July 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] uda134x: properly free allocated memory if uda134x_soc_probe fail with unknown codec type
by Axel Lin 16 Jul '10

16 Jul '10

If uda134x_soc_probe fail with unknown codec type, it should properly free allocated memory before return -EINVAL. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This case won't be executed in current implementation because there is a model checking in the begin of uda134x_soc_probe. But since the code is there, I think it's good to make it correct. Otherwise, we can simply remove the default case. sound/soc/codecs/uda134x.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 28aac53..9bff3c6 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -566,9 +566,10 @@ static int uda134x_soc_probe(struct platform_device *pdev) ARRAY_SIZE(uda1345_snd_controls)); break; default: + ret = -EINVAL; printk(KERN_ERR "%s unknown codec type: %d", __func__, pd->model); - return -EINVAL; + break; } if (ret < 0) { -- 1.5.4.3

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[alsa-devel] [PATCH 8/8] sound: asihpi: fix sign bug
by Kulikov Vasiliy 15 Jul '10

15 Jul '10

bytes_per_sec is unsigned, so if snd_pcm_format_width() return error we would not see it. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/asihpi/asihpi.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 91218f7..c80b0b8 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -460,6 +460,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); int err; u16 format; + int width; unsigned int bytes_per_sec; print_hwparams(params); @@ -512,9 +513,10 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, dpcm->hpi_buffer_attached); } bytes_per_sec = params_rate(params) * params_channels(params); - bytes_per_sec *= snd_pcm_format_width(params_format(params)); + width = snd_pcm_format_width(params_format(params)); + bytes_per_sec *= width; bytes_per_sec /= 8; - if (bytes_per_sec <= 0) + if (width < 0 || bytes_per_sec == 0) return -EINVAL; dpcm->bytes_per_sec = bytes_per_sec; -- 1.7.0.4

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[alsa-devel] ALSA Help?
by Matthew Hudson 15 Jul '10

15 Jul '10

Hello, I posted similar messages to the alsa-user mail list and to ubuntu forums ( http://ubuntuforums.org/showthread.php?t=1528442 ), but haven't been able to get help. Could any of you help with this? So, I've got a great Acer Aspire 7736. Seems to have a RealTek ALC888 card & Intel HDA. I've upgraded Alsa. I've used the Drivers at the RealTek site. I've checked alsamixer over and over and over. Yet, currently I cannot get sound through the headphone jack. Originally, I had problems w/ headphone/speakers playing simlutaneously. Somewhere in my adventures I fixed that, but then only one app would use sound during a login (UGH). So, I kept poking...as all good linux users do...now I'm afraid I've totally wonked it. I'll put all pertinent alsa-info up here for anyone who will help!! http://www.alsa-project.org/db/?f=58850a92a9ea0edbf28a8d014a875cdbd81b621d I guess I'm bright enough to really mess it up, but dumb enough not to have a clue how to fix it. It has NEVER been able to recognize a headphone plugged in the jack automatically. I hope someone knows... -- Soli Deo Gloria! Matt Hudson "There is a kind of happiness and wonder that makes you serious." C.S. Lewis

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[alsa-devel] [PATCH 1/3] ASoC: multi-component - Fix S3C compile error
by Seungwhan Youn 15 Jul '10

15 Jul '10

This patch fix compile error on using S3C I2S. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index ad92e10..5017e31 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -56,11 +56,11 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, switch (clk_id) { case S3C64XX_CLKSRC_PCLK: - iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_MUX: - iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + iismod |= S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_CDCLK: @@ -90,7 +90,7 @@ struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(dai); u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) return i2s->iis_cclk; else return i2s->iis_pclk; -- 1.6.2.5

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Re: [alsa-devel] [PATCH 1/2] Kconfig: SND_AC97_POWER_SAVE description improvement
by Jiri Kosina 15 Jul '10

15 Jul '10

On Wed, 14 Jul 2010, Michael Witten wrote: > The description has been expanded to explain the time-out > value provided by the power_save module parameter. Michael, thanks a lot for the patch. I'd rather let it go through Takashi still anyway (added to CC). Thanks. > > Signed-off-by: Michael Witten <mfwitten(a)gmail.com> > --- > sound/drivers/Kconfig | 24 +++++++++++++++++++++--- > 1 files changed, 21 insertions(+), 3 deletions(-) > > diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig > index 84714a6..3264600 100644 > --- a/sound/drivers/Kconfig > +++ b/sound/drivers/Kconfig > @@ -170,9 +170,25 @@ config SND_AC97_POWER_SAVE > AC97 codecs. In this mode, the power-mode is dynamically > controlled at each open/close. > > - The mode is activated by passing power_save=1 option to > - snd-ac97-codec driver. You can toggle it dynamically over > - sysfs, too. > + The mode is activated by passing 'power_save=X' to the > + snd-ac97-codec driver module, where 'X' is the time-out > + value, a nonnegative integer that specifies how many > + seconds of idle time the driver must count before it may > + put the AC97 into power-save mode; a value of 0 (zero) > + disables the use of this power-save mode. > + > + After the snd-ac97-codec driver module has been loaded, > + the 'power_save' parameter can be set via sysfs as follows: > + > + echo 10 > /sys/module/snd_ac97_codec/parameters/power_save > + > + In this case, the time-out is set to 10 seconds; setting > + the time-out to 1 second (the minimum activation value) > + isn't recommended because many applications try to reopen > + the device frequently. A value of 10 seconds would be a > + good choice for normal operations. > + > + See Documentation/sound/alsa/powersave.txt for more details. > > config SND_AC97_POWER_SAVE_DEFAULT > int "Default time-out for AC97 power-save mode" > @@ -182,4 +198,6 @@ config SND_AC97_POWER_SAVE_DEFAULT > The default time-out value in seconds for AC97 automatic > power-save mode. 0 means to disable the power-save mode. > > + See SND_AC97_POWER_SAVE for more details. > + > endif # SND_DRIVERS > -- > 1.7.2.rc1.9.g5b527 > -- Jiri Kosina SUSE Labs, Novell Inc.

