VirtualDub External Encoder Issues

[Shoal]

Hi all! I'm trying to get an external encoder in VirtualDub to work, but something is going wrong with the audio encoding using SoX. See the attached vdprof file which outputs MKV, h264 (x264), and WAV. The log keeps giving me the error message "cannot open test.mkv.wav". (I was naming the output file "test".) How can I fix this?

The goal is to output a 48kHz, 16-bit, stereo, PCM WAV file. I don't necessarily need to use SoX.

Any help would be greatly appreciated, thanks!

[sneaker]

Use this for SoX command (you cannot have input and output file name the same, VirtualDub sends using pipe):
-t wav - -t wav "%(tempaudiofile)"

In x264 command, change this for automatic resolution/fps:
--input-res %(width)x%(height) --fps %(fpsnum)/%(fpsden)

[Shoal]

It worked! Thanks! Though, honestly, I'm not sure how you came up with the new command line for SoX. Granted, I know very little about SoX. What does each part of it do? More specifically, why the extra hyphen, and what does "-t" mean?

[sneaker]

The first part ("-t wav -") tells SoX about the input options:
"-t wav" means it should expect wave format
"-" means it should expect input via pipe (not from a file). This is the same for x264, btw.

The second part are of course the output options:
"-t wav" means it should output wave format
"%(tempaudiofile)" is the file name of the output (the variable is filled out by VirtualDub automatically)

[Shoal]

Interesting, thanks! I would have never figured out the extra hyphen thing on my own!

[DarrellS]

Most command line encoders have a help file that lists all the commands. It was recommended to keep all encoders in C:\Tools along with a copy of cmd.exe. Then you can run commands from inside the folder and also makes it easier for users to exchange vdprof files without having to change the addresses of all the encoders.

Examples: double click cmd.exe an type...

x264.exe -h (for help file)
x264.exe --help (for long help file)

sox.exe -h

Microsoft Windows XP [Version 6.1.7601]
(C) Copyright 1985-2001 Microsoft Corp.

C:\Tools>sox.exe -h
sox.exe:      SoX v14.4.2

Usage summary: [gopts] [[fopts] infile]... [fopts] outfile [effect [effopt]]...

SPECIAL FILENAMES (infile, outfile):
-                        Pipe/redirect input/output (stdin/stdout); may need -t
-d, --default-device     Use the default audio device (where available)
-n, --null               Use the `null' file handler; e.g. with synth effect
-p, --sox-pipe           Alias for `-t sox -'

SPECIAL FILENAMES (infile only):
"|program [options] ..." Pipe input from external program (where supported)
http://server/file       Use the given URL as input file (where supported)

GLOBAL OPTIONS (gopts) (can be specified at any point before the first effect):
--buffer BYTES           Set the size of all processing buffers (default 8192)
--clobber                Don't prompt to overwrite output file (default)
--combine concatenate    Concatenate all input files (default for sox, rec)
--combine sequence       Sequence all input files (default for play)
-D, --no-dither          Don't dither automatically
--dft-min NUM            Minimum size (log2) for DFT processing (default 10)
--effects-file FILENAME  File containing effects and options
-G, --guard              Use temporary files to guard against clipping
-h, --help               Display version number and usage information
--help-effect NAME       Show usage of effect NAME, or NAME=all for all
--help-format NAME       Show info on format NAME, or NAME=all for all
--i, --info              Behave as soxi(1)
--input-buffer BYTES     Override the input buffer size (default: as --buffer)
--no-clobber             Prompt to overwrite output file
-m, --combine mix        Mix multiple input files (instead of concatenating)
--combine mix-power      Mix to equal power (instead of concatenating)
-M, --combine merge      Merge multiple input files (instead of concatenating)
--magic                  Use `magic' file-type detection
--multi-threaded         Enable parallel effects channels processing
--norm                   Guard (see --guard) & normalise
--play-rate-arg ARG      Default `rate' argument for auto-resample with `play'
--plot gnuplot|octave    Generate script to plot response of filter effect
-q, --no-show-progress   Run in quiet mode; opposite of -S
--replay-gain track|album|off  Default: off (sox, rec), track (play)
-R                       Use default random numbers (same on each run of SoX)
-S, --show-progress      Display progress while processing audio data
--single-threaded        Disable parallel effects channels processing
--temp DIRECTORY         Specify the directory to use for temporary files
-T, --combine multiply   Multiply samples of corresponding channels from all
                         input files (instead of concatenating)
--version                Display version number of SoX and exit
-V[LEVEL]                Increment or set verbosity level (default 2); levels:
                           1: failure messages
                           2: warnings
                           3: details of processing
                           4-6: increasing levels of debug messages
FORMAT OPTIONS (fopts):
Input file format options need only be supplied for files that are headerless.
Output files will have the same format as the input file where possible and not
overridden by any of various means including providing output format options.

