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• error:initializer element is not constant [closed]

  21 novembre 2011, par lqf

  I have downloaded ffmpeg's source code from 'www.ffmpeg.org' and created am empty project with ubuntu.Then I'm copied and added the ffmpeg's source code to the project.There are some errors when I compile it.

  The errors are:xxx error:initializer element is not constant xxx error:(near initializer for 'ff_eightbps_decoder.long_name') and appeared at ".long_name = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),"

• How to convert .mov videos to iPhone playable mp4 videos [migrated]

  21 novembre 2011, par Rinto George

  I am trying to convert .mov videos to mp4(should be playable in iPhone) using ffmpeg.I am using Linux CLI. I have tried the following command:

  -i source.MOV -s qvga -b 384k -vcodec libx264 -r 23.976 -acodec libfaac -ac 2 -ar 44100 -ab 64k -vpre baseline -crf 22 -deinterlace -o output.mp4 
  

  I get the output

  ffmpeg: unrecognized option '-o' 
  

• How to encode VP6 codec in ffmpeg ? [migrated]

  20 novembre 2011, par userffmpeg

  Can anyone tell me if there is a way to encode VP6 codec in ffmpeg? I used libvpx only to find out that it encodes using VP8...

• Can't record audio with ffmpeg linux

  20 novembre 2011, par FGraviton

  I'm trying to do a screencast with ffmpeg on OpenSUSE but the audio isn't working :

  ffmpeg -f oss -i /dev/audio -f x11grab -s $SCREEN -r 24 -b 100k -bf 2 -g 300 -i :0.0 -ar 22050 -ab 128k -acodec libmp3lame -vcodec libxvid -aspect 1.6 -sameq out.avi

  this one shows me that /dev/audio isn't there !!

  Any pointers ?

  Thanks Community,

• 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;
  }