Aften 0.0.8 is out

[canuckerfan]

does encwavtoac3 support stereo wav input or do I need to demux the stereo wav into 2 mono wav files?

[Boulder]

No need for that, it supports stereo WAVs just fine.

[canuckerfan]

but when I hit encode it gives me an error saying "Supported are minimum 2 and maximum 6 mono input files!"

[Boulder]

I just tried encoding a 32-bit stereo WAV with the latest version, no problems here.

[canuckerfan]

^maybe its my file... I tried the first file on this webpage ( M1F1-Alaw-AFsp.wav): http://www-mmsp.ece.mcgill.ca/Docume...E/Samples.html

and I got the same error. does that one work for you?

ps: the second one also gives me the same error(M1F1-AlawWE-AFsp.wav). somethings definately wrong with my setup. i am running v4.4 from the zipped package

[Boulder]

I think it's a problem with the encoding type in the WAV file.. try some "regular", i.e. PCM encoded WAV. For example this one is such: http://www-mmsp.ece.mcgill.ca/Docume...int16-AFsp.wav

[wisodev]

Originally Posted by canuckerfan

does encwavtoac3 support stereo wav input or do I need to demux the stereo wav into 2 mono wav files?

When you are using "Mmultiple mono input" mode the input files must be mono, if you turn off this mode then you can encode single multichannel input file.

[canuckerfan]

Originally Posted by Boulder

I think it's a problem with the encoding type in the WAV file.. try some "regular", i.e. PCM encoded WAV. For example this one is such: http://www-mmsp.ece.mcgill.ca/Docume...int16-AFsp.wav

I get the same error with that file as well.

Originally Posted by wisodev

When you are using "Mmultiple mono input" mode the input files must be mono, if you turn off this mode then you can encode single multichannel input file.

how do I disable mono input mode?

[tebasuna51]

Uncheck the 'Multiple mono input' box.

[canuckerfan]

Originally Posted by tebasuna51

Uncheck the 'Multiple mono input' box.

wow. I can't believe I missed something like that

thanks

edit: I have another problem now... my wav source is not sampled to 48k and I have to feed an avs script into encwavtoac3 so that it will convert it. here's my script:

#avisource("G:\temp\1.avi")+avisource("G:\temp\2.avi")
#SoundOut()

WavSource("G:\temp\bgf.wav")
ssrc(48000)

however as soon as I drop the avs into the program it waits about a second and gives an error. "Failed to initialize avisynth". then I have to end task it. I tried playing the avs file in mpc and works just fine... btw i'm running AviSynth 2.5.8

edit: I just found out that it only works with avs 2.5.7 and below as of Aug 2/'09. I guess that's still the case. I'll try it on 2.5.7 and give it a whirl.

[b66pak]

@canuckerfan or you can forget about drag&drop and use the add button...
_

[canuckerfan]

^I can't believe that made a difference haha. thanks, it works now.

[wisodev]

Originally Posted by b66pak

@canuckerfan or you can forget about drag&drop and use the add button...
_

Should be fixed in version 4.5

[Maxiuca]

I keep getting "Encoder Error: Failed to initialize encoder". Funny thing is that when I start the program for the first time after system restart I can get it to encode, but then this error keeps appearing in logs. I'm on 64-bit Vista Ultimate. Any ideas?

[wisodev]

Originally Posted by Maxiuca

I keep getting "Encoder Error: Failed to initialize encoder". Funny thing is that when I start the program for the first time after system restart I can get it to encode, but then this error keeps appearing in logs. I'm on 64-bit Vista Ultimate. Any ideas?

What is the input file format ? Are you using 64 or 32 bit version ? Is the program configuration preserved after restart ?

[Maxiuca]

Sorry for a false alarm, problem solved, my bad. I was trying to encode a 5.0 ac3 with LFE low-pass filter enabled. It was pure coincidence that the program started encoding after two reboots (different settings).

[Midzuki]

Let's say, I have a 3min 36sec stereo WAV file (extracted from an Audio CD), and compress it with Aften.exe @ 192kbps, and then use spdifer.exe to obtain a DDWAV stream. Instead of getting a 36.3MB .wav, spdifer produces a 34.8MB one (and if I try a different bitrate, e.g., 256kbps, the resulting SPDIFed .wav has only 17.4MB ). This problem *does not* happen with ac3 files generated by Soft Encode, for example.

[Midzuki]

OK, today I found this post by madshi:

http://forum.doom9.org/showpost.php?...&postcount=945

which made me give a try to the latest release of EncWAVToAC3. Sadly the "44.1kHz problem" seems to be even worse in the libaften-GUI encoder: the same .WAV source-file mentioned in my post above becomes a 5.77MB .ac3 encoded at 224kbps by EncWAVToAC3, however spdifer.exe obtains a
12-byte .WAV file from that. Again, spdifer keeps working correctly on files created by a "Dolby-certified" encoder...

[tebasuna51]

Originally Posted by Midzuki

OK, today I found this post by madshi:

http://forum.doom9.org/showpost.php?...&postcount=945

The problem don't exist now.

BTW, I can't understand the process.

You have a stereo wav file 3m:36s (36,3 MB), encoded with ac3 192 Kb/s (4.9 MB) and spdifer convert the file to ddwav (34.7 or 36.3 MB)
What is the benefit to have a lossy compressed audio with the same size than the lossless source?

With my stereo samples, encoded with Aften and Soft Encode, obtain the same wrong output size when use spdifer.

I have some 5.1 ddwav and the size isn't the size of the wav sources. Is the size of the ac3 x 2.2
ac3 file 31.5 M -> ddwav file 69.6 MB

With spdifer I obtain the same ratio with 5.1 ac3 files.
Maybe spdifer don't work properly with stereo files.

[Midzuki]

It depends on the spdifer's version one uses, I think. FWIW, I'm using the build 0.1a.

Anyway, I am not thinking in terms of "benefits", just in terms of "standard-compliance" --- if someone intends to create a multichannel AC3 Audio CD, it's better that they know they shouldn't be using Aften for doing that. Given that spdifer does work correctly with the 32kHz && 48kHz AC3 outputs from Aften, it seems to me, *now*, that the 44.1kHz encodes from Aften are not "correct enough". I remember that "many months ago", just for testing/learning purposes,
I created some stereo AC3s @ 44.1kHz with aften.exe, and realized that most of them were played back @ 48kHz when decoded by AC3Filter. Because such files were played back @ 44.1kHz when decoded by ffdshow, I (erroneously) concluded that the actual problem "had to be" in AC3Filter, "not" on the 44.1kHz files generated by Aften.exe.

P.S.: So, ¿¿¿ is there any better alternative to spdifer.exe ???

P.P.S.: ffdshow (r1376) audio processor is not an option,
because its AC3 encoder supports only 48kHz and 32kHz.