[alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS

[Srinivasan S]

Dear Clemens,

Many Thanks a lot for all your support w.r.t this thread so far

Am facing overrun & underrun issues, when I run the the above GSM application with the attached asound.conf

I have some understanding by googling & found why this overrun & underrun issues occurs ie., unable to transfer audio frames from hardware buffers to application buffers(for capture) & the viceversa(for playback), 

As I googled, I didn't get detailed information with bits & pieces here & there, could you please help me out in resolving this overrun & underrun issue ie., how to set this buffer size or only buffer_size is causing this issue or any other parameters like period_size, period_time values should also be considered inorder to fix this overrun & underrun issue

Now I have randomly set the buffer size ie., 4097 for 8Khz sampling rate

But I have no idea logically how this buffer size 4097 is working for 8Khz & when I copied the same buffer size to 48Khz, the audio gets cut & again overrun & underrun issue is seen

Could you please help me out how to fix this overrun & underrun issue logically, rather than I have that an understanding of programming some random values like 4097

Is it possible to fix this issue in asound.conf itself, rather than using multithreading in GSM application

So that fixing these types of issues henceforth will be faster & addressed logically & will be helpful for others as well


Kindly do the needful as early as possible

Many Thanks in advance once again



________________________________________
From: Srinivasan S
Sent: Wednesday, July 1, 2015 12:54 PM
To: Clemens Ladisch
Cc: alsa-devel at alsa-project.org; linux-audio-dev at lists.linuxaudio.org
Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way        call    to the real sound card UDA1345TS

Dear Clemens,

Many Thanks a lot for all your support w.r.t this thread so far

Am facing overrun & underrun issues on TI AM335x, when I run the the above GSM application with the attached asound.conf

I have some understanding by googling & found why this overrun & underrun issues occurs ie., unable to transfer audio frames from hardware buffers to application buffers(for capture) & the viceversa(for playback),

As I googled, I didn't get detailed information with bits & pieces here & there, could you please help me out in resolving this overrun & underrun issue ie., how to set this buffer size or only buffer_size is causing this issue or any other parameters like period_size, period_time values should also be considered inorder to fix this overrun & underrun issue

Now I have randomly set the buffer size ie., 4097 for 8Khz sampling rate

But I have no idea logically how this buffer size 4097 is working for 8Khz & when I copied the same buffer size to 48Khz, the audio gets cut & again overrun & underrun issue is seen

Could you please help me out how to fix this overrun & underrun issue logically, rather than I have that an understanding of programming some random values like 4097

Is it possible to fix this issue in asound.conf itself, rather than using multithreading in GSM application

So that fixing these types of issues henceforth will be faster & addressed logically & will be helpful for others as well


Kindly do the needful as early as possible

Many Thanks in advance once again




________________________________________
From: alsa-devel-bounces at alsa-project.org <alsa-devel-bounces at alsa-project.org> on behalf of Srinivasan S <srinivasan.s at tataelxsi.co.in>
Sent: Wednesday, June 3, 2015 11:35 AM
To: Clemens Ladisch
Cc: alsa-devel at alsa-project.org; linux-audio-dev at lists.linuxaudio.org
Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way        call    to the real sound card UDA1345TS

Dear Clemens

As I didn't find Jack mailing lists by googling, could you please
loop Jack mailing lists to this mail chain

Kindly do the needful as early as possible
Many Thanks in advance
________________________________________
From: Clemens Ladisch <clemens at ladisch.de>
Sent: Monday, June 1, 2015 8:09 PM
To: Srinivasan S
Cc: alsa-devel at alsa-project.org; linux-audio-dev at lists.linuxaudio.org
Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call   to the real sound card UDA1345TS

Srinivasan S wrote:
> 2. Could you please let me know, I have downloaded jack-1.9.10.tar.bz2, how this needs to be installed in my rootfs

IIRC Jack uses some non-standard build system.  Try asking on the Jack
mailing list how to cross-compile it.

Please note that the ALSA Jack plugin is part of the alsa-plugins
package.

And as I already mentioned, it is unlikely that Jack will use less CPU
than dshare.


Regards,
Clemens
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