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

s p stefanpiat at gmail.com
Tue Nov 10 21:37:55 CET 2015


hello

indeed
here is my command line
send : ffmpeg -f alsa -ac 1 -i hw:1,0 -acodec libopus -ar 48000 -b:a 256k
-ac 1 -f rtp rtp://hostname:1234 > rtpopus.sdp
receive : ffmpeg -i rtpopus.sdp -acodec copy -f alsa surround71:HD

and the console output :

$ 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
>   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
>   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 '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
>

if I specify the format like :
ffmpeg -f rtp -i rtpopus.sdp -acodec copy -f alsa surround71:HD

the output is

ffmpeg -f rtp -i micro1.sdp -acodec copy -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
>   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
> [rtp @ 0x562aac19cfa0] Unsupported RTP version packet received
> [rtp @ 0x562aac19cfa0] Received too short packet
>     Last message repeated 769283 times
>


2015-11-10 17:42 GMT+01:00 Moritz Barsnick <barsnick at gmx.net>:

> On Tue, Nov 10, 2015 at 17:15:56 +0100, s p wrote:
> > And as I am not able to decode the stream directly with ffmpeg, it looks
> > like there is a problem wiith the Header (I don,'t understand why it
> works
> > with ffplay then?)
>
> You forgot to show us the full command line and complete, uncut console
> output of your ffmpeg call.
>
> If ffplay can play it, ffmpeg should be able to demux and decode it.
>
> Moritz
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