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• How to use ffmpeg to merge multiple .mp4 files ? (same encoder, simple merge will work)

  6 août 2019, par Mars Robertson

  I’m trying :

  • How to concatenate two MP4 files using FFmpeg ?
  • https://superuser.com/questions/1059245/ffmpeg-join-two-mp4-files-with-ffmpeg-on-command-line

  ---

  Option A :

  ffmpeg -f concat -i list.txt -c copy output

  Line 1 : unknown keyword ’file’
  list.txt : Invalid data found when processing input

  cat list.txt
file './001-eclipse.mp4'
file './002-sundown.mp4'
file './003-robothear.mp4'
file './004-NOPE-boring-tower.mp4'

  Option B :

  ffmpeg -i "concat:001-eclipse.mp4|002-sundown.mp4|003-robothear.mp4|004-NOPE-boring-tower.mp4" -c copy output.mp4

  ffmpeg version 4.0.2 Copyright (c) 2000-2018 the FFmpeg developers
 built with Apple LLVM version 9.1.0 (clang-902.0.39.2)
 configuration: --prefix=/usr/local/Cellar/ffmpeg/4.0.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
 libavutil      56. 14.100 / 56. 14.100
 libavcodec     58. 18.100 / 58. 18.100
 libavformat    58. 12.100 / 58. 12.100
 libavdevice    58.  3.100 / 58.  3.100
 libavfilter     7. 16.100 /  7. 16.100
 libavresample   4.  0.  0 /  4.  0.  0
 libswscale      5.  1.100 /  5.  1.100
 libswresample   3.  1.100 /  3.  1.100
 libpostproc    55.  1.100 / 55.  1.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7feb1d001200] Found duplicated MOOV Atom. Skipped it
   Last message repeated 2 times
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'concat:001-eclipse.mp4|002-sundown.mp4|003-robothear.mp4|004-NOPE-boring-tower.mp4':
 Metadata:
   encoder         : Lavf58.12.100
   major_brand     : isom
   minor_version   : 512
   compatible_brands: isomiso2avc1mp41
 Duration: 00:00:13.64, start: 0.000000, bitrate: 161885 kb/s
   Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 5376x2688, 54015 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
   Metadata:
     handler_name    : VideoHandler
Output #0, mp4, to 'output.mp4':
 Metadata:
   compatible_brands: isomiso2avc1mp41
   major_brand     : isom
   minor_version   : 512
   encoder         : Lavf58.12.100
   Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 5376x2688, q=2-31, 54015 kb/s, 25 fps, 25 tbr, 12800 tbn, 12800 tbc (default)
   Metadata:
     handler_name    : VideoHandler
Stream mapping:
 Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
frame=  341 fps=0.0 q=-1.0 Lsize=   89943kB time=00:00:13.52 bitrate=54497.5kbits/s speed=90.2x
video:89937kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.005972%

  It contains only the first video :/

  ---

  Any ideas ?

  My version on Mac :

  ffmpeg -version
ffmpeg version 4.0.2 Copyright (c) 2000-2018 the FFmpeg developers
built with Apple LLVM version 9.1.0 (clang-902.0.39.2)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.0.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
libavutil      56. 14.100 / 56. 14.100
libavcodec     58. 18.100 / 58. 18.100
libavformat    58. 12.100 / 58. 12.100
libavdevice    58.  3.100 / 58.  3.100
libavfilter     7. 16.100 /  7. 16.100
libavresample   4.  0.  0 /  4.  0.  0
libswscale      5.  1.100 /  5.  1.100
libswresample   3.  1.100 /  3.  1.100
libpostproc    55.  1.100 / 55.  1.100

• Which player can play call center recoding file or convert

  26 août 2018, par Wida

  I’m working with call center recording file but it’s sound gibberish which mediainfo shows detail

  Format                                   : Wave
  
  File size                                : 239 KiB
  
  Duration                                 : 3 min 3 s
  
  Overall bit rate                         : 10.7 kb/s
  
  
  
  Audio
  
  Format                                   : G.723.1
  
  Codec ID                                 : A100
  
  Duration                                 : 3 min 3 s
  
  Bit rate                                 : 10.7 kb/s
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 8 000 Hz
  
  Stream size                              : 239 KiB (100%)

  and detail in from ffmpeg show as

  ffmpeg version 4.0 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.0 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
  
    libavutil      56. 14.100 / 56. 14.100
  
    libavcodec     58. 18.100 / 58. 18.100
  
    libavformat    58. 12.100 / 58. 12.100
  
    libavdevice    58.  3.100 / 58.  3.100
  
    libavfilter     7. 16.100 /  7. 16.100
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  1.100 /  5.  1.100
  
    libswresample   3.  1.100 /  3.  1.100
  
    libpostproc    55.  1.100 / 55.  1.100
  
  Input #0, wav, from 'test.wav':
  
