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• NodeJS piping with ffmpeg

  8 février 2014, par Gnap

  I wanted to do a HTTP live stream on a screen cast with using ffmpeg, nodejs and html5 . I wanted it to be as real time as possible. However, I find that my video received by the client was behind by 1 2 seconds (On Chrome/Chromium). I am using vp8/webm as my codec.

  I have eliminated the following factors as such :
  1) Network : I have tried serving and receiving the video file locally by stating the video source to be 127.0.0.1:PORT or localhost:PORT
  2) ffmpeg encoding speed:I have tried outputting the file locally, it the "delay" seems to be negligible.
  3) Chrome internal buffer. The buffer was accounted to be 0.07s 0.08s.

  On the nodeJS side, I have a child process that runs the ffmpeg command, and did a ffmpeg.stdout.pipe(res) ; <— ffmpeg is child_process.spawn(...)

  So it seems that the ffmpeg.std.pipe(res) of nodejs seems to be the one delaying the video stream. Am I correct in assuming so ? Is there anyway that I may reduce the delay ?

• HTTP Header for Duration of a MP4 for HTML 5 video

  9 mars 2014, par Mustafa

  I am trying to stream MP4 video as it is encoded from a webserver. I believe I used the appropriate flags, but it is not working correctly. When I download the video from my stream and open it with VLC, it properly shows the duration. Since a socket is not seekable, I assume it writes the metadata to end ? My Chrome browser always shows 8 seconds duration. The first 8 seconds plays at the normal speed, but afterwards the pause button turns into play button and the video plays very fast, probably as fast as it is recieved. However the audio is played at normal speed. I tried document.getElementById('myVid').duration = 20000 but it is a readonly field.

  I wonder, is there anyway to explicitly state the duration in HTTP headers or in any other way ? I cannot find any documentation about it.

  ffmpeg -i - -vcodec libx264 -acodec libvo_aacenc -ar 44100 -ac 2 -ab 128000 -f mp4 -movflags frag_keyframe+faststart pipe:1 -fflags +genpts -re -profile baseline -level 30 -preset fast

  To close-voters, that thinks it is not programming related, I use it in my own server I coded, and I need to set the duration programatically via JavaScript or setting the HTTP header. I believe it may be related to both ffmpeg or http headers, that's why I posted it here.

  app.get("/video/*", function(req,res){
  
      res.writeHead(200, {
  
          'Content-Type': 'video/mp4',
  
      });
  
      var dir = req.url.split("/").splice(2).join("/");
  
      var buf = new Buffer(dir, 'base64');
  
      var src = buf.toString();
  
  
  
      var Transcoder = require('stream-transcoder');
  
      var stream = fs.createReadStream(src);
  
      // I added my own flags to this module, they are at below:
  
      new Transcoder(stream)
  
          .videoCodec('libx264')
  
          .audioCodec("libvo_aacenc")
  
          .sampleRate(44100)
  
          .channels(2)
  
          .audioBitrate(128 * 1000)
  
          .format('mp4')
  
          .on('finish', function() {
  
              console.log("finished");
  
          })
  
          .stream().pipe(res);
  
  });

  exec function in that stream-transcoder module,

      a.push("-fflags");
  
      a.push("+genpts");
  
      a.push("-re");
  
      a.push("-profile");
  
      a.push("baseline");
  
      a.push("-level");
  
      a.push("30");
  
      a.push("-preset");
  
      a.push("fast");
  
      a.push("-strict");
  
      a.push("experimental");
  
      a.push("-frag_duration");
  
      a.push("" + 2 * (1000 * 1000));
  
      var child = spawn('ffmpeg', a, {
  
          cwd: os.tmpdir()
  
      });

• Decoding an mp3 file using FFmpeg but sound is glitchy

  28 avril 2017, par satyres

  After successfuly compiling the latest version of FFmpeg library and generated .a library in Ubuntu I’ve been struggling now for more than a week to play a simple mp3 file in Android without a success !
  The sound on my S4 working but it’s glitchy and stuttering
  I’ve followed this tutorial given by FFmpeg team in Github i’ve tried to use it in Android but no luck !
  here is the Native code.

