That sounds like what I hear from the online-converted PCM file.
That is where you are going wrong. That makes FFmpeg think it’s a native G.729 file not wrapped in WAV. The output sounds exactly like the output I get from running your 80088.wav through my G.729 decoder, then importing its PCM raw output into Audacity using Import Raw Data.
Use this command instead in FFmpeg 3.x.x:
ffmpeg -i source.g729 dest.wav
Gale
Now I’ll be damned. That works!
I still can’t quite understand how but it actually works!
It’s not exactly how I wanted it to work (just load the file into audacity and be fine with it) but having it run on ffmpeg prior to editing under audacity is perfectly acceptable, given the singularities of these files.
Now…why does it work this way? I am basically telling ffmpeg that it’s a WAV file but in fact it is a G.729 file.
Here’s the output of ffmpeg 3.3.1:
C:\Users\rcorreia\Downloads>d:\users\rcorreia\downloads\ffmpeg-3.3.1-win64-share
d\bin\ffmpeg.exe -i 80088.wav 80088_2.wav
ffmpeg version 3.3.1 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 6.3.0 (GCC)
configuration: --disable-static --enable-shared --enable-gpl --enable-version3
--enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --
enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0
r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-lib
bs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --ena
ble-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb -
-enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-lib
opus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --en
able-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc -
-enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-
libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --en
able-lzma --enable-zlib
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
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, wav, from '80088.wav':
Duration: 00:11:10.08, bitrate: 8 kb/s
Stream #0:0: Audio: g729 ([131][0][0][0] / 0x0083), 8000 Hz, mono, s16, 8 kb
/s
Stream mapping:
Stream #0:0 -> #0:0 (g729 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to '80088_2.wav':
Metadata:
ISFT : Lavf57.71.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, mono, s16, 1
28 kb/s
Metadata:
encoder : Lavc57.89.100 pcm_s16le
size= 10470kB time=00:11:10.08 bitrate= 128.0kbits/s speed= 305x
video:0kB audio:10470kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.000728%
C:\Users\rcorreia\Downloads>
Now, ffmpeg clearly states that this is a G.729 stream.
My ffmpeg doesn’t say it is compliant with G.729, so how come this works?
TIA
Cheers
“-i 80088.wav” just gives FFmpeg the input file name. It does not tell FFmpeg anything about what audio format is in the WAV. Later FFmpeg can figure that out for G.729 wrapped in WAV. Older FFmpeg can’t, even though it supports decoding of proper G.729 files.
As we have said, you can run Audacity on Linux in a virtual machine. If you get for example Ubuntu 16.04 you may not have to compile Audacity - the Ubuntu-supplied version of Audacity may have a late enough version of FFmpeg built in so that it recognises your G.729 embedded in a WAV file.
Alternatively you can put your files in a directory and run FFmpeg as a batch command.
Can we close this topic now? Steve and I have spent hours supporting you on this. Would you like to make a donation to Audacity http://www.audacityteam.org/donate/?
Gale
Just to confirm, on Windows 10, FFmpeg 2.2.2 cannot read the file. The most recent nightly, ffmpeg-20170601, from the command line, can convert the file.
The command I used, from the location of ffmpeg.exe
./ffmpeg.exe -i "input file" "output file"
Where “input file” and “output file” were fully qualified file names and “output file” had a file extension “.wav”