August 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [pulseaudio-discuss] [PATCH v0 07/20] Consider unknown availability in module-switch-on-port-available
by Raymond Yau 22 Aug '12

22 Aug '12

2012-8-22 下午5:13 於 "David Henningsson" <david.henningsson(a)canonical.com> 寫道： > > On 08/22/2012 10:08 AM, Mikel Astiz wrote: >> >> From: Mikel Astiz <mikel.astiz(a)bmw-carit.de> >> >> A port with the availability state flag set to unknown should not be >> considered an active port. >> --- >> src/modules/module-switch-on-port-available.c | 2 +- >> 1 files changed, 1 insertions(+), 1 deletions(-) >> >> diff --git a/src/modules/module-switch-on-port-available.c b/src/modules/module-switch-on-port-available.c >> index 1da612d..8c16821 100644 >> --- a/src/modules/module-switch-on-port-available.c >> +++ b/src/modules/module-switch-on-port-available.c >> @@ -85,7 +85,7 @@ static pa_bool_t try_to_switch_profile(pa_card *card, pa_device_port *port) { >> PA_IDXSET_FOREACH(sink, card->sinks, state2) { >> if (!sink->active_port) >> continue; >> - if (sink->active_port->available != PA_PORT_AVAILABLE_NO) >> + if (sink->active_port->available == PA_PORT_AVAILABLE_YES) >> found_active_port = TRUE; >> } >> >> > > I think this will break the following use case (on a normal laptop) : > > Assume we plug in HDMI. With this patch we will get inconsistent behaviour: if headphones are plugged in (available = yes), we won't switch to HDMI, but if not (speakers are available = unknown), we will switch to HDMI. > > if you look at those bug reports of no sound for those on-board ac97 codecs with ubuntu 12.04 https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1033996 how does pulseaudio determine which controls are effective in the headphone path and speaker path of those ac97 codecs ? the analog-output-stereo profile for the line out jack is missing ( 0.056| 0.000) D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-headphone is there any solution for those snd-intel8x0 and those onboard ac97 codecs with headphone playback volume control ( no ctljack control and no phantom port)? http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/a… http://git.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/al…

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[alsa-devel] [PATCH v3 1/2] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 22 Aug '12

22 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gernel sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled, and if all codecs are suspended (in D3), the HDA controller can also be suspended to a low power state by decreasing the power usage counter. And the user IO operation can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. TODO: Suspend the controller only if all codecs support CLKSTOP in D3, for pass-through operations and wakeup. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_intel.c | 52 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 49 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..4c30fd7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2402,11 +2406,19 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) + if (power_on) { + if (!chip->initialized) + pm_runtime_get_sync(&chip->pci->dev); azx_init_chip(chip, 1); + } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) + !bus->power_keep_link_on) { azx_stop_chip(chip); + + /* TODO: Suspend controller only if all codec support + stop-clock in D3, for wakeup consideration */ + pm_runtime_put_sync(&chip->pci->dev); + } } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2523,30 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3238,6 +3273,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3327,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3342,13 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + /* undo 'get' in azx_probe_continue() */ + pm_runtime_put_noidle(&pci->dev); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

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Re: [alsa-devel] [PATCH 1/5] ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific
by Hebbar, Gururaja 22 Aug '12

22 Aug '12

On Fri, Aug 17, 2012 at 23:43:44, Nori, Sekhar wrote: > Hi Gururaja, > > On 8/14/2012 6:12 PM, Hebbar, Gururaja wrote: > > Davinci McASP header & driver are shared by few OMAP platforms (like > > TI81xx, AM335x). Splitting asp header into Davinci platform specific > > header and Audio specific header helps to share them across platforms. > > > > Audio specific defines is moved to to common > > <linux/platform_data/davinci_asp.h> so that the header can be > > accessed by all related platforms. > > > > While here, correct the header usage (remove multiple header > > re-definitions and unused headers) and remove platform names from > > structures comments and enum > > > > Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> > > Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> > > This patch looks good to me. See a minor nit below. > > Acked-by: Sekhar Nori <nsekhar(a)ti.com> > > > diff --git a/arch/arm/mach-davinci/asp.h b/arch/arm/mach-davinci/asp.h > > new file mode 100644 > > index 0000000..8fd5238 > > --- /dev/null > > +++ b/arch/arm/mach-davinci/asp.h > > @@ -0,0 +1,49 @@ > > +/* > > + * TI DaVinci Audio definitions > > + */ > > +#ifndef __ASM_ARCH_DAVINCI_ASP_H > > +#define __ASM_ARCH_DAVINCI_ASP_H > > + > > +/* Bases of dm644x and dm355 register banks */ > > +#define DAVINCI_ASP0_BASE 0x01E02000 > > +#define DAVINCI_ASP1_BASE 0x01E04000 > > + > > +/* Bases of dm365 register banks */ > > +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 > > + > > +/* Bases of dm646x register banks */ > > +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 > > +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 > > Can you remove usage of tabs after 'define'. I understand this is coming > from existing code. Ok. I have got few more comments from Mark. Will add this into it. > > Thanks, > Sekhar > Regards, Gururaja

