Converting a call center recording to something us

2019-07-25 10:26发布

问题:

I have a call center recording (when played it sounds gibberish) for which the mediainfo shows info as

ion@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%)

The ffmpeg info shows this as

ion@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

So I converted this file to PCM using

ffmpeg -acodec g723_1 -i 48401-3405-48403--18042018170000.wav -acodec pcm_s16le -f wav outnew1.wav

But the audio still sound gibberish , I tried many variation and only Goldwave worked but that works on windows and with GUI not cli.

So how can I convert this file to something useful so that atleast I can listen to it , It feels like a challenge now.

Audio file : https://drive.google.com/open?id=1T54lKaI6IJmOqTPNOA_OkYRz89EQ5F2L

PS : Use VLC to play audio file