[FFmpeg-user] Task for all FFMPEG developers

Ryan Adlob ryanadlob at gmail.com
Thu Jun 21 13:55:49 EEST 2018


Hi,

I know it isn't right to send such a mail to you as it violates the meaning
of this channel but I am in dire need for your support.

The issue I have is that I have some audio files that are provided by a
call recording agency for which the media info shows info as

ion at aurora:~/Inbound$ mediainfo 48401-3405-48403--18042018170000.wav
General
Complete name                            : 48401-3405-48403--18042018170000.wav
Format                                   : Wave
File size                                : 327 KiB
Duration                                 : 4mn 11s
Overall bit rate                         : 10.7 Kbps

Audio
Format                                   : G.723.1
Codec ID                                 : A100
Duration                                 : 4mn 11s
Bit rate                                 : 10.7 Kbps
Channel(s)                               : 2 channels
Sampling rate                            : 8 000 Hz
Stream size                              : 327 KiB (100%)

and the ffmpeg shows info as

ion at aurora:~/Inbound$ ffmpeg -i 48401-3405-48403--18042018170000.wav
ffmpeg version N-91330-ga990184 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.9) 20160609
  configuration: --prefix=/home/ion/ffmpeg_build
--pkg-config-flags=--static
--extra-cflags=-I/home/ion/ffmpeg_build/include
--extra-ldflags=-L/home/ion/ffmpeg_build/lib --extra-libs='-lpthread
-lm' --bindir=/home/ion/bin --enable-gpl --enable-libaom
--enable-libass --enable-libfdk-aac --enable-libfreetype
--enable-libmp3lame --enable-libopus --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
  libavutil      56. 18.102 / 56. 18.102
  libavcodec     58. 20.103 / 58. 20.103
  libavformat    58. 17.100 / 58. 17.100
  libavdevice    58.  4.101 / 58.  4.101
  libavfilter     7. 25.100 /  7. 25.100
  libswscale      5.  2.100 /  5.  2.100
  libswresample   3.  2.100 /  3.  2.100
  libpostproc    55.  2.100 / 55.  2.100
Input #0, wav, from '48401-3405-48403--18042018170000.wav':
  Duration: 00:04:11.37, 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

When I play with VLC the whole audio sounds gibberish , What I want to do
is to decode this thing and make the audio understandable when its played.
So as of now I have found a solution where I open the file normally in
Goldwave and save it as wav 16bit PCM and then it plays normally but what I
wanna do is , I want FFMPEG to solve this issue.

So Can you guys please help  me out (No one on stackoverflow or forums
seems to have found the solution)

PS : The file is attached as google drive
https://drive.google.com/open?id=1T54lKaI6IJmOqTPNOA_OkYRz89EQ5F2L

Thanks


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