Hello,

I am currently developing a streaming solution for a client which uses FFmpeg 
to process an RTSP stream and convert it into HLS format. During the testing 
phase we have run into issues regarding audio processing and require some 
guidance on how we might possibly resolve it.

Specifically, the client's hardware uses a third party DVR which provides RTSP 
streams on request which normally do not include any data in audio sub-streams. 
On specific conditions the DVR will start feeding data into the audio streams 
(using audio codec PCM S16 BE) which we must be able to process, however FFmpeg 
stalls whenever no audio data is present and seems to wait until there is audio 
to consume to start outputting the *.ts segments. We need to be able to start 
consuming the stream in a timely/immediate fashion even when no audio is 
present, and must be able to consume the audio without additional input when 
present, since the third party hardware is entirely responsible for feeding the 
audio sub-stream and provides no additional tools to determine the presence or 
lack of audio.

For reference, this is the ffmpeg command we are using at the moment:

ffmpeg -probesize 50000 -analyzeduration 50000 -rtsp_transport tcp  -i 
"rtsp://<DVR_URL>" -c:v copy -c:a aac -hls_flags 
append_list+split_by_time+omit_endlist -hls_segment_filename 
"<SEGMENT_FILENAME>_%03d.ts" -hls_time 1 -hls_list_size 0 
"<PLAYLIST_FILENAME>.m3u8"
 We've tried -c:a with the appropriate codec instead of aac, but it produced no 
result.
Any help would be greatly appreciated!
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