Looks good. Yes, it might have took some time, but you learned more by doing it that way then reading about it.
Just a couple of comments. First, don't sweat it if you can't use 32-bit floating point format files. Yes, this would result in the most mathematically precise files, but the differences between 24 and 32 bit files are practically inaudible. The big difference occurs between 16 and 24 bits (yet some people claim they can't hear even that difference). Besides, programs such as Wavelab do their internal calculations at 32 bits, so the advantage of actually loading 32-bit files are pretty much irrelevant. I believe Wavelab has a setting for using 32-bit temporary files somewhere in its preference settings, so make sure you check that.
You mentioned that it's not necessary to use 48kHz audio. No, you don't need to if you're going to play the files back on a computer. However, if you're doing an audio-only DVD (not DVD-Audio but audio on a DVD-V)or the soundtrack for a DVD, it must be at 48 or 96kHz. That's a requirement of the DVD specification. However, if someone is going to do a surround CD (with an ac3wave or dtswav. Several guides on here explain this), the sampling rate must be 44.1kHz or it won't work.
If someone is doing a DVD-Audio project, the sampling rate can theoretically be anything. However, bandwidth limitations prevent a 24/96 5.1 DVD-Audio disc unless someone has the software for MLP compression.
Anyway, just wanted to throw in some information. Thanks for your guide. It's very helpful.
Regards,
Steve.
Quote:
Originally Posted by CrazyJ32
OK, first of all incase you cannot find the fLfR, CLFE, sLsR, and Limiter plugins in Wavelab then you need to put the V.I Suite 1.1 Installer and Classic Master Limiter files in C:\Program Files\Steinberg\Vstplugins
Now, if anyone else here has issues converting these wave files into a multichannel file, here is what I did: - In wavelab I set the bit-depth to 24bits¹ before encoding. I also used 48khz sample rate but you don't need to.
- I used Multiwav 1.0.0.1 to mux the files into a single, 6 channel wav.²
- I then used WAV to AC3 Encoder 0.9 to convert it to ac3 48khz³ 448kbps 5.1 surround. To do this I used the following settings:
- 448 kbps (any will do, but I needed this for my particular purpose)
- CBR (uncheck VBR, but its not necessary)
- Dolby Surround Mode: Dolby Surround Encoded
- Audio Coding Mode: 3/2 (L, R, C, Sl, SR)
- Specify use of LFE Channel: LFE Channel is present
- Engine: Aften (via drop-down menu)
The rest of the settings I left as default.
¹Unfortunately Multiwav does not support 32-bit floating algorithm.
²The wave file played as static in all media players except VLC.
³The default setting in my case. I don't know if this is the program default or if it was defaulted by the source sample rate. |