Command Line export seems to be only 16 bits??

[Kiers_Thebad]

Hi,

Audacity doesn't seem to encode the audio in 32 bit wav, so I decided to use an external program with the following command:


C:UsersPublicDocumentsffmpeg.exe -i - -y -acodec pcm_s32le "%f.wav"



Which then gave the output:

C:UsersPublicDocumentsffmpeg.exe -i - -y -acodec pcm_s32le "C:Guitar1.wav"

ffmpeg version git-N-30698-g39dbe9b, Copyright (c) 2000-2011 the FFmpeg developers
built on Jun 10 2011 22:10:00 with gcc 4.5.3
configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib --disable-outdev=sdl --pkg-config=pkg-config
libavutil 51. 8. 0 / 51. 8. 0
libavcodec 53. 7. 0 / 53. 7. 0
libavformat 53. 3. 0 / 53. 3. 0
libavdevice 53. 1. 1 / 53. 1. 1
libavfilter 2. 15. 0 / 2. 15. 0
libswscale 0. 14. 1 / 0. 14. 1
libpostproc 51. 2. 0 / 51. 2. 0
[wav @ 01AA9DE0] max_analyze_duration 5000000 reached at 5015510
Input #0, wav, from 'pipe:':
Duration: 00:14:30.00, bitrate: N/A
Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
Incompatible sample format 's16' for codec 'pcm_s32le', auto-selecting format 's32'
Output #0, wav, to 'C:Guitar1.wav':
Metadata:
encoder : Lavf53.3.0
Stream #0.0: Audio: pcm_s32le, 44100 Hz, 2 channels, s32, 2822 kb/s
Stream mapping:
Stream #0.0 -> #0.0
size= 25824kB time=00:01:15.00 bitrate=2820.7kbits/s
size= 49952kB time=00:02:25.00 bitrate=2822.0kbits/s
size= 74080kB time=00:03:35.01 bitrate=2822.4kbits/s
size= 98176kB time=00:04:45.00 bitrate=2821.9kbits/s
size= 122304kB time=00:05:55.00 bitrate=2822.2kbits/s
size= 142976kB time=00:06:55.01 bitrate=2822.2kbits/s
size= 165376kB time=00:08:00.00 bitrate=2822.4kbits/s
size= 189472kB time=00:09:10.01 bitrate=2822.0kbits/s
size= 213600kB time=00:10:20.01 bitrate=2822.2kbits/s
size= 232544kB time=00:11:15.00 bitrate=2822.2kbits/s
size= 256448kB time=00:12:24.33 bitrate=2822.4kbits/s
size= 279072kB time=00:13:30.00 bitrate=2822.4kbits/s
size= 298016kB time=00:14:25.01 bitrate=2822.3kbits/s
size= 299743kB time=00:14:30.00 bitrate=2822.4kbits/s

video:0kB audio:299742kB global headers:0kB muxing overhead 0.000022%






Which seems to indicate that the input file only has a bit width of 16 bits?!?
Is this a bug or just how audacity was designed?





EDIT: The recording on audacity was 44100Hz 32bit I forgot to mention...

[Gale Andrews]

Audacity doesn't seem to encode the audio in 32 bit wav

Try selecting "Other uncompressed files" when you export, click "Options" then choose "Signed 32 bit PCM" with WAV encoding (or what you want).

so I decided to use an external program with the following command:


C:UsersPublicDocumentsffmpeg.exe -i - -y -acodec pcm_s32le "%f.wav"



Which then gave the output:

C:UsersPublicDocumentsffmpeg.exe -i - -y -acodec pcm_s32le "C:Guitar1.wav"

ffmpeg version git-N-30698-g39dbe9b, Copyright (c) 2000-2011 the FFmpeg developers
built on Jun 10 2011 22:10:00 with gcc 4.5.3
Input #0, wav, from 'pipe:':
Duration: 00:14:30.00, bitrate: N/A
Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
Incompatible sample format 's16' for codec 'pcm_s32le', auto-selecting format 's32'
Output #0, wav, to 'C:Guitar1.wav':
Stream #0.0: Audio: pcm_s32le, 44100 Hz, 2 channels, s32, 2822 kb/s

Which seems to indicate that the input file only has a bit width of 16 bits?!?
Is this a bug or just how audacity was designed?

Looks like you will have to use standalone FFmpeg at the command line (DOS window) if you want the 32-bit input preserved.

I am not sure what the "s16" means (maybe an FFmpeg bug), but I ran the following with standalone ffmpeg from 32-bit mono float input and saved as 32-bit float. The result:

C:Program Files (x86)FFmpeg 2009>ffmpeg -i 32f.wav -acodec pcm_f32le 32fc.wav
Input #0, wav, from '32f.wav':
Duration: 00:00:30.00, bitrate: 1411 kb/s
Stream #0.0: Audio: pcm_f32le, 44100 Hz, 1 channels, s16, 1411 kb/s
Output #0, wav, to '32fc.wav':
Stream #0.0: Audio: pcm_f32le, 44100 Hz, 1 channels, flt, 1411 kb/s

but in Audacity from the same 32-bit content:

ffmpeg -i - -acodec pcm_f32le "E:aud32f.wav"
Input #0, wav, from 'pipe:':
Duration: N/A, bitrate: 5644 kb/s
Stream #0.0: Audio: pcm_s16le, 44100 Hz, 1 channels, s16, 705 kb/s
Output #0, wav, to 'E:aud32f.wav':
Stream #0.0: Audio: pcm_f32le, 44100 Hz, 1 channels, flt, 1411 kb/s

You see clearly that the input is 16-bit in Audacity. The content was deliberately clipped so I could see if the data above 0 dB was being preserved (32-bit float preserves this). The clipped content was preserved in the file produced in standalone FFmpeg but lost in the 32-bit float file Audacity produced using FFmpeg.

I wasn't aware of this specific case but it is known that Audacity imports most files that can only be imported via FFmpeg at 16-bit resolution irrespective of a 32-bit float setting in Audacity's Quality Preferences. Yet if you import a 32-bit WAV using FFmpeg (by setting the filter in the import window to "FFmpeg-compatible files") it does come in as 32-bit.

EDIT: The recording on audacity was 44100Hz 32bit I forgot to mention...

Well it was probably 16-bit padded out to 32-bit as there are almost no 32-bit recording devices.



Gale

[Kiers_Thebad]

Thanks"