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[alsa-devel] [RFC] Channel mapping API
by Takashi Iwai 21 Aug '12

21 Aug '12

Hi, this is a progress report of my longstanding TODO, the channel map API implementation. I'm going to cover this at Plumbers audio uconf, so we can discuss details there, too. The channel mapping API provides a method for user-space to query, get and set the channel map of a PCM stream. It's required for assigning channels properly for multi-channel streams. * KERNEL IMPLEMENTATION In my latest attempt, I implemented with control elements. A control element is created for each PCM substream with the corresponding device and substream index. Then it gives a TLV for querying maps, a read op for obtaining the current map, and optionally a write op for setting the map. The obvious merit by this way is that no extra kernel ABI is required. A couple of new helper functions are provided for assigning standard channel maps. Currently, HD-audio and AC97 drivers has some implementation. * ALSA-LIB IMPLEMENTATION The additional alsa-lib API functions look like: int **snd_pcm_query_chmaps(snd_pcm_t *pcm); int *snd_pcm_get_chmap(snd_pcm_t *pcm); int snd_pcm_set_chmap(snd_pcm_t *pcm, const int *map); snd_pcm_query_chmaps() returns the list of channel maps. A channel map is represented by an integer array, beginning with the channel map type, followed by the number of channels, and the position of each channel, e.g. { SND_CHMAP_FIXED, 4, SND_CHMAP_FL, SND_CHMAP_FR, SND_CHMAP_RL, SND_CHMAP_RR } snd_pcm_get_chmap() returns the currently assigned channel map for the given PCM stream. If the PCM is before prepared, it fills UNKNOWN. When a driver allows user to change the channel map, user can call snd_pcm_set_chmap(). For example, HDMI allows you to choose whether it's 4.0 or 3.1 output. * RESOURCES The latest kernel patchset is found in sound-unstable tree topic/tlv-chmap branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git topic/tlv-chmap The alsa-lib portion is found in a tree on github topic/chmap branch. git://github.com/tiwai/alsa-lib.git topic/chmap thanks, Takashi

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[alsa-devel] [PATCH] ASoC: wm_hubs: Allow configuration of MICBIAS power up delay via pdata
by Mark Brown 21 Aug '12

21 Aug '12

Sometimes the analogue circuitry connected to the microphone needs some time to settle after power up. Allow systems to configure this delay in the platform data, the driver will then insert the required delay during power up of paths that involve the microphone. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/pdata.h | 4 ++++ include/sound/wm8993.h | 4 ++++ sound/soc/codecs/wm8993.c | 2 ++ sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 35 +++++++++++++++++++++++++++++++---- sound/soc/codecs/wm_hubs.h | 4 ++++ 6 files changed, 47 insertions(+), 4 deletions(-) diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index f0361c0..fc87be4 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -164,6 +164,10 @@ struct wm8994_pdata { int num_micd_rates; struct wm8958_micd_rate *micd_rates; + /* Power up delays to add after microphone bias power up (ms) */ + int micb1_delay; + int micb2_delay; + /* LINEOUT can be differential or single ended */ unsigned int lineout1_diff:1; unsigned int lineout2_diff:1; diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h index eee19f6..8016fd8 100644 --- a/include/sound/wm8993.h +++ b/include/sound/wm8993.h @@ -32,6 +32,10 @@ struct wm8993_platform_data { unsigned int lineout1fb:1; unsigned int lineout2fb:1; + /* Delay to add for microphones to stabalise after power up */ + int micbias1_delay; + int micbias2_delay; + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 9fd80d6..94737a3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1520,6 +1520,8 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->pdata.lineout2fb, wm8993->pdata.jd_scthr, wm8993->pdata.jd_thr, + wm8993->pdata.micbias1_delay, + wm8993->pdata.micbias2_delay, wm8993->pdata.micbias1_lvl, wm8993->pdata.micbias2_lvl); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2d60ed3..a51431c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3233,6 +3233,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) pdata->lineout2fb, pdata->jd_scthr, pdata->jd_thr, + pdata->micb1_delay, + pdata->micb2_delay, pdata->micbias1_lvl, pdata->micbias2_lvl); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 8dc4e89..c72eaef 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -859,6 +859,28 @@ static int lineout_event(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + + switch (w->shift) { + case WM8993_MICB1_ENA_SHIFT: + if (hubs->micb1_delay) + msleep(hubs->micb1_delay); + break; + case WM8993_MICB2_ENA_SHIFT: + if (hubs->micb2_delay) + msleep(hubs->micb2_delay); + break; + default: + return -EINVAL; + } + + return 0; +} + void wm_hubs_update_class_w(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); @@ -1049,8 +1071,10 @@ SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), SND_SOC_DAPM_INPUT("IN2RP:VXRP"), -SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), @@ -1388,13 +1412,16 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, - int jd_scthr, int jd_thr, int micbias1_lvl, - int micbias2_lvl) + int jd_scthr, int jd_thr, + int micbias1_delay, int micbias2_delay, + int micbias1_lvl, int micbias2_lvl) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); hubs->lineout1_se = !lineout1_diff; hubs->lineout2_se = !lineout2_diff; + hubs->micb1_delay = micbias1_delay; + hubs->micb2_delay = micbias2_delay; if (!lineout1_diff) snd_soc_update_bits(codec, WM8993_LINE_MIXER1, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 14927f6..704e53f 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -55,6 +55,9 @@ struct wm_hubs_data { struct list_head dcs_cache; bool (*check_class_w_digital)(struct snd_soc_codec *); + int micb1_delay; + int micb2_delay; + bool lineout1_se; bool lineout1n_ena; bool lineout1p_ena; @@ -75,6 +78,7 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, int jd_scthr, int jd_thr, + int micbias1_dly, int micbias2_dly, int micbias1_lvl, int micbias2_lvl); extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); -- 1.7.10.4

