September 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] sound card matrix defaced
by Patrick Shirkey 22 Sep '11

22 Sep '11

Hi, Someone needs to restore the sound card matrix as it has been defaced by spammers. Cheers -- Patrick Shirkey Boost Hardware Ltd

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[alsa-devel] [PATCH] ASoC: Don't force bias on ground referenced devices
by Mark Brown 22 Sep '11

22 Sep '11

Currently we force all devices in the system to be at the same bias level. This is due to concerns about power or pop/click impacts from either ramping VMID or mismatching VMID on the analogue I/O lines between connected devices but does mean we power devices up more often than we really need to. If a device flags idle_bias_off this will usually mean that it's either all digital or ground referenced (in which case the idle and powered bias levels are identical) so this concern does not apply and we can save some power by leaving it off when not needed itself. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6cac045..2bde6b0c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1319,13 +1319,16 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - /* Force all contexts in the card to the same bias state */ + /* Force all contexts in the card to the same bias state if + * they're not ground referenced. + */ bias = SND_SOC_BIAS_OFF; list_for_each_entry(d, &card->dapm_list, list) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - d->target_bias_level = bias; + if (!d->idle_bias_off) + d->target_bias_level = bias; trace_snd_soc_dapm_walk_done(card); -- 1.7.6.3

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[alsa-devel] [PATCH] alsa-plugins: Pulse: only underrun if no more data has been written
by David Henningsson 22 Sep '11

22 Sep '11

With the new/upcoming version of PulseAudio (0.99.x) there is a protocol addition that makes it possible to handle underruns better in the pulse plugin. The attached patch implements that, but it has two flaws I wouldn't mind getting some help with if possible: 1) Since this uses the new API function pa_stream_get_underflow_index, it won't compile with current stable PulseAudio versions, only the upcoming version. 2) So now there are three possibilities for handle_underrun ( 0 = never, 1 = always, 2 = the new improved one that IMO should be used). Since handle_underrun is a bool in the config, it can not be set to "2", only 0 or 1. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH] ASoC: Add another DAPM stat for neighbour checks
by Mark Brown 21 Sep '11

21 Sep '11

The number of times we look at a potentially connected neighbour is just as important as the number of times we actually recurse into looking at that neighbour so also collect that statistic. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + include/trace/events/asoc.h | 7 +++++-- sound/soc/soc-dapm.c | 4 ++++ 3 files changed, 10 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0e2d017..bb59532 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -540,6 +540,7 @@ struct snd_soc_dapm_widget_list { struct snd_soc_dapm_stats { int power_checks; int path_checks; + int neighbour_checks; }; #endif diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 2e1adf6..ab26f8a 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -226,16 +226,19 @@ TRACE_EVENT(snd_soc_dapm_walk_done, __string( name, card->name ) __field( int, power_checks ) __field( int, path_checks ) + __field( int, neighbour_checks ) ), TP_fast_assign( __assign_str(name, card->name); __entry->power_checks = card->dapm_stats.power_checks; __entry->path_checks = card->dapm_stats.path_checks; + __entry->neighbour_checks = card->dapm_stats.neighbour_checks; ), - TP_printk("%s: %d power checks, %d path checks", __get_str(name), - (int)__entry->power_checks, (int)__entry->path_checks) + TP_printk("%s: checks %d power, %d path, %d neighbour", + __get_str(name), (int)__entry->power_checks, + (int)__entry->path_checks, (int)__entry->neighbour_checks) ); TRACE_EVENT(snd_soc_jack_irq, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 84d1d79..6cac045 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -677,6 +677,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { + DAPM_UPDATE_STAT(widget, neighbour_checks); + if (path->weak) continue; @@ -732,6 +734,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { + DAPM_UPDATE_STAT(widget, neighbour_checks); + if (path->weak) continue; -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Dynamically manage DBVDD2 and DBVDD3 on WM5100
by Mark Brown 21 Sep '11

21 Sep '11

Allow the DBVDD2 and DBVDD3 rails to be powered down when idle, helping fully power down connected devices when idle. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 82 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 81 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index a30b0bf..eff252b 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -55,6 +55,8 @@ struct wm5100_priv { struct regulator_bulk_data core_supplies[WM5100_NUM_CORE_SUPPLIES]; struct regulator *cpvdd; + struct regulator *dbvdd2; + struct regulator *dbvdd3; int rev; @@ -804,6 +806,52 @@ static int wm5100_cp_ev(struct snd_soc_dapm_widget *w, } } +static int wm5100_dbvdd_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct regulator *regulator; + int ret; + + switch (w->shift) { + case 2: + regulator = wm5100->dbvdd2; + break; + case 3: + regulator = wm5100->dbvdd3; + break; + default: + BUG(); + return 0; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(regulator); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", + w->shift, ret); + return ret; + } + return ret; + + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(regulator); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", + w->shift, ret); + return ret; + } + return ret; + + default: + BUG(); + return 0; + } +} + static void wm5100_log_status3(struct snd_soc_codec *codec, int val) { if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) @@ -881,6 +929,10 @@ SND_SOC_DAPM_SUPPLY("CP2", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_ENA_SHIFT, 0, SND_SOC_DAPM_SUPPLY("CP2 Active", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_BYPASS_SHIFT, 1, wm5100_cp_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DBVDD2", SND_SOC_NOPM, 2, 0, wm5100_dbvdd_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DBVDD3", SND_SOC_NOPM, 3, 0, wm5100_dbvdd_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM5100_MIC_BIAS_CTRL_1, WM5100_MICB1_ENA_SHIFT, 0, NULL, 0), @@ -1123,10 +1175,14 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "AIF1RX8", NULL, "SYSCLK" }, { "AIF2RX1", NULL, "SYSCLK" }, + { "AIF2RX1", NULL, "DBVDD2" }, { "AIF2RX2", NULL, "SYSCLK" }, + { "AIF2RX2", NULL, "DBVDD2" }, { "AIF3RX1", NULL, "SYSCLK" }, + { "AIF3RX1", NULL, "DBVDD3" }, { "AIF3RX2", NULL, "SYSCLK" }, + { "AIF3RX2", NULL, "DBVDD3" }, { "AIF1TX1", NULL, "SYSCLK" }, { "AIF1TX2", NULL, "SYSCLK" }, @@ -1138,10 +1194,14 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "AIF1TX8", NULL, "SYSCLK" }, { "AIF2TX1", NULL, "SYSCLK" }, + { "AIF2TX1", NULL, "DBVDD2" }, { "AIF2TX2", NULL, "SYSCLK" }, + { "AIF2TX2", NULL, "DBVDD2" }, { "AIF3TX1", NULL, "SYSCLK" }, + { "AIF3TX1", NULL, "DBVDD3" }, { "AIF3TX2", NULL, "SYSCLK" }, + { "AIF3TX2", NULL, "DBVDD3" }, { "MICBIAS1", NULL, "CP2" }, { "MICBIAS2", NULL, "CP2" }, @@ -2251,12 +2311,26 @@ static int wm5100_probe(struct snd_soc_codec *codec) goto err_core; } + wm5100->dbvdd2 = regulator_get(&i2c->dev, "DBVDD2"); + if (IS_ERR(wm5100->dbvdd2)) { + ret = PTR_ERR(wm5100->dbvdd2); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_cpvdd; + } + + wm5100->dbvdd3 = regulator_get(&i2c->dev, "DBVDD3"); + if (IS_ERR(wm5100->dbvdd3)) { + ret = PTR_ERR(wm5100->dbvdd3); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_dbvdd2; + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable core supplies: %d\n", ret); - goto err_cpvdd; + goto err_dbvdd3; } if (wm5100->pdata.ldo_ena) { @@ -2433,6 +2507,10 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); +err_dbvdd3: + regulator_put(wm5100->dbvdd3); +err_dbvdd2: + regulator_put(wm5100->dbvdd2); err_cpvdd: regulator_put(wm5100->cpvdd); err_core: @@ -2459,6 +2537,8 @@ static int wm5100_remove(struct snd_soc_codec *codec) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } + regulator_put(wm5100->dbvdd3); + regulator_put(wm5100->dbvdd2); regulator_put(wm5100->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -- 1.7.6.3

