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• avcodec/cbs : Fix potential overflow

  17 novembre 2019, par Andreas Rheinhardt

  avcodec/cbs : Fix potential overflow
  The number of bits in a PutBitContext must fit into an int, yet nothing
  
  guaranteed the size argument cbs_write_unit_data() uses in init_put_bits()
  
  to be in the range 0..INT_MAX / 8. This has been changed.
  Furthermore, the check 8 * data_size > data_bit_start that there is
  
  data beyond the initial padding when writing mpeg2 or H.264/5 slices
  
  could also overflow, so divide it by 8 to get an equivalent check
  
  without this problem.
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavcodec/cbs.c
  • [DH] libavcodec/cbs_h2645.c
  • [DH] libavcodec/cbs_mpeg2.c

• Concatenating two AAC decreasing number of frames packets

  29 décembre 2019, par Ahmed Hawary

  I am trying to concatenate two m4a (aac) files using the FFmpeg with the following command :

  ffmpeg -f concat -i input.txt -codec copy output.m4a

  the first file number of frames using afinfo on macOS :

  File type ID:   m4af
  
  Data format:     1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
  
  no channel layout.
  
  estimated duration: 8.473832 sec
  
  audio bytes: 68931
  
  audio packets: 367
  
  bit rate: 64710 bits per second
  
  packet size upper bound: 391
  
  maximum packet size: 391
  
  audio data file offset: 44
  
  not optimized
  
  audio 373696 valid frames + 2112 priming + 0 remainder = 375808
  
  format list:
  
  [ 0] format:      1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
  
  Channel layout: Mono

  The second file metadata :

  File type ID:   m4af
  
  Num Tracks:     1
  
  Data format:     1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
  
  no channel layout.
  
  estimated duration: 5.594558 sec
  
  audio bytes: 46077
  
  audio packets: 243
  
  bit rate: 65329 bits per second
  
  packet size upper bound: 340
  
  maximum packet size: 340
  
  audio data file offset: 44
  
  not optimized
  
  audio 246720 valid frames + 2112 priming + 0 remainder = 248832
  
  format list:
  
  [ 0] format:      1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
  
  Channel layout: Mono

  the resulted audio files metadate :

  File type ID:   m4af
  
  Num Tracks:     1
  
  Data format:     1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
  
  no channel layout.
  
  estimated duration: 14.070998 sec
  
  audio bytes: 122792
  
  audio packets: 607
  
  bit rate: 69696 bits per second
  
  packet size upper bound: 293
  
  maximum packet size: 293
  
  audio data file offset: 40
  
  not optimized
  
  audio 620531 valid frames + 1024 priming + 13 remainder = 621568
  
  format list:
  
  [ 0] format:      1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
  
  Channel layout: Mono

  The problem is that the total number of frames should be 367+293 = 610 and the resulted number of frames is 607. And the duration is 14.070998 sec instead of 14.06839 sec

  Any ideas if I am doing anything wrong here ? I need to precisely concatenate the two files without any loss or gain in the input frames.

• Slowing down audio using FFMPEG

  24 janvier 2020, par Maxim_A

  For example I have a source file which is the duration of 6.40 seconds.
  I divide this duration into 10 sections. Then each section is slowed down by a certain value. This works great.

  ffmpeg.exe -i preresult.mp4 -filter_complex 
  
  "[0:v]trim=0:0.5,setpts=PTS-STARTPTS[vv0]; 
  
  [0:v]trim=0.5:1,setpts=PTS-STARTPTS[vv1]; 
  
  [0:v]trim=1:1.5,setpts=PTS-STARTPTS[vv2]; 
  
  [0:v]trim=1.5:2,setpts=PTS-STARTPTS[vv3]; 
  
  [0:v]trim=2:2.5,setpts=PTS-STARTPTS[vv4]; 
  
  [0:v]trim=2.5:3,setpts=PTS-STARTPTS[vv5]; 
  
  [0:v]trim=3:3.5,setpts=PTS-STARTPTS[vv6]; 
  
  [0:v]trim=3.5:4,setpts=PTS-STARTPTS[vv7]; 
  
  [0:v]trim=4:4.5,setpts=PTS-STARTPTS[vv8]; 
  
  [0:v]trim=4.5:6.40,setpts=PTS-STARTPTS[vv9];
  
  
  
  [vv0]setpts=PTS*2[slowv0];
  
  [vv1]setpts=PTS*4[slowv1];
  
  [vv2]setpts=PTS*5[slowv2];
  
  [vv3]setpts=PTS*2[slowv3];
  
  [vv4]setpts=PTS*3[slowv4];
  
  [vv5]setpts=PTS*6[slowv5];
  
  [vv6]setpts=PTS*3[slowv6];
  
  [vv7]setpts=PTS*5[slowv7];
  
  [vv8]setpts=PTS*2[slowv8];
  
  [vv9]setpts=PTS*6[slowv9];
  
  
  
  [slowv0][slowv1][slowv2][slowv3][slowv4][slowv5][slowv6][slowv7][slowv8][slowv9]concat=n=10:v=1:a=0[v1]"  
  
  -r 30 -map "[v1]" -y result.mp4

  Then I needed to slow down along with the video and audio stream. In the documentation I found out about the atempo filter. The documentation says that the extreme boundaries of the value of this filter are from 0.5 to 100. To slow down by half, you need to use the value 0.5. I also learned that if you need to slow down the audio by 4 times, then you just need to apply two filters.

  [aa0]atempo=0.5[aslowv0] //Slowdown x2
  
  [aa0]atempo=0.5, atempo=0.5[aslowv0] //Slowdown x4

  Question 1 :
  How can i slow down audio an odd number of times ? for example, slow down audio by 3.5.7 times. There is no explanation of this point in the documentation.

  Question 2 :
  Do i understand correctly that if you slow down separately the audio stream and the video stream, they will have the same duration ?

  Thank you all in advance !