Audio problems using appsr and srtsink

[Carles Albert Bolaños Garcia]

Hello

I'm a software programer from Barcelona and my company develops 
professional video servers for broadcasting. We use propietary SDKs for 
managing video and audio, like MatroxDSX, Medialooks, BlackMagicSDK, ... 
For a recent project we are studying the possibility to use GStreamer 
form sending an SRT stream, but the resulting stream in some receivers, 
no all, like VLC, drops audio samples and I don't know the reason.

If I use this pipe, the resulting stream is Ok. Players like VLC are 
able to connect them using srt://127.0.0.1:5011?mode=caller and the 
works fine, with no errors and valid video and audio data

gst-launch-1.0 uridecodebin uri="file:///C:/temp/AT_0007 Corzo 
emboscado.mxf" name=decode ! videoconvert ! x264enc ! queue ! mpegtsmux 
name=mux ! queue ! srtsink uri=srt://127.0.0.1:5011?mode=listener 
decode. ! audioconvert ! avenc_aac ! queue ! mux.

But if I use appsrc the problem appears in some players, like VLC, but 
there are no errors in the log window

appsrc is-live=true do-timestamp=true ! videoconvert ! x264enc ! queue ! 
mpegtsmux name=mux ! queue ! srtsink 
uri=srt://127.0.0.1:5011?mode=listener appsrc is-live=true 
do-timestamp=true ! audioconvert ! avenc_aac ! queue ! mux.

I have done multiple tests with no success, removing queues, using UDP 
instead of SRT, saving to disk the trasport stream and playing back 
using VLC...
The "need-data" callbacks for audio and video are very easy The always 
push a buffer of one frame for video and the corresponding audio data 
for audio. The timestamps are ok.
If instead of send the data to a muxer we change the pipe to render the 
incoming video and audio, there are no problems, and the audio sounds 
ok. It really looks like  a timestamp problem, but I can't find the solution

This is the "need-data code"

bool want_video = false;

void cb_need_video_data(GstElement *src, guint unused_size, gpointer 
user_data)
{
   want_video = true;
   return;
}

void write_video_data(const void *data, int data_size)
{
   if(!want_video) return false;
   want_video = false;

   // Fetch num den factors
   GstStructure *s = gst_caps_get_structure(m_vCaps, 0);
   gint num = 0, den = 0;
   gst_structure_get_fraction(s, "framerate", &num, &den);

   GstBuffer *buffer = 
gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_READONLY, (gpointer) data, 
data_size, 0, data_size, NULL, NULL);
   static GstClockTime timestamp = 0;
   GST_BUFFER_PTS(buffer) = timestamp;
   GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale_int(GST_SECOND, 
den, num);
   // GST_BUFFER_DTS(buffer) = timestamp - GST_BUFFER_DURATION(buffer);
   timestamp += GST_BUFFER_DURATION(buffer);
   GstFlowReturn ret = gst_app_src_push_buffer((GstAppSrc *) vAppSrc, 
buffer);
}
And the same for audio. The incoming data comes from our video server

Can anybody help me? I'm working on Windows with the last GStreamer version.
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