Wanted: avs2wav

[kassandro]

Before we blame toolame, you should try first
lame --mp2input --decode my.mp3 | tooLame -s 48 -d n -c -o -e -b 224 - clip.mp2

If that doesn't work either, then it is a toolame problem.
In version 1.1 jcsston has added some other feature, which may cause problems (there is at least one report). Have you tried my version of avs2wav as well (the link can be found in this thread).

[matrix]

I'm sorry to bother you, but I don't get the command you posted. I just copy/paste your syntax, and I get an error from lame I belive, "cannot find my.mp3".
I tryed different ways, but can't get it to work. If I just take the name "my.mp3" ,I get something like:

Error reading headers in mp3 input file -.
Hit end of audio data
Avg slots/frame = -1.#IO; b/smp = -1.#J; bitrate = -1.#IO kbps

This is from avs2wav I guess.

I really appreciate your help, and I don't mean to be a pest, but I still need some help.

Regards.

[kassandro]

Originally posted by matrix
I'm sorry to bother you, but I don't get the command you posted. I just copy/paste your syntax, and I get an error from lame I belive, "cannot find my.mp3".
I tryed different ways, but can't get it to work. If I just take the name "my.mp3" ,I get something like:

Error reading headers in mp3 input file -.
Hit end of audio data
Avg slots/frame = -1.#IO; b/smp = -1.#J; bitrate = -1.#IO kbps

This is from avs2wav I guess.

I really appreciate your help, and I don't mean to be a pest, but I still need some help.

Regards.
"my.mp3" was only an arbitrary name for an mp3 file you have at hand. I simply wanted to test whether a wave file generated by lame causes the same problems as a wave file generated by avs2wav, when it is piped into toolame.
By the way, if you use something like

avs2wav.exe menu.avs -g1.5 -
tooLame -s 48 -d n -c -o -e -b 224 - clip.mp2

then the output of avs2wav is simply send to the DOS terminal, which is very messy and your pc may be blocked because it cannot cope with this enormous garbage flood.

[matrix]

"my.mp3" was only an arbitrary name for an mp3 file you have at hand
Yes I thought so.
I tryed many ways, but just doesn't work.
for ex.
lame --mp2input --decode one.mp3 - | tooLame -s 48 -d n -c -o -e -b 224 - one.mp2
is outputting an empty file. For some reason, it doesn't like it from stdin.
If instead, I save it to a file, and open that in toolame, it works.

Input File : 'one.wav' 44.1 kHz
Output File: 'one.mp2'
224 kbps MPEG-1 Layer II j-stereo Psy model 1
[De-emph:Off Copyright:Yes Original:Yes CRC:On]
[Padding:Normal Byte-swap:Off Chanswap:Off DAB:Off]
--------------------------------------------
Insufficient PCM input for one frame - fillout with zeros
Hit end of audio data
Avg slots/frame = 731.429; b/smp = 5.08; bitrate = 224.000 kbps

As you can see, it still gives me that insufficient pcm crap, but it creates the file.
Sorry, you mentioned it couple times, and I forgot to answer.
Yes, I did try your version. But it acts the same way.

I want to thank you again for your support.
And don't worry about it

Regards.

[oldchicken]

Im getting similar results as matrix.

@

kassandro

Could you post the syntax your using to output your avs2wav into oggenc, for some reason the pipe seems to be transposed. Oggenc keeps loading up first im using

avs2wav source.avs - | oggenc -b 350 -o audio.ogg

Im sure its something simple, please advice :)

btw im using xp with the latest patches could it be the culprit?

[State of Mind]

I'd like to use this application once you guys get all the quirks sorted out. I have 111 GBs of HD space and quite a bit of time on my hands, so I've never really desired to have an audio sampleserver (I wouldn't call it a frameserver as there aren't any "frames" in audio) and and I've never had a problem with having a large WAV file being written to my HD. However, it would be more convenient to have an audio sampleserver. Anyway, so a certain code would have to be placed in an AviSynth script in order to specify "serving" to an external audio encoder, wouldn't it? I have always used BeSweet to encode my DVD audio as AC3. I don't like processing audio and video at the same time because the lack of audio allows the media encoder to concentrate entirely on the video; something to do with bitrate control? Am I out on left field on this one? It's moreso what I am used to doing ever since I started making DVDs from my homemade MiniDV sources.
Anyway, so what are the current capabilities of this application? I've download AVS2WAV and the .exe is not even 10 KBs, which is cool. What can I do with it?

