Video corruption in rtsp server with appsink->appsrc

[Serj TorresSoldado]

It also coincides with the following events:

0:01:09.870444110 12808      0x10c9ca0 INFO              rtspclient
rtsp-client.c:2727:handle_request: client 0x11030b0: received a request
GET_PARAMETER rtsp://localhost:8554/test/ 1.0
0:01:11.740605331 12808 0x7f857804e8f0 INFO               GST_EVENT
gstevent.c:679:gst_event_new_caps: creating caps event application/x-rtcp
0:01:11.740733092 12808 0x7f857804e8f0 INFO               GST_EVENT
gstevent.c:760:gst_event_new_segment: creating segment event time segment
start=0:00:00.000000000, offset=0:00:00.000000000, stop=99:99:99.99999
9999, rate=1.000000, applied_rate=1.000000, flags=0x00,
time=0:00:00.000000000, base=0:00:00.000000000, position 0:00:00.000000000,
duration 99:99:99.999999999
0:01:11.740787716 12808 0x7f8570020850 INFO               GST_EVENT
gstevent.c:679:gst_event_new_caps: creating caps event application/x-rtcp
0:01:11.747962055 12808 0x7f857804ead0 INFO               GST_EVENT
gstevent.c:679:gst_event_new_caps: creating caps event application/x-rtcp
0:01:11.748070236 12808 0x7f857804ead0 INFO               GST_EVENT
gstevent.c:760:gst_event_new_segment: creating segment event time segment
start=0:00:00.000000000, offset=0:00:00.000000000, stop=99:99:99.99999
9999, rate=1.000000, applied_rate=1.000000, flags=0x00,
time=0:00:00.000000000, base=0:00:00.000000000, position 0:00:00.000000000,
duration 99:99:99.999999999
0:01:11.748110491 12808 0x7f8570020c00 INFO               GST_EVENT
gstevent.c:679:gst_event_new_caps: creating caps event application/x-rtcp
0:01:12.354714316 12808 0x7f8570003230 INFO              rtspstream
rtsp-stream.c:1616:on_new_ssrc: 0x7f8578038d70: new source 0x7f8570045ad0
0:01:12.354802661 12808 0x7f8570003230 INFO              rtspstream
rtsp-stream.c:1530:dump_structure: structure:
application/x-rtp-source-stats, ssrc=(uint)3989343839,
internal=(boolean)false, validated=(boolean
)false, received-bye=(boolean)false, is-csrc=(boolean)false,
is-sender=(boolean)false, seqnum-base=(int)-1, clock-rate=(int)-1,
rtcp-from=(string)127.0.0.1:35799, octets-sent=(guint64)0,
packets-sent=(guint64)0,
octets-received=(guint64)0, packets-received=(guint64)0,
bitrate=(guint64)0, packets-lost=(int)0, jitter=(uint)0,
sent-pli-count=(uint)0, recv-pli-count=(uint)0, sent-fir-count=(uint)0,
recv-fir-count=(uint)0, ha
ve-sr=(boolean)false, sr-ntptime=(guint64)0, sr-rtptime=(uint)0,
sr-octet-count=(uint)0, sr-packet-count=(uint)0, sent-rb=(boolean)false,
sent-rb-fractionlost=(uint)0, sent-rb-packetslost=(int)0, sent-rb-exthighe
stseq=(uint)0, sent-rb-jitter=(uint)0, sent-rb-lsr=(uint)0,
sent-rb-dlsr=(uint)0, have-rb=(boolean)false, rb-fractionlost=(uint)0,
rb-packetslost=(int)0, rb-exthighestseq=(uint)0, rb-jitter=(uint)0,
rb-lsr=(uint)
0, rb-dlsr=(uint)0, rb-round-trip=(uint)0;


On 29 March 2016 at 10:33, Serj TorresSoldado <torres.soldado at gmail.com>
wrote:

> I have just noticed this happens with the gst-rtsp-server examples as well.
>
> I am using the 1.6 branch.
>
> On 28 March 2016 at 21:50, Serj TorresSoldado <torres.soldado at gmail.com>
> wrote:
>
>> Hi All,
>>
>>  I am doing a sink -> src push. The sink is in a separate pipeline and
>> the src is created by gst-rtsp-server when creating the media pipeline.
>>
>>  I have tried both using a pull (need-data signal) and push (new-sample)
>> methods and the result is the same.
>>
>>  I am copying the buffer from the sink to the source. I am setting the
>> PTS on the copied buffer otherwise after the first client disconnects I am
>> unable to connect again.
>>
>>  GST_BUFFER_PTS(bufcpy) = client->timestamp_;
>>  client->timestamp_ += GST_BUFFER_DURATION(bufcpy);
>>  gst_app_src_push_buffer(client->appsrc_, bufcpy);
>>
>>  When cofiguring the client I have "played" with the following appsrc
>> properties but they don't seem to make a difference:
>>
>>   gst_util_set_object_arg(reinterpret_cast<GObject*>(appsrc.Get()),
>> "format", "time");
>>   g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "block", false,
>> nullptr);
>>   g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "is-live", true,
>> nullptr);
>>   g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "do-timestamp",
>> true, nullptr);
>> //  g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "stream-type",
>> "random-access", nullptr);
>> //  g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "max-bytes",
>> 1000, nullptr);
>>
>> Any help would be awesome, thanks.
>>
>> Serj
>>
>
>
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