    Duration: 00:03:03.84, bitrate: 10 kb/s
  
      Stream #0:0: Audio: g723_1 ([0][161][0][0] / 0xA100), 8000 Hz, mono, s16, 10 kb/s
  
  At least one output file must be specified

  I have used so many player : Audacity, VLC, WMP. But it’s sound gibberish.
  My case is very similar to

  Converting a call center recording to something useful

  in this link, he can fix by using Goldwave for converting file. I also tried Goldwave too but still didn’t work.

  So, Do I have to config more with Goldwave or any solution at least I want to listen to it

  I use MacOs but don’t mind if have to use window for open it.

  Ps. can listen sample audio in the link (my recoding is the same as the sample in this link)

  more detail from Switch by NCH Software

  format : WAVE
  
  codec : Comverse Infosys Ltd. G723 1
  
  Sample Rate : 8000
  
  Channels : 1 (Mono)

• Changing Audio Volume using FFmpeg Android Error

  23 août 2018, par kataroty

  I am using FFmpeg and I need to change the volume of the Audio.

  This is the command that should change the volume of the audio :

  ffmpeg -i input.wav -filter:a "volume=1.5" output.wav

  I read from somewhere that it had to overwrite the file if it already existed so either one had to delete the file or use -y. When I tried to do it without the -y command it never got to onSuccess() or ´onFailure(), but it always printed out the onStart() message.

  String[] cmdy = { "-i -y" , pcmtowavTempFile.toString(),  "-af", "volume=5", pcmtowavTempFile.toString()};
  
      ffmpeg.execute(cmdy, new ExecuteBinaryResponseHandler() {
  
  
  
          @Override
  
          public void onStart() {
  
              System.out.println("ayy: onStart");
  
          }
  
  
  
          @Override
  
          public void onSuccess(String message) {
  
              System.out.println("ayy: onSuccess");
  
              super.onSuccess(message);
  
          }
  
          @Override
  
          public void onFailure(String message) {
  
              System.out.println("ayy: onFailure " + message);
  
              super.onFailure(message);
  
          }
  
  });

  At the moment I get this error :

  ffmpeg version n4.0-39-gda39990 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.9.x (GCC) 20150123 (prerelease)
  
    configuration: --target-os=linux --cross-prefix=/root/bravobit/ffmpeg-android/toolchain-android/bin/arm-linux-androideabi- --arch=arm --cpu=cortex-a8 --enable-runtime-cpudetect --sysroot=/root/bravobit/ffmpeg-android/toolchain-android/sysroot --enable-pic --enable-libx264 --enable-ffprobe --enable-libopus --enable-libvorbis --enable-libfdk-aac --enable-libfreetype --enable-libfribidi --enable-libmp3lame --enable-fontconfig --enable-libvpx --enable-libass --enable-yasm --enable-pthreads --disable-debug --enable-version3 --enable-hardcoded-tables --disable-ffplay --disable-linux-perf --disable-doc --disable-shared --enable-static --enable-runtime-cpudetect --enable-nonfree --enable-network --enable-avresample --enable-avformat --enable-avcodec --enable-indev=lavfi --enable-hwaccels --enable-ffmpeg --enable-zlib --enable-gpl --enable-small --enable-nonfree --pkg-config=pkg-config --pkg-config-flags=--static --prefix=/root/bravobit/ffmpeg-android/build/armeabi-v7a --extra-cflags='-I/root/bravobit/ffmpeg-android/toolchain-android/include -U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -fno-strict-overflow -fstack-protector-all' --extra-ldflags='-L/root/bravobit/ffmpeg-android/toolchain-android/lib -Wl,-z,relro -Wl,-z,now -pie' --extra-cxxflags=
  
    libavutil      56. 14.100 / 56. 14.100
  
    libavcodec     58. 18.100 / 58. 18.100
  
    libavformat    58. 12.100 / 58. 12.100
  
    libavdevice    58.  3.100 / 58.  3.100
  
    libavfilter     7. 16.100 /  7. 16.100
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  1.100 /  5.  1.100
  
    libswresample   3.  1.100 /  3.  1.100
  
    libpostproc    55.  1.100 / 55.  1.100
  
  Unrecognized option 'i -y'.

  Which clearly states that it does not recognize option i -y, but also as I said earlier then if I remove the -y it never gets to onSuccess() or onFailure().

  help me to find a solution. Thanks.