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

[alsa-devel] [PATCH] ASoC: Add info callback for SX_TLV controls
by Charles Keepax 16 Oct '15

16 Oct '15

SX_TLV controls are intended for situations where the register behind the control has some non-zero value indicating the minimum gain and then gains increasing from there and eventually overflowing through zero. Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. This means when the info callback subtracts the minimum value from the maximum value to calculate the number of gain levels available it is actually under reporting the available levels. This patch fixes this issue by adding a new snd_soc_info_volsw_sx callback that does not subtract the minimum value. Fixes: 1d99f2436d0d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 6 ++++-- sound/soc/soc-ops.c | 28 ++++++++++++++++++++++++++++ 2 files changed, 32 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 98e080b..bd85564 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -86,7 +86,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array),\ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw_sx, \ .get = snd_soc_get_volsw_sx,\ .put = snd_soc_put_volsw_sx, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ @@ -156,7 +156,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw_sx, \ .get = snd_soc_get_volsw_sx, \ .put = snd_soc_put_volsw_sx, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ @@ -593,6 +593,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); #define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 100d92b..05977ae 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -207,6 +207,34 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_info_volsw); /** + * snd_soc_info_volsw_sx - Mixer info callback for SX TLV controls + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a single mixer control, or a double + * mixer control that spans 2 registers of the SX TLV type. SX TLV controls + * have a range that represents both positive and negative values either side + * of zero but without a sign bit. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + snd_soc_info_volsw(kcontrol, uinfo); + /* Max represents the number of levels in an SX control not the + * maximum value, so add the minimum value back on + */ + uinfo->value.integer.max += mc->min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_sx); + +/** * snd_soc_get_volsw - single mixer get callback * @kcontrol: mixer control * @ucontrol: control element information -- 1.7.2.5

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[alsa-devel] race condition between dmaengine_pcm_dma_complete and snd_pcm_release
by Matt Campbell 16 Oct '15

16 Oct '15

Hi All, I've been trying to track down a kernel crash I've been getting when closing ALSA devices. There is a race condition between the bottom half interrupt handler for the DMA system (dmaengine_pcm_dma_complete in pcm_dmaengine.c) and the releasing of the sub-stream resources it receives as an argument (when snd_pcm_release is called). An older thread from 2013 discussed this to a good extent so I wont go into the details here. I've been unable to track down either a patch to fix this or even a good solution that I could implement. I've spent a couple days trying to think of an elegant solution and come up with nothing so far. Any input would be appreciated. Link to original thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-October/067111.ht… It's worth nothing that the original thread points out how this can arise on multi-core systems. In my case I'm on a single core system, but with the real-time patch enabled and the userspace ALSA thread running at a higher priority than the kernel's tasklet thread which can reproduce this almost 100% of the time. ~Matt

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[alsa-devel] [PATCH] ALSA: timer: add config item to export PCM timer disabling for expert
by Jie Yang 16 Oct '15

