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• H.264 muxed to MP4 using libavformat not playing back

  14 mai 2015, par Brad Mitchell

  I am trying to mux H.264 data into a MP4 file. There appear to be no errors in saving this H.264 Annex B data out to an MP4 file, but the file fails to playback.

  I’ve done a binary comparison on the files and the issue seems to be somewhere in what is being written to the footer (trailer) of the MP4 file.

  I suspect it has to be something with the way the stream is being created or something.

  Init :

  AVOutputFormat* fmt = av_guess_format( 0, "out.mp4", 0 );
  
  oc = avformat_alloc_context();
  
  oc->oformat = fmt;
  
  strcpy(oc->filename, filename);

  Part of this prototype app I have is creating a png file for each IFrame. So when the first IFrame is encountered, I create the video stream and write the av header etc :

  void addVideoStream(AVCodecContext* decoder)
  
  {
  
      videoStream = av_new_stream(oc, 0);
  
      if (!videoStream)
  
      {
  
           cout << "ERROR creating video stream" << endl;
  
           return;        
  
      }
  
      vi = videoStream->index;    
  
      videoContext = videoStream->codec;      
  
      videoContext->codec_type = AVMEDIA_TYPE_VIDEO;
  
      videoContext->codec_id = decoder->codec_id;
  
      videoContext->bit_rate = 512000;
  
      videoContext->width = decoder->width;
  
      videoContext->height = decoder->height;
  
      videoContext->time_base.den = 25;
  
      videoContext->time_base.num = 1;    
  
      videoContext->gop_size = decoder->gop_size;
  
      videoContext->pix_fmt = decoder->pix_fmt;       
  
  
  
      if (oc->oformat->flags & AVFMT_GLOBALHEADER)
  
          videoContext->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  
  
      av_dump_format(oc, 0, filename, 1);
  
  
  
      if (!(oc->oformat->flags & AVFMT_NOFILE))
  
      {
  
          if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
  
          cout << "Error opening file" << endl;
  
      }
  
      avformat_write_header(oc, NULL);
  
  }

  I write packets out :

  unsigned char* data = block->getData();
  
  unsigned char videoFrameType = data[4];
  
  int dataLen = block->getDataLen();
  
  
  
  // store pps
  
  if (videoFrameType == 0x68)
  
  {
  
      if (ppsFrame != NULL)
  
      {
  
          delete ppsFrame; ppsFrameLength = 0; ppsFrame = NULL;
  
      }
  
      ppsFrameLength = block->getDataLen();
  
      ppsFrame = new unsigned char[ppsFrameLength];
  
      memcpy(ppsFrame, block->getData(), ppsFrameLength);
  
  }
  
  else if (videoFrameType == 0x67)
  
  {
  
      // sps
  
      if (spsFrame != NULL)
  
      {
  
          delete spsFrame; spsFrameLength = 0; spsFrame = NULL;
  
  }
  
      spsFrameLength = block->getDataLen();
  
      spsFrame = new unsigned char[spsFrameLength];
  
      memcpy(spsFrame, block->getData(), spsFrameLength);                 
  
  }                                           
  
  
  
  if (videoFrameType == 0x65 || videoFrameType == 0x41)
  
  {
  
      videoFrameNumber++;
  
  }
  
  if (videoFrameType == 0x65)
  
  {
  
      decodeIFrame(videoFrameNumber, spsFrame, spsFrameLength, ppsFrame, ppsFrameLength, data, dataLen);
  
  }
  
  
  
  if (videoStream != NULL)
  
  {
  
      AVPacket pkt = { 0 };
  
      av_init_packet(&pkt);
  
      pkt.stream_index = vi;
  
      pkt.flags = 0;                      
  
      pkt.pts = pkt.dts = 0;                                  
  
  
  
      if (videoFrameType == 0x65)
  
      {
  
          // combine the SPS PPS & I frames together
  
          pkt.flags |= AV_PKT_FLAG_KEY;                                                   
  
          unsigned char* videoFrame = new unsigned char[spsFrameLength+ppsFrameLength+dataLen];
  
          memcpy(videoFrame, spsFrame, spsFrameLength);
  
          memcpy(&videoFrame[spsFrameLength], ppsFrame, ppsFrameLength);
  
          memcpy(&videoFrame[spsFrameLength+ppsFrameLength], data, dataLen);
  
