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• Trying to concatenate multiple videos with multiple audio streams using ffmpeg

  25 avril 2019, par Toto

  I’m trying to concatenate 3 mp4 videos, each contains 3 audio streams using ffmpeg, since I couldn’t merge them with other software because of differences in the codecs private data.

  Since the files have different codecs I’m trying to use the concat video filter but I can’t seem to get it to work. The output turns out to be an unplayable file. I’m trying to specify the input files, and every media stream, then telling the software to concatenate it and copy it to the output file.

  ffmpeg -i A.mp4 -i B.mp4 -i C.mp4 -filter_complex  '[0:0] [0:1] [0:2] [0:3] [1:0] [1:1] [1:2] [1:3] [2:0] [2:1] [2:2] [2:3] concat=n=3:v=1:a=3 [v] [a1] [a2] [a3]'  -map '[v]' -map '[a1]' -map '[a2]'-map '[a3]' output.mp4

  The complete log :

  ffmpeg -i A.mp4 -i B.mp4 -i C.mp4 -filter_complex  '[0:0] [0:1] [0:2] [0:3] [1:0] [1:1] [1:2] [1:3] [2:0] [2:1] [2:2] [2:3] concat=n=3:v=1:a=3 [v] [a1] [a2] [a3]'  -map '[v]' -map '[a1]' -map '[a2]'-map '[a3]' output.mp4
  
  ffmpeg version 4.1.1 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20190212
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    libavutil      56. 22.100 / 56. 22.100
  
    libavcodec     58. 35.100 / 58. 35.100
  
    libavformat    58. 20.100 / 58. 20.100
  
    libavdevice    58.  5.100 / 58.  5.100
  
    libavfilter     7. 40.101 /  7. 40.101
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'A.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : HandBrake 0.10.1 2015030800
  
    Duration: 01:59:10.12, start: 0.000000, bitrate: 6868 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 6017 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 320 kb/s (default)
  
      Metadata:
  
        handler_name    : Stereo
  
      Stream #0:2(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 320 kb/s
  
      Metadata:
  
        handler_name    : Surround
  
      Stream #0:3(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        handler_name    : Stereo
  
  Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'B.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : HandBrake 0.10.1 2015030800
  
    Duration: 01:12:28.48, start: 0.000000, bitrate: 6846 kb/s
  
      Stream #1:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 6001 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #1:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 320 kb/s (default)
  
      Metadata:
  
        handler_name    : Stereo
  
      Stream #1:2(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 320 kb/s
  
      Metadata:
  
        handler_name    : Surround
  
      Stream #1:3(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        handler_name    : Stereo
  
  Input #2, mov,mp4,m4a,3gp,3g2,mj2, from 'C.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : HandBrake 0.10.1 2015030800
  
    Duration: 00:58:55.53, start: 0.000000, bitrate: 6850 kb/s
  
      Stream #2:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 6005 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #2:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 319 kb/s (default)
  
      Metadata:
  
        handler_name    : Stereo
  
      Stream #2:2(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 319 kb/s
  
      Metadata:
  
        handler_name    : Surround
  
      Stream #2:3(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        handler_name    : Stereo
  
  Stream mapping:
  
    Stream #0:0 (h264) -> concat:in0:v0
  
    Stream #0:1 (aac) -> concat:in0:a0
  
    Stream #0:2 (aac) -> concat:in0:a1
  
    Stream #0:3 (aac) -> concat:in0:a2
  
    Stream #1:0 (h264) -> concat:in1:v0
  
    Stream #1:1 (aac) -> concat:in1:a0
  
    Stream #1:2 (aac) -> concat:in1:a1
  
    Stream #1:3 (aac) -> concat:in1:a2
  
    Stream #2:0 (h264) -> concat:in2:v0
  
    Stream #2:1 (aac) -> concat:in2:a0
  
    Stream #2:2 (aac) -> concat:in2:a1
  
    Stream #2:3 (aac) -> concat:in2:a2
  
    concat:out:v0 -> Stream #0:0 (libx264)
  
    concat:out:a0 -> Stream #0:1 (aac)
  
    concat:out:a1 -> Stream #0:2 (aac)
  
    concat:out:a2 -> Stream #0:3 (aac)
  