[kassandro]

Originally posted by oldchicken

Could you post the syntax your using to output your avs2wav into oggenc, for some reason the pipe seems to be transposed. Oggenc keeps loading up first im using

avs2wav source.avs - | oggenc -b 350 -o audio.ogg

Im sure its something simple, please advice :)

btw im using xp with the latest patches could it be the culprit?
First make sure that you have the right version of avs2wav. I personally use the one, for which I have posted a link earlier in this thread (it should still be valid). I then use the following batch file (in fact I associate it with AVS-scripts, such that I can launch it with a right click):

set _input_=%1
set _output_=%_input_:.avs=.ogg%
C:\programme\eac\avs2wav %_input_% - | C:\programme\eac\oggenc -Q -m 8 -M 160 -q 2 --resample 44100 -o %_output_% -
C:\programme\eac\ogginfo %_output_%
pause

This script converts any AVS script into an ogg vorbis file, where the ending ".avs" is replaced by ".ogg". Here is a typical example of such a script:

video = mpeg2source("input.d2v")
audio=MPASource("input.mpa", normalize=false)
AudioDub(video, audio)
trim(0,10000)

avs2wav is not fast and you have to be patient, whence I only use it if it is really necessary.

Originally posted by State of Mind

Anyway, so what are the current capabilities of this application? I've download AVS2WAV and the .exe is not even 10 KBs, which is cool. What can I do with it?
The main task of avs2wav is to extract the audio of an AVS script and convert it into a .wav file. Alternatively the output of av2wav can be piped into an encoder to avoid big .wav files. There are some additional features described in this thread.

[oldchicken]

Ok i tried using that bat file that your using and it doesnt seem to read the avs files for input, in fact avs2wav crashes in the process. Just curious what OS are you using?

[oldchicken]

Here is what i managed to get from the console


C:\Documents and Settings\frisky>set _input_=

C:\Documents and Settings\frisky>set _output_=.avs=.ogg

C:\Documents and Settings\frisky>C:\mj1\avs2wav - | C:\mj1\oggenc -Q -m 8 -M
160 -q 2 --resample 44100 -o .avs=.ogg -
ERROR: Input file "(stdin)" is not a supported format

C:\Documents and Settings\frisky>C:\mj1\ogginfo .avs=.ogg
Error opening input file ".avs=.ogg": No such file or directory

C:\Documents and Settings\frisky>pause
Press any key to continue . . .

[kassandro]

Originally posted by oldchicken
Here is what i managed to get from the console


C:\Documents and Settings\frisky>set _input_=

C:\Documents and Settings\frisky>set _output_=.avs=.ogg

C:\Documents and Settings\frisky>C:\mj1\avs2wav - | C:\mj1\oggenc -Q -m 8 -M
160 -q 2 --resample 44100 -o .avs=.ogg -
ERROR: Input file "(stdin)" is not a supported format

C:\Documents and Settings\frisky>C:\mj1\ogginfo .avs=.ogg
Error opening input file ".avs=.ogg": No such file or directory

C:\Documents and Settings\frisky>pause
Press any key to continue . . .
Something must have gone here. All the "%", which are use to denote variables in batch files, are lost, which then results in nonsense.
Let's call my script avs2ogg (that's actually the name I use for it). If you run "avs2ogg my.avs", then the first two lines should look as follows:

C:\Documents and Settings\frisky>set _input_=my.avs

C:\Documents and Settings\frisky>set _output_=my.ogg

What system do you have? I have XP with service pack 1.

[oldchicken]

kassandro

I have managed to get it working from the command line with ogg2wav source.avs, unfortunately i get this wonderful message

Frame Buffer was too small for AVIStreamRead.

Then it resorts to creating a nice juicy wav for me, which brings great misery and tears of sadness (drama ack drama) at the end of the day i end up with a big wav and no ogg file what to do?

Also I have however been unable to get it to work via a right click when i try that then i get no action at all and those are the results i posted previously

By the way i am using xp with pack 2

[kassandro]

Originally posted by oldchicken
kassandro

I have managed to get it working from the command line with ogg2wav source.avs, unfortunately i get this wonderful message

Frame Buffer was too small for AVIStreamRead.

Then it resorts to creating a nice juicy wav for me, which brings great misery and tears of sadness (drama ack drama) at the end of the day i end up with a big wav and no ogg file what to do?

Also I have however been unable to get it to work via a right click when i try that then i get no action at all and those are the results i posted previously

By the way i am using xp with pack 2

You probably have a too old version of avs2wav. The version I use can be downloaded from here (http://home.pages.at/kassandro/avs2wav.rar).