  
  
          // overwrite the start code (00 00 00 01 with a 32-bit length)
  
          setLength(videoFrame, spsFrameLength-4);
  
          setLength(&videoFrame[spsFrameLength], ppsFrameLength-4);
  
          setLength(&videoFrame[spsFrameLength+ppsFrameLength], dataLen-4);
  
          pkt.size = dataLen + spsFrameLength + ppsFrameLength;
  
          pkt.data = videoFrame;
  
          av_interleaved_write_frame(oc, &pkt);
  
          delete videoFrame; videoFrame = NULL;
  
      }
  
      else if (videoFrameType != 0x67 && videoFrameType != 0x68)
  
      {   
  
          // Send other frames except pps & sps which are caught and stored                   
  
          pkt.size = dataLen;
  
          pkt.data = data;
  
          setLength(data, dataLen-4);                     
  
          av_interleaved_write_frame(oc, &pkt);
  
      }

  Finally to close the file off :

  av_write_trailer(oc);
  
  int i = 0;
  
  for (i = 0; i < oc->nb_streams; i++)
  
  {
  
      av_freep(&oc->streams[i]->codec);
  
      av_freep(&oc->streams[i]);      
  
  }
  
  
  
  if (!(oc->oformat->flags & AVFMT_NOFILE))
  
  {
  
      avio_close(oc->pb);
  
  }
  
  av_free(oc);

  If I take the H.264 data alone and convert it :

  ffmpeg -i recording.h264 -vcodec copy recording.mp4

  All but the "footer" of the files are the same.

  Output from my program :
  readrec recording.tcp out.mp4
  ** START * 01-03-2013 14:26:01 180000
  Output #0, mp4, to ’out.mp4’ :
  Stream #0:0 : Video : h264, yuv420p, 352x288, q=2-31, 512 kb/s, 90k tbn, 25 tbc
  * END ** 01-03-2013 14:27:01 102000
  Wrote 1499 video frames.

  If I try to convert using ffmpeg the MP4 file created using CODE :

  ffmpeg -i out.mp4 -vcodec copy out2.mp4
  
  ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
  
        built on Mar  7 2013 12:49:22 with suncc 0x5110
  
        configuration: --extra-cflags=-KPIC -g --disable-mmx
  
        --disable-protocol=udp --disable-encoder=nellymoser --cc=cc --cxx=CC
  
  libavutil      51. 54.100 / 51. 54.100
  
  libavcodec     54. 23.100 / 54. 23.100
  
  libavformat    54.  6.100 / 54.  6.100
  
  libavdevice    54.  0.100 / 54.  0.100
  
  libavfilter     2. 77.100 /  2. 77.100
  
  libswscale      2.  1.100 /  2.  1.100
  
  libswresample   0. 15.100 /  0. 15.100
  
  h264 @ 12eaac0] no frame!
  
      Last message repeated 1 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 23 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 74 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 64 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 34 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 49 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 24 times
  
  [h264 @ 12eaac0] Partitioned H.264 support is incomplete
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 23 times
  
  [h264 @ 12eaac0] sps_id out of range
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 148 times
  
  [h264 @ 12eaac0] sps_id (32) out of range
  
      Last message repeated 1 times
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 33 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 128 times
  
  [h264 @ 12eaac0] sps_id (32) out of range
  
      Last message repeated 1 times
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 3 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 3 times
  
  [h264 @ 12eaac0] slice type too large (0) at 0 0
  
  [h264 @ 12eaac0] decode_slice_header error
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 309 times
  
  [h264 @ 12eaac0] sps_id (32) out of range
  
      Last message repeated 1 times
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 192 times
  
  [h264 @ 12eaac0] Partitioned H.264 support is incomplete
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 73 times
  
  [h264 @ 12eaac0] sps_id (32) out of range
  
      Last message repeated 1 times
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 99 times
  
  [h264 @ 12eaac0] sps_id (32) out of range
  
      Last message repeated 1 times
  
  [h264 @ 12eaac0] no frame!
  
