[Live-devel] fNumchannels hardcoded
ssingh at neurosoft.in
ssingh at neurosoft.in
Wed Oct 30 18:26:39 PDT 2013
Hi,
I found that when I was checking my streaming using ffmpeg the ffmpeg
displays the channels as 1 and audio codec to be (null). I am using
AC3RTPSink in live555. I tried to debug the source of live555 and found
that the fNumChannels is not set while creating object (calling base
class AudioRTPSink) for AC3AudioRTPSink. Is that intentional or
something that we need to correct. I think it should be configurable and
should be a setter in AC3AudioRTPSink class. Also I am not sure why
ffmpeg display audio codec as (null). When i look at debug info for
ffmpeg it displays
===============================================
[rtsp @ 0085e9a0] SDP:
v=0
o=- 1383179174903374 1 IN IP4 10.20.22.20
s=Session streamed by "Ekomsys"
i=Streaming channel using Ekomsys streaming server
t=0 0
a=tool:LIVE555 Streaming Media v2011.12.23
a=type:broadcast
a=control:*
a=range:npt=0-
a=x-qt-text-nam:Session streamed by "Ekomsys"
a=x-qt-text-inf:Streaming channel using Ekomsys streaming server
m=audio 6666 RTP/AVP 96
c=IN IP4 239.255.42.42/255
b=AS:200
a=rtpmap:96 AC3/48000
a=control:track1
m=video 8888 RTP/AVP 96
c=IN IP4 239.255.42.42/255
b=AS:45000
a=rtpmap:96 MP4V-ES/90000
a=fmtp:96
profile-level-id=1;config=000001B001000001B58913000001000000012000C48D8800F50644191463000001B24C61766335352E31382E313032
a=control:track2
[rtsp @ 0085e9a0] audio codec set to: (null)
[rtsp @ 0085e9a0] audio samplerate set to: 48000
[rtsp @ 0085e9a0] audio channels set to: 1
[rtsp @ 0085e9a0] video codec set to: mpeg4
=====================================================
which makes me think that it recognizes that its AC3 format but still
responds (null) as audio codec.
I used following command "ffplay -rtsp_transport udp_multicast
rtsp://10.20.22.20:8554/Ekomsys_stream -loglevel debug"
Thanks,
Caduceus
More information about the live-devel
mailing list