-v|--volume FACTOR       Input file volume adjustment factor (real number)
--ignore-length          Ignore input file length given in header; read to EOF
-t|--type FILETYPE       File type of audio
-e|--encoding ENCODING   Set encoding (ENCODING may be one of signed-integer,
                         unsigned-integer, floating-point, mu-law, a-law,
                         ima-adpcm, ms-adpcm, gsm-full-rate)
-b|--bits BITS           Encoded sample size in bits
-N|--reverse-nibbles     Encoded nibble-order
-X|--reverse-bits        Encoded bit-order
--endian little|big|swap Encoded byte-order; swap means opposite to default
-L/-B/-x                 Short options for the above
-c|--channels CHANNELS   Number of channels of audio data; e.g. 2 = stereo
-r|--rate RATE           Sample rate of audio
-C|--compression FACTOR  Compression factor for output format
--add-comment TEXT       Append output file comment
--comment TEXT           Specify comment text for the output file
--comment-file FILENAME  File containing comment text for the output file
--no-glob                Don't `glob' wildcard match the following filename

AUDIO FILE FORMATS: 8svx aif aifc aiff aiffc al amb amr-nb amr-wb anb au avr awb caf cdda cdr cvs cvsd cvu dat dvms f32
f4 f64 f8 fap flac fssd gsm gsrt hcom htk ima ircam la lpc lpc10 lu mat mat4 mat5 maud mp2 mp3 nist ogg paf prc pvf raw
s1 s16 s2 s24 s3 s32 s4 s8 sb sd2 sds sf sl sln smp snd sndfile sndr sndt sou sox sph sw txw u1 u16 u2 u24 u3 u32 u4 u8
ub ul uw vms voc vorbis vox w64 wav wavpcm wv wve xa xi
PLAYLIST FORMATS: m3u pls
AUDIO DEVICE DRIVERS: waveaudio

EFFECTS: allpass band bandpass bandreject bass bend biquad chorus channels compand contrast dcshift deemph delay dither
divide+ downsample earwax echo echos equalizer fade fir firfit+ flanger gain highpass hilbert input# loudness lowpass mc
ompand noiseprof noisered norm oops output# overdrive pad phaser pitch rate remix repeat reverb reverse riaa silence sin
c spectrogram speed splice stat stats stretch swap synth tempo treble tremolo trim upsample vad vol
  * Deprecated effect    + Experimental effect    # LibSoX-only effect
EFFECT OPTIONS (effopts): effect dependent; see --help-effect

C:\Tools>

[jairovital]

The first part ("-t wav -") tells SoX about the input options:
"-t wav" means it should expect wave format
"-" means it should expect input via pipe (not from a file). This is the same for x264, btw.

The second part are of course the output options:
"-t wav" means it should output wave format
"%(tempaudiofile)" is the file name of the output (the variable is filled out by VirtualDub automatically)

Good explannation.

I'm having a similar problem when I try to encode using aec3to to encode audio, instead of neroaacenc.

When using neroaacenc, my Command Arguments are:

-q 0.15 -br 56 -normalize -ignorelength -if - -of "%(tempaudiofile)"

Input and output files are at the end of the line.