   void Java_com_example_home_hellondk_MainActivity_audio_1decode_1example(JNIEnv * env, jobject obj, jstring file, jbyteArray array) {
  
          jboolean isfilenameCopy;
  
           const char * filename = (*env)->GetStringUTFChars(env, file,
  
                   &isfilenameCopy);
  
  jclass cls = (*env)->GetObjectClass(env, obj);
  
           jmethodID play = (*env)->GetMethodID(env, cls, "playSound", "([BI)V");
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int len;
  
      FILE *f, *outfile;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
      AVFrame *decoded_frame = NULL;
  
  AVFormatContext* container=NULL;
  
      av_init_packet(&avpkt);
  
  
  
      printf("Decode audio file %s \n", filename);
  
  LOGE("Decode audio file %s\n", filename);
  
      /* find the MPEG audio decoder */
  
    /*  codec = avcodec_find_decoder(AV_CODEC_ID_MP3);
  
      if (!codec) {
  
          fprintf(stderr, "Codec not found\n");
  
          LOGE("Codec not found\n");
  
          exit(1);
  
      }*/
  
  int lError;
  
           if ((lError = avformat_open_input(&container, filename, NULL, NULL))
  
                   != 0) {
  
               LOGE("Error open source file: %d", lError);
  
               exit(1);
  
           }
  
           if ((lError =  avformat_find_stream_info(container,NULL)) < 0) {
  
               LOGE("Error find stream information: %d", lError);
  
               exit(1);
  
           }
  
           LOGE("Stage 1.5");
  
          LOGE("audio format: %s", container->iformat->name);
  
         LOGE("audio bitrate: %llu", container->bit_rate);
  
  
  
      int stream_id = -1;
  
          // To find the first audio stream. This process may not be necessary
  
          // if you can gurarantee that the container contains only the desired
  
          // audio stream
  
             LOGE("nb_streams: %d", container->nb_streams);
  
          int i;
  
          for (i = 0; i < container->nb_streams; i++) {
  
              if (container->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  
                  stream_id = i;
  
                   LOGE("stream_id: %d", stream_id);
  
                  break;
  
              }
  
          }
  
  
  
          AVCodecContext* codec_context = container->streams[stream_id]->codec;
  
          codec = avcodec_find_decoder(codec_context->codec_id);
  
           LOGE("stream_id: %d", stream_id);
  
          LOGE("codec %s", codec->name);
  
          if (!codec) {
  
              fprintf(stderr, "codec not found\n");
  
              exit(1);
  
          }
  
  
  
      c = avcodec_alloc_context3(codec);
  
      if (!c) {
  
          fprintf(stderr, "Could not allocate audio codec context\n");
  
            LOGE("Could not allocate audio codec context\n");
  
          exit(1);
  
      }
  
  
  
      /* open it */
  
      if (avcodec_open2(c, codec, NULL) < 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
           LOGE("Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      f = fopen(filename, "rb");
  
      if (!f) {
  
          fprintf(stderr, "Could not open %s\n", filename);
  
          LOGE("Could not open %s\n",filename);
  
          exit(1);
  
      }
  
  
  
  
  
  
  
          avpkt.data = inbuf;
  
          avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
  
          LOGE("Stage 5");
  
      /* decode until eof */
  
  
  
      while (1) {
  
  
  
      if ((len = av_read_frame(container, &avpkt)) < 0)
  
                         break;
  
  
  
          if (avpkt.stream_index == stream_id)
  
                       {
  
        if (!decoded_frame) {
  
                                               if (!(decoded_frame = av_frame_alloc())) {
  
                                                   fprintf(stderr, "Could not allocate audio frame\n");
  
                                                    LOGE("Could not allocate audio frame\n");
  
                                                   exit(1);
  
                                               }
  
                                           }
  
                              int got_frame = 0;
  
                                len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
  
                               LOGE("len=%d",len);
  
                               if (len < 0)
  
