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• mp4 plays only sound but no video on ios

  5 avril 2018, par Chen

  I have a video that I get from a third party provide in my App, I want to play using react-native-video. It works fine on simulator, but on a real iphone 7 it has only sound but no video.

  After further research I found, when I try to airdrop/move via itunes the video to my iphone. It shows something like the following :

  

  I have checked the codec with ffprobe and here is the output.
  ```

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'bad.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: mp42mp41
  
      creation_time   : 2018-03-28T21:30:13.000000Z
  
    Duration: 00:00:45.55, start: 0.000000, bitrate: 1373 kb/s
  
      Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 1042 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
  
      Metadata:
  
        creation_time   : 2018-03-28T21:30:13.000000Z
  
        handler_name    : Alias Data Handler
  
        encoder         : AVC Coding
  
      Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 317 kb/s (default)
  
      Metadata:
  
        creation_time   : 2018-03-28T21:30:13.000000Z
  
        handler_name    : Alias Data Handler
  
  [STREAM]
  
  index=0
  
  codec_name=h264
  
  codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
  
  profile=High
  
  codec_type=video
  
  codec_time_base=1001/60000
  
  codec_tag_string=avc1
  
  codec_tag=0x31637661
  
  width=1920
  
  height=1080
  
  coded_width=1920
  
  coded_height=1080
  
  has_b_frames=1
  
  sample_aspect_ratio=1:1
  
  display_aspect_ratio=16:9
  
  pix_fmt=yuv420p
  
  level=41
  
  color_range=tv
  
  color_space=bt709
  
  color_transfer=bt709
  
  color_primaries=bt709
  
  chroma_location=left
  
  field_order=unknown
  
  timecode=N/A
  
  refs=1
  
  is_avc=true
  
  nal_length_size=4
  
  id=N/A
  
  r_frame_rate=30000/1001
  
  avg_frame_rate=30000/1001
  
  time_base=1/30000
  
  start_pts=0
  
  start_time=0.000000
  
  duration_ts=1365364
  
  duration=45.512133
  
  bit_rate=1042237
  
  max_bit_rate=N/A
  
  bits_per_raw_sample=8
  
  nb_frames=1364
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=1
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  DISPOSITION:timed_thumbnails=0
  
  TAG:creation_time=2018-03-28T21:30:13.000000Z
  
  TAG:language=eng
  
  TAG:handler_name=Alias Data Handler
  
  TAG:encoder=AVC Coding
  
  [/STREAM]
  
  [STREAM]
  
  index=1
  
  codec_name=aac
  
  codec_long_name=AAC (Advanced Audio Coding)
  
  profile=LC
  
  codec_type=audio
  
  codec_time_base=1/48000
  
  codec_tag_string=mp4a
  
  codec_tag=0x6134706d
  
  sample_fmt=fltp
  
  sample_rate=48000
  
  channels=2
  
  channel_layout=stereo
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=0/0
  
  time_base=1/48000
  
  start_pts=0
  
  start_time=0.000000
  
  duration_ts=2184582
  
  duration=45.512125
  
  bit_rate=317375
  
  max_bit_rate=458625
  
  bits_per_raw_sample=N/A
  
  nb_frames=2135
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=1
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  DISPOSITION:timed_thumbnails=0
  
  TAG:creation_time=2018-03-28T21:30:13.000000Z
  
  TAG:language=eng
  
  TAG:handler_name=Alias Data Handler
  
  [/STREAM]

  Then I tried VLC on iOS the video played without problem (it has both video and sound), except at the beginning it shows a grey pixel screen.

  

  Next, what i did was using ffmpeg to re-encode the video stream via :

  ffmpeg -i bad.mp4 -c:a copy new.mp4

  After this new.mp4 plays with no problem.
  This is really confusing as the codec does not change and both videos works with no problem on my Mac.

  I have tried to play around with the b_frames, profile level on the new video to reproduce the problem but with no luck.

  I really do not want to process every video via ffmpeg after get it from the third party. Just wondering what might be the problem with the video ?

• How to have low buffer when capturing sound with FFmpeg

  8 mai 2018, par gagz

  I’m capturing sound from the jack input of a computer and sending that to a Icecast server, using FFmpeg.

  See the line :

  /usr/bin/ffmpeg -f alsa -i plughw:1 -c:a libmp3lame -b:a 96k -ar 32000 -content_type audio/mpeg -f mp3 icecast://source:password@myserver.net:8000/live

  On the Icecast side, I use default settings, and play the stream with mplayer.

  But the bandwidth on both sides is quite low (3G connections), thus after a while, the delay goes up to a few minutes, so I have to restart ffmpeg.

  I’m quiet sure now that ffmpeg keeps collecting data when frames are dropped by the network, but it keeps them until the connection comes back and then sends them to icecast.

  How can I tell FFmpeg to stay synchronised with the sound card ? Or have a maximum of 5 seconds delay ?

  Thank you !

• Android. Problems with AudioTrack class. Sound sometimes lost

  6 juin 2018, par bukka.wh

  I have found open source video player for Android, which uses ffmpeg to decode video.