If I use eac3to with the same line, I get an error:

Please specify the source and dest files first and then the options.  <ERROR>

So, following your tip, I change to:

-if - -of "%(tempaudiofile)" -q 0.15 -br 56 -normalize -ignorelength

...but still getting the same error:

Please specify the source and dest files first and then the options.  <ERROR>

At eac3to log file, I see that it puts output name correctly, but no input file name:

eac3to v3.31
command line: "B:\Programs\StaxRip\Apps\eac3to\eac3to.exe" -if - -of "D:\tests\sid.mkv.audio.aac" -q 0.15 -br 56 -normalize -ignorelength

Using Staxrip, eac3to Argument line is:

eac3to -quality=0.35 -normalize -resampleTo22050

without error.


What am I doing wrong?

[sneaker]

eac3to does not support input via pipe thus is not suited to be used with VirtualDub's "external encoders" feature.(*)

I recommend qaac encoder. In tests it was best aac encoder and also has some features like for example normalization.




((*) Maybe there are some tricks like writing special batch file but let's keep it simple..)

[raffriff42]

-- delete --

[jairovital]

eac3to does not support input via pipe thus is not suited to be used with VirtualDub's "external encoders" feature.(*)

I recommend qaac encoder. In tests it was best aac encoder and also has some features like for example normalization.




((*) Maybe there are some tricks like writing special batch file but let's keep it simple..)

Thank you, sneaker.

After too many attempts to use eac3to without success, something was wrong and you came up in a good time.
Yes, I do want normalization.
Why neroaacenc won't normalize?

I'll try qaac encoder and put here my results.

[jairovital]

I'm having troubles to match the correct command line...
May you, please, point me the right one?

I tried:

-q 0.15 -normalize -i - -o "%(tempaudiofile)"

and

-q 0.15 -normalize - -o "%(tempaudiofile)"

and got this:

The audio encoding proccess has prematurely exited with an error code of 2 (0000002).

Had found no log file after all; only a message window.

And how to encode using 56 kb bitrate?
this option?:

-c 56

[raffriff42]

You are using a neroaac command. No good with qaac.

User Nologic has kindly uploaded a "vdprof" with a working qaac encoder and posted it here. Vdprofs (*.vdprof, a special text file with encoder settings) are the best way to get external encoders working quickly. They are explained here:

https://forum.doom9.org/showthread.php?t=171141

TL;DR
1. Search the Web for .vdprof files (external encoder sets) for the file type you need.
2. Go to Options, External Encoders, Import and import the .vdprof file.
3. Open each imported encoder or muxer and fix the executable path if needed.

The qaac command Nologic used was:
--tvbr 91 -q 2 --threading -i -o "%(tempaudiofile)" -

The commands for qaac are here. I see normalizing requires two passes. To get 2-pass to work with VirtualDub external encoders requires advanced skills.

I have the skills but not the inclination. If I wanted to normalize I would process the audio separately with another program and bring the result into VirtualDub with Audio from another file...

[jairovital]

Thank you, raffriff42.

Actually, I already do what you said about process normalize separately.
But if it was possible to process in the same time at External Encoder, it would better.

I'll try your tips and post results later.

[jairovital]

When I start to make questions, I was just looking for a way to normalize audio in External Encoders, just like Staxrip does.
But I see now, it's not that simple.

User Nologic has kindly uploaded a "vdprof" with a working qaac encoder and posted it here. Vdprofs (*.vdprof, a special text file with encoder settings) are the best way to get external encoders working quickly. They are explained here:

https://forum.doom9.org/showthread.php?t=171141

Nologic profiles are very good. He also put batch and bin files in the pack.
Great help.
But there is no "normalize" command among all settings...

The qaac command Nologic used was:
--tvbr 91 -q 2 --threading -i -o "%(tempaudiofile)" -

This command is driving to the same error:

The audio encoding proccess has prematurely exited with an error code of 2 (0000002).

The commands for qaac are here. I see normalizing requires two passes. To get 2-pass to work with VirtualDub external encoders requires advanced skills.

This link is outdated.
Latest version is 2.64.

