Sur d’autres sites (6063)

• Ffmpeg - How can I create HLS multiple language streams, in multiple qualities ?

  28 avril 2022, par Daniel Ellis

  Preface

  


  I'm working on converting videos from 4k to multiple qualities with multiple languages but am having issues with the multiple languages overlaying, sometimes losing quality and sometimes being out of sync. (this is less of a problem in the German audio, as this is voice over anyhow)

  


  We as a team are complete noobs in terms of Video / Audio + HLS — I'm a front end developer who has no experience of this so apologies if my question is poorly phrased

  


  ---

  


  Videos

  


  I have the video in a 4k format and have removed the original sound as I have English and German audio files that need to be overlayed. I am then taking these files and throwing them together into a .ts file like this :

  


  $ ffmpeg -i ep03-ns-4k.mp4 -i nkit-ep3-de-output.m4a -i nkit-ep3-en-output.m4a \
> -thread 0 -muxdelay 0 -y \
> -map 0:v -map 1 -map 2  -movflags +faststart -refs 1 \
> -vcodec libx264 -acodec aac -profile:v baseline -level 30 -ar 44100 -ab 64k -f mpegts out.ts 


  


  This outputs a 4k out.ts video, with both audio tracks playing.

  


  The hard part

  


  This is where I'm finding it tricky, I now need to convert this single file into multiple quality levels (480, 720, 1080, 1920) and I attempt this with the following command :

  


  ffmpeg -hide_banner -y -i out.ts \
-crf 20 -sc_threshold 0 -g 48 -keyint_min 48 -ar 48000 \
-map 0:v:0 -map 0:v:0 -map 0:v:0 -map 0:v:0 \
-c:v:0 h264 -profile:v:0 main -filter:v:0 "scale=w=848:h=480:force_original_aspect_ratio=decrease" -b:v:0 1400k -maxrate:v:0 1498k -bufsize:v:0 2100k \
-c:v:1 h264 -profile:v:1 main -filter:v:1 "scale=w=1280:h=720:force_original_aspect_ratio=decrease" -b:v:1 2800k -maxrate:v:1 2996k -bufsize:v:1 4200k \
-c:v:2 h264 -profile:v:2 main -filter:v:2 "scale=w=1920:h=1080:force_original_aspect_ratio=decrease" -b:v:2 5600k -maxrate:v:2 5992k -bufsize:v:2 8400k \
-c:v:3 h264 -profile:v:3 main -filter:v:3 "scale=w=3840:h=1920:force_original_aspect_ratio=decrease" -b:v:3 11200k -maxrate:v:3 11984k -bufsize:v:3 16800k \
-var_stream_map "v:0 v:1 v:2 v:3" \
-master_pl_name master.m3u8 \
-f hls -hls_time 4 -hls_playlist_type vod -hls_list_size 0 \
-hls_segment_filename "%v/episode-%03d.ts" "%v/episode.m3u8"


  


  This creates the required qualities, but I'm now at a loss of how this might work with the audio

  


  Audio

  


  For the audio I run this command :

  


  ffmpeg -i out.ts -threads 0 -muxdelay 0 -y -map 0:a:0 -codec copy -f segment -segment_time 4 -segment_list_size 0 -segment_list audio-de/audio-de.m3u8 -segment_format mpegts audio-de/audio-de_%d.aac
ffmpeg -i out.ts -threads 0 -muxdelay 0 -y -map 0:a:1 -codec copy -f segment -segment_time 4 -segment_list_size 0 -segment_list audio-en/audio-en.m3u8 -segment_format mpegts audio-en/audio-en_%d.aac



  


  This creates the required audio segments.

  


  The question

  


  I realise this is quite an ask, but is there anything wrong with our inputs ? Is there a way that this can be done a bit more streamlined ?

  


  Any answers are greatly appreciated.

  


• there is no sound after adding a logo to a video moviepy

  1er juin 2021, par NKG

  I have a video with sound.
Then, using moviepy I am adding a logo.png on the video.
The video with the logo has sound, but when I upload it onto instagram there is no sound(
P.S. the original video uploaded onto the instagram has sound.

