appsrc with audio data fails with error

[stic at free.fr]

ok found it.
For those how may have the same issue, I changed format to GST_FORMAT_TIME and now it works.

Cheers

----- Mail original -----
De: stic at free.fr
À: "Discussion of the development of and with GStreamer" <gstreamer-devel at lists.freedesktop.org>
Envoyé: Mardi 8 Juillet 2014 22:59:04
Objet: appsrc with audio data fails with error

Hi gstreamer friends,

I try to send live raw audio data in my pipeline using appsrc, encode it and output it on a rtsp stream, but it fails.
I think it is not related to how I use rtsp because it doesn't work also with other pipelines I tested.
I also already use another appsrc element with rtsp for video data without any issue.

The audio is output for something like 1 second, then it fails with following errors:

W/GStreamer+basesink(12329): 0:03:04.921607999 0x68c27400 gstbasesink.c:3264:gst_base_sink_chain_unlocked:<multiudpsink22> warning: Internal data flow problem.
W/GStreamer+basesink(12329): 0:03:04.922194999 0x68c27400 gstbasesink.c:3264:gst_base_sink_chain_unlocked:<multiudpsink22> warning: Received buffer without a new-segment. Assuming timestamps start from 0.
W/GStreamer+rtspmedia(12329): 0:03:04.923596999 0x68de56c0 jni/gst-rtsp-server/rtsp-media.c:1795:default_handle_message 0x6a1a5240: got warning Internal data flow problem. (gstbasesink.c(3264): gst_base_sink_chain_unlocked (): /GstPipeline:media-pipeline/GstMultiUDPSink:multiudpsink22:
W/GStreamer+rtspmedia(12329): Received buffer without a new-segment. Assuming timestamps start from 0.)

Here is my sample pipeline:
"appsrc is-live=true do-timestamp=true min-latency=0 max-latency=10000000 name=asource ! audio/x-raw,rate=48000,channels=2,format=S16LE ! faac ! queue ! rtpmp4apay pt=97 name=pay0 " // audio mp4a

Any idea about what could be my problem would be really appreciated.

Thanks a lot.
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