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[alsa-devel] [PATCH 0/3] ASoC: tpa6130a2: model handling cleanup
by Peter Ujfalusi 21 Sep '11

21 Sep '11

Hello, Small cleanup for the tpa6130a2 driver for model handling: Remove the model_id from platform_data, and use the device name/device_data to distinguish between the supported models of TPA. Regards, Peter --- Peter Ujfalusi (3): ASoC: tpa6130a2: Model support cleanup OMAP: RX51: No need to configure the tpa_model ASoC: tpa6130a2: Remove model_id from platform data arch/arm/mach-omap2/board-rx51-peripherals.c | 1 - include/sound/tpa6130a2-plat.h | 6 ------ sound/soc/codecs/tpa6130a2.c | 12 +++++++++--- 3 files changed, 9 insertions(+), 10 deletions(-) -- 1.7.6.1

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Re: [alsa-devel] [PATCH] ALSA: snd-aloop: make preallocated buffer size configurable
by Takashi Iwai 21 Sep '11

21 Sep '11

At Wed, 21 Sep 2011 08:26:27 -0500, Pierre-Louis Bossart wrote: > > > > Default of 2MB of physically-contiguous memory can be > > > too high for embedded solutions, make the size configurable. > > > > Hm, maybe better to have an option for global limit of preallocation? > > The problem is not necessarily specific to snd-aloop. > > Not sure I understand the comment. This code is essentially the same as the > configuration option for HDAudio, and the parameter value defines the amount > of preallocated memory, with a limit to 32MB. I meant to have a kconfig option to limit the upper-bound of pre-alloc buffer sizes for all drivers. That is, a patch like below (untested) Takashi --- diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 475455c..525ed28 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -204,6 +204,15 @@ config SND_PCM_XRUN_DEBUG sound clicking when system is loaded, it may help to determine the process or driver which causes the scheduling gaps. +config SND_PCM_MAX_PREALLOC_SIZE + int "Maximum pre-allocated buffer size" + range 0 32768 + default 0 + depends on EMBEDDED + help + Specifies the default upper-limit of PCM pre-allocation + buffer sizes in kB. Zero means unlimited. + config SND_VMASTER bool diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 150cb7e..a09425c 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -270,6 +270,9 @@ int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream, int type, struct device *data, size_t size, size_t max) { + if (CONFIG_SND_PCM_MAX_PREALLOC_SIZE && + max > CONFIG_PCM_SND_MAX_PREALLOC_SIZE * 1024) + max = CONFIG_SND_PCM_PREALLOC_SIZE * 1024; substream->dma_buffer.dev.type = type; substream->dma_buffer.dev.dev = data; return snd_pcm_lib_preallocate_pages1(substream, size, max);

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[alsa-devel] [RFC v2 PATCH 1/1] ASoC: soc-core: symmetry checking for each DAIs separately
by Dong Aisheng 21 Sep '11