      Last message repeated 197 times
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 12e3100] decoding for stream 0 failed
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 12e3100] Could not find codec parameters
  
  (Video: h264 (avc1 / 0x31637661), 393539 kb/s)
  
  out.mp4: could not find codec parameters

  I really do not know where the issue is, except it has to be something to do with the way the streams are being set up. I’ve looked at bits of code from where other people are doing a similar thing, and tried to use this advice in setting up the streams, but to no avail !

  ---

  The final code which gave me a H.264/AAC muxed (synced) file is as follows. First a bit of background information. The data is coming from an IP camera. The data is presented via a 3rd party API as video/audio packets. The video packets are presented as the RTP payload data (no header) and consist of NALU’s that are reconstructed and converted to H.264 video in Annex B format. AAC audio is presented as raw AAC and is converted to adts format to enable playback. These packets have been put into a bitstream format that allows the transmission of the timestamp (64 bit milliseconds since Jan 1 1970) along with a few other things.

  This is more or less a prototype and is not clean in any respects. It probably leaks bad. I do however, hope this helps anyone else out trying to achieve something similar to what I am.

  Globals :

  AVFormatContext* oc = NULL;
  
  AVCodecContext* videoContext = NULL;
  
  AVStream* videoStream = NULL;
  
  AVCodecContext* audioContext = NULL;
  
  AVStream* audioStream = NULL;
  
  AVCodec* videoCodec = NULL;
  
  AVCodec* audioCodec = NULL;
  
  int vi = 0;  // Video stream
  
  int ai = 1;  // Audio stream
  
  
  
  uint64_t firstVideoTimeStamp = 0;
  
  uint64_t firstAudioTimeStamp = 0;
  
  int audioStartOffset = 0;
  
  
  
  char* filename = NULL;
  
  
  
  Boolean first = TRUE;
  
  
  
  int videoFrameNumber = 0;
  
  int audioFrameNumber = 0;

  Main :

  int main(int argc, char* argv[])
  
  {
  
      if (argc != 3)
  
      {   
  
          cout &lt;&lt; argv[0] &lt;&lt; " <stream playback="playback" file="file"> <output mp4="mp4" file="file">" &lt;&lt; endl;
  
          return 0;
  
      }
  
      char* input_stream_file = argv[1];
  
      filename = argv[2];
  
  
  
      av_register_all();    
  
  
  
      fstream inFile;
  
      inFile.open(input_stream_file, ios::in);
  
  
  
      // Used to store the latest pps &amp; sps frames
  
      unsigned char* ppsFrame = NULL;
  
      int ppsFrameLength = 0;
  
      unsigned char* spsFrame = NULL;
  
      int spsFrameLength = 0;
  
  
  
      // Setup MP4 output file
  
      AVOutputFormat* fmt = av_guess_format( 0, filename, 0 );
  
      oc = avformat_alloc_context();
  
      oc->oformat = fmt;
  
      strcpy(oc->filename, filename);
  
  
  
      // Setup the bitstream filter for AAC in adts format.  Could probably also achieve
  
      // this by stripping the first 7 bytes!
  
      AVBitStreamFilterContext* bsfc = av_bitstream_filter_init("aac_adtstoasc");
  
      if (!bsfc)
  
      {       
  
          cout &lt;&lt; "Error creating adtstoasc filter" &lt;&lt; endl;
  
          return -1;
  
      }
  
  
  
      while (inFile.good())
  
      {
  
          TcpAVDataBlock* block = new TcpAVDataBlock();
  
          block->readStruct(inFile);
  
          DateTime dt = block->getTimestampAsDateTime();
  
          switch (block->getPacketType())
  
          {
  
              case TCP_PACKET_H264:
  
              {       
  
                  if (firstVideoTimeStamp == 0)
  
                      firstVideoTimeStamp = block->getTimeStamp();
  
                  unsigned char* data = block->getData();
  
                  unsigned char videoFrameType = data[4];
  
                  int dataLen = block->getDataLen();
  
  
  
                  // pps
  
                  if (videoFrameType == 0x68)
  
                  {
  
                      if (ppsFrame != NULL)
  
                      {
  
                          delete ppsFrame; ppsFrameLength = 0;
  
                          ppsFrame = NULL;
  
                      }
  
                      ppsFrameLength = block->getDataLen();
  
                      ppsFrame = new unsigned char[ppsFrameLength];
  
                      memcpy(ppsFrame, block->getData(), ppsFrameLength);
  
