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• Assertion error when encoding video using ffmpeg library on visual studio c++

  18 juin 2015, par Leon Phan

  I have a project, that i need to received data from a rtsp link, decode,resize video, encode it and save to .mp4 file.

  • I Using ffmpeg from http://ffmpeg.zeranoe.com/builds/
  • I am using visual studio 2010, run on windows 7 and windows 8.
  • This is rtsp link i am using : rtsp ://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov
  • I decode and save to ppm file is ok. But encode isn’t.
  • I following there tutorials : http://dranger.com/ffmpeg/tutorial01.html

  this is my code :

  AVPacket destPacket;
  
  av_init_packet(&destPacket);
  
  destPacket.data = NULL;
  
  destPacket.size = 0;
  
  while(av_read_frame(i_format_context, &packet)>=0) {
  
      if(packet.stream_index==video_stream_index) {
  
          int rest = avcodec_decode_video2(copyInputCodec, pFrame, &frameFinished, &packet);
  
          if(frameFinished) {
  
              av_init_packet(&destPacket);
  
              destPacket.data = NULL;
  
              destPacket.size = 0;
  
              av_free_packet(&destPacket);
  
              //deocde here
  
              int ret_scale = sws_scale(sws_ctx,pFrame->data,pFrame->linesize,0,copyInputCodec->height,encodeFrame->data,encodeFrame->linesize);
  
              encodeFrame->pts=i;
  
              int ret_enc = avcodec_encode_video2(copyOutputCodec,&destPacket,encodeFrame,&encodeFinishsed);//<--- Problem here.
  
              getchar();
  
          }
  
      }
  
  }

  And this is output error

  [swscaler @ 00607880] Warning : data is not aligned ! This can lead to a
  speedlo ss
  Assertion avctx->codec->encode2 failed at /home/kyle/software/ffmpeg/source/ffmp
  eg-git/libavcodec/utils.c:2134

  I try to run many sample code to encode video, the result is same.

  Thanks for help.

  PS : sorry about my English skill.

• Encode audio to aac with libavcodec

  22 octobre 2015, par ryan

  I’m using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac
  (libfaac support enabled). I do this by calling avcodec_encode_audio
  repeatedly with codec_context->frame_size samples each time. The first four
  calls return successfully, but the fifth call never returns. When I use gdb
  to break, the stack is corrupt.

  If I use audacity to export the pcm data to a .wav file, then I can use
  command-line ffmpeg to convert to aac without any issues, so I’m sure it’s
  something I’m doing wrong.

  I’ve written a small test program that duplicates my problem. It reads the
  test data from a file, which is available here :
  http://birdie.protoven.com/audio.pcm ( 2 seconds of signed 16 bit LE pcm)

  I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I’m also encoding video, and writing both to an mp4.

  ffmpeg version info :

  FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers
  
    built on Mar  3 2010 15:40:46 with gcc 4.4.1
  
    configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared
  
    libavutil     50.10. 0 / 50.10. 0
  
    libavcodec    52.55. 0 / 52.55. 0
  
    libavformat   52.54. 0 / 52.54. 0
  
    libavdevice   52. 2. 0 / 52. 2. 0
  
    libswscale     0.10. 0 /  0.10. 0
  
    libpostproc   51. 2. 0 / 51. 2. 0

  Is there something I’m not setting, or setting incorrectly in my codec
  context, maybe ? Any help is greatly appreciated !

  Here is my test code :

  #include 
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  
  
  void EncodeTest(int sampleRate, int channels, int audioBitrate,
  
      uint8_t *audioData, size_t audioSize)
  
  {
  
      AVCodecContext  *audioCodec;
  
      AVCodec *codec;
  
      uint8_t *buf;
  
      int bufSize, frameBytes;
  
  
  
      avcodec_register_all();
  
  
  
      //Set up audio encoder
  
      codec = avcodec_find_encoder(CODEC_ID_AAC);
  
      if (codec == NULL) return;
  
      audioCodec = avcodec_alloc_context();
  
      audioCodec->bit_rate = audioBitrate;
  
      audioCodec->sample_fmt = SAMPLE_FMT_S16;
  
      audioCodec->sample_rate = sampleRate;
  
      audioCodec->channels = channels;
  
      audioCodec->profile = FF_PROFILE_AAC_MAIN;
  
      audioCodec->time_base = (AVRational){1, sampleRate};
  
      audioCodec->codec_type = CODEC_TYPE_AUDIO;
  
      if (avcodec_open(audioCodec, codec) &lt; 0) return;
  
  
  
      bufSize = FF_MIN_BUFFER_SIZE * 10;
  
      buf = (uint8_t *)malloc(bufSize);
  
      if (buf == NULL) return;
  
  
  
      frameBytes = audioCodec->frame_size * audioCodec->channels * 2;
  
      while (audioSize >= frameBytes)
  
      {
  
          int packetSize;
  
  
  
          packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData);
  
          printf("encoder returned %d bytes of data\n", packetSize);
  
          audioData += frameBytes;
  
          audioSize -= frameBytes;
  
      }
  
  }
  
  
  
  int main()
  
  {
  
      FILE *stream = fopen("audio.pcm", "rb");
  
      size_t size;
  
      uint8_t *buf;
  
  
  
      if (stream == NULL)
  
      {
  
          printf("Unable to open file\n");
  
          return 1;
  
      }
  
  
  
      fseek(stream, 0, SEEK_END);
  
      size = ftell(stream);
  
      fseek(stream, 0, SEEK_SET);
  
      buf = (uint8_t *)malloc(size);
  
      fread(buf, sizeof(uint8_t), size, stream);
  
      fclose(stream);
  
  
  
      EncodeTest(32000, 2, 448000, buf, size);
  
  }

• configure : Allow log2 with MSVC 2013 onwards.

  11 mai 2014, par Matt Oliver

  configure : Allow log2 with MSVC 2013 onwards.
  Only MSVC 2010 in x64 mode, in the static msvcrt, had a
  
  stray log2 function (which wasn’t available in the headers).
  MSVC 2013 has got a proper log2 function though.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] configure