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• avformat/webpenc : Fix memleak when trailer is never written

  18 mars 2021, par Andreas Rheinhardt

  avformat/webpenc : Fix memleak when trailer is never written
  When the trailer is never written (or when a stream switches from
  
  non-animation mode to animation mode mid-stream), a cached packet
  
  (if existing) would leak. Fix this by adding a deinit function.
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavformat/webpenc.c

• Could you please guide me on how to play a single sample of an audio file in C# using FFmpeg ?

  11 juillet 2023, par hello world

  What is the recommended approach for playing a single sample of an audio file in C# using FFmpeg ? I would like to incorporate FFmpeg into my C# application to play a specific sample from an audio file. Could someone provide an example or guide me on how to achieve this ? Any help would be appreciated.

  


  using System;
using System.Diagnostics;

public class AudioPlayer
{
    private string ffmpegPath;

    public AudioPlayer(string ffmpegPath)
    {
        this.ffmpegPath = ffmpegPath;
    }

    public void PlayAudioSample(string audioFilePath, TimeSpan samplePosition)
    {
        // Prepare FFmpeg process
        Process ffmpegProcess = new Process();
        ffmpegProcess.StartInfo.FileName = ffmpegPath;
        ffmpegProcess.StartInfo.Arguments = $"-ss {samplePosition} -i \"{audioFilePath}\" -t 1 -acodec pcm_s16le -f wav -";
        ffmpegProcess.StartInfo.RedirectStandardOutput = true;
        ffmpegProcess.StartInfo.RedirectStandardError = true;
        ffmpegProcess.StartInfo.UseShellExecute = false;
        ffmpegProcess.StartInfo.CreateNoWindow = true;

        // Start FFmpeg process
        ffmpegProcess.Start();

        // Play audio sample
        using (var audioOutput = new NAudio.Wave.WaveOutEvent())
        {
            using (var audioStream = new NAudio.Wave.RawSourceWaveStream(ffmpegProcess.StandardOutput.BaseStream, new NAudio.Wave.WaveFormat(44100, 16, 2)))
            {
                audioOutput.Init(audioStream);
                audioOutput.Play();
                while (audioOutput.PlaybackState == NAudio.Wave.PlaybackState.Playing)
                {
                    System.Threading.Thread.Sleep(100);
                }
            }
        }

        // Wait for FFmpeg process to exit
        ffmpegProcess.WaitForExit();
    }
}


  


• FFMPEG UDP to DASH I am wondering if it could be written more efficiently ?

  18 mars 2019, par c7borg

  I have the below working but it is quite cpu intensive I’ve just moved to ffmpeg 3.4 and was wondering if it could be written more efficently ?

  The below takes a multicast stream from our local LAN avoids the choppy footage by using the scenecut then adjusts the audio with the async to keep it in time and uses the yadif to deinterlace to provide better quality. This command/script also trims the maximum number of segments otherwise with a live stream it would fill up the directory.

  If anyone has any improvements I’d much appreciate it

  I also add subtitles using -filter_complex "[0:v][0:s]overlay" but this conflicts with the -vf yadif option.

  ffmpeg -i \
  
  "udp://@239.192.4.5:1234?overrun_nonfatal=1&fifo_size=50000000" \
  
  -acodec aac -strict -2 -vcodec libx264 \
  
  -vf yadif \
  
  -af aresample=async=1 \
  
  -x264opts 'keyint=25:scenecut=-1' \
  
  -window_size 10 -extra_window_size 10 \
  
  -f dash /var/www/html/stream/out.mpd

  If it can’t be written more efficiently at least this may help others as it took me a long time to get this far. For reference I use shaka player in chromium for the client side