock_work, + mod_delayed_work(system_power_efficient_wq, &rt5645->rcclock_work, msecs_to_jiffies(200)); return ret; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: avoid using divisible by 3 for DMIC clk
by Bard Liao 10 Nov '15

10 Nov '15

Few codecs will meet no DMIC clock output issue when select a divided number which is divisible by 3. To prevent this issue, the patch ignore the numbers when calculating the DMIC clock divider. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rl6231.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index aca479f..18b4292 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -80,6 +80,8 @@ int rl6231_calc_dmic_clk(int rate) } for (i = 0; i < ARRAY_SIZE(div); i++) { + if ((div[i] % 3) == 0) + continue; /* find divider that gives DMIC frequency below 3MHz */ if (3000000 * div[i] >= rate) return i; -- 1.8.1.1.439.g50a6b54

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