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• How to open a remote radio stream with ffmpeg's `avformat_open_input` without segfault ?

  19 août 2020, par Keyboard embossed forhead

  I'm at the beginning stage of writing a small app to stream internet radio stations. For the moment I'm just trying to get the detected info of the input stream. Whilst I am successful in getting all the stream's details via the the command line tool (ffmpeg -i ${URL}), calling the library's avformat_open_input(...) method call results in a SEGFAULT (a stack overflow to be precise when checked in valgrind).

  


  Passing a local file url works fine though in both the command line utility and the library call.

  


  Here's a minimal example :

  


  int test() {
    const char      * station_url = "http://stream.srg-ssr.ch/m/rsc_de/aacp_96";
    const char      * test_file   = "test.mp3"; //works
    AVFormatContext * av_ctx      = avformat_alloc_context();
    int               ret         = 0;

    avformat_network_init();

    if( ( ret = avformat_open_input( &av_ctx, station_url, NULL, NULL ) ) < 0 ) { //SEGFAULT   
        printf( "Could not open file '%s': %i", station_url, ret );
        return -1;
    }

    printf( "Format %s, duration %ld us", av_ctx->iformat->long_name, av_ctx->duration );

    avformat_network_deinit();
    return 0;
}


  


  If anyone with experience in dealing with acquiring remote streams using ffmpeg libraries in C has some insights I'd be grateful. Thanks in advance.

  


  I'm using ffmpeg v4.3.1 on Linux.

  


• ppc : reduce overreads when loading 8 pixels in altivec dsp functions

  13 février 2014, par Janne Grunau

  ppc : reduce overreads when loading 8 pixels in altivec dsp functions
  Altivec can only load naturally aligned vectors. To handle possibly
  
  unaligned data a second vector is loaded from an offset of the original
  
  location and the data is recovered through a vector permutation.
  
  Overreads are minimal if the offset for second load points to the last
  
  element of data. This is 7 for loading eight 8-bit pixels and overreads
  
  are reduced from 16 bytes to 8 bytes if the pixels are 64-bit aligned.
  
  For unaligned pixels the overread is reduced from 23 bytes to 15 bytes
  
  in the worst case.
  

  • [DBH] libavcodec/ppc/dsputil_altivec.c

• 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