[oldchicken]

Yes sir you are correct looks like i was using an old version. I downloaded so many versions i guess i ended up confusing myself. Thanks for the assist, i finally got it working now :)

Only thing is i cant figure out how you pulled off the nifty trick of being able to do it with a right click. I have it setup in xp as an option to open avs files with the bat but i get no love. Any suggestions there?

Thanks again for your help! :D

[raymod2]

You probably have a too old version of avs2wav. The version I use can be downloaded from here (http://home.pages.at/kassandro/avs2wav.rar).

@kassandro: I just noticed a problem with avs2wav. When I use the following command I get a pop at the end of the clip:

avs2wav filter.avs - | faac -o audio.mp4 -

But if I use a 2-step process with an intermediate WAV file I do not get the pop:

avs2wav filter.avs audio.wav
faac -o audio.mp4 audio.wav

Is it possible some extraneous data is getting passed through standard input that doesn't belong in the output file?

[tebasuna51]

avs2wav don't work for me.
I test avs2wav.rar, from kassandro's post, who say me:
"Don't found MSVCP70.dll"
After search and put MSVCP70.dll I get:
avs2wav v1.0 by Jory Stone <jcsston@toughguy.net>
Input: d:\Programa\Audio\0\Audio.avs
Output: -
Scanning for Audio Stream...
Found Audio Stream
Using stdout for WAV Output

Writing WAV Headers...
Scanning...
And I am still waiting the end.
Using Ctrl-C to stop the program I have a file with the correct DataLength expected but without data.

I found avs2wav-v1.1.zip, this version say me:
avs2wav v1.1 by Jory Stone <jcsston@toughguy.net>, stdout patch by kassandro
Input: d:\Programa\Audio\0\Audio.avs
Output: -
Scanning for Audio Stream...
Found Audio Stream
Using stdout for WAV Output

Writing WAV Headers...
Gaining with a factor of 1.0000 ...
Took 0.00 seconds for the writing pass
And end, but the output file is always empty: a wav header with 0 data and DataLength = 0.

Bepipe works without problems for me.

[MatMaul]

same problem than tebasuna51 for me

[vlada]

I downloaded AVS2WAV from http://home.pages.at/kassandro/avs2wav.rar and then I downloaded msvcr70.dll and MSVCP70.DLL libraries. And I created AC3 without any problems using following command
avs2wav.exe "input.avs" - | aften.exe -b 224 -s 1 -m 1 -w 45 -dnorm 27 - "output.ac3"

[Pookie]

Look for sh0dan's new SoundOut plugin - a cool replacement for the hit and miss avs2wav.exe and the .net reliant Bepipe.exe

http://forum.doom9.org/showthread.php?t=120025

[tebasuna51]

I downloaded AVS2WAV from http://home.pages.at/kassandro/avs2wav.rar and then I downloaded msvcr70.dll and MSVCP70.DLL libraries. And I created AC3 without any problems using following command
avs2wav.exe "input.avs" - | aften.exe -b 224 -s 1 -m 1 -w 45 -dnorm 27 - "output.ac3"

Downloaded also msvcr70.dll and using kassandro/avs2wav.rar (v1.0) the process never end, using v1.1 empty output.

@Pookie, I also wait the sh0dan's new SoundOut plugin.

[LoRd_MuldeR]

Sorry for digging out an old thread, but I recently made a stripped-down and clean-up version of avs2wav (http://home.pages.at/kassandro/avs2wav.rar), which can be found here:
http://code.google.com/p/mulder/source/browse/trunk/Utils/avs2wav#avs2wav%2Favs2wav

It now also supports Unicode filenames. Well, at least in theory. For some reason AVIFileOpenW() will succeed with Unicode file names (even if the file does not exist), but will then fail to find any streams (even if the file does exist). At least this was my experience on Windows 7. Nonetheless Unicode names for the output Wave are no problem...

BTW: I also had a look at the "native" Avisynth API, which I would prefer over the VFW interface, but it seems there is no Unicode support at all. Or does it handle UTF-8 strings? :confused:


[UPDATES]

2011-05-22: Now using GetShortPathName() internally, which should solve the problem with Unicode file names most of the time.
2011-05-24: Now the application throws a warning, if the Wave file could not be finalized correctly or if the final size exceeds the 4 GB limit.

[Gavino]

I also had a look at the "native" Avisynth API, which I would prefer over the VFW interface, but it seems there is no Unicode support at all. Or does it handle UTF-8 strings? :confused:
See the thread (ongoing over the last 5 years, and still not fully resolved)
Foreign Language characters in filenames (http://forum.doom9.org/showthread.php?t=110467)