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

[s p]

hello,

I made a mistake for the last test, i used a different system and forgot to
change hostname.
stream was send to wrong place then.

so it works now without the "-f rtp" and without "-c:a copy"
this was my problem

thank you for helping me to solve this !



2015-11-11 11:00 GMT+01:00 s p <stefanpiat at gmail.com>:

> 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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>>
>
>