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• ffmpeg to combine two mov files from iPhone

  8 octobre 2013, par Jason Stallings

  I'm trying to combine two videos recorded on an iPhone into one file with ffmpeg.

  I've tried everything I could find and I can't get anything to work right.

  My current line is

  ffmpeg -i 'concat:output.mov|capturedvideo.MOV' -vcodec copy -acodec copy output2.mov

  This currently won't work. The end result needs to be played on an iPhone.

• ffmpeg metadata, multiline text into comment field

  15 avril 2013, par decss

  my question is next : i use ffmpeg to update metadata tags like this way

  ffmpeg -i "in.m4a" -acodec copy -metadata artist="artist text"
  
  -metadata comment="comment text" "out.m4a"

  Well, "artist text" and "comment text" is usual plain text, but comment field allow to put multiline text. Each line must have new line character to do it. Something like this :

  -metadata comment="source: lastfm \r\n tags: tag1, tag2 \r\n ..."

  But all characters that i use (\r\n, %nl%, ^N) brings no result

  So i have no idea how to put multiline text into a comment field via ffmpeg
  Hope you will help me : ) Thanks !

• Setting up audio queue for ffmpeg rtsp stream playing

  20 novembre 2011, par illew

  I'm working on an rtsp streaming(AAC format) client for iOS using ffmpeg. Right now I can only say my app is workable, but the streaming sound is very noisy and even a little distorted, far worse than when it's played by vlc or mplayer.

  The stream is read by av_read_frame(), decoded by avcodec_decode_audio3(). Then I just send the decoded raw audio to Audio Queue.

  When decoding a local aac file with my app, the sound seemed not so noisy at all. I know initial encoding would dramatically affect the result. However at least I should try to have it sounded like other streaming clients...

  Many parts in my implementation/modification actually came from try and error. I believe I'm doing something wrong in setting up Audio Queue, and the callback function for filling Audio Buffer.

  Any hints, suggestions or help are greatly appreciated.

  // —info of test materials dumped by av_dump_format() —

  Metadata:
  
      title           : /demo/test.3gp
  
    Duration: 00:00:30.11, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Audio: aac, 32000 Hz, stereo, s16
  
  aac  Advanced Audio Coding 

  // — the Audio Queue setup procedure —

  - (void) startPlayback
  
  {
  
      OSStatus err = 0;
  
      if(playState.playing) return;
  
  
  
      playState.started = false;
  
  
  
      if(!playState.queue) 
  
      {
  
  
  
          UInt32 bufferSize;
  
  
  
  
  
          playState.format.mSampleRate = _av->audio.sample_rate;
  
          playState.format.mFormatID = kAudioFormatLinearPCM;
  
          playState.format.mFormatFlags = kAudioFormatFlagsCanonical;
  
          playState.format.mChannelsPerFrame = _av->audio.channels_per_frame;
  
          playState.format.mBytesPerPacket = sizeof(AudioSampleType) *_av->audio.channels_per_frame;
  
          playState.format.mBytesPerFrame = sizeof(AudioSampleType) *_av->audio.channels_per_frame;
  
          playState.format.mBitsPerChannel = 8 * sizeof(AudioSampleType);
  
  
  
          playState.format.mFramesPerPacket = 1;        
  
          playState.format.mReserved = 0;
  
  
  
  
  
          pauseStart = 0;
  
          DeriveBufferSize(playState.format,playState.format.mBytesPerPacket,BUFFER_DURATION,&bufferSize,&numPacketsToRead);
  
          err= AudioQueueNewOutput(&playState.format, aqCallback, &playState, NULL, kCFRunLoopCommonModes, 0, &playState.queue);
  
  
  
          if(err != 0)
  
          {
  
              printf("AQHandler.m startPlayback: Error creating new AudioQueue: %d \n", (int)err);
  
          }
  
  
  
          for(int i = 0 ; i < NUM_BUFFERS ; i ++)
  
          {
  
              err = AudioQueueAllocateBufferWithPacketDescriptions(playState.queue, bufferSize, numPacketsToRead , &playState.buffers[i]);
  
  
  
              if(err != 0)
  
                  printf("AQHandler.m startPlayback: Error allocating buffer %d", i);
  
              fillAudioBuffer(&playState,playState.queue, playState.buffers[i]);
  
          }
  
  
  
      }
  
  
  
      startTime = mu_currentTimeInMicros();
  
  
  
      err=AudioQueueStart(playState.queue, NULL);
  
  
  
      if(err)
  
      {
  
  
  
          char sErr[4];
  
          printf("AQHandler.m startPlayback: Could not start queue %ld %s.", err, FormatError(sErr,err));
  
  
  
          playState.playing = NO;
  
      } 
  
      else
  
      {
  
          AudioSessionSetActive(true);
  
          playState.playing = YES;
  
      }           
  
  }

  // — callback for filling audio buffer —

  static int ct = 0;
  
  static void fillAudioBuffer(void *info,AudioQueueRef queue, AudioQueueBufferRef buffer)
  
