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

[alsa-devel] Joining the development
by Ramon Fried 13 Dec '10

13 Dec '10

Hello, First, I'm sorry if I posted it in the wrong place. I would like to participate in the development of ALSA drivers. If there is any project that need assitance I would be happy to contribute. Please notify me. Warm regards, Ramon Fried.

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[alsa-devel] low volume with hda base card (see attached alsa-info output
by Halim Sahin 13 Dec '10

13 Dec '10

Hi, I need to set all mixer stuff to 100% and the output isn't ok. Volume is too low. When I switch to another input on my amp. it's about 500% louder. Please have a look to attached ala-info output. I tested with opensuse 11.2 and also with a newer kernel for the distro 2.6.34. Can you please help? Thx. Halim -- Halim Sahin E-Mail: halim.sahin (at) t-online.de

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[alsa-devel] [PATCH] ASoC: Fix widgets for WM8994/58 AIF2 source control
by Mark Brown 13 Dec '10

13 Dec '10

The compiler really ought to have been warning about unreferenced variables... Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index da48802..b599616 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -780,13 +780,13 @@ SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME, SOC_ENUM("AIF1ADCL Source", aif1adcl_src), SOC_ENUM("AIF1ADCR Source", aif1adcr_src), -SOC_ENUM("AIF2ADCL Source", aif1adcl_src), -SOC_ENUM("AIF2ADCR Source", aif1adcr_src), +SOC_ENUM("AIF2ADCL Source", aif2adcl_src), +SOC_ENUM("AIF2ADCR Source", aif2adcr_src), SOC_ENUM("AIF1DACL Source", aif1dacl_src), SOC_ENUM("AIF1DACR Source", aif1dacr_src), -SOC_ENUM("AIF2DACL Source", aif1dacl_src), -SOC_ENUM("AIF2DACR Source", aif1dacr_src), +SOC_ENUM("AIF2DACL Source", aif2dacl_src), +SOC_ENUM("AIF2DACR Source", aif2dacr_src), SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), -- 1.7.1

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Re: [alsa-devel] [Alsa-user] regarding setting of fragment size in usbaudio driver
by Bill Unruh 13 Dec '10

13 Dec '10

On Mon, 13 Dec 2010, Amit Nagal wrote: > Hi , > > Thanx for the reply . > > Actually on my embedded target , i am not using arecord , but reading > from (/dev/dsp0) capture data . He asked you if arecord works, not whether you are using it. Why are you using the OSS emulation in alsa, rather than using the native alsa? That adds an extra layer and more code. > > i have 2 question : > > i used SNDCTL_DSP_GETBLKSIZE ioctl to find the fragment size of driver > . i found it to be 16k . > > 1) now if the application reads 4k data from userspace , read call > will remain blocked for audio driver's fragment size usb data read ( > 16k) > or app_request_size(4k) ? So read in all 16K into your buffer. But you do not say how your application reads that 4k in? Why do you not publish the fragment of you code that you use so people can see what y ou do. As I understand OSS, the driver reads the usb and puts the output into a buffer. YOur call to the oss system, then reads from that buffer and delivers it to you. That should not block, unless there is nothing in the buffer (ie you forgot to tell the driver to start reading the sound card). > > 2)time taken for read(4k) = time_taken(fragment size read ) or > time_taken(app_request_size=4k) ? > > thanx & regards > amit nagal > > > On Mon, Dec 13, 2010 at 6:04 PM, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Amit Nagal wrote: >>> i am using a OSS based usbaudio application for usb audio streaming . >>> >>> my usb audio device is capture only device . >>> >>> i want to set the fragment size of usbaudio alsa driver to reduce >>> latency problem coming in my application . >>> >>> currently even if my application reads 4k data from userspace , i >>> observed driver will buffer for 100 ms data ( period size data )= 16 k >>> at 44100 frquency >>> >>> and after driver reads 16k data , read call will return with 4k data >>> . so read() call remains blocked for 100ms which is causing some >>> latency problem . >> >> There is no extra buffer in the driver. >> There seems to be something wrong with your algorithm. >> >> Does recording with the arecord tool work? >> >> >> Regards, >> Clemens >> > > ------------------------------------------------------------------------------ > Oracle to DB2 Conversion Guide: Learn learn about native support for PL/SQL, > new data types, scalar functions, improved concurrency, built-in packages, > OCI, SQL*Plus, data movement tools, best practices and more. > http://p.sf.net/sfu/oracle-sfdev2dev > _______________________________________________ > Alsa-user mailing list > Alsa-user(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user > -- William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273 Physics&Astronomy | Advanced Research | Fax: +1(604)822-5324 UBC, Vancouver,BC | Program in Cosmology | unruh(a)physics.ubc.ca Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/

