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• Libav and xaudio2 - audio not playing

  25 août 2013, par siz

  I am trying to get audio playing with libav using xaudio2. The xaudio2 code I am using works with an older ffmpeg using avcodec_decode_audio2, but that has been deprecated for avcodec_decode_audio4. I have tried following various libav examples, but can't seem to get the audio to play. Video plays fine (or rather it just plays right fast now, as I haven't coded any sync code yet).

  Firstly audio gets init, no errors, video gets init, then packet :

  while (1) {
  
      //is this packet from the video or audio stream?
  
      if (packet.stream_index == player.v_id) {
  
          add_video_to_queue(&packet);
  
      } else if (packet.stream_index == player.a_id) { 
  
          add_sound_to_queue(&packet);
  
      } else {
  
          av_free_packet(&packet);
  
      }
  
  }

  Then in add_sound_to_queue :

  int add_sound_to_queue(AVPacket * packet) {
  
  AVFrame *decoded_frame = NULL;
  
  int done = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
  int got_frame = 0;
  
  
  
  if (!decoded_frame) {
  
      if (!(decoded_frame = avcodec_alloc_frame())) {
  
          printf("[ADD_SOUND_TO_QUEUE] Out of memory\n");
  
          return -1;
  
      }
  
  } else {
  
      avcodec_get_frame_defaults(decoded_frame);
  
  }
  
  
  
  if (avcodec_decode_audio4(player.av_acodecctx, decoded_frame, &got_frame, packet) < 0)  {
  
      printf("[ADD_SOUND_TO_QUEUE] Error in decoding audio\n");
  
      av_free_packet(packet);
  
      //continue;
  
      return -1;
  
  }
  
  if (got_frame) {
  
      int data_size;
  
      if (packet->size > done) {
  
          data_size = done;
  
      } else {
  
          data_size = packet->size;
  
      }
  
      BYTE * snd = (BYTE *)malloc( data_size * sizeof(BYTE));
  
  
  
      XMemCpy(snd,
  
          AudioBytes, 
  
          data_size * sizeof(BYTE)
  
      );
  
  
  
      XMemSet(&g_SoundBuffer,0,sizeof(XAUDIO2_BUFFER));
  
  
  
      g_SoundBuffer.AudioBytes = data_size;
  
      g_SoundBuffer.pAudioData = snd;
  
      g_SoundBuffer.pContext = (VOID*)snd;
  
  
  
      XAUDIO2_VOICE_STATE state;
  
      while( g_pSourceVoice->GetState( &state ), state.BuffersQueued > 60 ) {
  
          WaitForSingleObject( XAudio2_Notifier.hBufferEndEvent, INFINITE );
  
      }
  
  
  
      g_pSourceVoice->SubmitSourceBuffer( &g_SoundBuffer );
  
  }
  
  return 0;
  
  }

  I can't seem to figure out the problem, I have added error messages in init, opening video, codec handling etc. As mentioned before the xaudio2 code is working with an older ffmpeg, so maybe I have missed something with the avcodec_decode_audio4 ?

  If this snappet of code isn't enough, I can post the whole code, these are just the places in the code I think the problem would be :(

• How to copy audio stream using FFMpeg API ( not a command line tool )

  12 août 2013, par Jindong Jung

  I'm developing some Video Editing Apps on Android.

  the objective of the app is "Editing Videos on Android".

  and...

  I'm just completed making video file using some images.
  but.. I can't attach audio into the video.

  my method is same as follows.

  1.VideoStream, audio stream creation using AVFormatContext

  2.Movie encoding in video stream was successful

  3.Encode codec open in audio stream was successful

  4.Set sample format to AV_SAMPLE_FMT_FLTP

  5.Sample rate and channel was set same as source audio

  6.Choose appropriate Decoder and read packet

  7.Convert packets using swr_converter, setting same as sample format

  8.Encode converted data

  9.memory deallocation

  10.END !

  ---

  Problem is here :

  Video of finally created video file was normally played. but the Audio wasn't.

  It heared like weird. It have many noises and plays slowly.

  I've googled with many keywords but they only say about "FFmpeg command line usage".

  I wanna make with FFMpeg API. not a Command line tool.

  Please help.

• ffmpeg commands to concatenate different type and resolution videos into 1 video and can be played in android

  26 octobre 2015, par Aalap

  I want to concatinate 4 different videos of 4 different resolution and type into 1 video which can be played in android. I am using ffmpeg ported on android using https://github.com/guardianproject/android-ffmpeg

  So I have these 4 different types of videos
  1)

  ./ffmpeg -i 1.mp4 
  
  Video: h264 (High), yuv420p, 1920x1080, 16959 kb/s, 29.85 fps, 90k tbr, 90k tbn, 180k tbc
  
  Audio: aac, 48000 Hz, stereo, s16, 106 kb/s

  2)

  ffmpeg -i 2.mp4
  
  Video: h264 (Constrained Baseline), yuv420p, 640x480, 3102 kb/s, 29.99 fps, 90k tbr, 90k tbn, 180k tbc
  
  Audio: aac, 48000 Hz, stereo, s16, 93 kb/s

  3)

  ffmpeg -i 3.3gp
  
  Video: h263, yuv420p, 1408x1152 [PAR 12:11 DAR 4:3], 2920 kb/s, 15 fps, 15 tbr, 15360 tbn, 29.97 tbc
  
  Audio: amrnb, 8000 Hz, 1 channels, flt, 12 kb/s

  4)

  ffmpeg -i 4.3gp
  
  Video: h264 (High), yuv420p, 352x288 [PAR 12:11 DAR 4:3], 216 kb/s, 24 fps, 24 tbr, 24 tbn, 48 tbc

  Audio : aac, 44100 Hz, stereo, s16, 92 kb/s

  So I am converting them to mpegts using following commands

  ./ffmpeg -i 1.mp4 -c:v libx264 -vf scale=1920:1080 -r 60 -c:a aac -ar 48000 -b:a 160k -strict experimental -f mpegts 1.ts
  
  ./ffmpeg -i 2.mp4 -c:v libx264 -vf scale=1920:1080 -r 60 -c:a aac -ar 48000 -b:a 160k -strict experimental -f mpegts 2.ts
  
  ./ffmpeg -i 3.3gp -c:v libx264 -vf scale=1920:1080 -r 60 -c:a aac -ar 48000 -b:a 160k -strict experimental -f mpegts 3.ts
  
  ./ffmpeg -i 4.3gp -c:v libx264 -vf scale=1920:1080 -r 60 -c:a aac -ar 48000 -b:a 160k -strict experimental -f mpegts 4.ts

  then concatenating the .ts files into f.ts and then creating a final .mp4 file from it using

  cat 1.ts 2.ts 3.ts 4.ts > f.ts
  
  ./ffmpeg -i f.ts -c copy -bsf:a aac_adtstoasc output.mp4

  But my f.ts also doesnt seem to play correctly in VLC on linux, it plays first 2 mp4’s video + audio and it plays last .3gp’s audio only.(Same for output.mp4 too) Could you please help me in figuring out what am I missing ?

  Thanks in advance