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

[alsa-devel] How to structure CPU DAI
by jonsmirl＠gmail.com 19 Nov '12

19 Nov '12

The LPC3130 has two independent I2S channels but it is not two copies of the same hardware block. Instead it is a single hardware block implementing the two channels. Some setup registers contain setup bits for both channels. So I have one CPU DAI with two independent CODECs hung off from it. So in a function like hw_params() for the CPU DAI, how do I know which I2S channel I'm supposed to be using? At the point where I created the soc_card I know the right channel. But I can't use the card pointer in the CPU DAI, since there are two cards and only on CPU DAI. The card pointer points to the last one registered. Can I put the info into the substream? Or should I register two CPU DAIs from a single I2S device driver and give then two different names? A device tree like this describes the hardware.. i2s@16000000 { #address-cells = <2>; #size-cells = <0>; compatible = "nxp,lpc31xx-i2s"; reg = <0x16000000 0x1000>; i2s0:i2s@0 { interrupts = <13 15>; reg = <0x80 0x180>; /*receive-slave;*/ /* default is master */ }; i2s1:i2s@1 { interrupts = <14 16>; reg = <0x100 0x200>; /*receive-slave;*/ /* default is master */ }; }; sound@1 { compatible = "ea,ea3131-uda1380"; audio-codec = <&uda1380-0>; i2s-controller = <&i2s0>; }; sound@2 { compatible = "ea,ea3131-uda1380"; audio-codec = <&uda1380-1>; i2s-controller = <&i2s1>; }; -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] USB hardware that supports implicit feedback?
by Daniel Griscom 18 Nov '12

18 Nov '12

[I posted this to the user list with no response; perhaps it was too technical.] I'm having trouble getting implicit feedback working with my custom capture/playback USB audio class device, even with the latest kernel. I presume it's a problem with my device's configuration, but can't figure out exactly what the problem is (after a whole lotta investigation). Specifically, my device captures an AES/EBU data stream (at a frame rate dictated by an external device), sends it to the computer via USB and ALSA, and then takes a processed audio stream from the computer via ALSA and USB, and sends it out at the same frame rate. The output stream should have the exact same frame rate as the input stream, governed by implicit feedback. I understand that only recently has ALSA included support for implicit feedback. I'm running with kernel 3.6.6, which I believe includes the support, but still can't get it working. I'd really love to see something functioning in this mode so I can determine where the problem is. Is there any commercial USB hardware out there that supports implicit feedback? I assume the ALSA developers are testing against something, but I can't find anything appropriate. Or, is there source available that runs on some USB-capable micro board that supports it? Thanks, Dan -- Daniel T. Griscom griscom(a)suitable.com Suitable Systems http://www.suitable.com/ 1 Centre Street, Suite 204 (781) 665-0053 Wakefield, MA 01880-2400

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[alsa-devel] [patch] sound: oss/sb_audio: cap value in sb201_audio_set_speed()
by Dan Carpenter 18 Nov '12

18 Nov '12

We set "s" before we have capped "speed" so it could be the wrong value. This could lead to a divide by zero bug. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- I fixed a similar bug in this file in Aug. I'm not sure how I missed this one. I don't see any others I missed. diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index b2b3c01..048439a 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -442,7 +442,7 @@ static int sb201_audio_set_speed(int dev, int speed) { sb_devc *devc = audio_devs[dev]->devc; int tmp; - int s = speed * devc->channels; + int s; if (speed > 0) { @@ -452,6 +452,7 @@ static int sb201_audio_set_speed(int dev, int speed) speed = 44100; if (devc->opened & OPEN_READ && speed > 15000) speed = 15000; + s = speed * devc->channels; devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; tmp = 256 - devc->tconst; speed = ((1000000 + tmp / 2) / tmp) / devc->channels;

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[alsa-devel] [PATCH] sound: use bitmap_weight
by Joe Perches 17 Nov '12

17 Nov '12

Use bitmap_weight to count the total number of bits set in bitmap. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/usb/endpoint.c | 7 +------ 1 files changed, 1 insertions(+), 6 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 34de6f2..51a9aa3 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -485,15 +485,10 @@ __exit_unlock: static int wait_clear_urbs(struct snd_usb_endpoint *ep) { unsigned long end_time = jiffies + msecs_to_jiffies(1000); - unsigned int i; int alive; do { - alive = 0; - for (i = 0; i < ep->nurbs; i++) - if (test_bit(i, &ep->active_mask)) - alive++; - + alive = bitmap_weight(&ep->active_mask, ep->nurbs); if (!alive) break;

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[alsa-devel] About the VT1602 codec iic timing
by l9jj＠163.com 17 Nov '12

17 Nov '12

Hello guys,I am debugging the audio Codec VT1602 which producted by VIA,dos anyone have its datasheet.i am not sure what its iic read timing is.i got a copy of it but do not found anything about the timing from the descriptions. Thanks for any suggestion. Best regards.