  


  there is a code bellow

  


  import moviepy.editor as mp


INPUT_FILE_PATH = rf'input\video.mp4'
OUTPUT_FILE_PATH = rf'output\video.mp4'

video = mp.VideoFileClip(INPUT_FILE_PATH)


logo = (mp.ImageClip("logo.png")
        .set_duration(video.duration)
        .resize(width=width / 3)
        .margin(right=width // 20, top=5 * height // 8, opacity=0)  # (optional) logo-border padding
        .set_pos(("right", "top")))

final = mp.CompositeVideoClip([video, logo])


final.write_videofile(OUTPUT_FILE_PATH, fps=30, codec="libx264", audio_fps=22050, audio_bitrate="31k")


  


  Maybe I need add some params to output video, But I don't know what params

  


• Xuggler Encoding video of Desktop With Audio - audio has gaps

  2 novembre 2012, par Chris

  I am using Xuggler to convert images captured from the java Robot class and sound read from TargetDataLine class and encoding this into a video. I am then attempting to http stream this video data (after writing my header) to a flash client via http (Socket OutputStream) but it plays and stutters (never just playing smoothly) no matter what buffer value I use on the client side.

  I am asking for help and showing my java code because I suspect it might be to do with how I am encoding the video or something about sending data via http socket which i am not getting..

  ByteArrayURLHandler ba = new ByteArrayURLHandler();
  
  final IRational FRAME_RATE = IRational.make(30); 
  
  final int SECONDS_TO_RUN_FOR = 20; 
  
  final Robot robot = new Robot(); 
  
  final Toolkit toolkit = Toolkit.getDefaultToolkit(); 
  
  final Rectangle screenBounds = new Rectangle(toolkit.getScreenSize()); 
  
  IMediaWriter writer;
  
  
  
  writer = ToolFactory.makeWriter(
  
      XugglerIO.map( 
  
          XugglerIO.generateUniqueName(out, ".flv"), 
  
          out 
  
      )); 
  
  
  
  writer.addListener(new MediaListenerAdapter() { 
  
      public void onAddStream(IAddStreamEvent event) { 
  
          event.getSource().getContainer().setInputBufferLength(1000);
  
          IStreamCoder coder = event.getSource().getContainer().getStream(event.getStreamIndex()).getStreamCoder();
  
          if (coder.getCodecType() == ICodec.Type.CODEC_TYPE_AUDIO) { 
  
              coder.setFlag(IStreamCoder.Flags.FLAG_QSCALE, false);   
  
              coder.setBitRate(32000); 
  
              System.out.println("onaddstream"+ coder.getPropertyNames().toString());
  
          } 
  
          if (coder.getCodecType() == ICodec.Type.CODEC_TYPE_VIDEO) { 
  
              // coder.setBitRate(64000); 
  
              // coder.setBitRateTolerance(64000); 
  
          } 
  
      } 
  
  });
  
  
  
  writer.addVideoStream(videoStreamIndex, videoStreamId, 1024, 768);
  
  final int channelCount = 1;       
  
  
  
  int audionumber =   writer.addAudioStream(audioStreamIndex, audioStreamId,1, 44100);
  
  int bufferSize = (int)audioFormat.getSampleRate()   *audioFormat.getFrameSize();//*6;///6;
  
  byte[] audioBuf;// = new byte[bufferSize]; 
  
  
  
  int i = 0;
  
  
  
  final int audioStreamIndex = 1;
  
  final int audioStreamId = 1;
  
  BufferedImage screen, bgrScreen;
  
  long startTime = System.nanoTime();
  
  while(keepGoing)
  
  { 
  
  
  
      audioBuf = new byte[bufferSize]; 
  
      i++;
  
  
  
      screen = robot.createScreenCapture(screenBounds); 
  
  
  
      bgrScreen = convertToType(screen, BufferedImage.TYPE_3BYTE_BGR); 
  
      long nanoTs = System.nanoTime()-startTime; 
  
      writer.encodeVideo(0, bgrScreen, nanoTs, TimeUnit.NANOSECONDS);
  
      audioBuf = new byte[line.available()];
  
      int nBytesRead = line.read(audioBuf, 0, audioBuf.length); 
  
  
  
      IBuffer iBuf = IBuffer.make(null, audioBuf, 0, nBytesRead);
  
  
  
      IAudioSamples smp = IAudioSamples.make(iBuf,1,IAudioSamples.Format.FMT_S16);
  
      if (smp == null) {
  
          return;
  
      }
  
  
  
      long numSample = audioBuf.length / smp.getSampleSize();
  
  
  
      smp.setComplete(true, numSample,(int)
  
      audioFormat.getSampleRate(), audioFormat.getChannels(),
  
      IAudioSamples.Format.FMT_S16, nanoTs/1000);
  
  
  
      writer.encodeAudio(1, smp);
  
  
  
      writer.flush();
  
  }