  {
  
  
  
      int lengthCopied = INT32_MAX;
  
      int dts= 0;
  
      int isDone = 0;
  
  
  
      buffer->mAudioDataByteSize = 0;
  
      buffer->mPacketDescriptionCount = 0;
  
  
  
      OSStatus err = 0;
  
      AudioTimeStamp bufferStartTime;
  
  
  
      AudioQueueGetCurrentTime(queue, NULL, &bufferStartTime, NULL);
  
  
  
  
  
      PlayState *ps = (PlayState *)info;
  
  
  
      if (!ps->started)
  
          ps->started = true;
  
  
  
      while(buffer->mPacketDescriptionCount < numPacketsToRead && lengthCopied > 0)
  
      {
  
          lengthCopied = getNextAudio(_av,
  
                          buffer->mAudioDataBytesCapacity-buffer->mAudioDataByteSize,
  
                          (uint8_t*)buffer->mAudioData+buffeg->mAudioDataByteSize,
  
                          &dts,&isDone);
  
  
  
          ct+= lengthCopied;
  
  
  
          if(lengthCopied < 0 || isDone) 
  
          {
  
              printf("nothing to read....\n\n");
  
              PlayState *ps = (PlayState *)info;
  
              ps->finished = true;
  
              ps->started = false;
  
              break;
  
          }
  
  
  
          if(aqStartDts < 0) aqStartDts = dts;
  
  
  
          if(buffer->mPacketDescriptionCount ==0)
  
          {
  
              bufferStartTime.mFlags = kAudioTimeStampSampleTimeValid;
  
              bufferStartTime.mSampleTime = (Float64)(dts-aqStartDts);//* _av->audio.frame_size;
  
  
  
              if (bufferStartTime.mSampleTime <0 ) 
  
                  bufferStartTime.mSampleTime = 0;
  
  
  
              printf("AQHandler.m fillAudioBuffer: DTS for %x: %lf time base: %lf StartDTS: %d\n", 
  
                      (unsigned int)buffer, 
  
                      bufferStartTime.mSampleTime, 
  
                      _av->audio.time_base, 
  
                      aqStartDts);
  
  
  
          }
  
  
  
          buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mStartOffset = buffer->mAudioDataByteSize;
  
          buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mDataByteSize = lengthCopied;
  
  
  
  
  
  
  
          buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mVariableFramesInPacket = 0;
  
  
  
          buffer->mPacketDescriptionCount++;
  
  
  
          buffer->mAudioDataByteSize += lengthCopied;
  
      }
  
  
  
      int audioBufferCount, audioBufferTotal,  videoBufferCount, videoBufferTotal;
  
      bufferCheck(_av,&videoBufferCount, &videoBufferTotal, &audioBufferCount, &audioBufferTotal);
  
  
  
      if(buffer->mAudioDataByteSize)
  
      {
  
  
  
          err = AudioQueueEnqueueBufferWithParameters(queue, buffer, 0, NULL, 0, 0, 0, NULL, &bufferStartTime, NULL);
  
  
  
          if(err)
  
          {
  
              char sErr[10];
  
              printf("AQHandler.m fillAudioBuffer: Could not enqueue buffer 0x%x: %d %s.", buffer, err, FormatError(sErr, err));
  
  
  
          }
  
  
  
      }
  
  
  
  }
  
  
  
  
  
  
  
  
  
  int getNextAudio(video_data_t* vInst, int maxlength, uint8_t* buf, int* pts, int* isDone) 
  
  {
  
  
  
      struct video_context_t  *ctx = vInst->context;
  
      int    datalength            = 0;
  
  
  
      while(ctx->audio_ring.lock || (ctx->audio_ring.count <= 0 && ((ctx->play_state & STATE_DIE) != STATE_DIE)))
  
      {
  
  
  
          if (ctx->play_state & STATE_EOF) return -1;        
  
          usleep(100);
  
      }
  
  
  
      *pts = 0;
  
      ctx->audio_ring.lock = kLocked;
  
  
  
      if(ctx->audio_ring.count>0 && maxlength > ctx->audio_buffer[ctx->audio_ring.read].size)
  
      {    
  
          memcpy(buf, ctx->audio_buffer[ctx->audio_ring.read].data,ctx->audio_buffer[ctx->audio_ring.read].size);
  
  
  
          *pts = ctx->audio_buffer[ctx->audio_ring.read].pts;
  
  
  
          datalength = ctx->audio_buffer[ctx->audio_ring.read].size;
  
  
  
          ctx->audio_ring.read++;        
  
          ctx->audio_ring.read %= ABUF_SIZE;        
  
          ctx->audio_ring.count--;
  
  
  
      }
  
      ctx->audio_ring.lock = kUnlocked;
  
  
  
      if((ctx->play_state & STATE_EOF) == STATE_EOF && ctx->audio_ring.count == 0) *isDone = 1;
  
  
  
      return datalength;
  
  }