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, 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@alsa-project.org; linux-audio-dev@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@alsa-project.org alsa-devel-bounces@alsa-project.org on behalf of Srinivasan S srinivasan.s@tataelxsi.co.in Sent: Wednesday, June 3, 2015 11:35 AM To: Clemens Ladisch Cc: alsa-devel@alsa-project.org; linux-audio-dev@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@ladisch.de Sent: Monday, June 1, 2015 8:09 PM To: Srinivasan S Cc: alsa-devel@alsa-project.org; linux-audio-dev@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:
- 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 _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Srinivasan S wrote:
Am facing overrun & underrun issues, when I run the the above GSM application with the attached asound.conf
The sound card and the GSM streams are not synchronized. You need to compensate for the drift between the clocks, typically by resampling. (Jack's alsa_in/alsa_out would automatically do this.)
Regards, Clemens
Dear Clemens,
Thanks a lot for your support Clemens
As you suggested, When I try to use alsa jack plugin, am facing the below error
root@lifeline:/# vi etc/asound.conf pcm.rawjack { type jack playback_ports { 0 system:playback_1 1 system:playback_2 } capture_ports { 0 system:capture_1 1 system:capture_2 } }
pcm.jack { type plug slave { pcm "rawjack" } hint { description "JACK Audio Connection Kit" } }
pcm.!default { type plug slave { pcm "rawjack" } }
root@lifeline:/opt/tunstall/audio# aplay -D pcm.jack TangoForTajMusic11.wav ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_jack.so aplay: main:722: audio open error: No such device or address root@lifeline:/opt/tunstall/audio#
I feel order to resolve this error,
1. I feel that there is no libasound_module_pcm_jack.so, I have downloaded the alsa-plugins-1.0.29 could you please help me out how this can be cross compiled for my TI AM335x platform
2. Could you please let me know to cross compile & install the package libasound2-dev
Could you please help me out in resolving this issue
Kindly do the needful as early as possible
Many Thanks a lot for your support once again w.r.t this thread so far
________________________________________ From: Clemens Ladisch clemens@ladisch.de Sent: Wednesday, July 1, 2015 7:41 PM To: Srinivasan S Cc: alsa-devel@alsa-project.org; linux-audio-dev@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:
Am facing overrun & underrun issues, when I run the the above GSM application with the attached asound.conf
The sound card and the GSM streams are not synchronized. You need to compensate for the drift between the clocks, typically by resampling. (Jack's alsa_in/alsa_out would automatically do this.)
Regards, Clemens
Dear Clemens
Could you please kindly help me out to cross_compile this alsa_plugins, so that I can resolve the below error & & get jack plugins working as you suggested earlier
Many Thanks in advance Srinivasan S ________________________________________ From: Srinivasan S Sent: Wednesday, July 8, 2015 8:55 PM To: Clemens Ladisch Cc: alsa-devel@alsa-project.org; linux-audio-dev@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,
Thanks a lot for your support Clemens
As you suggested, When I try to use alsa jack plugin, am facing the below error
root@lifeline:/# vi etc/asound.conf pcm.rawjack { type jack playback_ports { 0 system:playback_1 1 system:playback_2 } capture_ports { 0 system:capture_1 1 system:capture_2 } }
pcm.jack { type plug slave { pcm "rawjack" } hint { description "JACK Audio Connection Kit" } }
pcm.!default { type plug slave { pcm "rawjack" } }
root@lifeline:/opt/tunstall/audio# aplay -D pcm.jack TangoForTajMusic11.wav ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/dlmisc.c:252:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_jack.so aplay: main:722: audio open error: No such device or address root@lifeline:/opt/tunstall/audio#
I feel order to resolve this error,
1. I feel that there is no libasound_module_pcm_jack.so, I have downloaded the alsa-plugins-1.0.29 could you please help me out how this can be cross compiled for my TI AM335x platform
2. Could you please let me know to cross compile & install the package libasound2-dev
Could you please help me out in resolving this issue
Kindly do the needful as early as possible
Many Thanks a lot for your support once again w.r.t this thread so far
________________________________________ From: Clemens Ladisch clemens@ladisch.de Sent: Wednesday, July 1, 2015 7:41 PM To: Srinivasan S Cc: alsa-devel@alsa-project.org; linux-audio-dev@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:
Am facing overrun & underrun issues, when I run the the above GSM application with the attached asound.conf
The sound card and the GSM streams are not synchronized. You need to compensate for the drift between the clocks, typically by resampling. (Jack's alsa_in/alsa_out would automatically do this.)
Regards, Clemens
participants (2)
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Clemens Ladisch -
Srinivasan S