16 Oct '15

PCM timer is not always used. For embedded device, we need an interface to disable it when it is not needed, to shrink the kernel size and memory footprint, here add CONFIG_SND_PCM_TIMER for it. When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected, about 25KB saving bonus we can get. Please be noted that when disabled, those stubs who using pcm timer (e.g. dmix, dsnoop & co) may work incorrectlly. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/pcm.h | 8 +++++++- sound/core/Kconfig | 13 ++++++++++++- sound/core/Makefile | 3 ++- sound/core/pcm_lib.c | 2 ++ sound/core/pcm_native.c | 36 ++++++++++++++++-------------------- 5 files changed, 39 insertions(+), 23 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee..79cf943 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1117,10 +1117,16 @@ static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substrea * Timer interface */ +#ifdef CONFIG_SND_PCM_TIMER void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream); void snd_pcm_timer_init(struct snd_pcm_substream *substream); void snd_pcm_timer_done(struct snd_pcm_substream *substream); - +#else +static inline void +snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} +#endif /** * snd_pcm_gettime - Fill the timespec depending on the timestamp mode * @runtime: PCM runtime instance diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6c96fee..e3e9491 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -4,7 +4,7 @@ config SND_TIMER config SND_PCM tristate - select SND_TIMER + select SND_TIMER if SND_PCM_TIMER config SND_PCM_ELD bool @@ -93,6 +93,17 @@ config SND_PCM_OSS_PLUGINS support conversion of channels, formats and rates. It will behave like most of new OSS/Free drivers in 2.4/2.6 kernels. +config SND_PCM_TIMER + bool "PCM timer interface" if EXPERT + default y + help + If you disable this option, pcm timer will be inavailable, so + those stubs used pcm timer (e.g. dmix, dsnoop & co) may work + incorrectlly. + + For some embedded device, we may disable it to reduce memory + footprint, about 20KB on x86_64 platform. + config SND_SEQUENCER_OSS bool "OSS Sequencer API" depends on SND_SEQUENCER diff --git a/sound/core/Makefile b/sound/core/Makefile index 3354f91..48ab4b8 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,8 +13,9 @@ snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o -snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ +snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o +snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d45645..6dc4277 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1883,8 +1883,10 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; +#ifdef CONFIG_SND_PCM_TIMER if (substream->timer_running) snd_timer_interrupt(substream->timer, 1); +#endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); if (runtime->transfer_ack_end) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 75888dd..d5672ed 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -486,6 +486,16 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, int state) snd_pcm_stream_unlock_irq(substream); } +static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, + int event) +{ +#ifdef CONFIG_SND_PCM_TIMER + if (substream->timer) + snd_timer_notify(substream->timer, event, + &substream->runtime->trigger_tstamp); +#endif +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -1042,9 +1052,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } static struct action_ops snd_pcm_action_start = { @@ -1092,9 +1100,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) if (runtime->status->state != state) { snd_pcm_trigger_tstamp(substream); runtime->status->state = state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP); } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1208,18 +1214,12 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) snd_pcm_trigger_tstamp(substream); if (push) { runtime->status->state = SNDRV_PCM_STATE_PAUSED; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MPAUSE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MCONTINUE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE); } } @@ -1267,9 +1267,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) snd_pcm_trigger_tstamp(substream); runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1373,9 +1371,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->status->state = runtime->status->suspended_state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); } static struct action_ops snd_pcm_action_resume = { -- 2.1.4

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[alsa-devel] [PATCH V3 0/4] BAT: some fixes
by han.lu＠intel.com 16 Oct '15

16 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> some fixes and adjusts. Lu, Han (4): BAT: Merge message strings BAT: Use colon instead of comma for separation BAT: Adjust parameters and strings BAT: Use dynamic temp file bat/bat.c | 77 +++++++++++++++++++++++++++++++++--------------------------- bat/common.h | 2 +- 2 files changed, 43 insertions(+), 36 deletions(-) -- 1.9.1

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[alsa-devel] [RFC PATCH] ASoC: Modify check condition of multiple bindings of components
by Koro Chen 16 Oct '15

16 Oct '15

The patch "ASoC: Prevent components from being bound to multiple cards" adds check to prevent multiple cards from using the same component. However, snd_soc_register_platform() or snd_soc_register_codec() will also create components, and sharing the same platform by multiple cards is then refused. This happens with a platform having multiple independent DAIs that share the same DMA controller. Relax the condition by checking component->registered_as_component, which is only true in case of snd_soc_register_component() and will be false for components created by snd_soc_register_platform() or snd_soc_register_codec(). Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42575b0..eca169a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1106,7 +1106,8 @@ static int soc_probe_component(struct snd_soc_card *card, return 0; if (component->card) { - if (component->card != card) { + if (component->card != card && + component->registered_as_component) { dev_err(component->dev, "Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n", card->name, component->card->name); -- 1.7.9.5

