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• avcodec/snow : Hardcode table to save space

  2 décembre 2020, par 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()

  1er septembre 2019, par 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 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 ?