Here a comparison.

I have the skills but not the inclination. If I wanted to normalize I would process the audio separately with another program and bring the result into VirtualDub with Audio from another file...

You're right.
Going back to normalize audio with SoundForge before Vdub actions.
Or Staxrip's way.


You gave me an enormous amount of references to read and learn about.
Thanks.
I'll try to read them.

Meanwhile, if you know an easy way to normalize audio with External Encoders, let's hear that.

[sneaker]

The commands for qaac are here. I see normalizing requires two passes. To get 2-pass to work with VirtualDub external encoders requires advanced skills.

qaac will create a temp file for this so it should be possible to use normalization even with pipe. Of course that needs extra space (you can set --tmpdir) and time.

[sneaker]

The qaac command Nologic used was:
--tvbr 91 -q 2 --threading -i -o "%(tempaudiofile)" -

This command is driving to the same error:

The audio encoding proccess has prematurely exited with an error code of 2 (0000002).

In the audio encoder options there is a second tab. Set VirtualDub to send "WAV" there.

If there are still problems look into VirtualDub's log (F8). (Maybe need to redirect stdout to log.)

[raffriff42]

jairovital, you are welcome, it's a pleasure dealing with you. I think you're ready...

((*) Maybe there are some tricks like writing special batch file but let's keep it simple..)

...for batch files. They're a little harder to set up and to use, but there is a big advantage: you get a full error log rather than the deaded "process has prematurely exited" message.

I believe DarrellS (who posted earlier) goes back as far and knows at least as much about vdprofs and batch files as I do. Budman1 too. One, or maybe both of them figured out how to do 2-pass video*, and if you can find and examine their batch files, you can probably find a way to do 2-pass audio.

*EDIT so did I, which I forgot about because I never use it - see here. But I think there are better solutions out there...

[sneaker]

They're a little harder to set up and to use, but there is a big advantage: you get a full error log rather than the deaded "process has prematurely exited" message.

You can also read the errors in VirtualDub's log. By default it only redirects stderr there but you can also set to redirect stdout. It depends on the encoding software which one you "need".

[raffriff42]

You can also read the errors in VirtualDub's log. By default it only redirects stderr there but you can also set to redirect stdout.

I did not know that! I suppose you have to use the command-line VirtualDub, vdub.exe or vdub64.exe. The GUI log (last time I checked) truncates at around 300 lines, so earlier events get lost.

[sneaker]

I didn't know it truncates. Is this a common problem? Probably not for x264/qaac/mkvmerge. Maybe ffmpeg with lots of warnings/errors.

[raffriff42]

That was my whole motivation for making batch files - some dang program filling up the log with progress messages. I think it was ffmpeg, yes. (The limit seems to be 600 lines actually, which is not a lot when each event takes a minimum of 3 lines)

[sneaker]

FWIW: ffmpeg progress can be turned off with -nostats. And other messages can be silenced by choosing a different verbosity level.

[jairovital]

In the audio encoder options there is a second tab. Set VirtualDub to send "WAV" there.

If there are still problems look into VirtualDub's log (F8). (Maybe need to redirect stdout to log.)

WAV radiobutton is already checked.
But still getting the same error.

Vdub log (F8) says the same error message:

[E] Error: The audio encoding process has prematurely exited with an error 
    code of 2 (00000002). Check the log for possible error messages.

Using -log argment, log file said that:

ERROR: CoreAudioToolbox.dll: No such module found.

When copying qaac264 to path, there is no CoreAudioToolbox.dll.
Just that:
libsoxconvolver64.dll
libsoxr64.dll
qaac64.exe
refalac64.exe

[jairovital]

Looking for CoreAudioToolbox.dll at Google, I found this.
Put it in the same qaac folder, but error now is code 00001:

The audio encoding process has prematurely exited with an error code of 1 (00000001)

and then

Write error occurred on file "<anonymous pipe>"

[sneaker]

You need certain dlls from Apple.