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[alsa-devel] [PATCH MC Next Gen 0/2] Add mixer and control interface media entities
by Shuah Khan 15 Oct '15

15 Oct '15

This patch series adds the following: 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. This patch series is dependent on an earlier patch series: 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 graph with mixer and control interface nodes: https://drive.google.com/folderview?id=0B0NIL0BQg-Alb3JFb2diMXRoQlU&usp=sha… New file: graph_mixer.png Shuah Khan (2): sound/usb: Create media mixer function and control interface entities media: au0828 create link between ALSA Mixer and decoder drivers/media/usb/au0828/au0828-core.c | 12 +++++ drivers/media/usb/au0828/au0828.h | 1 + sound/usb/card.c | 5 ++ sound/usb/media.c | 85 ++++++++++++++++++++++++++++++++++ sound/usb/media.h | 4 ++ sound/usb/mixer.h | 1 + sound/usb/usbaudio.h | 1 + 7 files changed, 109 insertions(+) -- 2.1.4

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[alsa-devel] DSD: Playing silence before and after actual audio data
by Martin Pietryka 15 Oct '15

15 Oct '15

Hi guys, I'm trying to figure out the best way to insert ~40ms worth of silence pattern before and after playing back DSD data. There seems to be support inside ALSA for a DSD silence pattern, but as far as I have seen it's only used when an underrun might occur (silence_threshold, silence_size). Is there some possibility inside ALSA to basically insert silence before and after playback, or how hard would it be to implement something like this? I have looked a bit for myself and for me it seems that the modifications needed inside ALSA to achieve this feature are substantial, but of course I also might have missed something. Martin

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[alsa-devel] Audio Mini Summit 2015, 8th October, Dublin
by Mark Brown 15 Oct '15

15 Oct '15

[This message is BCCed to everyone currently on the signup list] As previously announce the audio mini-summit for this year will be this Thursday (8th August) at CCD in Dublin. Topics we currently have on the list for discussion are: - Power saving and general system design decisions (is what we're doing lining up with modern hardware with things like multiple output streams?). - What to do with HDMI and its integration with graphics. - Test tools (Intel BAT and others). - BATCH flag. - ACPI simple-card and general generic card support. - Topology & media controller integration. - Kernel tinification. - Android <-> alsa-lib configuration translation. - HDA restructuring. - PCM core simplification - the code is currently quite complex! - Soundwire driver support plans. If you're planning to attend and haven't done so already please sign up at: https://docs.google.com/spreadsheets/d/1bi-8GHsqlzt41FR20WiMuWILe_cZowFireV… Once I have details on the room & so on I will follow up to this mail with both. We'll also try and sort out some dinner on Thursday night, if there are any special requirements for that please say (I'm assuming that we will have at least some vegetarians).

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[alsa-devel] [PATCH] ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers
by Luis de Bethencourt 15 Oct '15

15 Oct '15

These platform drivers have a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/bcm/bcm2835-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 03fa1cb..8c435be 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -862,6 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = { {}, }; +MODULE_DEVICE_TABLE(of, bcm2835_i2s_of_match); + static struct platform_driver bcm2835_i2s_driver = { .probe = bcm2835_i2s_probe, .driver = { -- 2.4.6

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Re: [alsa-devel] Sound not working after new kernel installed
by James Cameron 15 Oct '15

15 Oct '15

On Thu, Oct 15, 2015 at 01:17:15PM +0200, Simon Wolk wrote: > The Kernel changed from 3.2.0 to 3.19.0 for me. By installing the > 3.19. the sound stopped working. Yes, Ubuntu kernels change like that, but you can easily run more recent kernel versions, see Ubuntu Wiki page: https://wiki.ubuntu.com/Kernel/MainlineBuilds While you wait for help, you can install latest v4.2.3 built 3rd October 2015. You can also test kernels between 3.2 and 3.19, if you don't get any other help. -- James Cameron http://quozl.linux.org.au/

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