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[alsa-devel] [PATCH 0/2] alsa-plugins: fix PulseAudio prebuf
by Rémi Denis-Courmont 16 Nov '12

16 Nov '12

Hello, The following changes since commit 9df5dc7dd628b0d692cf896c210a134e85e88915: Release v1.0.26 (2012-09-06 09:52:58 +0200) make the ALSA<->PulseAudio PCM plugin abide by sw_params. This fixes frequently broken playback with VLC, or potentially any other application that expects playback to start within a short or configured time. Best regards, ---------------------------------------------------------------- Rémi Denis-Courmont (2): pcm_pulse: do not trigger immediately at start pcm_pulse: set prebuf parameter according to software parameters pulse/pcm_pulse.c | 37 ++++++++++++++++++++++++++++--------- 1 file changed, 28 insertions(+), 9 deletions(-) -- Rémi Denis-Courmont http://www.remlab.net/

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[alsa-devel] [PATCH 0/3] Updated names for ASoC OMAP HDMI drivers
by Ricardo Neri 16 Nov '12

16 Nov '12

Hi Mark, Liam, Now that the OMAPDSS HDMI driver will create the platform device for HDMI audio [1], the ASoC drivers need to be updated as well. Also, I took the opportunity to give the drivers more meaningful and descriptive names. The last patch in the series makes the CPU DAI driver look cleaner by obtaining its resources from the platform device and not from hard-coded value inside the driver. Also, I still have pending a couple of patches to update the platform device creation. I wanted to submit these patches first in case there are comments about the naming I propose. BR, Ricardo [1]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg79795.html Ricardo Neri (3): ASoC: OMAP: HDMI: Update machine driver name ASoC: OMAP: HDMI: Update CPU DAI driver name ASoC: OMAP: HDMI: Obtain DMA port from resources sound/soc/omap/omap-hdmi-card.c | 4 ++-- sound/soc/omap/omap-hdmi.c | 5 ++--- sound/soc/omap/omap-hdmi.h | 2 -- 3 files changed, 4 insertions(+), 7 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: codecs/wm8978: Remove remuxing of GPIO1 pin
by Eric Millbrandt 16 Nov '12

16 Nov '12

Remove remuxing GPIO1. Leave control of this up to the platform device. Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- sound/soc/codecs/wm8978.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index e329187..2fcc6da 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -529,9 +529,6 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) return idx; wm8978->mclk_idx = idx; - - /* GPIO1 into default mode as input - before configuring PLL */ - snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); } else { return -EINVAL; } -- 1.7.2.5

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[alsa-devel] [PATCH v2] ASoC: bells: Up to 512fs
by Dimitris Papastamos 16 Nov '12

16 Nov '12

Optimize performance by providing a 512fs based clkin. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 3d7ace7..b5065ee 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -28,7 +28,7 @@ #define MCLK_RATE 24576000 #define SYS_AUDIO_RATE 44100 -#define SYS_MCLK_RATE (SYS_AUDIO_RATE * 256) +#define SYS_MCLK_RATE (SYS_AUDIO_RATE * 512) #define DAI_AP_DSP 0 #define DAI_DSP_CODEC 1 -- 1.8.0

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[alsa-devel] [PATCH] ASoC: arizona: Fix typo - Swap value in 48k_rates[] and 44k1_rates[]
by Dimitris Papastamos 16 Nov '12

16 Nov '12

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index c03b65a..054967d 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -268,7 +268,7 @@ EXPORT_SYMBOL_GPL(arizona_out_ev); static unsigned int arizona_sysclk_48k_rates[] = { 6144000, 12288000, - 22579200, + 24576000, 49152000, 73728000, 98304000, @@ -278,7 +278,7 @@ static unsigned int arizona_sysclk_48k_rates[] = { static unsigned int arizona_sysclk_44k1_rates[] = { 5644800, 11289600, - 24576000, + 22579200, 45158400, 67737600, 90316800, -- 1.8.0

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