1. Download makeportable.zip
2. Download iTunes (64) installer (do not install)
3. Download and install or use portable 7zip
4. Put "makeportable.cmd", iTunes installer (and 7zip portable exe) into same folder and run "makeportable.cmd"
5. Put all files from "QTfiles" into same folder as "qaac.exe" (or all files from "QTfiles64" into same folder as "qaac64.exe")

[raffriff42]

regarding <anonymous pipe>, double check your hyphens. They mean the anonymous pipes stdin or stdout. For example,

--tvbr 91 -q 2 --threading -i -o "%(tempaudiofile)" -
...............................................this ^ is really important

[Bruce Banner]

You are using a neroaac command. No good with qaac.

User Nologic has kindly uploaded a "vdprof" with a working qaac encoder and posted it here. Vdprofs (*.vdprof, a special text file with encoder settings) are the best way to get external encoders working quickly. They are explained here:

https://forum.doom9.org/showthread.php?t=171141

TL;DR
1. Search the Web for .vdprof files (external encoder sets) for the file type you need.
2. Go to Options, External Encoders, Import and import the .vdprof file.
3. Open each imported encoder or muxer and fix the executable path if needed.

The qaac command Nologic used was:
--tvbr 91 -q 2 --threading -i -o "%(tempaudiofile)" -

The commands for qaac are here. I see normalizing requires two passes. To get 2-pass to work with VirtualDub external encoders requires advanced skills.

I have the skills but not the inclination. If I wanted to normalize I would process the audio separately with another program and bring the result into VirtualDub with Audio from another file...

Hi. I just downloaded that vdprof file but when I tried importing it, I got the following error message:

The file contains an external encoder set called "Animated GIF" that already exists.

What should I do?

EDIT: I figured it out. I just highlighted the Animated GIF encoder set in VirtualDub and removed it. But now I'm having another problem. I tried creating an mp4 and got an error message. Here's the log.
[*] Beginning preview operation.

[i] Dub: Recompressing using format: RGB888.
[*] Ending operation.
[*] Beginning dub operation.

[i] Dub: Input (decompression) format is: YUV420.

[i] Dub: Output (compression) format is: YUV420.

[i] AudioEnc: qaac 2.47, CoreAudioToolbox 7.9.8.3

[i] AudioEnc: test.mp4.Audio.mp4

[i] AudioEnc: AAC-LC Encoder, TVBR q91, Quality 96

[i] AudioEnc: 0:00.960 (0.9x)

[i] AudioEnc: 0:01.000 (0.9x)

[i] AudioEnc: 48048/-1 samples processed in 0:01.078

[i] AudioEnc: Overall bitrate: 88.3316kbps

[i] AudioEnc: Optimizing...done

[i] Mux: AVC-H264 import - frame size 1432 x 1076 at 23.976 FPS

[i] Mux: Importing AVC-H264: |==== | (20/100)

[i] Mux: Importing AVC-H264: |===== | (28/100)

[i] Mux: Importing AVC-H264: |====== | (30/100)

[i] Mux: Importing AVC-H264: |====== | (32/100)

[i] Mux: Importing AVC-H264: |====== | (34/100)

[i] Mux: Importing AVC-H264: |======== | (42/100)

[i] Mux: Importing AVC-H264: |======== | (44/100)

[i] Mux: Importing AVC-H264: |========= | (47/100)

[i] Mux: Importing AVC-H264: |========= | (48/100)

[i] Mux: Importing AVC-H264: |=========== | (57/100)

[i] Mux: Importing AVC-H264: |=========== | (59/100)

[i] Mux: Importing AVC-H264: |============ | (62/100)

[i] Mux: Importing AVC-H264: |============ | (63/100)

[i] Mux: Importing AVC-H264: |============== | (73/100)

[i] Mux: Importing AVC-H264: |=============== | (76/100)

[i] Mux: Importing AVC-H264: |=============== | (79/100)

[i] Mux: Importing AVC-H264: |================ | (81/100)

[i] Mux: Importing AVC-H264: |================= | (88/100)

[i] Mux: Importing AVC-H264: |================= | (89/100)

