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Old 5th March 2007, 19:04   #841  |  Link
xc3ll
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Juhu, I patched it with aften 0.06, and it fixed the problem. Thanks! The only troubling thing now is that BeHappy seems to think its 2ch audio... I'll have to look at that more closely...

EDIT:
double checked it, and it is 2ch audio... the audio sounds fine tho. Are you getting 6ch audio Juhu?

Last edited by xc3ll; 5th March 2007 at 19:18.
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Old 5th March 2007, 19:06   #842  |  Link
Deckard2019
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Same kind of problem with SoundOut, which uses Aften too. My AVS looks like xc3ll one.
If I choose another format in SoundOut, it works.
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Old 5th March 2007, 19:19   #843  |  Link
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Check out this build of BeHappy. Works great.
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Old 5th March 2007, 19:45   #844  |  Link
xc3ll
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I keep getting the 2ch problem with BeHappy. Not sure how to fix it either.

JnZ, also tried your method, but WaveWizard says it doesn't recognize the file type. I did get BeHappy to output a 6ch ac3 file tho.

This is what I did:
1)Use graphedit settings provided by JnZ, hit play.
2) Use sox settings provided by JnZ
3)Make .avs file with the following line:

directshowsource("dump.wav",video=false)

where dump.wav is the file you just made in sox
4)Input into BeHappy using Juhu's settings.
5)Done!

The only problem so far is that I can only make 448Kb/s audio. I tried the version of behappy that chumbo linked to, but it didn't work. I can't test mappings of sound either... I'm just guessing that they're right for now. The strangest thing is that the audio is MUCH shorter than it should be. I'm pretty sure this is happening when I make the wave. I tested the output of the DD+ stream in graphedit, and it sounds fine, but the wav plays 3x faster than it should be.

Either 1)the dump isn't working correctly(its only 1.7gb) or 2)sox isn't working correctly.

Any ideas?

Last edited by xc3ll; 5th March 2007 at 21:24.
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Old 5th March 2007, 21:48   #845  |  Link
MichalHabart
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Quote:
Originally Posted by xc3ll View Post
I keep getting the 2ch problem with BeHappy. Not sure how to fix it either.

JnZ, also tried your method, but WaveWizard says it doesn't recognize the file type. I did get BeHappy to output a 6ch ac3 file tho.

This is what I did:
1)Use graphedit settings provided by JnZ, hit play.
2) Use sox settings provided by JnZ
3)Make .avs file with the following line:

directshowsource("dump.wav",video=false)

where dump.wav is the file you just made in sox
4)Input into BeHappy using Juhu's settings.
5)Done!

The only problem so far is that I can only make 448Kb/s audio. I tried the version of behappy that chumbo linked to, but it didn't work. I can't test mappings of sound either... I'm just guessing that they're right for now. The strangest thing is that the audio is MUCH shorter than it should be. I'm pretty sure this is happening when I make the wave. I tested the output of the DD+ stream in graphedit, and it sounds fine, but the wav plays 3x faster than it should be.

Either 1)the dump isn't working correctly(its only 1.7gb) or 2)sox isn't working correctly.

Any ideas?
If raw pcm has same size as "soxed" wav file, then sox is working properly. How long is that movie you are trying to demux? For 99minutes long movie pcm is about 4900MB.
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Old 5th March 2007, 22:00   #846  |  Link
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Quote:
Originally Posted by xc3ll View Post
I keep getting the 2ch problem with BeHappy. Not sure how to fix it either.
Any ideas?
I doubt there's something to fix, Behappy is just the "transporter".
It takes what Sonic audio decoder gives him and feeds your chosen encoder with it.

As some already previously mentioned, you have to specify your system you're using 5.1 output. That means 2 visits to your control panel

#1 : general audio settings ->advanced audio parameters :


That leads us to #2 : sonic cineplayer audio parameters :


if you don't choose 5.1 output on step #1, there are chances you won't see the 5.1 option on step #2....

now the 6 channels should follow the pipe .evo->sonic audio->behappy->aften.
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Old 6th March 2007, 00:51   #847  |  Link
earl
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hi guys, i have read here a lot.. but i do not understat what i must do toplay .evo (vc1 with ddplus) file in the mpc
i must link a filter from scenarist to the mpc? but how, and which one?
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Old 6th March 2007, 00:56   #848  |  Link
xc3ll
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Juhu, thanks A LOT, I'm finally getting 6ch audio. However, I've noticed that even though I've done the time stretch, my audio still goes too slow. By the time it gets to the 4 minute mark, it's already 1/2 a second too slow, and it gets to be about 3 seconds too slow at the end. My calculated is 149158 and counted is 149216, which is a 2.5 second difference. It just doesn't seem to be doing the time stretch, because I've tried doing it multiple times with the same result.

earl: get graphedit(google it and you'll easily find it), and set it up like this:


That'll play it fine. You could also get PowerDVD, it will also play .evo

Last edited by xc3ll; 6th March 2007 at 05:51. Reason: audio spanning not working correctly
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Old 6th March 2007, 11:41   #849  |  Link
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Quote:
Originally Posted by JnZ View Post
Hm strange, try this:
1) DTS/AC3/DD+ Decoder -> Sonic Audio decoder 4.2 -> Default Direct Show device (Def. DS device is in Audio Renderers), and try to play it...on soundcard you must hear sound.
2) Do you use Sonic Audio Decoder 4.2?
1. Nothing, no sound, no progress bar
(With dump at the end of the graph, the progress bar is working)
2. Yes.
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Old 6th March 2007, 12:35   #850  |  Link
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Quote:
Originally Posted by Pelican9 View Post
1. Nothing, no sound, no progress bar
(With dump at the end of the graph, the progress bar is working)
2. Yes.
When you press play, what do you have in the filter properties window of the Default Directsound Device ?
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Old 6th March 2007, 13:40   #851  |  Link
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Quote:
Originally Posted by Darth Pinous View Post
When you press play, what do you have in the filter properties window of the Default Directsound Device ?
There are three pages (Audio renderer, Advanced, Audio input pin (rendered)).
Nothing changes when I press the play.

