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• 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


  


• FFmpeg raw h.264 set pts value

  19 mai 2020, par thecaptain0220

  I am currently using ffmpeg to convert a custom container media format to mp4. It is straightforward to dump all the h.264 frames to one file and the aac audio to another. Then I can combine the two and create an mp4 file with ffmpeg.

  



  The problem is that the video source isn't always perfect. From time to time frames are dropped or late etc. This causes an A/V sync issue since the pts is generated using a constant rate by ffmpeg. The source format I am using has the PTS value but I cant figure out a way to pass it to ffmpeg with the raw h.264 frames.

  



  I suppose it would be possible to create a demuxer for the custom format, but it seems like a lot effort. I looked into ffmpeg's .nut container format thinking that I might be able to convert from the custom container to .nut first. Unfortunately it seems more complex than it looks on the surface.

  



  It seems like there should be an easy way to pass a frame and its PTS value to ffmpeg, but I haven't come across it yet. Any help would be appreciated.

  



  Here is the ffmpeg command I am using

  



  ffmpeg -f s16le -ac 1 -ar 48k -i source.audio -framerate 20 -i source.video -c:a aac -b:a 64k -r 20 -c:v h264_nvenc -rc:v vbr_hq -cq:v 19 -n out.mp4


  


• Compiling ffmpeg for iOS 5.1

  11 septembre 2012, par Vlad Bogdan

  I followed jold's steps from here :

  problem compiling ffmpeg for iFrameExtractor

  When i try jold's solution i get the following error for armv6 and armv7. i386 works perfect.

  lipo : can't open input file : armv6/libavcodec.a (No such file or directory)

  lipo : can't open input file : armv6/libavutil.a (No such file or directory

  lipo : can't open input file : armv6/libswscale.a (No such file or directory)

  When the script ends i have only libadevice.a and libavformat.a in /armv7 and /armv6.

  Why ?