When I'm trying to output to a file it gets stuck at [rtmp @ 0xb481460] New incoming chunk size = 4096 and when I stop rtmplite I'm getting the rest.
./ffmpeg -loglevel debug -i rtmp://127.0.0.1/audio/test -f mp3 test.mp3 ffmpeg version N-54023-g6fe419b Copyright (c) 2000-2013 the FFmpeg developers built on Jun 14 2013 05:12:40 with gcc 4.6 (Debian 4.6.3-1) configuration: --prefix=/root/ffmpeg-static/32bit --arch=x86_32 --extra-cflags='-m32 -I/root/ffmpeg-static/32bit/include -static' --extra-ldflags='-m32 -L/root/ffmpeg-static/32bit/lib -static' --extra-libs='-lxml2 -lexpat -lfreetype' --enable-static --disable-shared --disable-ffserver --disable-doc --enable-bzlib --enable-zlib --enable-postproc --enable-runtime-cpudetect --enable-libx264 --enable-gpl --enable-libtheora --enable-libvorbis --enable-libmp3lame --enable-gray --enable-libass --enable-libfreetype --enable-libopenjpeg --enable-libspeex --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-version3 --enable-libvpx libavutil 52. 35.101 / 52. 35.101 libavcodec 55. 16.100 / 55. 16.100 libavformat 55. 8.102 / 55. 8.102 libavdevice 55. 2.100 / 55. 2.100 libavfilter 3. 77.101 / 3. 77.101 libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Splitting the commandline. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. Reading option '-i' ... matched as input file with argument 'rtmp:// 127.0.0.1/audio/test'. Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'. Reading option 'test.mp3' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input file rtmp://127.0.0.1/audio/test. Successfully parsed a group of options. Opening an input file: rtmp://127.0.0.1/audio/test. [rtmp @ 0xb481460] Handshaking... [rtmp @ 0xb481460] Type answer 3 [rtmp @ 0xb481460] Server version 1.2.3.4 [rtmp @ 0xb481460] Proto = rtmp, path = /audio/test, app = audio, fname = test [rtmp @ 0xb481460] Server bandwidth = 1073741824 [rtmp @ 0xb481460] Creating stream... [rtmp @ 0xb481460] Sending play command for 'test' [rtmp @ 0xb481460] New incoming chunk size = 4096 [flv @ 0xb480f40] Format flv probed with size=2048 and score=100 [flv @ 0xb480f40] File position before avformat_find_stream_info() is 13 [flv @ 0xb480f40] Could not find codec parameters for stream 0 (Video: none): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [flv @ 0xb480f40] Could not find codec parameters for stream 1 (Audio: none, 0 channels): unspecified sample format Consider increasing the value for the 'analyzeduration' and 'probesize' options [flv @ 0xb480f40] File position after avformat_find_stream_info() is 13 rtmp://127.0.0.1/audio/test: could not find codec parameters [AVIOContext @ 0xb481e00] Statistics: 13 bytes read, 0 seeks [rtmp @ 0xb481460] Deleting stream... On Mon, Jun 24, 2013 at 12:36 AM, pablo platt <[email protected]> wrote: > Tested with ffmpeg.static.32bit.2013-06-14.tar.gz from > http://ffmpeg.gusari.org/static/ under ubuntu 12.04 32bit. > > I'm using rtmplite with the test client. > > test:~/rtmplite$ python rtmp.py -d > > I'm Using the textClient that comes with rtmplite to create a video+audio > stream: > test:~/rtmplite/testClient/bin-debug$ python -m SimpleHTTPServer > > I open the testClient example with Firefox in > http://localhost:8000/testClient.html > > Result: > > ./ffmpeg -loglevel debug -i rtmp://127.0.0.1/audio/test -f mp3 pipe:1 > > ffmpeg version N-54023-g6fe419b Copyright (c) 2000-2013 the FFmpeg developers > built on Jun 14 2013 05:12:40 with gcc 4.6 (Debian 4.6.3-1) > configuration: --prefix=/root/ffmpeg-static/32bit --arch=x86_32 > --extra-cflags='-m32 -I/root/ffmpeg-static/32bit/include -static' > --extra-ldflags='-m32 -L/root/ffmpeg-static/32bit/lib -static' > --extra-libs='-lxml2 -lexpat -lfreetype' --enable-static --disable-shared > --disable-ffserver --disable-doc --enable-bzlib --enable-zlib > --enable-postproc --enable-runtime-cpudetect --enable-libx264 --enable-gpl > --enable-libtheora --enable-libvorbis --enable-libmp3lame --enable-gray > --enable-libass --enable-libfreetype --enable-libopenjpeg --enable-libspeex > --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-version3 > --enable-libvpx > > libavutil 52. 35.101 / 52. 35.101 > libavcodec 55. 16.100 / 55. 16.100 > libavformat 55. 8.102 / 55. 8.102 > libavdevice 55. 2.100 / 55. 2.100 > libavfilter 3. 77.101 / 3. 77.101 > libswscale 2. 3.100 / 2. 3.100 > > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > Splitting the commandline. > Reading option '-loglevel' ... matched as option 'loglevel' (set logging > level) with argument 'debug'. > > Reading option '-i' ... matched as input file with argument > 'rtmp://127.0.0.1/audio/test'. > Reading option '-f' ... matched as option 'f' (force format) with argument > 'mp3'. > > Reading option 'pipe:1' ... matched as output file. > Finished splitting the commandline. > Parsing a group of options: global . > Applying option loglevel (set logging level) with argument debug. > Successfully parsed a group of options. > > Parsing a group of options: input file rtmp://127.0.0.1/audio/test. > Successfully parsed a group of options. > Opening an input file: rtmp://127.0.0.1/audio/test. > > [rtmp @ 0xa8a7460] Handshaking... > [rtmp @ 0xa8a7460] Type answer 3 > [rtmp @ 0xa8a7460] Server version 1.2.3.4 > [rtmp @ 0xa8a7460] Proto = rtmp, path = /audio/test, app = audio, fname = test > [rtmp @ 0xa8a7460] Server bandwidth = 1073741824 > > [rtmp @ 0xa8a7460] Creating stream... > [rtmp @ 0xa8a7460] Sending play command for 'test' > [rtmp @ 0xa8a7460] New incoming chunk size = 4096 > > > > On Sun, Jun 23, 2013 at 8:31 PM, Paul B Mahol <[email protected]> wrote: > >> On 6/23/13, pablo platt <[email protected]> wrote: >> > I'm using static builds from: >> > http://ffmpeg.gusari.org/static/ >> > >> > I've tried: >> > ./ffmpeg -loglevel debug -i rtmp://127.0.0.1/audio/test -f mp3 - >> > and >> > ./ffmpeg -loglevel debug -i rtmp://127.0.0.1/audio/test -f mp3 pipe:1 >> > >> > I don't see anything in the shell >> >> Really? Paste full uncut console output. >> >> > >> > >> > On Sun, Jun 23, 2013 at 3:57 AM, pablo platt <[email protected]> >> wrote: >> > >> >> Sorry but the mailing list is called libav. >> >> How can I install ffmpeg on ubuntu 12.04 and above? >> >> >> >> >> >> On Sun, Jun 23, 2013 at 3:42 AM, Paul B Mahol <[email protected]> >> wrote: >> >> >> >>> On 6/23/13, pablo platt <[email protected]> wrote: >> >>> > Hi, >> >>> > >> >>> > I'm trying to transcode a live RTMP stream to mp3. >> >>> > I can't make avconv write the output to stdout. >> >>> > >> >>> > I was trying: >> >>> > avconv -i rtmp://127.0.0.1/audio/test -f mp3 - >> >>> > and >> >>> > avconv -i rtmp://127.0.0.1/audio/test -f mp3 pipe:1 >> >>> > >> >>> > But all I'm getting is a message that the speex header is missing. >> >>> > When I'm stopping the RTMP server I see some more message but not >> >>> getting >> >>> > mp3 packets. >> >>> > >> >>> > I'm using ubuntu 12.04 64bit with the libav-tools deb package. >> >>> > For testing I'm using the python rtmp server >> >>> > http://code.google.com/p/rtmplite/ >> >>> > >> >>> > How can I make the output go to stdout? >> >>> >> >>> You are asking for help on ffmpeg mailing list for avconv >> >>> which is from libav (ffmpeg fork). >> >>> >> >>> If you want help for avconv don't ask here as this is >> >>> mailing list for ffmpeg only (real ffmpeg not fake one). >> >>> _______________________________________________ >> >>> Libav-user mailing list >> >>> [email protected] >> >>> http://ffmpeg.org/mailman/listinfo/libav-user >> >>> >> >> >> >> >> > >> _______________________________________________ >> Libav-user mailing list >> [email protected] >> http://ffmpeg.org/mailman/listinfo/libav-user >> > >
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