                  }
  
                  else if (videoFrameType == 0x67)
  
                  {
  
                      // sps
  
                      if (spsFrame != NULL)
  
                      {
  
                          delete spsFrame; spsFrameLength = 0;
  
                          spsFrame = NULL;
  
                      }
  
                      spsFrameLength = block->getDataLen();
  
                      spsFrame = new unsigned char[spsFrameLength];
  
                      memcpy(spsFrame, block->getData(), spsFrameLength);                   
  
                  }                                           
  
  
  
                  if (videoFrameType == 0x65 || videoFrameType == 0x41)
  
                  {
  
                      videoFrameNumber++;
  
                  }
  
                  // Extract a thumbnail for each I-Frame
  
                  if (videoFrameType == 0x65)
  
                  {
  
                      decodeIFrame(h264, spsFrame, spsFrameLength, ppsFrame, ppsFrameLength, data, dataLen);
  
                  }
  
                  if (videoStream != NULL)
  
                  {
  
                      AVPacket pkt = { 0 };
  
                      av_init_packet(&amp;pkt);
  
                      pkt.stream_index = vi;
  
                      pkt.flags = 0;           
  
                      pkt.pts = videoFrameNumber;
  
                      pkt.dts = videoFrameNumber;           
  
                      if (videoFrameType == 0x65)
  
                      {
  
                          pkt.flags = 1;                           
  
  
  
                          unsigned char* videoFrame = new unsigned char[spsFrameLength+ppsFrameLength+dataLen];
  
                          memcpy(videoFrame, spsFrame, spsFrameLength);
  
                          memcpy(&amp;videoFrame[spsFrameLength], ppsFrame, ppsFrameLength);
  
  
  
                          memcpy(&amp;videoFrame[spsFrameLength+ppsFrameLength], data, dataLen);
  
                          pkt.data = videoFrame;
  
                          av_interleaved_write_frame(oc, &amp;pkt);
  
                          delete videoFrame; videoFrame = NULL;
  
                      }
  
                      else if (videoFrameType != 0x67 &amp;&amp; videoFrameType != 0x68)
  
                      {                       
  
                          pkt.size = dataLen;
  
                          pkt.data = data;
  
                          av_interleaved_write_frame(oc, &amp;pkt);
  
                      }                       
  
                  }
  
                  break;
  
              }
  
  
  
          case TCP_PACKET_AAC:
  
  
  
              if (firstAudioTimeStamp == 0)
  
              {
  
                  firstAudioTimeStamp = block->getTimeStamp();
  
                  uint64_t millseconds_difference = firstAudioTimeStamp - firstVideoTimeStamp;
  
                  audioStartOffset = millseconds_difference * 16000 / 1000;
  
                  cout &lt;&lt; "audio offset: " &lt;&lt; audioStartOffset &lt;&lt; endl;
  
              }
  
  
  
              if (audioStream != NULL)
  
              {
  
                  AVPacket pkt = { 0 };
  
                  av_init_packet(&amp;pkt);
  
                  pkt.stream_index = ai;
  
                  pkt.flags = 1;           
  
                  pkt.pts = audioFrameNumber*1024;
  
                  pkt.dts = audioFrameNumber*1024;
  
                  pkt.data = block->getData();
  
                  pkt.size = block->getDataLen();
  
                  pkt.duration = 1024;
  
  
  
                  AVPacket newpacket = pkt;                       
  
                  int rc = av_bitstream_filter_filter(bsfc, audioContext,
  
                      NULL,
  
                      &amp;newpacket.data, &amp;newpacket.size,
  
                      pkt.data, pkt.size,
  
                      pkt.flags &amp; AV_PKT_FLAG_KEY);
  
  
  
                  if (rc >= 0)
  
                  {
  
                      //cout &lt;&lt; "Write audio frame" &lt;&lt; endl;
  
                      newpacket.pts = audioFrameNumber*1024;
  
                      newpacket.dts = audioFrameNumber*1024;
  
                      audioFrameNumber++;
  
                      newpacket.duration = 1024;                   
  
  
  
                      av_interleaved_write_frame(oc, &amp;newpacket);
  
                      av_free_packet(&amp;newpacket);
  