The first shows:
wformat tag: 1
nchannels: 2
nsamplepersec: 48000
navgbytespersec: 192000
nblockalign: 4
rate: 1.00

Advanced shows nothing (I mean all '-' and '0')

The last shows:
Major type: Audio
sub type: PCMaudio
Format: waveformatex: 48.000 kHz 16 bit
stereo
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Old 6th March 2007, 14:28   #852  |  Link
Pelican9
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Quote:
Originally Posted by idamien View Post
Feedback:

Just tried SUPRead v0.25b. Access violation when opening app doesn´t happen anymore but is still present when opening .sup file...
EVOdemux creates wrong .sup files if you uncheck any stream.
I'm working on it.

Edit:
Done.
Download the newest beta of EVOdemux and demux the subtitle again.

Last edited by Pelican9; 6th March 2007 at 15:06.
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Old 6th March 2007, 16:56   #853  |  Link
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Quote:
Originally Posted by Pelican9 View Post
1. Nothing, no sound, no progress bar
(With dump at the end of the graph, the progress bar is working)
2. Yes.
Hmm, strange, something wrong in Your system...that filter must "play" (when connected to DShow audio device).

We are trying to "catch" RAW data, which goes to soundcard redirecting them to the file (dump). So if you can't play this filter on soundcard, not able to "save" any raw data to file too.
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Old 6th March 2007, 17:04   #854  |  Link
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If sombody interested to making 5.1 OGG channel audio from DD+ (which is the best IMHO), here is guide:

1) Make graph filter: DD+ src-> Sonic Audio 4.2 -> Dump
2) Use SoX: sox -r48000 -S -t .raw -c 6 -3 -s xxx.raw -2 y:\xxx.wav
3) Use wavewizard and separate it to six mono WAVes
4) Wrote MUX file with 0-2-1-4-5-3 channel order. (OGG have different ch. order, than 5.1ch WAVe)
5) Use BeSweet and make WAVe file (test.wav).
6) Use aoTuV encoder: venc -q6 test.wav (-q switch means quality, -q6 gives average 460kbps)

Note: It's possible to use oggenc directly to RAW data: oggenc.exe -r -B24 -C6 -R48000 -q6 xxx.raw, but this create bad channel-ordered OGG.
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Old 6th March 2007, 17:21   #855  |  Link
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Quote:
Originally Posted by Pelican9 View Post
There are three pages (Audio renderer, Advanced, Audio input pin (rendered)).
Nothing changes when I press the play.

The first shows:
wformat tag: 1
nchannels: 2
nsamplepersec: 48000
navgbytespersec: 192000
nblockalign: 4
rate: 1.00

Advanced shows nothing (I mean all '-' and '0')

The last shows:
Major type: Audio
sub type: PCMaudio
Format: waveformatex: 48.000 kHz 16 bit
stereo

Seems to me that the sonic audio decoder sends crap to your soundcard... Did you install Sonic 4.1 filters ?
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Old 6th March 2007, 18:18   #856  |  Link
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EVOdemux v0.623

Changes for v0.623

- Display H.264 video stream info (finished)
- Rebuild change the DSI packet's audio and subpicture information and PTM info too
- Fixed bug with MLP and LPCM stream handling during Rebuild
- Fixed bug with 'Cancelled' message
- Fixed bug with 'Invalid floating point operation' after demux
- Fixed bug with DTS HD info
- Fixed bug with .sup files
- Fixed minor bugs with opening files and closing application

EVOdemux v0.623


Quote:
Originally Posted by Darth Pinous View Post
Seems to me that the sonic audio decoder sends crap to your soundcard... Did you install Sonic 4.1 filters ?
No. I've installed 4.2 only.
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Old 6th March 2007, 18:36   #857  |  Link
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Quote:
Originally Posted by Pelican9 View Post
No. I've installed 4.2 only.
I've tried to install 4.2 only, but I couldn't connect Sonic with orbitlee filter in Graphedit, and I had repeated errors when closing Sonic configuration panel.

After installing Scenarist 4.1 (which doesn't affect 4.2 Sonic filters), I could connect Sonic Decoder with orbitlee filter and render DD+ files... Maybe that will work for you too ...?
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Old 6th March 2007, 20:13   #858  |  Link
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I've registered scenarist 4.1 filters (as the application doesn't work with mce) and 4.2.
All i get is a 6-channel output of which 4 channels are crap.
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Old 6th March 2007, 21:32   #859  |  Link
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Im also getting that the orbitlee dtsac3source filter cannot connect to sonic audio filter 4.2. Can someone advise me what is wrong, can it be a conflict with filters? , 4.1 connects to orbitlee but not 4.2.

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Old 6th March 2007, 21:48   #860  |  Link
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This might not belong in this thread but here it goes anyway. Since we are discussing "Sonic Cinemaster Audio Decoder 4.2".

When I use it to decode for example a LPCM 5.1 stream I would like to preserve the same dynamic range as the original. There is a setting in Sonic Cinemaster called "environment" that adjusts dymaic range.

My question is if "Quiet" means "Full Dymanic Range = like the original" or if it artificially increases dynamic range even further. Or should I go with "Normal"?
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