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• ffmpeg : split mp3, encode aac and join produce artifacts and empty space

  18 juin 2016, par aganeiro

  Source mp3

      ffprobe -show_frames -select_streams a -print_format csv -show_entries  
  
   frame=index,pkt_dts_time ~/demo_files/000.orig.5352357791787324393.mp3
  
  frame,0.000000
  
  frame,0.026122
  
  frame,0.052245
  
  frame,0.078367

  every part I make with command, -ss position and -t time I got and calculate from previous ffprobe output

      /home/xxx/bin/ffmpeg -analyzeduration 50000000 -probesize 50000000  
  
  -ss 0.000000 -i /home/xxx/demo_files/000.orig.5352357791787324393.mp3  
  
  -s 0 -t 0.926276 -flags +global_header -c:a libfdk_aac -strict -2  
  
  -b:a 64k -ac 2 -ar 44100 -vn -f mpegts -y /tmp/p0.ts
  
  
  
      /home/xxx/bin/ffmpeg -analyzeduration 50000000 -probesize 50000000  
  
  -ss 1.018776 -i /home/xxx/demo_files/000.orig.5352357791787324393.mp 
  
  -s 0 -t 0.900153 -flags +global_header -c:a libfdk_aac -strict -2  
  
  -b:a 64k -ac 2 -ar 44100 -vn -f mpegts -y /tmp/p1.ts

  it’s produce

  [mp3 @ 0x39ca980] Estimating duration from bitrate, this may be inaccurate
  
      Input #0, mp3, from '/home/xxx/demo_files/000.orig.5352357791787324393.mp3':
  
      Duration: 00:05:17.20, start: 0.000000, bitrate: 320 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
  
      [mpegts @ 0x39ccea0] Using AVStream.codec to pass codec  
  
  parameters to muxers is deprecated, use AVStream.codecpar instead.
  
      [mpegts @ 0x39ccea0] frame size not set
  
      Output #0, mpegts, to '/tmp/p0.ts':
  
        Metadata:
  
          encoder         : Lavf57.38.100
  
          Stream #0:0: Audio: aac (libfdk_aac), 44100 Hz, stereo, s16, 64 kb/s
  
          Metadata:
  
            encoder         : Lavc57.46.100 libfdk_aac
  
      Stream mapping:
  
        Stream #0:0 -> #0:0 (mp3 (native) -> aac (libfdk_aac))
  
      Press [q] to stop, [?] for help
  
      size=      10kB time=00:00:00.92 bitrate=  92.3kbits/s speed=39.8x    
  
      video:0kB audio:8kB subtitle:0kB other streams:0kB global  
  
  headers:0kB muxing overhead: 24.619143%
  
        Duration: 00:00:00.63, start: 1.400000, bitrate: 127 kb/s

  Part info

      ffmpeg -hide_banner -i /tmp/p0.ts 2>&1 |grep -P 'Duration|Stream'
  
      Duration: 00:00:00.95, start: 1.400000, bitrate: 90 kb/s
  
      Stream #0:0[0x100]: Audio: aac (LC) ([15][0][0][0] / 0x000F),  
  
  44100 Hz, stereo, fltp, 68 kb/s

  Then I join all parts together with

      /home/xxx/bin/ffmpeg -i /tmp/p0.ts -i /tmp/p1.ts -i /tmp/p2.ts  
  
  -i /tmp/p3.ts -i /tmp/p4.ts -i /tmp/p5.ts -filter_complex  
  
  "[0:a]asetpts=PTS-STARTPTS[a0];[1:a]asetpts=PTS-STARTPTS[a1];  
  
  [2:a]asetpts=PTS-STARTPTS[a2];[3:a]asetpts=PTS-STARTPTS[a3];  
  
  [4:a]asetpts=PTS-STARTPTS[a4];[5:a]asetpts=PTS-STARTPTS[a5];  
  
  [a0][a1][a2][a3][a4][a5] concat=n=6:v=0:a=1 [a]"  
  
  -map [a] -strict experimental -fflags +genpts -flags +global_header  
  
  -c libfdk_aac -bsf:a aac_adtstoasc -y /tmp/res.m4a

  waveform of original and joined on the left
  

  So, as you can see joined have delays and waveforms starte later. Why ? maybe it depens that all encoded parts have start time 1.400000, ?? How to set start time to 0 on encode ?

  Also I tried to cut empty space on joining with filter_complex but result stil not good and contains artifacts because trim position looks different in every part.

      /home/xxx/bin/ffmpeg -i /tmp/p0.ts -i /tmp/p1.ts -i /tmp/p2.ts  
  
  -i /tmp/p3.ts -i /tmp/p4.ts -i /tmp/p5.ts -filter_complex  
  
  "[0:a]atrim=0.020000,asetpts=PTS-STARTPTS[a0];  
  
  [1:a]atrim=0.020000,asetpts=PTS-STARTPTS[a1];  
  
  [2:a]atrim=0.020000,asetpts=PTS-STARTPTS[a2];  
  
  [3:a]atrim=0.020000,asetpts=PTS-STARTPTS[a3];  
  
  [4:a]atrim=0.020000,asetpts=PTS-STARTPTS[a4];  
  
  [5:a]atrim=0.020000,asetpts=PTS-STARTPTS[a5];  
  
  [a0][a1][a2][a3][a4][a5] concat=n=6:v=0:a=1 [a]"  
  
  -map [a] -strict experimental -fflags +genpts  
  
  -flags +global_header -c libfdk_aac -bsf:a aac_adtstoasc  
  
  -y /tmp/res.m4a

  Whyyyy and how to solve it ?

• avcodec/huffman : beautify : add space between #include and filename.

  22 juillet 2016, par Yong Lei

  avcodec/huffman : beautify : add space between #include and filename.
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavcodec/huffman.c

• What is the ideal image format extraction for YUV color space and lossless compression from mxf video format using ffmpeg

  15 février 2016, par DragonDance27

  I would like to be able to work on frames from an .mxf video file. The image format of the video is JPEG 2000, using a YUV color space with 4:2:2 subsampling and lossless compression.

  My intentions are to extract frames from this video using ffmpeg. Extracted frames would then be processed in Matlab (at the moment I’m interested in performing colorization).

  I want to extract the frames with as minimal data loss as possible and I would like to work in the YUV color space. I understand PNG involves a lossless process, but only involves the RGB color space - so not an option.

  I think I can extract jpeg2000 frames in the YUV color space, but I’m not sure if I’m losing data from the compression process. I attempted the following code in ffmpeg :

  ffmpeg -i video.mxf -r 1/5 out%03d.jp2

  ... however, the extracted jp2 files are unreadable in various software, including HiView which is a specialised JPEG 2000 software.

  Quesiton 1 : Is this jpeg 2000 extraction method lossless ? What am I doing wrong ?

  I also considered extracting the images in the tiff format where I can achieve the YUV and lossless requirements. I attempted the following code in ffmpeg :

  ffmpeg -i video.mxf -vcodec tiff f%10d.tif

  ... however, the extracted tiff files are unreadable in software such as paint, paint.net and windows photo viewer.

  Question 2 : Is this tiff extraction method correct ? What am I doing wrong ?

  Question 3 : What is an ideal image format that covers my YUV and lossless requirements ?