21 Sep '11

The orginal code does not cover the case that one DAI such as codec may be shared between other two DAIs(CPU). When do symmetry checking, altough the codec DAI requires symmetry, the two CPU DAIs may still be configured to run on different rates. We change to check each DAI's state separately instead of only checking the dai link to prevent this issue. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- Change since v1: Address the comments from Lars-Peter Clausen. * add the dai as an parameter for soc_pcm_apply_symmetry Thanks for the suggestion. --- include/sound/soc-dai.h | 3 +++ include/sound/soc.h | 2 -- sound/soc/soc-pcm.c | 40 +++++++++++++++++++++++++--------------- 3 files changed, 28 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5ad5f3a..12d98b4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,6 +242,9 @@ struct snd_soc_dai { void *playback_dma_data; void *capture_dma_data; + /* Symmetry data - only valid if symmetry is being enforced */ + unsigned int rate; + /* parent platform/codec */ union { struct snd_soc_platform *platform; diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..5449139 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -849,8 +849,6 @@ struct snd_soc_pcm_runtime { unsigned int complete:1; unsigned int dev_registered:1; - /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..8eb0f07 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,15 +27,13 @@ #include <sound/soc.h> #include <sound/initval.h> -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, + struct snd_soc_dai *soc_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && + if (!soc_dai->driver->symmetric_rates && !rtd->dai_link->symmetric_rates) return 0; @@ -43,19 +41,19 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. */ - if (!rtd->rate) { - dev_warn(&rtd->dev, + if (!soc_dai->rate) { + dev_warn(soc_dai->dev, "Not enforcing symmetric_rates due to race\n"); return 0; } - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + dev_dbg(soc_dai->dev, "Symmetry forces %dHz rate\n", soc_dai->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); + soc_dai->rate, soc_dai->rate); if (ret < 0) { - dev_err(&rtd->dev, + dev_err(soc_dai->dev, "Unable to apply rate symmetry constraint: %d\n", ret); return ret; } @@ -185,8 +183,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); + if (cpu_dai->active) { + ret = soc_pcm_apply_symmetry(substream, cpu_dai); + if (ret != 0) + goto config_err; + } + + if (codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, codec_dai); if (ret != 0) goto config_err; } @@ -288,8 +292,12 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - if (!cpu_dai->active && !codec_dai->active) - rtd->rate = 0; + /* clear the corresponding DAIs rate when inactive */ + if (!cpu_dai->active) + cpu_dai->rate = 0; + + if (!codec_dai->active) + codec_dai->rate = 0; /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. @@ -447,7 +455,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* store the rate for each DAIs */ + cpu_dai->rate = params_rate(params); + codec_dai->rate = params_rate(params); out: mutex_unlock(&rtd->pcm_mutex); -- 1.7.0.4

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[alsa-devel] ASoC/DAPM: How to make the DAPM routing complete?
by Bas Vermeulen 21 Sep '11

21 Sep '11

I am using a slightly modified version of the 88pm860x-codec.c driver from 3.x in an android modified 2.6.35 kernel. I have had to make some changes to allow both capture and playback work on the same (I2S) interface, but so far things are working. I am however having some problems getting the DAPM framework to set the correct switches needed to get the complete route to actually start. What would be needed to get this working as it should? I can post or send the driver on request, but basically the same question rises for the current driver in the 3.x kernel. I want/need to go from HS1/HS2 to DIN for playback, and from MIC1 to DOUT for capture. In order to do that, I have to set the following switches: amixer -q sset 'DAC HS1 Mux' 'Right' amixer -q sset 'DAC HS2 Mux' 'Left' amixer -q sset 'I2S Mic Mux' 'ADC' amixer -q sset 'ADC Left Mux' 'ADCL' I would like to make this automatic, but I am unsure how to express this in my dapm routing table. Note that in order to be able to use playback and capture independently on the I2S interface (within reason, it is not possible to use a different rate for instance), I had to add a SUPPLY control to the dapm widgets that turns on the I2S clock when needed. Any help, hints or pointers to the right manual to read would be appreciated. Bas Vermeulen

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Re: [alsa-devel] [PATCH] ASoC: Add driver for rt5631
by Mark Brown 21 Sep '11

21 Sep '11

On Wed, Sep 07, 2011 at 11:16:35AM +0800, johnnyhsu(a)realtek.com wrote: > From d291cbcdd5c1faebd4c37b4564b33c322df51987 Mon Sep 17 00:00:00 2001 > From: Johnny Hsu <johnnyhsu(a)realtek.com> > Date: Wed, 7 Sep 2011 11:05:19 +0800 > Subject: [PATCH] ASoC: Add driver for rt5631 Applied, thanks. It'd be really helpful if where things aren't clear you could reply to code reviews - now that it's clear what the index register stuff is doing I suspect the sysfs stuff would be OK though it ought to go in debugfs not sysfs.

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