2 channel and 5.1 question

[aicjofs]

Ok I have read the conversion guide(great guide) and did some searching on this site, and didn't find a definitive answer for this.

What kind of sound quality differences exist between converting a 2 channel source to 5.1(or 5.0 w/o LFE), versus playing a 2 channel into an amp or reciever with Dolby Pro Logic II?

I can see an argument for both. Obviously you can tweak your own settings for your own encode, where Pro Logic II is fixed(more or less), but why spend hours encoding when Pro Logic can do it on the fly(this might answer it's own question if I think about it, i.e. on the fly might not be as good)

Either way of course only I can tell, so I will be doing some experiments this week. I am just looking for some initial feedback on what others think.

[KpeX]

IMHO, it depends mostly on your amp / pro logic device. Since pro Logic (II) is really intended to work on a 2 channel source that has multichannels matrixed in, some do a better job than others of extrapolating 5.1 from a 'true' 2 channel source, such as most music. For example, my amp, when extrapolating 2 channels, feeds most of the signal into the center channel, which is not good for music, because there is little surround effect and the center channel speaker is not designed for music. That's why I started getting into stereo upmixes. Your best bet is to do some experiments with pro logic and the several upmixing methods out there (forum search will help), and see what you like better for your source material.

[kempfand]

Immediate reaction: If Pro Logic II was 'as good' as conversion methods (be it KpeX, SAD5.1, Ambisonics, etc), people evidently wouldn't spend hours with these methods :p

The Ambisonic method with Bidule will be quite fast, once the new release (with the off-line mode, i.e. converting faster than real-time, as fast as your CPU(s) will do) will be available. Still all methods require some work to get them 'up & running'.

Let us know about your listening-tests and what you think.

Kind regards,

Andreas

[aicjofs]

I have been following the tutorial by EoH and keep running into some troubles. First, I am encoding a 44,100 16bit 2 channel wav 4:32. After running it through Bidule I get a 44,100 16bit 6 channel wav 21:31, at least this is the time reported by windows, dbAmp, and winamp. I checked to make sure Bidule is at the right sampling frequency and triple checked module settings. When played with winamp it is in 6 segments, some very slow, some normal speed, a definite LFE channel track too. I know the 6 the channel wave conforms to standards but is not played by all software. Is this normal or should I get a 4:32 clip? Progressing further, when I run BeSweet 1.5b23 using command line I get 6 21:31 mono waves. This is where I stop. Should I continue on, it just doesn't seem right. I really expected the 6 mono waves to be 4:32. I know DSPguru wants logfiles but I don't think this is a BeSweet problem at all. I think it's my Bidule config. Finally one BeSweet question. My mono files are not corretly named. It seems the tutorial thread has alot of discussion on the pinouts in Bidule but no matter how I configure the pins the BeSweet "real" LFE channel is always named surround right. I remember DSPguru saying (somewhere I can't find again) he was going to create a switch to rename them. I haved switched the known LFE on emigrator to 3 different pins on the audio recorder and can't seem to get LFE named anything but surround right. I also tried upsampling to 48,000 32 bit with the aforementioned issue. Here is 1 of the 3 bidule pin mappings which MUST be wrong can someone point me in the right direction here. Thanks in advance. http://home.comcast.net/~aicjofs/bidule.jpg

[daphy]

Hi
The channel settings (http://home.wanadoo.nl/appyhappy/43.jpg) in the guide seamed a little diffrent to your settings. Check this one more time! LFE Channel can be added but you use the wrong ping (read the guide at this point one more time ;)

Bidule:
what settings (see preferences) do you use 44.1/48 (this might be the first problem -> you have to set to 44.1 for DD/DTS-CD (PCM)

if the source is 16 bit -> output file in recorder should be the same, if you want to upsample, then upsample the source into 32/44.1 not the result.
The 6wav containing result from bidule must be about 4-5 times of the size from the source file - splitting is necessary because this file (it selfe) makes no sense if you wouldn´t split and transcode it to DD/DTS!

Hope this helps a little ;)

CYA Daphy