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Re: [alsa-devel] 10 Sec sleep in wait_for_avail_min function
by Clemens Ladisch 13 Dec '10

13 Dec '10

Nitin Arora wrote: > I need to stream audio from my audio device to the host through USB. > While audio is being streamed and songs are being played on the host > and during that I plug out the audio device, Disconnect event takes > too long to get reported and displayed on the screen (on GUI). > > I dug this problem a little deeper and found that when my application is > continuous invoking read on the audio device to capture and audio data > it follows this path: > > .... .. snd_pcm_lib_read -> snd_pcm_lib_read1 -> snd_pcm_capture_avail > > and in the case when there is no data left (MY DISCONNECT CASE) > it invokes as follows: > > .... .. snd_pcm_lib_read -> snd_pcm_lib_read1 -> wait_for_avail_min > > and in wait_for_avail_min it sleeps for 10 sec. This ten-second timeout is intended for devices where the interrupt does not work for some reason. In the case of USB disconnection, the driver should stop the stream immediately. Does this happen if you try to record with the arecord tool? Regards, Clemens -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] Custom IOCTL call
by Radivoje Jovanovic 13 Dec '10

13 Dec '10

I am writing custom IOCTL calls for my card. I have made hwdep interface and set ioctl ops call for it. But, now the question is which device I need to open in my user space app to talk to correct ioctl? I have tried /dev/snd/controlC0 and /dev/snd/pcmC0D0p Both of theses returned invalid IOCTL. Cheers Ogi

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[alsa-devel] Unable to set ALSA buffer size
by Irfan Shaikh 13 Dec '10

13 Dec '10

Hello All, I am trying to set my alsa buffer size as 1024*8. But my playback device is not able to set to requested ALSA buffer size. snd_pcm_hw_params_set_periods(handle, params, 8, 0); snd_pcm_hw_params_set_period_size(handle, params, 1024, 0); snd_pcm_uframes_t frames1; frames1=1024*8; retVal=snd_pcm_hw_params_set_buffer_size_near(handle, params, &frames1); snd_pcm_hw_params_get_buffer_size(params, (snd_pcm_uframes_t *) &val); /*Expected Output*/ ==============>>>>>>> buffer size = 8192 frames /*Output*/ ==============>>>>>>> buffer size = 1325 frames Return value for above APIs is coming as zero. but still i am unable to get expected result. Is there any other forcible way to do the above. Regards, SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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[alsa-devel] Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
by Jonathan Gardner 13 Dec '10

