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• ffmpeg trimming audio WAV files and setting timecode

  14 juillet 2022, par user19551045

  I am currently trying to cut an audio file to match the length of a video (without combining the two...just looking at timecodes) and produce a trimmed audio file that has a timecode that will match up with the video, the video is considered the absolute truth.

  


  Currently, the issue is that the timecodes from the original audio file do not get carried over into the new cropped audio file. So, the starting timecode is now 00:00:00:00 instead of say 07:20:02:14. Even using the -timecode commands and trying to hardcode the timecode that way doesn't seem to do the trick. I am wondering if there is any way around this ? I just want to do as minimal to the raw audio as possible...just change the audio file's length while setting the timecodes so the new audio will line up with the video. Any thoughts/suggestions welcome !

  


  Currently I have tried two options that don't seem to work :
using ffmpeg cmds :

  


  
        cmd2 = r'{} -ss "{}" -i "{}" -codec copy -timecode "{}" "{}"'.format(
            FFMPEG_PATH,
            abs(tc_diff_in_seconds),
            audio_path,
            "17074647",
            out_path
        )


  


  and also using pydub :

  


          current_audio = AudioSegment.from_wav("{}".format(audio_path))
        start_time_in_milli = abs(tc_diff_in_seconds*1000)
        end_time_in_milli = start_time_in_milli + video_dur_in_seconds * 1000
        trimmed_audio = current_audio[start_time_in_milli:end_time_in_milli]
        trimmed_audio.export('{}'.format(out_path), format='WAV', parameters=["-timecode", "17:07:46:47"])


  


  Any thoughts/suggestions welcome ! Thanks

  


• Convert webm (or any other) format's chunks to mp4

  12 décembre 2016, par Cheyenne Forbes de Avapno

  Is it possible to get webm ( or other format ) chucks from a http post (upload) on my sever (i know how to do this).... then feed them as chucks (chunks recieved from browser) to gstreamer or ffmpeg to be converted to mp4 with reduced quality without loading the entire file in memory or to disk before saving the converted mp4 ? Why I dont want them to be loaded fully into memory or disk ? scalability

• Convert mp3/audio source to RTSP or something that is live streamable

  20 mars 2019, par spezticle

  I’ve got an mp3 that’s continually being written to. I’m trying to get some HTML5 tag or something to play this file and it will, but it stops at the end of the file as it was loaded in the moment the webpage was pulled up. If I refresh the page, the new length is reflected but position is lost.

  This is streaming, but not live.

  I’m using ffmpeg to push an audio source from a local computer to a remote web server. That webserver receives the file with ffmpeg and writes it to the mp3. Is there no better way of doing this that isn’t absurdly complicated ?