[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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