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• FFmpeg on iPhone - Modifying Video Orientation

  6 avril 2015, par Matthew McGoogan

  I’m messing with h264 videos loaded with FFmpeg on the iPhone 3GS. The problem is any videos recorded in "Portrait" orientation have a transformation matrix applied to them causing them to display rotated 90 degrees counter-clock.

  From what I understand thus far, I just need to modify the transform matrix in the ’tkhd’ atom. The problem is I am having trouble accessing or modifying this data. I checked out the FFmpeg implementation for :

  static int mov_read_tkhd(MOVContext *c, ByteIOContext *pb, MOVAtom atom)

  which clearly shows how the matrix is accessed in avformat but when I try to access the header bytes using the same functions I am not getting any rational values. Even if I were to successfully pull the matrix I’m not sure how to replace it ? FFmpeg has functions for retrieving and appending to the track header but nothing for replace it seems ?

  Any help would be greatly appreciated.

  Thanks,
  Matt.

• flv reencode to mp4 for iphone/ipod via ffmpeg and x264 (quality issue)

  3 octobre 2011, par zeroasterisk

  There are a lot of questions on this topic, and I've read most of them and most of the google search results I could come up with.

  When I use FFMPEG to convert a FLV to a iphone3 compatble MP4 file, it just doesn't preserver enough of the quality. Yes, I've worked the hell out of -sameq and -b and -bt settings, text just isn't readable.

  Next I tried to split the video out and process it directly, using these instructions :
  https://sites.google.com/site/linuxencoding/x264-encoding-guide

  The problem is myplayer (via ffmpeg) was not able to determine the duration of the FLV (even though the metadata was set).

  (I assume) Because of that unknown duration, when I create the MP4 file, the resulting x264 file plays through super-fast while the audio plays at the normal rate.

  user@server:/tmp# mplayer -nosound -benchmark -sws 9 -vf dsize=640:480:0,scale=0:0,expand=640:480 -vo yuv4mpeg:file=>(x264 --demuxer y4m --crf 0 --preset slow --threads auto --output output.264 - 2>x264.log) 'input.flv'
  
  MPlayer 1.0rc4-4.4.5 (C) 2000-2010 MPlayer Team
  
  mplayer: could not connect to socket
  
  mplayer: No such file or directory
  
  Failed to open LIRC support. You will not be able to use your remote control.
  
  
  
  Playing input.flv.
  
  libavformat file format detected.
  
  [flv @ 0x1202460]Estimating duration from bitrate, this may be inaccurate
  
  [lavf] stream 0: video (vp6f), -vid 0
  
  [lavf] stream 1: audio (nellymoser), -aid 0
  
  VIDEO:  [VP6F]  1680x992  0bpp  1000.000 fps   33.4 kbps ( 4.1 kbyte/s)
  
  Clip info:
  
   audiocodecid: 6
  
   audiodatarate: 86
  
   audiosamplerate: 44100
  
   audiosamplesize: 16
  
   audiosize: 6097005
  
   canSeekToEnd: true
  
   datasize: 8609138
  
   duration: 567
  
   framerate: 2
  
   hasAudio: true
  
   hasCuePoints: false
  
   hasKeyframes: true
  
   hasMetadata: true
  
   hasVideo: true
  
   height: 992
  
   lasttimestamp: 567
  
   metadatacreator: flvtool++ (Facebook, Motion project, dweatherford)
  
   stereo: false
  
   totalframes: 1043
  
   videocodecid: 4
  
   videodatarate: 33
  
   videosize: 2316256
  
   width: 1680
  
  Using (default) progressive frame mode.Opening video filter: [expand w=640 h=480]
  
  Expand: 640 x 480, -1 ; -1, osd: 0, aspect: 0.000000, round: 1
  
  Opening video filter: [scale w=0 h=0]
  
  Opening video filter: [dsize=640:480:0]
  
  ==========================================================================
  
  Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
  
  Selected video codec: [ffvp6f] vfm: ffmpeg (FFmpeg VP6 Flash)
  
  ==========================================================================
  
  Audio: no sound
  
  Starting playback...
  
  Movie-Aspect is undefined - no prescaling applied.
  
  [swscaler @ 0x7f0c738b9620]Lanczos scaler, from yuv420p to yuv420p using MMX2
  
  VO: [yuv4mpeg] 640x480 => 641x480 Planar YV12

  I have also tried specifying FPS, but no change in results

  user@server:/tmp# mplayer -nosound -fps 25-benchmark -sws 9 -vf dsize=640:480:0,scale=0:0,expand=640:480 -vo yuv4mpeg:file=>(x264 --demuxer y4m --fps 25 --crf 0 --preset slow --threads auto --output output.264 - 2>x264.log) 'input.flv'

  Can someone tell me how to either :

  1. fix my split A/V processing/timing/duration issues ?
  2. improve the
     quality of the FFMPEG conversion of FLV to iphone3 compatible
     format ?

• iPhone ffmpeg dev using libav to decode from AMR to ACC codec

  10 octobre 2011, par VictorT

  It seems to be that, the AMR support of AudioQueue has been disappeared since iOS 4.3 was released. I can't use audio frame received from RSTP server with old way :

  audioFormat.mFormatID = kAudioFormatAMR; 
  
  int err = AudioQueueNewOutput(&audioFormat, MyAudioQueueOutputCallback, self, NULL, kCFRunLoopCommonModes, 0, &audioQueue);

  As a result I received an error in last string.

  Maybe someone know how to decode AMR AVPacket into raw buffer and encode it with AAC or MP3 using LIBAV ?

  I've tried to use

  avcodec_decode_audio3

  It works and I can get raw buffer but when I'm trying to encode it with

  avcodec_encode_audio

  I get 0 as result

  This is my method to encode buffer :

  - (AVPacket) encodeRawFrame:(const short *) in_buffer withSize:(unsigned int) in_buf_byte_size
  
  {
  
      AVPacket res;
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int count, out_size, outbuf_size, frame_byte_size;
  
      uint8_t *outbuf;
  
  
  
      avcodec_init();
  
      avcodec_register_all();
  
  
  
      printf("Audio encoding\n");
  
  
  
      codec = avcodec_find_encoder(CODEC_ID_AAC);
  
      if (!codec) {
  
          fprintf(stderr, "codec not found\n");
  
          return res;
  
      }
  
  
  
      c= avcodec_alloc_context();
  
  
  
      c->bit_rate = 64000;
  
      c->sample_rate = 24000;
  
      c->channels = 2;
  
  
  
      if (avcodec_open(c, codec) < 0) 
  
      {
  
          fprintf(stderr, "could not open codec\n");
  
      }
  
      else
  
      {
  
          frame_byte_size=c->frame_size*2*2;
  
          count = in_buf_byte_size/frame_byte_size;
  
  
  
          fprintf(stderr, "Number of frames: %d\n", count);
  
  
  
          outbuf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
          outbuf = (uint8_t*) malloc(outbuf_size);
  
  
  
          out_size = avcodec_encode_audio(c, outbuf, outbuf_size, &in_buffer[frame_byte_size*i]);
  
          if(out_size >= 0)
  
          {
  
              res.size = outbuf_size;
  
              res.data = malloc(outbuf_size);                
  
          }
  
  
  
          free(outbuf);
  
      }
  
  
  
  
  
      avcodec_close(c);
  
      av_free(c);
  
      return res;
  
  }

  After encoding "out_size" is always 0 and result buffer is empty.

  Thanks.