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 <[email protected]>: > 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 <[email protected]>: > >> 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 >> _______________________________________________ >> ffmpeg-user mailing list >> [email protected] >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > _______________________________________________ ffmpeg-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/ffmpeg-user
