Audio via shmem and preserve clock for adder plugin?

[Simon Brandner]

Hello together,

I would like to realize interprocess audio streaming with gstreamer -
as lightweight as possible, so things like UDP are preferred to be
avoided.
Therefore I am using the shmem plugin with  GStreamer version 0.10.36
under Ubuntu 12.04. (in Ubuntu 12.10 the shmem plugin seems to have
been patched into "good" - I have not tried with that yet).

As audio-source I use:
gst-launch-0.10 filesrc location=../Music/title.mp3  !  mad !
audioconvert ! "audio/x-raw-int,endianness=1234,signed=(boolean)true,width=(int)16,depth=(int)16,rate=(int)44100,channels=(int)2"
 ! shmsink socket-path=/tmp/shmsink shm-size=10000000
wait-for-connection=false preroll-queue-len=5

The playing part is:
gst-launch-0.10 shmsrc socket-path=/tmp/shmsink !
"audio/x-raw-int,endianness=1234,signed=(boolean)true,width=(int)16,depth=(int)16,rate=(int)44100,channels=(int)2"
! audioconvert ! alsasink

This constellation just works fine.

For full functionality I would also need an adder to combine multiple
audio sources in a pipeline, so for one source it is looking like:
gst-launch-0.10 shmsrc socket-path=/tmp/shmsink !
"audio/x-raw-int,endianness=1234,signed=(boolean)true,width=(int)16,depth=(int)16,rate=(int)44100,channels=(int)2"
! audioconvert ! adder !  alsasink

// Unfortunately then, I get a crackling in the sound and several
error messages:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...

** (gst-launch-0.10:32308): CRITICAL **: gst_audio_buffer_clip:
assertion `segment->format == GST_FORMAT_TIME || segment->format ==
GST_FORMAT_DEFAULT' failed
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSinkClock

** (gst-launch-0.10:32308): CRITICAL **: gst_audio_buffer_clip:
assertion `segment->format == GST_FORMAT_TIME || segment->format ==
GST_FORMAT_DEFAULT' failed

** (gst-launch-0.10:32308): CRITICAL **: gst_audio_buffer_clip:
assertion `segment->format == GST_FORMAT_TIME || segment->format ==
GST_FORMAT_DEFAULT' failed //This message comes cyclically, just as
the crackling noise.


I would assume that through the shmem, a clock/timestamp information
is lost, because a single pipeline without a shmem in beetween works
fine with the adder. The message "New clock: GstAudioSinkClock" also
points to this, and that the adder requires one. How could I tunnel
the timestamp through the shmem, or create a valid new one? Could a
small buffer eliminate the problem?

Any way to solve the problem differently would also be welcome.


Thank you.
Regards,

Simon



PS:
Not very relevant for now, but a short try with GStreamer 1.0.2 even
failed without shmem and adder:
gst-launch-1.0 filesrc location=../Music/title.mp3  !  mad !
audioconvert ! "audio/x-raw,endianness=1234,signed=(boolean)true,width=(int)16,depth=(int)16,rate=(int)44100,channels=(int)2"
 ! alsasink
"ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0:
Internal data flow error.
Additional debug info:
gstbasesrc.c(2791): gst_base_src_loop ():
/GstPipeline:pipeline0/GstFileSrc:filesrc0:
streaming task paused, reason not-negotiated (-4)"
My hope was, it would be more simple there..
_______________________________________________
gstreamer-devel mailing list
[hidden email]
http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

[Tristan Matthews-2]

2012/11/12 Simon Brandner <[hidden email]>:

> Hello together,
>
> I would like to realize interprocess audio streaming with gstreamer -
> as lightweight as possible, so things like UDP are preferred to be
> avoided.
> Therefore I am using the shmem plugin with  GStreamer version 0.10.36
> under Ubuntu 12.04. (in Ubuntu 12.10 the shmem plugin seems to have
> been patched into "good" - I have not tried with that yet).
>
> As audio-source I use:
> gst-launch-0.10 filesrc location=../Music/title.mp3  !  mad !
> audioconvert ! "audio/x-raw-int,endianness=1234,signed=(boolean)true,width=(int)16,depth=(int)16,rate=(int)44100,channels=(int)2"
>  ! shmsink socket-path=/tmp/shmsink shm-size=10000000
> wait-for-connection=false preroll-queue-len=5
>
> The playing part is:
> gst-launch-0.10 shmsrc socket-path=/tmp/shmsink !
> "audio/x-raw-int,endianness=1234,signed=(boolean)true,width=(int)16,depth=(int)16,rate=(int)44100,channels=(int)2"
> ! audioconvert ! alsasink
>
> This constellation just works fine.
>
> For full functionality I would also need an adder to combine multiple
> audio sources in a pipeline, so for one source it is looking like:
> gst-launch-0.10 shmsrc socket-path=/tmp/shmsink !
> "audio/x-raw-int,endianness=1234,signed=(boolean)true,width=(int)16,depth=(int)16,rate=(int)44100,channels=(int)2"
> ! audioconvert ! adder !  alsasink
>
> // Unfortunately then, I get a crackling in the sound and several
> error messages:
> Setting pipeline to PAUSED ...
> Pipeline is PREROLLING ...
>
> ** (gst-launch-0.10:32308): CRITICAL **: gst_audio_buffer_clip:
> assertion `segment->format == GST_FORMAT_TIME || segment->format ==
> GST_FORMAT_DEFAULT' failed
> Pipeline is PREROLLED ...
> Setting pipeline to PLAYING ...
> New clock: GstAudioSinkClock
>
> ** (gst-launch-0.10:32308): CRITICAL **: gst_audio_buffer_clip:
> assertion `segment->format == GST_FORMAT_TIME || segment->format ==
> GST_FORMAT_DEFAULT' failed
>
> ** (gst-launch-0.10:32308): CRITICAL **: gst_audio_buffer_clip:
> assertion `segment->format == GST_FORMAT_TIME || segment->format ==
> GST_FORMAT_DEFAULT' failed //This message comes cyclically, just as
> the crackling noise.

Have you tried with the JACK plugins? If you're looking to do
interprocess audio,
you may have more success with them.

> PS:
> Not very relevant for now, but a short try with GStreamer 1.0.2 even
> failed without shmem and adder:
> gst-launch-1.0 filesrc location=../Music/title.mp3  !  mad !
> audioconvert ! "audio/x-raw,endianness=1234,signed=(boolean)true,width=(int)16,depth=(int)16,rate=(int)44100,channels=(int)2"
>  ! alsasink
> "ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0:
> Internal data flow error.
> Additional debug info:
> gstbasesrc.c(2791): gst_base_src_loop ():
> /GstPipeline:pipeline0/GstFileSrc:filesrc0:
> streaming task paused, reason not-negotiated (-4)"
> My hope was, it would be more simple there..

Do you have the same error if you leave the caps out?

--
Tristan Matthews
web: http://tristanswork.blogspot.com
_______________________________________________
gstreamer-devel mailing list
[hidden email]
http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel