[Libav-user] mp3 live stream to stdout
pablo platt
pablo.platt at gmail.com
Mon Jun 24 00:22:43 CEST 2013
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 <pablo.platt at gmail.com> 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 <onemda at gmail.com> wrote:
>
>> On 6/23/13, pablo platt <pablo.platt at gmail.com> 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 <pablo.platt at gmail.com>
>> 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 <onemda at gmail.com>
>> wrote:
>> >>
>> >>> On 6/23/13, pablo platt <pablo.platt at gmail.com> 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
>> >>> Libav-user at ffmpeg.org
>> >>> http://ffmpeg.org/mailman/listinfo/libav-user
>> >>>
>> >>
>> >>
>> >
>> _______________________________________________
>> Libav-user mailing list
>> Libav-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/libav-user
>>
>
>
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