                  }   
  
                  else
  
                  {
  
                      cout &lt;&lt; "Error filtering aac packet" &lt;&lt; endl;
  
  
  
                  }
  
              }
  
              break;
  
  
  
          case TCP_PACKET_START:
  
              break;
  
  
  
          case TCP_PACKET_END:
  
              break;
  
          }
  
          delete block;
  
      }
  
      inFile.close();
  
  
  
      av_write_trailer(oc);
  
      int i = 0;
  
      for (i = 0; i &lt; oc->nb_streams; i++)
  
      {
  
          av_freep(&amp;oc->streams[i]->codec);
  
          av_freep(&amp;oc->streams[i]);       
  
      }
  
  
  
      if (!(oc->oformat->flags &amp; AVFMT_NOFILE))
  
      {
  
          avio_close(oc->pb);
  
      }
  
  
  
      av_free(oc);
  
  
  
      delete spsFrame; spsFrame = NULL;
  
      delete ppsFrame; ppsFrame = NULL;
  
  
  
      cout &lt;&lt; "Wrote " &lt;&lt; videoFrameNumber &lt;&lt; " video frames." &lt;&lt; endl;
  
  
  
      return 0;
  
  }
  
  </output></stream>

  The stream stream/codecs are added and the header is created in a function called addVideoAndAudioStream(). This function is called from decodeIFrame() so there are a few assumptions (which aren’t necessarily good)
  1. A video packet comes first
  2. AAC is present

  The decodeIFrame was kind of a separate prototype by where I was creating a thumbnail for each I Frame. The code to generate thumbnails was from : https://gnunet.org/svn/Extractor/src/plugins/thumbnailffmpeg_extractor.c

  The decodeIFrame function passes an AVCodecContext into addVideoAudioStream :

  void addVideoAndAudioStream(AVCodecContext* decoder = NULL)
  
  {
  
      videoStream = av_new_stream(oc, 0);
  
      if (!videoStream)
  
      {
  
          cout << "ERROR creating video stream" << endl;
  
          return;       
  
      }
  
      vi = videoStream->index;   
  
      videoContext = videoStream->codec;       
  
      videoContext->codec_type = AVMEDIA_TYPE_VIDEO;
  
      videoContext->codec_id = decoder->codec_id;
  
      videoContext->bit_rate = 512000;
  
      videoContext->width = decoder->width;
  
      videoContext->height = decoder->height;
  
      videoContext->time_base.den = 25;
  
      videoContext->time_base.num = 1;
  
      videoContext->gop_size = decoder->gop_size;
  
      videoContext->pix_fmt = decoder->pix_fmt;       
  
  
  
      audioStream = av_new_stream(oc, 1);
  
      if (!audioStream)
  
      {
  
          cout << "ERROR creating audio stream" << endl;
  
          return;
  
      }
  
      ai = audioStream->index;
  
      audioContext = audioStream->codec;
  
      audioContext->codec_type = AVMEDIA_TYPE_AUDIO;
  
      audioContext->codec_id = CODEC_ID_AAC;
  
      audioContext->bit_rate = 64000;
  
      audioContext->sample_rate = 16000;
  
      audioContext->channels = 1;
  
  
  
      if (oc->oformat->flags & AVFMT_GLOBALHEADER)
  
      {
  
          videoContext->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
          audioContext->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
      }
  
  
  
      av_dump_format(oc, 0, filename, 1);
  
  
  
      if (!(oc->oformat->flags & AVFMT_NOFILE))
  
      {
  
          if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
  
              cout << "Error opening file" << endl;
  
          }
  
      }
  
  
  
      avformat_write_header(oc, NULL);
  
  }

  As far as I can tell, a number of assumptions didn’t seem to matter, for example :
  1. Bit Rate. The actual video bit rate was 262k whereas I specified 512kbit
  2. AAC channels. I specified mono, although the actual output was Stereo from memory

  You would still need to know what the frame rate (time base) is for the video & audio.

  Contrary to a lot of other examples, when setting pts & dts on the video packets, it was not playable. I needed to know the time base (25fps) and then set the pts & dts according to that time base, i.e. first frame = 0 (PPS, SPS, I), second frame = 1 (intermediate frame, whatever its called ;)).

