[FFmpeg-trac] #6603(ffmpeg:new): Regression bug when using amerge filter and WAV files.

[FFmpeg]

#6603: Regression bug when using amerge filter and WAV files.
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             Reporter:  Atarikid  |                     Type:  defect
               Status:  new       |                 Priority:  normal
            Component:  ffmpeg    |                  Version:  git-master
             Keywords:            |               Blocked By:
             Blocking:            |  Reproduced by developer:  0
Analyzed by developer:  0         |
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 I found a regression bug with the latest builds 3.3.0 or higher. Works
 fine with 3.2.2 and lower.

 When using amerge with WAV files, FFmpeg always throws an error. Using any
 other audio file (read: non .WAV) it works fine.

 Here is the full ffmpeg output

 stef-MacBook-Pro:~ stef $ /Users/stef/Desktop/ffmpeg-multibit265 -i
 /Users/stef/Desktop/issue/test.mp4 -c:a aac -ab 128k -c:v libx264 -crf 20
 -r 25 -s 480x270 -aspect 16:9 -pix_fmt yuv420p -filter_complex
 "amovie=/Users/stef/Desktop/issue/_left.wav[0];amovie=/Users/stef/Desktop/issue/_right.wav[1];[0][1]amerge=inputs=2"
 -sn -y /Users/stef/Movies/test.mov
 ffmpeg version 3.3 Copyright (c) 2000-2017 the FFmpeg developers
   built with Apple LLVM version 8.1.0 (clang-802.0.42)
   configuration: --prefix=/Volumes/tempdisk/sw --as=yasm --enable-gpl
 --enable-version3 --enable-pthreads --disable-ffplay --disable-ffserver
 --disable-shared --enable-static --enable-libvpx --disable-decoder=libvpx
 --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis
 --enable-libx264 --enable-libx265 --enable-libxvid --enable-zlib --enable-
 avfilter --enable-fontconfig --enable-libfreetype --enable-libass
 --enable-libvidstab --enable-libsnappy --enable-filters --enable-postproc
 --enable-runtime-cpudetect --disable-indev=qtkit --disable-
 indev=x11grab_xcb
   libavutil      55. 58.100 / 55. 58.100
   libavcodec     57. 89.100 / 57. 89.100
   libavformat    57. 71.100 / 57. 71.100
   libavdevice    57.  6.100 / 57.  6.100
   libavfilter     6. 82.100 /  6. 82.100
   libswscale      4.  6.100 /  4.  6.100
   libswresample   2.  7.100 /  2.  7.100
   libpostproc    54.  5.100 / 54.  5.100
 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
 '/Users/stef/Desktop/issue/test.mp4':
   Metadata:
     major_brand     : isom
     minor_version   : 512
     compatible_brands: isomiso2avc1mp41
     creation_time   : 2017-08-20T17:16:53.000000Z
     encoder         : Lavf57.76.100
   Duration: 00:00:03.07, start: -0.001451, bitrate: 536 kb/s
     Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
 480x270 [SAR 1:1 DAR 16:9], 439 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc
 (default)
     Metadata:
       creation_time   : 2017-08-20T17:16:53.000000Z
       handler_name    : VideoHandler
     Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz,
 stereo, fltp, 96 kb/s (default)
     Metadata:
       creation_time   : 2017-08-20T17:16:53.000000Z
       handler_name    : SoundHandler
 [Parsed_amovie_0 @ 0x7feec7d009e0] Channel layout is not set in output
 stream 0, guessed channel layout is 'mono'
 [Parsed_amovie_1 @ 0x7feec7d03e00] Channel layout is not set in output
 stream 0, guessed channel layout is 'mono'
 Stream mapping:
   amerge (graph 0) -> Stream #0:0 (aac)
   Stream #0:0 -> #0:1 (h264 (native) -> h264 (libx264))
 Press [q] to stop, [?] for help
 [Parsed_amovie_0 @ 0x7feec7f30340] Channel layout is not set in output
 stream 0, guessed channel layout is 'mono'
 [Parsed_amovie_1 @ 0x7feec7c18760] Channel layout is not set in output
 stream 0, guessed channel layout is 'mono'
 [Parsed_amerge_2 @ 0x7feec7e02c40] Input channel layouts overlap: output
 layout will be determined by the number of distinct input channels
 [aac @ 0x7feec9801200] more samples than frame size
 (avcodec_encode_audio2)
 Audio encoding failed



 The [Parsed_amovie are nothing serious. It is the [aac @ 0x7fa3c0808c00]
 more samples than frame size (avcodec_encode_audio2) that is holding it.

 Note:
 If changing -c:a aac to -c:a pcm_s16le it works fine. Set it to any other
 audio codec it fails as it does with aac.

 Again, when using FFmpeg 3.2.2 it is working fine as expected.

 Attached the needed file to reproduce the issue.

--
Ticket URL: <https://trac.ffmpeg.org/ticket/6603>
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