13 Dec '10

I'm a developer who spends most of his time in Python writing web apps but I know my way around C. I've been using Linux for over 10 years now and I thought it's time to get serious about making Linux work for the hardware I use. I'm self-employed now, so I don't have to deal with any legal departments and can help out to my heart's desire. Disclaimer: I intend to learn how to help debug these issues and hack on some fairly low-level pieces of software and get a clear sense of how hardware really works. Mostly this is a "scratch-my-own-itch" thing, but it's always bothered me that I've never truly understood what happens below the KDE interface. I recently bought a Gigabyte 880GM-USB3 motherboard. http://www.gigabyte.us/products/product-page.aspx?pid=3412#sp I think "Realtek ALC892" describes the hardware for audio. I'm running Fedora 13 with KDE which seems to have alsa 1.0.23 running. I don't know much about hardware or audio, but I do know that it's not working 100% for me with this motherboard. There are two issues that stand out. First, with my older motherboard, I could control the various output and input levels of the channels. With this motherboard, I only have one channel "Internal Audio Analog Stereo". This board is supposed to support up to Dolby 7.1 surround sound. I want to make this computer my hub for all my devices. Obviously this isn't possible. Second, I never really got MIDI working, at least not that I recall in my 10 years of Linux. I know MIDI is possible, I just don't know what it takes. I've been playing with musescore which apparently plays the notes over MIDI so you can hear what you wrote, which would be nice. RTFM is certainly acceptable advice, if you can show me which FM is preferred. I'll also write some documentation if it helps. -- Jonathan Gardner jgardner(a)jonathangardner.net

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[alsa-devel] "Unknown symbol" with alsa snapshot (tiwai)
by Simos Xenitellis 12 Dec '10

12 Dec '10

Hi, I am trying out the latest snapshot of Alsa with Ubuntu 10.10. It compiles without errors. However, when the modules are to be loaded, there are Unknown symbol errors: [ 332.243113] snd: Unknown symbol unregister_sound_special (err 0) [ 332.243379] snd: Unknown symbol register_sound_special_device (err 0) [ 332.245467] snd_timer: Unknown symbol snd_info_register (err 0) [ 332.245631] snd_timer: Unknown symbol snd_info_create_module_entry (err 0) [ 332.245839] snd_timer: Unknown symbol snd_info_free_entry (err 0) [ 332.246210] snd_timer: Unknown symbol __snd_printk (err 0) [ 332.246373] snd_timer: Unknown symbol snd_iprintf (err 0) [ 332.246607] snd_timer: Unknown symbol snd_ecards_limit (err 0) [ 332.246834] snd_timer: Unknown symbol snd_oss_info_register (err 0) [ 332.246997] snd_timer: Unknown symbol snd_unregister_device (err 0) [ 332.247224] snd_timer: Unknown symbol snd_device_new (err 0) [ 332.247625] snd_timer: Unknown symbol snd_register_device_for_dev (err 0) Is this a module loading order issue? Has there been a change in the latest kernel that causes the problem? Simos

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[alsa-devel] [PATCH v2 04/16] ASoC: twl6040: Set default gains to minimun value
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Updated default values to improve power consumption. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b575fd3..27d6d53 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -86,19 +86,19 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x1B, /* TWL6040_LINEGAIN 0x0F */ 0x00, /* TWL6040_HSLCTL 0x10 */ 0x00, /* TWL6040_HSRCTL 0x11 */ - 0x00, /* TWL6040_HSGAIN 0x12 */ - 0x00, /* TWL6040_EARCTL 0x13 */ + 0xFF, /* TWL6040_HSGAIN 0x12 */ + 0x1E, /* TWL6040_EARCTL 0x13 */ 0x00, /* TWL6040_HFLCTL 0x14 */ - 0x00, /* TWL6040_HFLGAIN 0x15 */ + 0x1D, /* TWL6040_HFLGAIN 0x15 */ 0x00, /* TWL6040_HFRCTL 0x16 */ - 0x00, /* TWL6040_HFRGAIN 0x17 */ + 0x1D, /* TWL6040_HFRGAIN 0x17 */ 0x00, /* TWL6040_VIBCTLL 0x18 */ 0x00, /* TWL6040_VIBDATL 0x19 */ 0x00, /* TWL6040_VIBCTLR 0x1A */ 0x00, /* TWL6040_VIBDATR 0x1B */ 0x00, /* TWL6040_HKCTL1 0x1C */ 0x00, /* TWL6040_HKCTL2 0x1D */ - 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x02, /* TWL6040_GPOCTL 0x1E */ 0x00, /* TWL6040_ALB 0x1F */ 0x00, /* TWL6040_DLB 0x20 */ 0x00, /* not used 0x21 */ -- 1.7.0.4

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