[FFmpeg-user] decode multiple rtp stream for multichannel usage

[s p]

hello

when I remove "-c:a copy" ffmpeg console says "output file is empty,
nothing was encoded"
I also succeeded to decode this stream with gstreamer, where I can specify
some informations about the stream without using an sdp file
isn't it possible to do so with ffmpeg ?

see the updated command line and console output,
I also add the content of the .sdp file

# ffmpeg -i stream1-aac.sdp -f alsa surround71:HD

ffmpeg version 2.8.1 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 5.2.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static
--disable-stripping --enable-avisynth --enable-avresample
--enable-fontconfig --enable-gnutls --enable-gpl --enable-ladspa
--enable-libass --enable-libbluray --enable-libfreetype --enable-libfribidi
--enable-libgsm --enable-libmodplug --enable-libmp3lame
--enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg
--enable-libopus --enable-libpulse --enable-libschroedinger
--enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora
--enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid
--enable-shared --enable-version3 --enable-x11grab --host-cflags='"-fPIC"'
  libavutil      54. 31.100 / 54. 31.100
  libavcodec     56. 60.100 / 56. 60.100
  libavformat    56. 40.101 / 56. 40.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 40.101 /  5. 40.101
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc    53.  3.100 / 53.  3.100
Input #0, sdp, from 'stream1-aac.sdp':
  Metadata:
    title           : No Name
  Duration: N/A, bitrate: N/A
    Stream #0:0: Audio: aac, 44100 Hz, mono, fltp
Output #0, alsa, to 'surround71:HD':
  Metadata:
    title           : No Name
    encoder         : Lavf56.40.101
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
    Metadata:
      encoder         : Lavc56.60.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
stream1-aac.sdp: Connection timed out
size=N/A time=00:00:00.00 bitrate=N/A
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames
parameters if used)


SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 192.168.42.101
t=0 0
a=tool:libavformat 56.40.101
m=audio 16384 RTP/AVP 97
b=AS:264
a=rtpmap:97 MPEG4-GENERIC/44100/1
a=fmtp:97
profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdelt$





2015-11-11 10:24 GMT+01:00 Moritz Barsnick <barsnick at gmx.net>:

> Okay, sorry for not seeing that those are two attempts. My bad.
>
> On Tue, Nov 10, 2015 at 21:37:55 +0100, s p wrote:
> > $ ffmpeg -i rtpopus.sdp -c:a copy -ar 44100 -f alsa surround71:HD
> > > ffmpeg version 2.8.1 Copyright (c) 2000-2015 the FFmpeg developers
> [...]
> > > Input #0, sdp, from 'rtpopus.sdp':
> > >   Metadata:
> > >     title           : No Name
> > >   Duration: N/A, start: 0.000000, bitrate: N/A
> > >     Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
> > > [alsa @ 0x564ee5d982c0] sample format 0x4f505553 is not supported
> > > Output #0, alsa, to 'surround71:HD':
> > >   Metadata:
> > >     title           : No Name
> > >     encoder         : Lavf56.40.101
> > >     Stream #0:0: Audio: opus, 48000 Hz, stereo
> > > Stream mapping:
> > >   Stream #0:0 -> #0:0 (copy)
> > > Could not write header for output file #0 (incorrect codec parameters
> ?):
> > > Function not implemented
>
> You are trying to copy ("-c:a copy") an opus stream to an alsa device.
>
> ffmpeg complains:
> > > [alsa @ 0x564ee5d982c0] sample format 0x4f505553 is not supported
> [...]
> > > Could not write header for output file #0 (incorrect codec parameters
> ?):
>
> Does your alsa implementation decode opus by itself? I think not. You
> need to let ffmpeg decode opus for you. ffmpeg will probably choose the
> correct codec for alsa if you just omit "-c:a ...".
>
> Moritz
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