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• avcodec/snow: Hardcode table to save space

  2 December 2020, by Andreas Rheinhardt

  avcodec/snow: Hardcode table to save space
  The size of ff_qexp is only 32 bytes, but the code to generate it at
  
  runtime takes 47 bytes (GCC 9.3, x64, -O3 in an av_cold function); so
  
  just hardcode it.
  Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavcodec/snow.c
  • [DH] libavcodec/snow.h
  • [DH] libavcodec/snowdata.h

• avcodec/bgmc: Check input space in ff_bgmc_decode_init()

  1 September 2019, by Michael Niedermayer

  avcodec/bgmc: Check input space in ff_bgmc_decode_init()
  Fixes: Infinite loop
  
  Fixes: 16608/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-5636229827133440
  Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
  
  Reviewed-by: Thilo Borgmann <thilo.borgmann@mail.de>
  
  Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavcodec/alsdec.c
  • [DH] libavcodec/bgmc.c
  • [DH] libavcodec/bgmc.h

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

  18 June 2016, by 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?