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

[Andrea Lo Pumo]

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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