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Re: [alsa-devel] DMA interrupt not getting triggered(nailed the problem need solution)
by ANISH KUMAR 08 Feb '10

08 Feb '10

> If the initial hardware state happens to result in LRCLK being where the >lrsync() function is trying to get it then the driver will start up, but >the misclocking of the configuration will result in ALSA detecting an >error which will eventually result in a restart of the audio stream. From the lrsync function i know that it is just trying to check whether a particular bit is set or not.So according to my understanding this lrsync function will timeout if the clock itself is not coming. Please forgive for not understanding your reply. But i am not able to understand "where the lrsync() function is trying to get-FROM YOUR REPLY"?? And misclocking of the configuration??Please explain this term(googling didn't help much).

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[alsa-devel] [PATCH -next] ASoC: pandora: Add DAC regulator support
by Grazvydas Ignotas 08 Feb '10

08 Feb '10

Pandora's external DAC is connected to VSIM TWL4030 supply, so let's start switching it too to save more power. Also DAC got it's own DAPM handler and the delay was removed (it gave on audible improvement). Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 37 +++++++++++++++++++++++++++++++------ 1 files changed, 31 insertions(+), 6 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 68980c1..b978411 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -22,7 +22,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/gpio.h> -#include <linux/delay.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> @@ -40,6 +40,8 @@ #define PREFIX "ASoC omap3pandora: " +static struct regulator *omap3pandora_dac_reg; + static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, unsigned int fmt) { @@ -106,21 +108,31 @@ static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_CBS_CFS); } -static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, +static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) { + regulator_enable(omap3pandora_dac_reg); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); - gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); } else { - gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); - mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + regulator_disable(omap3pandora_dac_reg); } return 0; } +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); + else + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + + return 0; +} + /* * Audio paths on Pandora board: * @@ -130,7 +142,9 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM, + 0, 0, omap3pandora_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -306,8 +320,18 @@ static int __init omap3pandora_soc_init(void) goto fail2; } + omap3pandora_dac_reg = regulator_get(&omap3pandora_snd_device->dev, "vcc"); + if (IS_ERR(omap3pandora_dac_reg)) { + pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n", + dev_name(&omap3pandora_snd_device->dev), + PTR_ERR(omap3pandora_dac_reg)); + goto fail3; + } + return 0; +fail3: + platform_device_del(omap3pandora_snd_device); fail2: platform_device_put(omap3pandora_snd_device); fail1: @@ -320,6 +344,7 @@ module_init(omap3pandora_soc_init); static void __exit omap3pandora_soc_exit(void) { + regulator_put(omap3pandora_dac_reg); platform_device_unregister(omap3pandora_snd_device); gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); -- 1.6.3.3

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Re: [alsa-devel] DMA interrupt not getting triggered(nailed the problem need solution)
by ANISH KUMAR 08 Feb '10

08 Feb '10

>If the initial hardware state happens to result in LRCLK being where the >lrsync() function is trying to get it then the driver will start up, but >the misclocking of the configuration will result in ALSA detecting an >error which will eventually result in a restart of the audio stream. Mark in my case it is not restarting the audio stream(though it is trying to call trigger again for restarting) because in s3c_snd_lrsync function again i am getting time out(ETIMEDOUT). iiscon = readl(s3c_i2s.regs + S3C_IISCON); if (iiscon & S3C_IISCON_LRI) break; This LRI register bit is not getting set properly causing timeout.How this situation is not coming always(4/10)?? And who should suppose to set it? If my machine driver had a problem then this should have come always?? Thanks for your reply.

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[alsa-devel] ALSA Driver 1.0.22.1
by Darsen Lu 07 Feb '10

07 Feb '10

I just installed alsa-drivers 1.0.22.1 (the latest version to my understanding) on Ubuntu 9.10. After that I started noticing conflicts between different sound-playback applications. For example, if I watch a Flash video on Firefox, stop it, and then start playing an MP3 file on Totem (a Media Player), I don't hear any sound. Once I close Firefox the music from Totem starts coming out. This never happened before on my Linux system. I'd be very grateful if someone could give me advices! My /var/log/syslog shows the following whenever this happens: Feb 7 02:42:36 darsen-laptop pulseaudio[1789]: alsa-sink.c: Error opening PCM device front:0: Device or resource busy Feb 7 02:42:36 darsen-laptop pulseaudio[1789]: alsa-sink.c: Error opening PCM device front:0: Device or resource busy I also noticed the following message during bootup and was asked to report this issue to ALSA Developers. Feb 7 02:17:31 darsen-laptop pulseaudio[1789]: ratelimit.c: 179 events suppressed Feb 7 02:17:35 darsen-laptop pulseaudio[1789]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Feb 7 02:17:35 darsen-laptop pulseaudio[1789]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Feb 7 02:17:35 darsen-laptop pulseaudio[1789]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Feb 7 02:17:36 darsen-laptop pulseaudio[1789]: ratelimit.c: 182 events suppressed The good news is my microphone started working with the newly-installed ALSA drivers (It did not work with ALSA v1.0.20) My Linux kernel version is 2.6.31-19.general. I installed it using the Ubuntu Package Manager. On the other hand the ALSA driver was compiled from source code. ALSA was configured with the following options sudo ./configure -with-kernel=/usr/src/linux-headers-`uname -r` --with-cards=hda-intel "lspci -v" shows the following 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Dell Device 02cf Flags: bus master, fast devsel, latency 0, IRQ 31 Memory at f6afc000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel I confirmed that the correct version of driver was loaded into the kernel by typing "cat /proc/asound/version": Advanced Linux Sound Architecture Driver Version 1.0.22.1. Compiled on Feb 7 2010 for kernel 2.6.31-19-generic (SMP). Thanks, Darsen