  AAC I also had to make the assumption that it was 16000 hz. 1024 samples per AAC packet (You can also have AAC @ 960 samples I think) to determine the audio "offset". I added this to the pts & dts. So the pts/dts are the sample number that it is to played back at. You also need to make sure that the duration of 1024 is set in the packet before writing also.

  —

  I have found additionally today that Annex B isn’t really compatible with any other player so AVCC format should really be used.

  These URLS helped :
  Problem to Decode H264 video over RTP with ffmpeg (libavcodec)
  http://aviadr1.blogspot.com.au/2010/05/h264-extradata-partially-explained-for.html

  When constructing the video stream, I filled out the extradata & extradata_size :

  // Extradata contains PPS & SPS for AVCC format
  
  int extradata_len = 8 + spsFrameLen-4 + 1 + 2 + ppsFrameLen-4;
  
  videoContext->extradata = (uint8_t*)av_mallocz(extradata_len);
  
  videoContext->extradata_size = extradata_len;
  
  videoContext->extradata[0] = 0x01;
  
  videoContext->extradata[1] = spsFrame[4+1];
  
  videoContext->extradata[2] = spsFrame[4+2];
  
  videoContext->extradata[3] = spsFrame[4+3];
  
  videoContext->extradata[4] = 0xFC | 3;
  
  videoContext->extradata[5] = 0xE0 | 1;
  
  int tmp = spsFrameLen - 4;
  
  videoContext->extradata[6] = (tmp >> 8) & 0x00ff;
  
  videoContext->extradata[7] = tmp & 0x00ff;
  
  int i = 0;
  
  for (i=0;iextradata[8+i] = spsFrame[4+i];
  
  videoContext->extradata[8+tmp] = 0x01;
  
  int tmp2 = ppsFrameLen-4;   
  
  videoContext->extradata[8+tmp+1] = (tmp2 >> 8) & 0x00ff;
  
  videoContext->extradata[8+tmp+2] = tmp2 & 0x00ff;
  
  for (i=0;iextradata[8+tmp+3+i] = ppsFrame[4+i];

  When writing out the frames, don’t prepend the SPS & PPS frames, just write out the I Frame & P frames. In addition, replace the Annex B start code contained in the first 4 bytes (0x00 0x00 0x00 0x01) with the size of the I/P frame.

• Mix Audio tracks with offset in SOX

  4 août 2012, par Laramie

  From ASP.Net, I am using FFMPEG to convert flv files on a Flash Media Server to wavs that I need to mix into a single MP3 file. I originally attempted this entirely with FFMPEG but eventually gave up on the mixing step because I don't believe it it possible to combine audio only tracks into a single result file. I would love to be wrong.

  I am now using FFMPEG to access the FLV files and extract the audio track to wav so that SOX can mix them. The problem is that I must offset one of the audio tracks by a few seconds so that they are synchronized. Each file is one half of a conversation between a student and a teacher. For example teacher.wav might need to begin 3.3 seconds after student.wav. I can only figure out how to mix the files with SOX where both tracks begin at the same time.

  My best attempt at this point is :

  ffmpeg -y -i rtmp://server/appName/instance/student.flv -ac 1 student.wav 
  
  ffmpeg -y -i rtmp://server/appName/instance/teacher.flv -ac 1 teacher.wav 
  
  
  
  sox -m student.wav teacher.wav combined.mp3 splice 3.3

  These tools (FFMEG/SoX) were chosen based on my best research, but are not required. Any working solution would allow an ASP.Net service to input the two FMS flvs and create a combined MP3 using open-source or free tools.

  EDIT :
  I was able to offset the files using the delay switch in SOX.

  sox -M student.wav teacher.wav combined.mp3 delay 2.8

  I'm leaving the question open in case someone has a better approach than the combined FFMPEG/SOX solution.

• Winamp and the March of GUI

  1er juillet 2012, par Multimedia Mike — General, ars technica, gui, user interface, winamp

  Ars Technica recently published a 15-year retrospective on the venerable Winamp multimedia player, prompting bouts of nostalgia and revelations of "Huh ? That program is still around ?" from many readers. I was among them.

