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I'm trying to save an audio (or video) feed to local using FFmpeg. Here are the command and arguments that I'm using.

ffmpeg -i http://7359.live.streamtheworld.com:80/CONTINENTALAAC_SC  -vn -ar 22050 -acodec flac -ss 0 -t 6000 test1.flac

I then iterate that in Bash, but I've got an overlap between the end of one file and the next.

Is there a way to get that feed chunked every 10 minutes?

This is t shell output:

    ffmpeg -i http://7359.live.streamtheworld.com:80/CONTINENTALAAC_SC  -vn -ar 22050 -acodec flac -ss 0 -t 6 test1.flac
FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  built on Jan 29 2012 23:56:18 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
  configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  libavutil     50.15. 1 / 50.15. 1
  libavcodec    52.72. 2 / 52.72. 2
  libavformat   52.64. 2 / 52.64. 2
  libavdevice   52. 2. 0 / 52. 2. 0
  libavfilter    1.19. 0 /  1.19. 0
  libswscale     0.11. 0 /  0.11. 0
  libpostproc   51. 2. 0 / 51. 2. 0
[aac @ 0x928e850]max_analyze_duration reached
[aac @ 0x928e850]Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'http://7359.live.streamtheworld.com:80/CONTINENTALAAC_SC':
  Duration: N/A, bitrate: 29 kb/s
    Stream #0.0: Audio: aac, 44100 Hz, 2 channels (FC), s16, 29 kb/s
File 'test1.flac' already exists. Overwrite ? [y/N] y
Output #0, flac, to 'test1.flac':
  Metadata:
    encoder         : Lavf52.64.2
    Stream #0.0: Audio: flac, 22050 Hz, 2 channels, s16, 64 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
Press [q] to stop encoding
size=     175kB time=6.06 bitrate= 236.6kbits/s
video:0kB audio:167kB global headers:0kB muxing overhead 4.824872%

Console print out with new arguments as suggested by LordNeckbeard:

ffmpeg -i http://7359.live.streamtheworld.com:80/CONTINENTALAAC_SC -vn -codec:a flac -map 0 -f segment -segment_list out.list -segment_time 00:10:00.00 out%03d.flac
FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  built on Jan 29 2012 23:56:18 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
  configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  libavutil     50.15. 1 / 50.15. 1
  libavcodec    52.72. 2 / 52.72. 2
  libavformat   52.64. 2 / 52.64. 2
  libavdevice   52. 2. 0 / 52. 2. 0
  libavfilter    1.19. 0 /  1.19. 0
  libswscale     0.11. 0 /  0.11. 0
  libpostproc   51. 2. 0 / 51. 2. 0
[aac @ 0x8c8e850]max_analyze_duration reached
[aac @ 0x8c8e850]Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'http://7359.live.streamtheworld.com:80/CONTINENTALAAC_SC':
  Duration: N/A, bitrate: 23 kb/s
    Stream #0.0: Audio: aac, 44100 Hz, 2 channels (FC), s16, 23 kb/s
Unrecognized option 'codec:a'
Improve this question
2
  • 1
    What happens when you place the -ss before the -i? (Also, please always add full, uncut console output when asking FFmpeg questions, thanks)
    – slhck
    Feb 5, 2013 at 17:47
  • Works in the same way or at least with no noticiable difference
    – xulen
    Feb 6, 2013 at 19:14

1 Answer 1

Reset to default
1

Try the segmenter muxer.

ffmpeg -i input -vn -codec:a flac -map 0 -f segment -segment_list out.list \
-segment_time 00:10:00.00 out%03d.flac

Copying the audio stream instead of re-encoding would be worth considering, but it is impossible to provide a non-guessing example without your ffmpeg console output.

Improve this answer
6
  • Ffmpeg yells about I/O error occurred. Also ive tryed to apply those arguments with another input, a local file, and then it yells: ffmpeg: unrecognized option '-codec:a'
    – xulen
    Feb 6, 2013 at 13:40
  • @user144738 Your ffmpeg command and complete console output is missing. Please update your question with this information.
    – llogan
    Feb 6, 2013 at 18:00
  • @user144738 Update your FFmpeg. Yours is now more than two years old. Download a static build from the homepage, for example.
    – slhck
    Feb 6, 2013 at 19:50
  • Thank you both, you make me happier. Works fine with newest static version (built feb 6 2013). The only thing that ive changed, because wasnt working for me, was the argument "-codec:a" to "-acodec".
    – xulen
    Feb 7, 2013 at 4:58
  • @xulen With a recent build, this should work. codec:a and c:a mean the same, and acodec is just an alias to the former.
    – slhck
    Feb 7, 2013 at 11:22

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