Can't Batch Encode Wav Files about axiom HOT 4 CLOSED

I've released an update v1.7.9.4, let me know if it fixes it.

Press the arrow button in Axiom to update through PowerShell, or download from here
https://github.com/MattMcManis/Axiom/releases

It looks like I needed to remove -b:a %A for wav. Leaving it blank lets PCM use the FFmpeg default, which I think is the max bit rate. You can still select the bit rate, bit depth, and sample rate manually if needed.

For reference, here's the script it generated that worked. I batch converted mp3 to wav.
(I'm using environment variable for FFmpeg so it just shows ffmpeg instead of the full path to the exe).

cd /d "C:\Users\Matt\Desktop\mp3\" 

&& for %f in (*) do (echo) 

&& 

ffmpeg -y 

-i "C:\Users\Matt\Desktop\mp3\%~f" 

-vn 
-sn 

-c:a pcm_s24le 
-map 0:a:0? 

-map_metadata 0 

-f wav 

-threads 7 

"C:\Users\Matt\Desktop\wav\%~nf.wav"

It works flawlessly converted all 159 files without an issue.

from axiom.

Hi, thanks.
Can you post the script that it generates?

from axiom.

Hi, thanks.
Can you post the script that it generates?

This happens on every file during converting

C:\Users\sjomm\Desktop\StreamTool-master\mcoutput>(echo) && "C:\Users\sjomm\Desktop\FFMpeg Gui\ffmpeg\bin\ffmpeg.exe" -y -i "C:\Users\sjomm\Desktop\StreamTool-master\mcoutput\voice_00.xma.wav" -vn -sn -c:a pcm_s16le -b:a %A -rematrix_maxval 1.0 -ac 2 -map 0:a:0? -map_metadata 0 -f wav -threads 0 "C:\Users\sjomm\Desktop\SH 2 NEW MIX\CONV\voice_00.xma.wav"
ECHO is on.
ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9.2.1 (GCC) 20200122
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
libavutil 56. 31.100 / 56. 31.100
libavcodec 58. 54.100 / 58. 54.100
libavformat 58. 29.100 / 58. 29.100
libavdevice 58. 8.100 / 58. 8.100
libavfilter 7. 57.100 / 7. 57.100
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.0 : 5.1
Input #0, wav, from 'C:\Users\sjomm\Desktop\StreamTool-master\mcoutput\voice_00.xma.wav':
Duration: 00:05:56.27, bitrate: 4608 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s16, 4608 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[pcm_s16le @ 0000021b572e30c0] [Eval @ 000000a6da5fdf70] Undefined constant or missing '(' in '%A'
[pcm_s16le @ 0000021b572e30c0] Unable to parse option value "%A"
[pcm_s16le @ 0000021b572e30c0] Error setting option b to value %A.
Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
Conversion failed!

And this is the script keep in mind i'm just letting the GUI handle everything. if you do the files one by one it does work.

cd /d "C:\Users\sjomm\Desktop\StreamTool-master\mcoutput"

&& for %f in (*) do (echo)

&&

"C:\Users\sjomm\Desktop\FFMpeg Gui\ffmpeg\bin\ffmpeg.exe" -y

-i "C:\Users\sjomm\Desktop\StreamTool-master\mcoutput%~f"

-vn
-sn

-c:a pcm_s16le
-b:a %A -rematrix_maxval 1.0 -ac 2
-map 0:a:0?

-map_metadata 0

-f wav

-threads 0

"C:\Users\sjomm\Desktop\SH 2 NEW MIX\CONV%~nf.wav"

from axiom.

I've released an update v1.7.9.4, let me know if it fixes it.

Press the arrow button in Axiom to update through PowerShell, or download from here
https://github.com/MattMcManis/Axiom/releases

It looks like I needed to remove -b:a %A for wav. Leaving it blank lets PCM use the FFmpeg default, which I think is the max bit rate. You can still select the bit rate, bit depth, and sample rate manually if needed.

For reference, here's the script it generated that worked. I batch converted mp3 to wav.
(I'm using environment variable for FFmpeg so it just shows ffmpeg instead of the full path to the exe).

cd /d "C:\Users\Matt\Desktop\mp3\" 

&& for %f in (*) do (echo) 

&& 

ffmpeg -y 

-i "C:\Users\Matt\Desktop\mp3\%~f" 

-vn 
-sn 

-c:a pcm_s24le 
-map 0:a:0? 

-map_metadata 0 

-f wav 

-threads 7 

"C:\Users\Matt\Desktop\wav\%~nf.wav"

from axiom.