[i] Mux: Importing AVC-H264: |================== | (92/100)

[i] Mux: Importing AVC-H264: |================== | (93/100)

[i] Mux: Importing AVC-H264: |=================== | (96/100)

[i] Mux: Importing AVC-H264: |=================== | (98/100)

[i] Mux: Importing AVC-H264: |=================== | (95/100)

[i] Mux:

[i] Mux: AVC Import results: 24 samples - Slices: 1 I 6 P 17 B - 1 SEI - 1 IDR

[i] Mux: Stream uses forward prediction - stream CTS offset: 2 frames

[i] Mux: [Importer] Unknown input file type for "test.mp4.m4a"

[i] Mux: Error importing test.mp4.m4a: Requested URL is not valid or cannot be
found

[E] Error: CLI: The multiplexing process failed with error code 1 (00000001).
Check the log for possible error messages.
[*] Ending operation.

[raffriff42]

from the log:
[i] Mux: [Importer] Unknown input file type for "test.mp4.m4a"

Check your audio encoder output file (example "%(outputname).Audio.mp3", but it can vary; the extension has to match the content of course) and your muxer input (should be "%(tempaudiofile)" with no added extension)

[Bruce Banner]

from the log:
[i] Mux: [Importer] Unknown input file type for "test.mp4.m4a"

Check your audio encoder output file (example "%(outputname).Audio.mp3", but it can vary; the extension has to match the content of course) and your muxer input (should be "%(tempaudiofile)" with no added extension)

It's

%(outputname).Audio.mp4

When you say the extension has to match the content, what do you mean by the content?

My command arguments are

--tvbr 91 -q 2 --threading -i -o "%(tempaudiofile)" -

[raffriff42]

One common issue is you have to enter the correct file extension in the final Save As dialog. It isn't filled in automatically, not correctly anyway.

Check the muxer settings just to be sure it's looking for "%(tempaudiofile)" and not something else.

>When you say the extension has to match the content, what do you mean by the content?
Just reminding people not to use .mp3 from my example, unless that's what the file really is.

Quoting myself from here,

'Output filename' is the filename of the temporary file that the encoder creates. By default it will be %(outputname).audio or %(outputname).video, but it is recommended to change '.audio' and '.video' to the correct extension as some multiplexers rely on extensions for file type detection.

Here's the log with highlights added, and with the redundant progress messages removed. The name seems correct at the audio encoder, but completely different at the muxer:

[*] Beginning dub operation.

[i] Dub: Input (decompression) format is: YUV420.

[i] Dub: Output (compression) format is: YUV420.

[i] AudioEnc: qaac 2.47, CoreAudioToolbox 7.9.8.3

[i] AudioEnc: test.mp4.Audio.mp4

[i] AudioEnc: AAC-LC Encoder, TVBR q91, Quality 96

[i] AudioEnc: 0:00.960 (0.9x)

[i] AudioEnc: 0:01.000 (0.9x)

[i] AudioEnc: 48048/-1 samples processed in 0:01.078

[i] AudioEnc: Overall bitrate: 88.3316kbps

[i] AudioEnc: Optimizing...done

[i] Mux: AVC-H264 import - frame size 1432 x 1076 at 23.976 FPS

[i] Mux: Importing AVC-H264: |==== | (20/100)



[i] Mux: Importing AVC-H264: |=================== | (96/100)

[i] Mux: Importing AVC-H264: |=================== | (98/100)

[i] Mux: Importing AVC-H264: |=================== | (95/100)

[i] Mux:

[i] Mux: AVC Import results: 24 samples - Slices: 1 I 6 P 17 B - 1 SEI - 1 IDR

[i] Mux: Stream uses forward prediction - stream CTS offset: 2 frames

[i] Mux: [Importer] Unknown input file type for "test.mp4.m4a"

[i] Mux: Error importing test.mp4.m4a: Requested URL is not valid or cannot be
found

[E] Error: CLI: The multiplexing process failed with error code 1 (00000001).
Check the log for possible error messages.[*] Ending operation.