View Full Version : Can cook encode 48kHz?
datascab
13th December 2003, 17:55
I dint think it could then i read a post which pointed to the settings in realone player (audio quality).
I think this was just for playback though.
Can i configure cook to do 48Khz instead of the default 44.1kHz?
THanks
Data
Sirber
20th December 2003, 03:24
cook stereo or cook surround?
datascab
20th December 2003, 07:39
Hey,
Yeah cook surround is the one.
Dont know if there would be an audible difference but the ac3 streams are usually 48kHz so it would avoid the resampling.
THanks
Data
Sirber
20th December 2003, 15:02
use "surround" if the source is AC3. It supports 48k.
datascab
21st December 2003, 07:17
Hey Sirber,
I always choose multichannel surround to preserve the 5.1 arrangement but as far as i know the cook codec only encodes at 44.1kHz.
The audio input streams are usually 48kHz.
I think when I posted this to Dark on video-tools.com he said that although producer can accept 48kHz it will down sample it to 44.1kHz.
He suggested to get besweet to do the down sampling as its of better quality than when producer does it.
I'm not sure but i think realaudio cook codec is designed for 44.1kHz only, but i just wanted to ask on here to see what other people said.
Thanks for your help with this.
Data
karl_lillevold
21st December 2003, 08:47
He suggested to get besweet to do the down sampling as its of better quality than when producer does it.
This is unlikely. Producer's resampling filter is very high quality, and it is very fast, but it is fast without any shortcuts being done. Besweet's resampling is probably high quality too, but since I don't know, I would probably recommend Producer's resampling, because it is faster. Either method works well. With regards to downsampling 48 -> 44.1 kHz, and when, I will find out the right answer. Off the top of my head, I thought cook can encode at 48 kHz, but I will check with the right audio codec team member.
datascab
22nd December 2003, 22:33
Hey Karl,
Thanks for your post. I'd really appreciate if you could find out. Am i right in thinking that avoiding any resampling would always be preferable assuming the playback format is supported.
Thanks
Data
karl_lillevold
22nd December 2003, 23:03
ah yes, I did find out. Even though cook could have encoded at 48 kHz natively, there were system issues that forced Producer to always resample to 44.1 kHz. These issues may go away, and we will consider encoding at 48 kHz when the input is 48 kHz.
Sirber
22nd December 2003, 23:04
Cook @ 48kHz @5.1 works #1 in matroska, but I had some difficulties in RM.
karl_lillevold
22nd December 2003, 23:13
@Sirber: i think what you saw was internal Producer resampling. Works for both Matroska and RM. Your RM problem may have been playback related.
datascab
23rd December 2003, 03:21
Hi again,
Does this mean we need to wait for another version before producer will encode at 48kHz?
I'm using the latest AutoRV9 and it seems to downsample if I dont have the force 44.1kHz when i give it a 48kHz 5.1 ac3.
Thanks
Data
karl_lillevold
23rd December 2003, 08:49
Originally posted by datascab
Does this mean we need to wait for another version before producer will encode at 48kHz?
Yes, but don't hold your breath. It will not be included in the next release, but I will enter a request for a enhancement. Thanks for the feedback.
If you let Besweet do the resampling, Producer will not resample. If you do not let Besweet reample, Producer will. I bet the quality will be indistinguishable.
As a sidenote, soundcards like Audigy 1 and 2, always resample to 48 kHz, which does not seem to bother most people, so the problem of resampling even when un-necessary, is not uncommon.
datascab
23rd December 2003, 09:11
Hi,
Thanks for the reply. From what you've said I guess the resampling process isnt lossy?
I have a Turtle Beach Santa Cruz so I dont know off hand what my card does (although I like it heaps more than my old sblive 1024.
Thanks for the help.
Data
karl_lillevold
23rd December 2003, 09:36
it's lossy, since there will be rounding differences, but very minor compared to codec compression schemes. Still, avoiding it would still be best. With a high quality resampler, like the one in Producer, it should be inaudible (audio experts, please correct me..) The resampler used by Creative is of lower quality than the resampler in Producer, and can be heard by audio enthusisasts with really high quality DACs and headphones, and suitable test material.
tiki4
23rd December 2003, 10:22
If someone is interested in resampling issues with Creative soundcards maybe the best idea is to take a look into the HA forums. Resampling was discussed often and widely there. As a wild guess I'd say that 95 % of people don't notice the difference on movie material.
Also I think that most modern sound cards resample internally to 48 kHz but Creative's don't seem to use the best algorithm. BeSweet uses ssrc to resample which is one of the best out there. DSPGuru cites this site (http://www.ne.jp/asahi/fa/efu/fsconv/fsconv_2.html) which is in Japanese it seems.
Anyway, preventing resampling would still be the best solution as you have two times a lossy process (48 -> 44 while encoding and 44 -> 48 while listening).
Regards,
tiki4
ChristianHJW
27th December 2003, 01:24
Just for your information, if the AC3 is resampled in BeSweet, this can be done with a 32 bit FP precision if the output of Azid is set to outputting at this bitdepth, SSRC can handle it well and will even use dithering for the 16 bit output ....
tiki4
28th December 2003, 21:47
@Christian:
Can you give an example command line, because if I remember correctly BeSweet shows 'floating point process: no' if you choose stereo WAV or 5.1 AIFF output. It just shows 'floating point process' when encoding directly to MP3, MP2 or Vorbis. Please correct me, if I'm wrong.
tiki4
vBulletin® v3.8.4, Copyright ©2000-2010, Jelsoft Enterprises Ltd.