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Re: [alsa-devel] mx27_3ds : Sound doesn't work anymore on linux-next
by Gaëtan Carlier 21 Aug '12

21 Aug '12

Just some more informations, process enter in D+ state and this is so impossible to kill them. On 08/17/2012 05:16 PM, Gaëtan Carlier wrote: > Hi, > It is several days that I tried to make sound work on mx27_3ds platform > from linux-next (daily cloned/pulled) but each time I try start start > aplay or alsamixer, the process freeze and CTRL+C has not effect. I can > open another SSH session and execute other commands but alsa commands > always freeze. > The sound card is correctly installer at boot : > > imx_mc13783 imx_mc13783.0: mc13783-hifi <-> imx-ssi.0 mapping ok > ALSA device list: > #0: imx_mc13783 > > I only apply the patch "[PATCH 1/2] ARM: mx27pdk: Add audio support". I > attach my config file. > > I try to apply patch on a previous commit when I told that this patch > was working but it didn't succeed the is still freezing. > I try to make it working on another mx27-based platform but same > problem. I try to update Alsa from version 1.0.23 to 1.0.25 but same > behaviour. > What is wrong ? > > Regards, > Gaëtan Carlier. Regards, Gaëtan.

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Re: [alsa-devel] [PATCH] ASoC: Remove unused variable
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 04:51:59PM +0200, Emil Goode wrote: > This patch removes a unused struct snd_soc_dai. > > Sparse is warning: > sound/soc/soc-compress.c:156:22: warning: > unused variable ‘codec_dai’ [-Wunused-variable] Already fixed by someone else.

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[alsa-devel] [PATCH 0/3] ASoC: omap-mcbsp: Sidetone related changes
by Peter Ujfalusi 21 Aug '12

21 Aug '12

Hello, When we boot with DT blob the sidetone is not supported at the moment. Prepare the McBSP code for the time when we are going to be able to support ST with DT booted kernel. Meanwhile do not block the audio card if the st_data is missing from the port, but print a warning about it. This will allow us to debug the DT booted kernels with audio. Mark: This series as such does not depend on the McBSP DT support, but it can be attached to that branch. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcbsp: Check mcbsp->id instead of cpu_dai->id when adding ST controls ASoC: omap-mcbsp: Only print warning if the st_data is missing for the port ASoC: omap-mcbsp: Use macro to create the McBSP2/3 ST controls sound/soc/omap/omap-mcbsp.c | 47 ++++++++++++++++++------------------------ 1 files changed, 20 insertions(+), 27 deletions(-) -- 1.7.8.6

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Re: [alsa-devel] [PATCH] ASoC: soc-compress: Remove unused variable
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 12:20:22PM +0530, Sachin Kamat wrote: > codec_dai is not used in the function. > > sound/soc/soc-compress.c: In function ‘soc_compr_set_params’: > sound/soc/soc-compress.c:156:22: warning: > unused variable ‘codec_dai’ [-Wunused-variable] Applied, thanks.

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[alsa-devel] [PATCH] ASoC: compress - fix code alignment
by Vinod Koul 21 Aug '12

21 Aug '12

Reported-by: Fengguang Wu <wfg(a)linux.intel.com> Signed-off-by: Namarta Kohli <namartax.kohli(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/soc/soc-compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 88d85ba..cc0562d 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -104,7 +104,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (platform->driver->compr_ops && platform->driver->compr_ops->free) platform->driver->compr_ops->free(cstream); - cpu_dai->runtime = NULL; + cpu_dai->runtime = NULL; if (cstream->direction == SND_COMPRESS_PLAYBACK) { if (!rtd->pmdown_time || codec->ignore_pmdown_time || -- 1.7.0.4

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