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  • ffmpeg conversion - keep audio bitrate

    17 mars 2020, par clausvdb

    I’m using ffmpeg to extract the audio from different video formats (flv, mp4) and convert it to mp3.

    %~dp0ffmpeg.exe -i %1 -ar 44100 -ac 2 -ab 128k "%~dpn1.mp3"

    This works just fine. However, in my input files, the audio bitrate varies, and I want to adjust the output bitrate accordingly. Even by extensive Google searching, I didn’t find any hint how to just keep the original bitrate.

    What I would need would be something like :

    -ab copy

    Which, of course, does not work.

    Is there anything that will work ?

    P.S : As you might have figured from the formatting above, I’m using a windows batch file. There would be the hack to use % dp0ffmpeg.exe -i, get the audio bitrate by grep and insert it in the command line. I just think there has to be an easier and more elegant way.

  • ffmpeg conversion - keep audio bitrate

    2 août 2015, par clausvdb

    I’m using ffmpeg to extract the audio from different video formats (flv, mp4) and convert it to mp3.

    %~dp0ffmpeg.exe -i %1 -ar 44100 -ac 2 -ab 128k "%~dpn1.mp3"

    This works just fine. However, in my input files, the audio bitrate varies, and I want to adjust the output bitrate accordingly. Even by extensive Google searching, I didn’t find any hint how to just keep the original bitrate.

    What I would need would be something like :

    -ab copy

    Which, of course, does not work.

    Is there anything that will work ?

    P.S : As you might have figured from the formatting above, I’m using a windows batch file. There would be the hack to use % dp0ffmpeg.exe -i, get the audio bitrate by grep and insert it in the command line. I just think there has to be an easier and more elegant way.

  • How to tell ffmpeg to encode only the latest frames from input pipe (live streaming) ?

    12 mars 2018, par Valdir

    Similar question on another language : How to get the latest frames in ffmpeg, not the next frame

    Well, basically I receive a vp8 webm chunks through sockets and send it to ffmpeg stdin. My problem is that when the network slows down or temporarily stops, ffmpeg output freezes until the expected chunks arrives. When that happens I can see the output "as expected".

    The real problem is that for every glitch, ffmpeg starts to transcoding from the point it stopped, so let’s say I’m streaming my webcam and at 01:01 I have a network glitch. My chunks are still sent from browser, but never received. At this point the video live streaming (on client’s browser) is at let’s say 01:10, and after the network starts to work as expected, ffmpeg starts receiving all chunks at once since 01:01 and encodes it from that point.

    What I want is ffmpeg to encode only the latest chunks in order to keep in sync with the original video being broadcasted. So I want it to discard that part that lagged.

    It seems like the original problem comes from the socket connection, which sends packets in order and can delay depending on the connection.

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