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• hpeldsp : Increase put_no_rnd_pixels_tab[][] size

  4 novembre 2014, par Michael Niedermayer

  hpeldsp : Increase put_no_rnd_pixels_tab[][] size
  CC : libav-stable@libav.org
  
  Bug-Id : CID 991852
  

  • [DBH] libavcodec/hpeldsp.h

• Call ffmpeg in c++ with system() function fails

  2 septembre 2014, par zhen lee

  I write a c++ program which needs to convert some(say:10) mp4 videos to flv videos.
  I use ffmpeg in my program for each video like this :

  system("ffmpeg -i video -filter:v yadif -ar 44100 -sameq -y -f flv temp.flv")

  however,it turns out :only first video will be converted successfully,the others will fail.
  it means :
  when i change the input order of which video to convert and re-execution the program,it behave the same:only the first video(will be different each time as i changed the input video order) will be converted successfully.

  The error message like :

  [h264 @ 0xaee0740] concealing 45 DC, 45 AC, 45 MV errors
  [h264 @ 0xaee0ce0] AVC : nal size 305665
  Last message repeated 1 times
  [h264 @ 0xaee0ce0] no frame !
  [h264 @ 0xaee1280] AVC : nal size 572993
  Last message repeated 1 times
  [h264 @ 0xaee1280] no frame !
  [aac @ 0xad9ccc0] channel element 0.13 is not allocated
  Error while decoding stream #0:1
  [aac @ 0xad9ccc0] channel element 0.13 is not allocated
  Error while decoding stream #0:1
  [aac @ 0xad9ccc0] channel element 0.13 is not allocated
  Error while decoding stream #0:1
  ......
  The most strange thing is :when i run ffmpeg command in bash shell,all video will be converted successfully .

  After google it,I have try these(certainly failed) :

  1. remove -sameq option,the result is same ;

  2. write ffmpeg commod in a shell script ConvertToFlv.sh like :

     /usr/local/bin/ffmpeg -i "$dir/$1" -filter:v yadif -ar 44100 -sameq -y -f flv "$dir/temp.fiv"

     then call this script in program like

     system("ConvertToFlv.sh"+video)

     or

     system("sh ConvertToFlv.sh"+video)

     The result is same.

  The ffmpeg configure is :

    ffmpeg version 0.9.1.git Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Dec 17 2012 16:17:30 with gcc 4.1.2 20080704 (Red Hat 4.1.2-48)
  
    configuration: --enable-gpl --enable-postproc --enable-nonfree --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads --enable-libxvid --enable-libx264 --enable-libmp3lame --enable-libfaac --disable-ffserver --disable-ffplay
  
    libavutil      51. 41.100 / 51. 41.100
  
    libavcodec     54.  4.100 / 54.  4.100
  
    libavformat    54.  1.100 / 54.  1.100
  
    libavdevice    53.  4.100 / 53.  4.100
  
    libavfilter     2. 62.101 /  2. 62.101
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0.  7.100 /  0.  7.100
  
    libpostproc    52.  0.100 / 52.  0.100

  and my machine envirment is :

  Linux master 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:14 EDT 2010 x86_64 x86_64 x86_64 GNU/Linux

  I’m irritable now,I hope someone can give me some advice,really appreciate it.

• Android how to increase ffmpeg mp4 perfromance ?

  4 janvier 2013, par testCoder

  I have detected that function avcodec_decode_audio3 works slow with mp4 format, here my code cycle for decoding audio :

  while (av_read_frame(av_format_context, &packet) >= 0 && is_play == 1) {
  
          if (av_codec_context->codec_type == AVMEDIA_TYPE_AUDIO
  
                  && is_play == 1) {
  
              int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
              int size = packet.size;
  
              int n;
  
              int dataLength = size;
  
              int decoded = 0;
  
              while (size > 0) {
  
  
  
                  //start measure time
  
                  gettimeofday(&tvBegin, NULL);
  
  
  
                  int len = avcodec_decode_audio3(av_codec_context,
  
                          (int16_t *) pAudioBuffer, &out_size, &packet);
  
  
  
                  //stop measure time
  
                  gettimeofday(&tvEnd, NULL);
  
                  timeval_subtract(&tvDiff, &tvEnd, &tvBegin);
  
  
  
                  LOGI("%d", tvDiff.tv_usec / 1000);
  
                  LOGI("len='%d'", len);
  
                  LOGI("out_size='%d'", out_size);
  
  
  
                  if (len < 0) {
  
                      break;
  
                      return 1;
  
                  }
  
                  if (out_size > 0) {
  
  
  
  
  
  
  
                      jbyte *bytes = (*env)->GetByteArrayElements(env, array,
  
                              NULL);
  
                      memcpy(bytes, (int16_t *) pAudioBuffer, out_size);
  
                      (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
                      (*env)->CallVoidMethod(env, obj, play, array, out_size,
  
                              is_play);
  
  
  
                  }
  
                  size -= len;
  
              }
  
  
  
          }
  
          if (packet.data)
  
              av_free_packet(&packet);
  
  
  
      }

  But with other formats like flac and mp3 it works fine. avcodec_decode_audio3 take about 1-2 milisecounds for decoding mp3 frame with out_size = 4608 but with the same frame size in mp4 decoding take about 6-7 millisecounds. I got my build script from here.

  Does it normal behavior ? Is any way to increase performance of decoding mp4 ?