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• Is there a way to include a transition video when using ffmpeg's select filter ?

  19 février 2021, par Tim

  I wanted to take some segments of an input video and make a highlight reel automatically using ffmpeg. I was originally experimenting with trimming and concating but it was difficult because the audio was falling out of sync and also I have a variable number of highlights that I want.

  


  I discovered from the top comment here that the 'select' filter is very powerful. I have a python program that inputs all the parts I want selected into the command and run it in the terminal and it works perfect. Only issue is that I want a quick transition video to play in between each of these highlights. Is this not possible with select ? Do I have to return to using trim and concat ? Thank you !

  


  Edit : for reference, here is an example ffmpeg command I am running

  


  ffmpeg -y -i video.mp4 -i audio.mp4 -vf "select='between(t,56,69)+between(t,60,135)+between(t,73,132)+between(t,152,163)+between(t,251,278)+between(t,600,700)+between(t,774,872)', setpts=N/FRAME_RATE/TB " -af "aselect='between(t,56,69)+between(t,60,135)+between(t,73,132)+between(t,152,163)+between(t,251,278)+between(t,600,700)+between(t,774,872)',asetpts=N/SR/TB" output.mp4


  


• FFmpeg : How to apply volume filter fast in ffmpeg

  29 novembre 2018, par Prashant_Sarin

  i am using below code for applying volume filter in ffmpeg, but its taking too long.Let’s say the audio itself is for 5 minutes then applying this filter takes 3 minutes. Can someone help to speed up the process.

  my command is as :

  var command = "-i $audioPath -af volume=enable='between(t,5,10)':volume=0.25,volume=enable='between(t,15,20)':volume=0.25 $outputFile"

  logs are :

  2018-11-29 15:17:16.912 17250-17250/com.example.prashants.ffmpegtest 
  
  D/com.example.prashants.ffmpegtest.service.AudioOverlayService: onProgress - 
  
  ---AudioVolume---- video:0kB audio:3030kB subtitle:0kB other streams:0kB 
  
  global headers:0kB muxing overhead: 1.236199%
  
  2018-11-29 15:17:16.913 17250-17250/com.example.prashants.ffmpegtest 
  
  D/com.example.prashants.ffmpegtest.service.AudioOverlayService: onProgress - 
  
  ---AudioVolume---- bench: utime=168.977s
  
  2018-11-29 15:17:16.915 17250-17250/com.example.prashants.ffmpegtest 
  
  D/com.example.prashants.ffmpegtest.service.AudioOverlayService: onProgress - 
  
  ---AudioVolume---- bench: maxrss=65116kB
  
  2018-11-29 15:17:16.917 17250-17250/com.example.prashants.ffmpegtest 
  
  D/com.example.prashants.ffmpegtest.service.AudioOverlayService: onProgress - 
  
  ---AudioVolume---- [aac @ 0xaf5e4400] Qavg: 15833.816
  
  2018-11-29 15:17:16.920 17250-17250/com.example.prashants.ffmpegtest 
  
  D/com.example.prashants.ffmpegtest.service.AudioOverlayService: AudioVolume 
  
  success--------- -benchmark -threads 4 -i 
  
  /storage/emulated/0/FFmpegTest/looped.m4a -af 
  
  volume=enable='between(t,5,10)+between(t,15,20)':volume=0.25:eval=frame 
  
  /storage/emulated/0/FFmpegTest/filtered_audio.m4a

  that’s the log from my android studio console

• FFMPEG C++ Volume Filter

  18 mai 2020, par Hrethric

  I seem to be having a bit of trouble using FFMPEG audio filters in C++ code. If I only have "abuffer" and "abuffersink" filters, and grab the audio frame from the filtergraph, it sounds perfect. Once I add another filter into the graph (in this case, it's a "volume" filter), there is a lot of noise introduced. I can't figure out what would be causing this.

  



  This isn't the case for all filters - "aecho" works just fine, for example. Any thoughts ? Here's the relevant code :

  



  Filter Creation

  



  char args[512];
int ret = 0;

_filterGraph = avfilter_graph_alloc();

// abuffer must be the first filter used -- it feeds data into the filter graph
/******************
ABUFFER FILTER
*******************/
_abufferFilter = avfilter_get_by_name("abuffer");

/*buffer audio source : decoded frames will be
inserted here. */
if (!_inAudioCodecContext->channel_layout)
{
    _inAudioCodecContext->channel_layout = av_get_default_channel_layout(_inAudioStream->codec->channels);
}

snprintf(args, sizeof(args),
    "sample_rate=%d:sample_fmt=%s:channel_layout=0x%" PRIx64,
    _inAudioCodecContext->sample_rate,
    av_get_sample_fmt_name(_inAudioCodecContext->sample_fmt),
    _inAudioCodecContext->channel_layout);

ret = avfilter_graph_create_filter(&_abufferFilterCtx, _abufferFilter, "abuffer", args, NULL, _filterGraph);
char *errorCode = new char[256];
av_strerror(ret, errorCode, 256);

/******************
VOLUME FILTER
*******************/
snprintf(args, sizeof(args),
    "%f",
    2.0f);
_volumeFilter = avfilter_get_by_name("volume");
ret = avfilter_graph_create_filter(&_volumeFilterCtx, _volumeFilter, "volume", args, NULL, _filterGraph);
char *errorCode = new char[256];
av_strerror(ret, errorCode, 256);

/******************
ABUFFERSINK FILTER
*******************/
// abuffersink must be the last filter used -- it gets data out of the filter graph
_abuffersinkFilter = avfilter_get_by_name("abuffersink");
ret = avfilter_graph_create_filter(&_abufferSinkFilterCtx, _abuffersinkFilter, "abuffersink", NULL, NULL, _filterGraph);

// Link the source buffer to the volume filter
// If I link this to the sink buffer and comment out the next line
// Audio sounds perfect
ret = avfilter_link(_abufferFilterCtx, 0, _volumeFilterCtx, 0);
// Link the volume filter to the sink buffer
ret = avfilter_link(_volumeFilterCtx, 0, _abufferSinkFilterCtx, 0);
ret = avfilter_graph_config(_filterGraph, NULL);

return ret;


  



  Read frames from buffer

  



  // Read a frame from the audio stream/file
ret = av_read_frame(_inFormatContext, &_packet);
int frameFinished = 0;
// Decode the resulting packet into a single frame for processing
int length = avcodec_decode_audio4(_inAudioCodecContext, _audioFrame, &frameFinished, &_packet);

if (frameFinished)
{
    // Insert the frame into the source filter
    ret = av_buffersrc_write_frame(_abufferFilterCtx, _audioFrame);
    while (true)
    {
        // Pull a frame from the filter graph
        ret = av_buffersink_get_frame(_abufferSinkFilterCtx, _audioFrame);

        // EOF or EAGAIN is expected when filtering frames, set the error to "0"
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
        {
            ret = 0;
            break;
        }
        if (ret < 0)
        {
            break;
        }
    }

    // This keeps going, doing some resampling and conversion based on codec output selection, but that isn't relevant to the issue