[Bug 781824] New: RTP audio packets are sent as a burst, not in real-time, with rawaudioparse

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=781824

            Bug ID: 781824
           Summary: RTP audio packets are sent as a burst, not in
                    real-time, with rawaudioparse
    Classification: Platform
           Product: GStreamer
           Version: git master
                OS: Linux
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: wegherfe at gmail.com
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

Created attachment 350526
  --> https://bugzilla.gnome.org/attachment.cgi?id=350526&action=edit
Sample audio file used to reproduce the issue

Hi all.

I have to implement a RTP streaming pipeline for S16LE 48KHz raw audio content,
sourced from a Unix named pipe. My pipeline is the following:

gst-launch-1.0 blocksize=1388 location=/home/wegherfe/pipe-sink ! queue !
"audio/x-unaligned-raw, format=S16LE, channels=2, rate=48000,
layout=interleaved" ! rawaudioparse use-sink-caps=true ! audiorate !
audioconvert dithering=0 noise-shaping=0 ! rtpL16pay pt=99 ! queue ! udpsink
host=192.168.1.2 port=6004

In short, the pipeline sets the named pipe as file source, raw parses it with
caps, converts to S16 BE format and encapsulates data into RTP payload. The two
queues are used to split job into 3 tasks: reading, processing, transmitting.

In order to test the pipeline, just create the Unix named pipe (as root) and
write raw audio content into it (I have attached the sample raw audio I used in
my tests: it says "front center", taken from my Ubuntu PC: <nabble_a
href="audio.raw">audio.raw</nabble_a>):

mknode /home/wegherfe/pipe-sink p
cat audio.raw > /home/wegherfe/pipe-sink

The server pipeline and test commands are run on my laptop, which is connected
to a small embedded device, working as RTP client, which has limited buffering
and processing capabilities.

The problem is the following: audio is played fluently and properly at first
invocation of cat command, but it is played with glitches and bad timing
sometimes at following invocations. Usually, the first word "front" is eaten,
cannot be heard well, while "center" is usually fine.

By running tcpdump, I noticed that, with following invocations of cat command,
UDP packets are sent as a burst and not in real-time. I think this causes issue
to my client device, which has a little buffer.

Notices:
1) if I remove rawaudioparse and use wavparse, audio is always ok:

gst-launch-1.0 filesrc location=/home/wegherfe/Front_Center.wav ! wavparse !
queue ! audioresample ! audioconvert dithering=0 noise-shaping=0 ! rtpL16pay
pt=99 ! queue ! udpsink host=192.168.1.2 port=6004

so I think the problem is in timestamps generated by rawaudioparse;

2) if I set blocksize to smaller values, the client device does not work
properly
3) queue elements seems to mitigate the issue a bit

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