                               {
  
                                   LOGE("Error decoding audio\n");
  
                                   continue;
  
                               }
  
  
  
                               if (got_frame)
  
                               {
  
                                    LOGE("begin frame decode\n");
  
                                    int data_size = av_samples_get_buffer_size(NULL, c->channels,decoded_frame->nb_samples,c->sample_fmt, 1);
  
                                  if (data_size>0)
  
                                  {
  
                                    LOGE("after frame decode %d\n",data_size);
  
  
  
                                   jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
                                    memcpy(bytes, decoded_frame->data[0], data_size);
  
                                    (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
                                    (*env)->CallVoidMethod(env, obj, play, array, data_size);
  
  
  
                                   }
  
                                   else
  
                                   {
  
                                   fprintf(stderr, "Failed to calculate data size\n");
  
                                    exit(1);
  
  
  
                                   }
  
  
  
                               }
  
  
  
                               avpkt.size -= len;
  
                               avpkt.data += len;
  
                               avpkt.pts = AV_NOPTS_VALUE;
  
                              if (avpkt.size < AUDIO_REFILL_THRESH)
  
                              {
  
                                  memmove(inbuf, avpkt.data, avpkt.size);
  
                                  avpkt.data = inbuf;
  
                                  len = fread(avpkt.data + avpkt.size, 1, AUDIO_INBUF_SIZE - avpkt.size, f);
  
                                  if (len > 0)
  
                                      avpkt.size += len;
  
                              }
  
  
  
                         }
  
  
  
  
  
  
  
  
  
      }
  
  
  
  
  
      fclose(f);
  
  
  
      avcodec_free_context(&c);
  
  av_frame_free(&decoded_frame);
  
      }

  The Java code :

  package com.example.home.hellondk;
  
  
  
  import android.media.AudioFormat;
  
  import android.media.AudioManager;
  
  import android.media.AudioTrack;
  
  import android.media.MediaPlayer;
  
  import android.support.v7.app.AppCompatActivity;
  
  import android.os.Bundle;
  
  import android.util.Log;
  
  
  
  import java.io.File;
  
  import java.io.FileNotFoundException;
  
  import java.io.FileOutputStream;
  
  import java.io.IOException;
  
  
  
  public class MainActivity extends AppCompatActivity {
  
      static {
  
          System.loadLibrary("MyLibraryPlayer");
  
      }
  
      public native void createEngine();
  
  
  
      public native void audio_decode_example(String outfilename, byte[] array);
  
  
  
  
  
  
  
      private AudioTrack track;
  
      private FileOutputStream os;
  
  
  
      @Override
  
      protected void onCreate(Bundle savedInstanceState) {
  
          super.onCreate(savedInstanceState);
  
          setContentView(R.layout.activity_main);
  
          createEngine();
  
  
  
   /*        MediaPlayer mp = new MediaPlayer();
  
          mp.start();*/
  
  
  
          int bufSize = AudioTrack.getMinBufferSize(32000,
  
                  AudioFormat.CHANNEL_CONFIGURATION_STEREO,
  
                  AudioFormat.ENCODING_PCM_16BIT);
  
  
  
          track = new AudioTrack(AudioManager.STREAM_MUSIC,
  
                  32000,
  
                  AudioFormat.CHANNEL_CONFIGURATION_STEREO,
  
                  AudioFormat.ENCODING_PCM_16BIT,
  
                  bufSize,
  
                  AudioTrack.MODE_STREAM);
  
  
  
          byte[] bytes = new byte[bufSize];
  
  
  
  
  
     audio_decode_example("/storage/emulated/0/test.mp3", bytes);
  
  
  
      }
  
  
  
      void playSound(byte[] buf, int size) {
  
          //android.util.Log.v("ROHAUPT", "RAH Playing");
  
          if (track.getPlayState() != AudioTrack.PLAYSTATE_PLAYING)
  
              track.play();
  
          track.write(buf, 0, size);
  
  
  
  
  
      }
  
  }

  Thank you so much for your help.
  Kind regards