  Press [q] to stop, [?] for help
  
  [libx264 @ 000001930c898980] using SAR=1/1
  
  [libx264 @ 000001930c898980] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 000001930c898980] profile High, level 4.0, 4:2:0, 8-bit
  
  [libx264 @ 000001930c898980] 264 - core 157 r2935 545de2f - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
  
  Output #0, mp4, to 'output.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf58.20.100
  
      Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 29.97 fps, 30k tbn, 29.97 tbc (default)
  
      Metadata:
  
        encoder         : Lavc58.35.100 libx264
  
      Side data:
  
        cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
  
      Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc58.35.100 aac
  
      Stream #0:2: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 341 kb/s
  
      Metadata:
  
        encoder         : Lavc58.35.100 aac
  
      Stream #0:3: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc58.35.100 aac
  
  [aac @ 000001930d3a88c0] skip_data_stream_element: Input buffer exhausted before END element found
  
  Error while decoding stream #0:1: Invalid data found when processing input
  
  [aac @ 000001930c89a4c0] Number of bands (48) exceeds limit (47).ate=6048.4kbits/s dup=1 drop=0 speed=0.916x
  
  Error while decoding stream #2:1: Invalid data found when processing input
  
  frame=450571 fps= 28 q=-1.0 Lsize=11204203kB time=04:10:34.13 bitrate=6105.1kbits/s dup=1 drop=0 speed=0.939x
  
  video:10071486kB audio:1099256kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.299544%
  
  [libx264 @ 000001930c898980] frame I:4679  Avg QP:20.30  size:103274
  
  [libx264 @ 000001930c898980] frame P:125629 Avg QP:22.82  size: 41052
  
  [libx264 @ 000001930c898980] frame B:320263 Avg QP:25.08  size: 14590
  
  [libx264 @ 000001930c898980] consecutive B-frames:  2.3%  6.0%  8.8% 83.0%
  
  [libx264 @ 000001930c898980] mb I  I16..4: 16.4% 78.4%  5.2%
  
  [libx264 @ 000001930c898980] mb P  I16..4:  7.0% 17.1%  1.1%  P16..4: 43.0%  9.7%  4.9%  0.0%  0.0%    skip:17.3%
  
  [libx264 @ 000001930c898980] mb B  I16..4:  0.8%  1.7%  0.1%  B16..8: 39.4%  3.6%  0.7%  direct: 4.3%  skip:49.4%  L0:43.8% L1:50.2% BI: 6.0%
  
  [libx264 @ 000001930c898980] 8x8 transform intra:68.6% inter:84.9%
  
  [libx264 @ 000001930c898980] coded y,uvDC,uvAC intra: 40.3% 64.9% 17.2% inter: 12.9% 24.2% 0.4%
  
  [libx264 @ 000001930c898980] i16 v,h,dc,p: 40% 22% 10% 28%
  
  [libx264 @ 000001930c898980] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 15% 23%  5%  6%  7%  5%  6%  5%
  
  [libx264 @ 000001930c898980] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 16% 16%  6%  9%  9%  7%  6%  5%
  
  [libx264 @ 000001930c898980] i8c dc,h,v,p: 48% 20% 26%  6%
  
  [libx264 @ 000001930c898980] Weighted P-Frames: Y:8.6% UV:6.4%
  
  [libx264 @ 000001930c898980] ref P L0: 58.8% 12.5% 19.8%  8.4%  0.4%
  
  [libx264 @ 000001930c898980] ref B L0: 88.5%  9.1%  2.4%
  
  [libx264 @ 000001930c898980] ref B L1: 96.9%  3.1%
  
  [libx264 @ 000001930c898980] kb/s:5487.92
  
  [aac @ 000001930c899280] Qavg: 212.043
  
  [aac @ 000001930d1cfbc0] Qavg: 211.329
  
  [aac @ 000001930d1ce080] Qavg: 228.685

  This is the output when trying the concat demuxer, which also produces an unplayable file.

  ffmpeg -f concat -safe 0 -i mylist.txt -map 0 -c copy concat2.mp4
  
  ffmpeg version 4.1.1 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20190212
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    libavutil      56. 22.100 / 56. 22.100
  
    libavcodec     58. 35.100 / 58. 35.100
  
    libavformat    58. 20.100 / 58. 20.100
  
    libavdevice    58.  5.100 / 58.  5.100
  
    libavfilter     7. 40.101 /  7. 40.101
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0000025ca95f2dc0] Auto-inserting h264_mp4toannexb bitstream filter
  