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 05 Feb '10

05 Feb '10

The following changes since commit cb67286d6629ecb5bfc44071d664cf1cbd01a350: Peter Ujfalusi (1): ASoC: TWL4030: Module unloading fix are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Grazvydas Ignotas (2): ASoC: pandora: Add APLL supply to fix audio output ASoC: pandora: Add DAC regulator support Mark Brown (1): Merge branch 'for-2.6.33' into for-2.6.34 sound/soc/omap/omap3pandora.c | 43 ++++++++++++++++++++++++++++++++++++---- 1 files changed, 38 insertions(+), 5 deletions(-)

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[alsa-devel] ASoC update for 2.6.33
by Mark Brown 05 Feb '10

05 Feb '10

The following changes since commit 5bbd4953a4fb5d8d597b4a53b8da97eee320b634: Anuj Aggarwal (1): ASoC: AM3517: ASoC driver not getting compiled are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Grazvydas Ignotas (1): ASoC: pandora: Add APLL supply to fix audio output sound/soc/omap/omap3pandora.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-)

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[alsa-devel] [PATCH] ASoC: pandora: Add APLL supply to fix audio output
by Grazvydas Ignotas 05 Feb '10

05 Feb '10

Pandora's external DAC is using 256*Fs output from the TWL4030 codec, and TWL4030 needs to have APLL enabled for it's 256*Fs output to function. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- I noticed this a bit late because TWL4030 codec used to have APLL enable hardcoded until that was clean up by not-so-old patch. Would be nice to see this in next -rc, as without this audio won't work completely on pandora (again).. sound/soc/omap/omap3pandora.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 71b2c16..68980c1 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { }; static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"PCM DAC", NULL, "APLL Enable"}, {"Headphone Amplifier", NULL, "PCM DAC"}, {"Line Out", NULL, "PCM DAC"}, {"Headphone Jack", NULL, "Headphone Amplifier"}, -- 1.6.3.3

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Re: [alsa-devel] [Q] playing / recording mono on a stereo-only hardware
by Guennadi Liakhovetski 05 Feb '10

05 Feb '10

(re-adding the list) On Tue, 2 Feb 2010, Michael Trimarchi wrote: > Hi, > > Guennadi Liakhovetski wrote: > > On Sat, 30 Jan 2010, Michael Trimarchi wrote: > > > > > > > Guennadi Liakhovetski wrote: > > > > > > > Hi > > > > > > > > I know this question belongs rather to alsa-users, and I did ask it > > > > there, > > > > but got no solution so far. Maybe someone here can help. > > > > > > > > The recently posted SuperH SIU driver only supports stereo playback and > > > > capture (channels_min = channels_max = 2). Is there a way to tell > > > > alsa-lib > > > > to duplicate the mono-channel during playback and to mix recorded > > > > channels > > > > (or use one of them), if mono recording is requested? > > > > > > > I'm not an expert but I think that you can use plughw support of alsa-lib > > > > > > pcm.record { > > > type plug > > > slave { > > > pcm record_hw > > > } > > > route_policy average > > > rate_converter "linear" > > > } > > > > > > I think that automatically change rate and channel for supporting mono > > > channel > > > too > > > > > > > Thanks for the snippet, but, I'm afraid, it didn't help either. > > > Sorry can you explain a little bit more why it doesn't help you? I certainly gladly would explain this, if only I knew... It just doesn't change anything. Are you using it anywhere? If yes, with which hardware configuration and software (alsa-lib) versions? Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

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[alsa-devel] RFC: workaround for 'azx_get_response timeout, switching to polling mode...'
by Maxim Levitsky 05 Feb '10

05 Feb '10

Hi, Although I thought that polling mode is harmless, it sometimes isn't. I found out that after few dozens of hibernate cycles, the timeout would happen again kicking out first MSI, and then whole RRIB (single_cmd). However, I also found out that if I resend the command that caused the timeout, it would complete normally. Even futher, it is possible to poll for this command, and send next normally. It just works. So interrupts do work, but sometimes (very rarely) are missed. It might be even a hardware bug. The patch I send allows to delay switch to polling till we get 3 such timeouts in a row. I tested this approach for about 130 hibernate cycles. Best regards, Maxim Levitsky

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Re: [alsa-devel] DMA interrupt not getting triggered
by ANISH KUMAR 05 Feb '10

05 Feb '10

>Which board are you trying this on? S3C6410 >There may be some improvements with the DMA code in newer kernels, it would >be worth trying 2.6.33 and seeing if that has the same problem. There was >a problem with the circular DMA code failing to be fully merged which caused >problems including a resource exhaustion. Ben it would be very difficult to take the entire change.There are lot of efforts already gone into to make it stable. Can you please tell me which all changes exactly is required to get rid of this problem?As to what all the files that needs merging as i am not a kernel expert. or Can you give me some patch that would solve this problem?

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