  
  
  

  I remember first using Winamp in 1997. I remember finding a few of these new files called MP3s online and being able to play the first 20 seconds using the official Fraunhofer Windows player— full playback required the fully licensed version. Then I searched for another player and came up with Winamp. The first version I ever used was v1.05 in the summer of 1997. I remember checking the website often for updates and trying out every single one. I can’t imagine doing that nowadays— programs need to auto-update themselves (which Winamp probably does now ; I can’t recall the last time I used the program).

  Video Underdog
  The last time Winamp came up on my radar was early in 2003 when a new version came with support for a custom, proprietary multimedia audio/video format called Nullsoft Video (NSV). I remember the timeframe because the date is indicated in the earliest revision of my NSV spec document (back when I was maintaining such docs in a series of plaintext files). This was cobbled together from details I and others in the open source multimedia community sorted out from sample files. It was missing quite a few details, though.

  Then, Winamp founder Justin Frankel — introduced through a colleague on the xine team — emailed me his official NSV format and told me I was free to incorporate details into my document just as long as it wasn’t obvious that I had the official spec. This put me in an obnoxious position of trying to incorporate details which would have been very difficult to reverse engineer without the official doc. I think I coped with the situation by never really getting around to updating my doc in any meaningful way. Then, one day, the official spec was released to the world anyway, and it is now mirrored here at multimedia.cx.

  I don’t think the format ever really caught on in any meaningful way, so not a big deal. (Anytime I say that about a format, I always learn it saw huge adoption is some small but vocal community.)

  What’s Wrong With This Picture ?
  What I really wanted to discuss in this post was the matter of graphical user interfaces and how they have changed in the last 15 years.
  
  I still remember when I first downloaded Winamp v1.05 and tried it on my Windows machine at the time. Indignantly, the first thought I had was, "What makes this program think it’s so special that it’s allowed to violate the user interface conventions put forth by the rest of the desktop ?" All of the Windows programs followed a standard set of user interface patterns and had a consistent look and feel... and then Winamp came along and felt it could violate all those conventions.

  I guess I let the program get away with it because it was either that or only play 20-second clips from the unregistered Fraunhofer player. Though incredibly sterile by comparison, the Fraunhofer player, it should be noted, followed Windows UI guidelines to the letter.

  As the summer of 1997 progressed and more Winamp versions were released, eventually one came out (I think it was v1.6 or so) that supported skins. I was excited because there was a skin that made the program look like a proper Windows program— at least if you used the default Windows color scheme, and had all of your fonts a certain type and size.

  Skins were implemented by packaging together a set of BMP images to overlay on various UI elements. I immediately saw a number of shortcomings with this skinning approach. A big one was UI lock-in. Ironically, if you skin an app and wish to maintain backwards compatibility with the thousands of skins selflessly authored by your vibrant community (seriously, I couldn’t believe how prolific these things were), then you were effectively locked into the primary UI. Forget about adding a new button anywhere.

  Another big problem was resolution-independence. Basing your UI on static bitmaps doesn’t scale well with various resolutions. Winamp had its normal mode and it also had double-sized mode.

  Skins proliferated among many types of programs in the late 1990s. I always treasured this Suck.com (remember them ? that’s a whole other nostalgia trip) essay from April, 2000 entitled Skin Cancer. Still, Winamp was basically the standard, and the best, and I put away my righteous nerd rage and even dug through the vast troves of skins. I remember settling on Swankamp for a good part of 1998, probably due to the neo-swing revival at the time.

  
  
  

  Then again, if Winamp irked me, imagine my reaction when I was first exposed to the Sonique Music Player in 1998 :

  
  
  

  The New UI Order
  Upon reflection, I realize now that I had a really myopic view of what a computer GUI should be. I thought the GUIs were necessarily supposed to follow the WIMP (windows, icons, mouse, pointer) paradigm and couldn’t conceive of anything different. For a long time, I couldn’t envision a useful GUI on a small device (like a phone) because WIMP didn’t fit well on such a small interface (even though I saw various ill-fated attempts to make it work). This thinking seriously crippled me when I was trying to craft a GUI for a custom console media player I was developing as a hobby many years ago.

  I’m looking around at what I have open on my Windows 7 desktop right now. Google Chrome browser, Apple iTunes, Adobe Photoshop Elements, and VMware Player are 4 programs which all seem to have their own skins. Maybe Winamp doesn’t look so out of place these days.