Using rtpjitterbuffer to handle RTP packets based on their timestamp only (mode 0)

[Andrea Lo Pumo]

By manually setting the PTS (Presentation Timestamp) and using the
liveadder plugin before wavenc and not using rtpjitterbuffer, it actually
works: the missing packets which correspond to silence are rightly
considered as silence.
I found the liveadder plugin by mistake. I would suggest to add a reference
to liveadder in the rtpjitterbuffer documentation. Something like: "if you
would like to create silence for missing RTP packets, use the liveadder
plugin".

Pipeline:
    appsrc caps="application/x-rtp, media=(string)audio,
clock-rate=(int)8000, encoding-name=(string)PCMA" emit-signals=true
format=3 block=true ! decodebin ! liveadder ! wavenc ! filesink
location=/tmp/out-file.wav

However, I would like rtpjitterbuffer to set the PTS for me using the RTP
timestamps. I am not able to achieve this. Even if I use the liveadder
plugin, the pipeline blocks at EOS.

Pipeline:
    appsrc caps="application/x-rtp, media=(string)audio,
clock-rate=(int)8000, encoding-name=(string)PCMA" emit-signals=true
format=3 block=true ! rtpjitterbuffer mode=0 latency=0
max-dropout-time=3600000 ! decodebin ! liveadder ! wavenc ! filesink
location=/tmp/out-file.wav

I tried with and without format=3 and with latency=0.

Best regards.




Il giorno gio 3 ott 2019 alle ore 18:36 Andrea Lo Pumo <alopumo at movia.biz>
ha scritto:

> I played with gstreamer-java, using an appsrc as the source of RTP packets
> given to rtpjitterbuffer (instead of rtpfilesrc).
> In this test, I set the Presentation Timestamp of all generated buffers
> (in the right way, i.e. (timestamp-firstTimestamp)/160*20 milliseconds).
> And I removed "mode=0" as an option to rtpjitterbuffer.
> However, the behaviour, is the same. After, the source has finished (
> appsrc.endOfStream() ), the pipeline remains blocked.
> This does not happen if the streams does not have jumps in the rtp
> timestamp.
>
> Il giorno gio 3 ott 2019 alle ore 09:50 Andrea Lo Pumo <alopumo at movia.biz>
> ha scritto:
>
>> P.S. here attached the source code of rtpfilesrc
>>
>> Il giorno mer 2 ott 2019 alle ore 18:14 Andrea Lo Pumo <alopumo at movia.biz>
>> ha scritto:
>>
>>> I have a dump of RTP traffic (a single ssrc stream) and I would like to
>>> decode it with gstreamer.
>>> What I am trying works, but it fails if there are jumps in the
>>> timestamps because there is silence in between (no packets sent during the
>>> silence). The result is an audio without the right silence.
>>> Note that the RTP timestamp advances correctly, so one knows how much
>>> time passed before the jump and after it.
>>> Thus, I was trying "rtpjitterbuffer mode=0" however, when I use mode=0,
>>> I get the following:
>>> ....
>>> INFO         rtpjitterbuffer
>>> gstrtpjitterbuffer.c:2559:calculate_expected:<rtpjitterbuffer0> lost
>>> packets (1, #23086->#23086) duration too large 0:00:00.800000000 >
>>> 0:00:00.200000000, consider 0 lost (0:00:00.200000000)
>>> INFO         rtpjitterbuffer
>>> gstrtpjitterbuffer.c:2559:calculate_expected:<rtpjitterbuffer0> lost
>>> packets (2, #23112->#23113) duration too large 0:00:00.740000000 >
>>> 0:00:00.200000000, consider 1 lost (0:00:00.293333332)
>>> ...
>>> INFO                 basesrc
>>> gstbasesrc.c:2839:gst_base_src_loop:<rtpfilesrc0> pausing after
>>> gst_base_src_get_range() = eos
>>> INFO                    task
>>> gsttask.c:316:gst_task_func:<rtpfilesrc0:src> Task going to paused
>>>
>>> and at this point it remains blocked.
>>>
>>> This is the pipeline I am using:
>>>
>>>     gst-launch-1.0 rtpfilesrc location=/tmp/input.rtp !
>>> "application/x-rtp, media=(string)audio, clock-rate=(int)8000,
>>> encoding-name=(string)PCMA" ! rtpjitterbuffer mode=0
>>> max-rtcp-rtp-time-diff=-1 ! decodebin ! wavenc ! filesink
>>> location=/tmp/out.wav
>>>
>>> Rtpfilesrc is a plugin I have written that reads an RTP dump and becomes
>>> a source of UDP packets. I used PushSrc as the base class and have
>>> overridden "fill".
>>> In init() I am using the following configuration:
>>>   gst_base_src_set_live (GST_BASE_SRC (rtpfilesrc), TRUE);
>>>   gst_base_src_set_format (GST_BASE_SRC (rtpfilesrc), GST_FORMAT_TIME);
>>>   // do not make rtpfilesrc set timestamps on outgoing buffers based on
>>> the running_time when they were captured
>>>   gst_base_src_set_do_timestamp (GST_BASE_SRC (rtpfilesrc), FALSE);
>>>
>>> What should I do?
>>>
>>> Thank you in advance.
>>>
>>
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