  I have some problems with audio, that sometimes plays with jerks, but video picture is shown well. The basic idea of player is that audio and video are decoded in two different streams, and then in the third stream the are passed back, video picture is shown on SurfaceView and video sound is passed in byte array to AudioTrack and then plays. But sometimes sound is lost or playing with jerks. Can anyone give me start point for what to do (some basic concepts). May be I should change buffer size for AudioTrack or add some flags to it. Here is a piece of code, where AudioTrack class is created.

  private AudioTrack prepareAudioTrack(int sampleRateInHz,
  
          int numberOfChannels) {
  
  
  
      for (;;) {
  
          int channelConfig;
  
          if (numberOfChannels == 1) {
  
              channelConfig = AudioFormat.CHANNEL_OUT_MONO;
  
          } else if (numberOfChannels == 2) {
  
              channelConfig = AudioFormat.CHANNEL_OUT_STEREO;
  
          } else if (numberOfChannels == 3) {
  
              channelConfig = AudioFormat.CHANNEL_OUT_FRONT_CENTER
  
                      | AudioFormat.CHANNEL_OUT_FRONT_RIGHT
  
                      | AudioFormat.CHANNEL_OUT_FRONT_LEFT;
  
          } else if (numberOfChannels == 4) {
  
              channelConfig = AudioFormat.CHANNEL_OUT_QUAD;
  
          } else if (numberOfChannels == 5) {
  
              channelConfig = AudioFormat.CHANNEL_OUT_QUAD
  
                      | AudioFormat.CHANNEL_OUT_LOW_FREQUENCY;
  
          } else if (numberOfChannels == 6) {
  
              channelConfig = AudioFormat.CHANNEL_OUT_5POINT1;
  
          } else if (numberOfChannels == 8) {
  
              channelConfig = AudioFormat.CHANNEL_OUT_7POINT1;
  
          } else {
  
              channelConfig = AudioFormat.CHANNEL_OUT_STEREO;
  
          }
  
          try {
  
              Log.d("MyLog","Creating Audio player");
  
              int minBufferSize = AudioTrack.getMinBufferSize(sampleRateInHz,
  
                      channelConfig, AudioFormat.ENCODING_PCM_16BIT);
  
              AudioTrack audioTrack = new AudioTrack(
  
                      AudioManager.STREAM_MUSIC, sampleRateInHz,
  
                      channelConfig, AudioFormat.ENCODING_PCM_16BIT,
  
                      minBufferSize, AudioTrack.MODE_STREAM);
  
              return audioTrack;
  
          } catch (IllegalArgumentException e) {
  
              if (numberOfChannels > 2) {
  
                  numberOfChannels = 2;
  
              } else if (numberOfChannels > 1) {
  
                  numberOfChannels = 1;
  
              } else {
  
                  throw e;
  
              }
  
          }
  
      }
  
  }

  And this is a piece of native code where sound bytes are written to AudioTrack

  int player_write_audio(struct DecoderData *decoder_data, JNIEnv *env,
  
      int64_t pts, uint8_t *data, int data_size, int original_data_size) {
  
  struct Player *player = decoder_data->player;
  
  int stream_no = decoder_data->stream_no;
  
  int err = ERROR_NO_ERROR;
  
  int ret;
  
  AVCodecContext * c = player->input_codec_ctxs[stream_no];
  
  AVStream *stream = player->input_streams[stream_no];
  
  LOGI(10, "player_write_audio Writing audio frame")
  
  
  
  jbyteArray samples_byte_array = (*env)->NewByteArray(env, data_size);
  
  if (samples_byte_array == NULL) {
  
      err = -ERROR_NOT_CREATED_AUDIO_SAMPLE_BYTE_ARRAY;
  
      goto end;
  
  }
  
  
  
  if (pts != AV_NOPTS_VALUE) {
  
      player->audio_clock = av_rescale_q(pts, stream->time_base, AV_TIME_BASE_Q);
  
      LOGI(9, "player_write_audio - read from pts")
  
  } else {
  
      int64_t sample_time = original_data_size;
  
      sample_time *= 1000000ll;
  
      sample_time /= c->channels;
  
      sample_time /= c->sample_rate;
  
      sample_time /= av_get_bytes_per_sample(c->sample_fmt);
  
      player->audio_clock += sample_time;
  
      LOGI(9, "player_write_audio - added")
  
  }
  
  enum WaitFuncRet wait_ret = player_wait_for_frame(player,
  
          player->audio_clock + AUDIO_TIME_ADJUST_US, stream_no);
  
  if (wait_ret == WAIT_FUNC_RET_SKIP) {
  
      goto end;
  
  }
  
  
  
  LOGI(10, "player_write_audio Writing sample data")
  
  
  
  jbyte *jni_samples = (*env)->GetByteArrayElements(env, samples_byte_array,
  
          NULL);
  
  memcpy(jni_samples, data, data_size);
  
  (*env)->ReleaseByteArrayElements(env, samples_byte_array, jni_samples, 0);
  
  
  
  LOGI(10, "player_write_audio playing audio track");
  
  ret = (*env)->CallIntMethod(env, player->audio_track,
  
          player->audio_track_write_method, samples_byte_array, 0, data_size);
  
  jthrowable exc = (*env)->ExceptionOccurred(env);
  
  if (exc) {
  
      err = -ERROR_PLAYING_AUDIO;
  
      LOGE(3, "Could not write audio track: reason in exception");
  
      // TODO maybe release exc
  
      goto free_local_ref;
  
  }
  
  if (ret < 0) {
  
      err = -ERROR_PLAYING_AUDIO;
  
      LOGE(3,
  
              "Could not write audio track: reason: %d look in AudioTrack.write()", ret);
  
      goto free_local_ref;
  
  }
  
  
  
  free_local_ref:
  
  LOGI(10, "player_write_audio releasing local ref");
  
  (*env)->DeleteLocalRef(env, samples_byte_array);
  
  
  
  end: return err;

  }

  I will be pleased for any help !!!! Thank you very much !!!!