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[alsa-devel] [RFC][PATCH 0/3] ASoC: FSI - codecs settings
by Kuninori Morimoto 15 Jul '10

15 Jul '10

Dear Mark, Liam These patches are FSI - codecs settings Kuninori Morimoto (3): ASoC: ak4642: make sure snd_soc_codec value for ak4642 ASoC: da7210: make sure snd_soc_codec value for da7210 ASoC: Add fsi-codec support But I'm not sure this type of fix is correct or not. So, please check and teach me. Now, FSI driver has 2 connections like below. fsi - ak4642 fsi - da7210 In future, I will send "fsi - HDMI" pair. On AP4 board, I would like to use "fsi - ak4642" and "fsi - HDMI" in same time. To use it, I guess I should modify snd_soc_card :: dai_link correct ? I expected this is correct, I tried to use "fsi - ak4642" and "fsi - da7210" in same time for test. But when I used "soc_codec_dev_ak4642" for snd_soc_device :: codec_dev, da7210 didn't work. because i2c address of ak4642 is used for da7210. Same things happen on "soc_codec_dev_da7210" - ak4642. My patches solve above issue for me. But I'm not sure this is correct way. Best regards -- Kuninori Morimoto

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[alsa-devel] controlling access to the sound card
by Jason Butler 14 Jul '10

14 Jul '10

I have three applications that want to use the sound card, two audio stream players, and a voip phone. Is it possible to use alsa so that if a call comes in on the phone alsa will disconnect the audio players, give exclusive access to the voip phone, and then when the phone is done reconnect the audio players to the sound card. Can this be done using alsa and if so how? Could it be done using just the alsa command line tools and config files and if so how?

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[alsa-devel] [PATCH] wm8727: add a missing return in wm8727_platform_probe
by Axel Lin 14 Jul '10

14 Jul '10

otherwise the error path will always be executed. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8727.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 1072621..9d1df26 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -127,6 +127,8 @@ static __devinit int wm8727_platform_probe(struct platform_device *pdev) goto err_codec; } + return 0; + err_codec: snd_soc_unregister_codec(codec); err: -- 1.7.0.4

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Re: [alsa-devel] [PATCH 4/4] ASoC: DaVinci: pcm, fix underrun by using sram
by troy.kisky＠boundarydevices.com 14 Jul '10

14 Jul '10

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; } On Tue 13/07/10 6:00 AM , "Nori, Sekhar" nsekhar(a)ti.com sent: Hi Troy, On Tue, Nov 17, 2009 at 05:22:34, Troy Kisky wrote: > Fix underruns by using dma to copy 1st to sram > in a ping/pong buffer style and then copying from > the sram to the ASP. This also has the advantage > of tolerating very long interrupt latency on dma > completion. > > Signed-off-by: Troy Kisky [...] > +static int request_ping_pong(struct snd_pcm_substream *substream, > + struct davinci_runtime_data *prtd, > + struct snd_dma_buffer *iram_dma) > +{ > + dma_addr_t asp_src_ping; > + dma_addr_t asp_dst_ping; > + int link; > + struct davinci_pcm_dma_params *dma_data = prtd->params; > + > + /* Request ram master channel */ > + link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, > + davinci_pcm_dma_irq, substream, > + EVENTQ_1); What is the reason for choosing EVENTQ_1 for this channel? EVENTQ_0 is already being used for ASP channel. I imagine it will be much easier to tune the queue usage in the system if all of audio data was using the same queue. I am working on a patch which lets platform specify the event queues for audio DMA. I am not sure if I really need to make a provision for two different queues to be specified - that's why I ask. Thanks, Sekhar The reason is so that the IRAM data can be fetched and used while EVENTQ_1 fetches the next buffer of data from sdram into IRAM. Troy

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[alsa-devel] [PATCH] ASoC: Handle read failures in codec_reg
by Mark Brown 14 Jul '10

14 Jul '10

When a device is powered down volatile registers can't be read so attempts to display codec_reg will show error values, and obviously it is also possible for there to be hardware errors too. Check for errors from reads and display them more clearly when formatting codec_reg. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 22 +++++++++++++++++----- 1 files changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96f9a8e..844ae82 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static int run_delayed_work(struct delayed_work *dwork) /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { - int i, step = 1, count = 0; + int ret, i, step = 1, count = 0; if (!codec->reg_cache_size) return 0; @@ -101,12 +101,24 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (count >= PAGE_SIZE - 1) break; - if (codec->display_register) + if (codec->display_register) { count += codec->display_register(codec, buf + count, PAGE_SIZE - count, i); - else - count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); + } else { + /* If the read fails it's almost certainly due to + * the register being volatile and the device being + * powered off. + */ + ret = codec->read(codec, i); + if (ret >= 0) + count += snprintf(buf + count, + PAGE_SIZE - count, + "%4x", ret); + else + count += snprintf(buf + count, + PAGE_SIZE - count, + "<no data: %d>", ret); + } if (count >= PAGE_SIZE - 1) break; -- 1.7.1

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