  Input #0, concat, from 'mylist.txt':
  
    Duration: N/A, start: 0.000000, bitrate: 6850 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 6017 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 320 kb/s
  
      Metadata:
  
        handler_name    : Stereo
  
      Stream #0:2(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 320 kb/s
  
      Metadata:
  
        handler_name    : Surround
  
      Stream #0:3(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        handler_name    : Stereo
  
  Output #0, mp4, to 'concat2.mp4':
  
    Metadata:
  
      encoder         : Lavf58.20.100
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 6017 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 320 kb/s
  
      Metadata:
  
        handler_name    : Stereo
  
      Stream #0:2(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 320 kb/s
  
      Metadata:
  
        handler_name    : Surround
  
      Stream #0:3(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        handler_name    : Stereo
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #0:1 -> #0:1 (copy)
  
    Stream #0:2 -> #0:2 (copy)
  
    Stream #0:3 -> #0:3 (copy)
  
  Press [q] to stop, [?] for help
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0000025ca95f9480] Auto-inserting h264_mp4toannexb bitstream filter4x
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0000025ca95f9480] Auto-inserting h264_mp4toannexb bitstream filter5x
  
  frame=450570 fps=1251 q=-1.0 Lsize=12591157kB time=04:10:34.11 bitrate=6860.8kbits/s speed=41.7x
  
  video:11029679kB audio:1527897kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.267411%

• add subtitle to video created from mp3 file and an image using ffmpeg

  1er février 2018, par no name

  I have a simple image and an mp3 file all I want to do is to add a lyrics to this mp3 file in a video as an output with fade out style.
  so i used this command to do so from this this blog
  

  ffmpeg -loop 1 -y -i test.jpg -i test.mp3 -vf subtitles=”f=test.srt:force_style=’FontName=Arial,FontSize=14″ -shortest test1.mp4

  when i used this command i got this eror

      ffmpeg -loop 1 -y -i test.jpg -i test.mp3 -vf subtitles="f=test.srt:force_style=’FontName=Arial,FontSize=14″ -shortest test1.mp4
  
  ffmpeg version N-89940-gb1af0e23a3 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.2.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libmfx --enable-amf --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
  
    libavutil      56.  7.100 / 56.  7.100
  
    libavcodec     58.  9.100 / 58.  9.100
  
    libavformat    58.  7.100 / 58.  7.100
  
    libavdevice    58.  0.101 / 58.  0.101
  
    libavfilter     7. 11.101 /  7. 11.101
  
    libswscale      5.  0.101 /  5.  0.101
  
    libswresample   3.  0.101 /  3.  0.101
  
    libpostproc    55.  0.100 / 55.  0.100
  
  Trailing options were found on the commandline.
  
  Input #0, image2, from 'test.jpg':
  
    Duration: 00:00:00.04, start: 0.000000, bitrate: 17290 kb/s
  
      Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 960x959 [SAR 1:1 DAR 960:959], 25 fps, 25 tbr, 25 tbn, 25 tbc
  
  Input #1, mp3, from 'test.mp3':
  
    Metadata:
  
      comment         : converted by convert2mp3.net
  
      title           : Celestial Empire [ Worlds 2017 ] [ Pick and b
  
      artist          : Raffael Gruber Matthias Ullrich
  
    Duration: 00:02:33.13, start: 0.025057, bitrate: 193 kb/s
  
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc57.28
  
  At least one output file must be specified

  iam using windows 10, last version of ffmpeg, so what makes this problem. also how to make the style of the lyrics to be something like fade in/out and so on. if it’s possible
  Thanks in advance.
  update
  when i use this command

  ffmpeg -loop 1 -y -i test.jpg -i test.mp3 -vf "subtitles=test.srt:force_style='FontName=Arial,FontSize=14'" -shortest test1.mp4

  it give me this error

  Press [q] to stop, [?] for help
  
  [Parsed_subtitles_0 @ 049af980] Shaper: FriBidi 0.19.7 (SIMPLE)
  
  [Parsed_subtitles_0 @ 049af980] Using font provider directwrite
  
  [libx264 @ 049d2440] height not divisible by 2 (960x959)
  
  Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
  
  Conversion failed!

  i searched this error and i found a solution by add this part before output file so the final command is

  ffmpeg -loop 1 -y -i test.jpg -i test.mp3 -vf "subtitles=test.srt:force_style='FontName=Arial,FontSize=14'" -shortest -vf "scale=trunc(iw/2)*2:trunc(ih/2)*2" test1.mp4

  i got no error and it worked but the output video not contains any lyrics which i added

  Update
  
  last thing i tried and it works first i use this command to convert the image and mp3 file to a mp4 video

  ffmpeg -loop 1 -i test.jpg -i test.mp3 -shortest -acodec copy -vf "scale=trunc(iw/2)*2:trunc(ih/2)*2" video.mp4

  but this command take a long time to process to convert mp3 of 2:55 minute it with image.
  then i start to add lyrics or subtitle using this command

  ffmpeg -i video.mp4 -vf "subtitles=test.srt:force_style='FontName=Arial,FontSize=14,PrimaryColour=&H0000ff&'" -c:a copy output.mp4

  this worked perfect and give the output but it take a long time to process i don’t know if this normal.
  now last thing how can i set the style of the subtitle so it be fade out or something

• How can I convert audio to WAVE_FORMAT_PCM using FFmpeg ?

  16 juin, par Eric Stdlib

  I am using Python's wave module to read audio, and using FFmpeg to convert audio from other types to wav. However, I am encountering some problem.

  



  I wrote v.py to generate an silence audio file a.wav

  



  import sys, wave, math
import numpy as np

wave_data = np.zeros(44100).astype(np.short)

f = wave.open('a.wav', 'wb')
f.setnchannels(1)
f.setsampwidth(2)
f.setframerate(96000)
f.writeframes(wave_data.tostring())
f.close()


  



  Then I used FFmpeg to "copy" a.wav to b.wav (though it seems to encode / decode the file), but I can only read a.wav with Python ; b.wav cannot be opened.

  



  [user@localhost tmp]$ ffmpeg -i a.wav b.wav&#xA;Guessed Channel Layout for Input Stream #0.0 : mono&#xA;Input #0, wav, from &#x27;a.wav&#x27;:&#xA;  Duration: 00:00:00.46, bitrate: 1536 kb/s&#xA;    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, mono, s16, 1536 kb/s&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))&#xA;Press [q] to stop, [?] for help&#xA;Output #0, wav, to &#x27;b.wav&#x27;:&#xA;  Metadata:&#xA;    ISFT            : Lavf57.71.100&#xA;    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, mono, s16, 1536 kb/s&#xA;    Metadata:&#xA;      encoder         : Lavc57.89.100 pcm_s16le&#xA;size=      86kB time=00:00:00.45 bitrate=1537.8kbits/s speed= 706x    &#xA;video:0kB audio:86kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.115646%&#xA;[user@localhost tmp]$ python3&#xA;Python 3.6.4 (default, Jan 23 2018, 22:25:37) &#xA;[GCC 7.2.1 20170915 (Red Hat 7.2.1-2)] on linux&#xA;Type "help", "copyright", "credits" or "license" for more information.&#xA;>>> import wave&#xA;>>> wave.open(&#x27;a.wav&#x27;)&#xA;&#xA;>>> wave.open(&#x27;b.wav&#x27;)&#xA;Traceback (most recent call last):&#xA;  File "<stdin>", line 1, in <module>&#xA;  File "/usr/lib64/python3.6/wave.py", line 499, in open&#xA;    return Wave_read(f)&#xA;  File "/usr/lib64/python3.6/wave.py", line 163, in __init__&#xA;    self.initfp(f)&#xA;  File "/usr/lib64/python3.6/wave.py", line 143, in initfp&#xA;    self._read_fmt_chunk(chunk)&#xA;  File "/usr/lib64/python3.6/wave.py", line 260, in _read_fmt_chunk&#xA;    raise Error(&#x27;unknown format: %r&#x27; % (wFormatTag,))&#xA;wave.Error: unknown format: 65534&#xA;>>> &#xA;</module></stdin>

  



  How should I change the command of FFmpeg to convert the file to WAVE_FORMAT_PCM, so that I can read b.wav with Python ?