how to transcode dts files with foobar2000

[Kurtnoise]

Hi,

Everybody knows (I hope at least) that the freeware to transcode dts files amount on a hand ;) We can have azidts (http://www.doom9.org/Soft21/Audio/azidts.zip) and dtsdec (http://www.rarewares.org/files/others/dtsdec.zip) from the libdca librairy.
The first one seems to be great because we can use it with besweet.dll. So, we can make a transcode of dts to aac/ac3/mp2/mp3/vorbis/wav very quickly. But the most disadvantage is this tool need the WinDVD Intervideo filter :( and unfortunately WinDVD player isn't a freeware...:rolleyes:
The second one dtsdec is a CLI (Command Line Interface). So, first we need to transcode dts to wav and then wav to anything following ours desires. In conclusion we have made 2 steps and this tool is more or less buggy :scared:

Fortunately today we have a new way ;) And it seems to work pretty good. This way is Foobar (http://www.foobar2000.org/), the famous audio player with his new plugin foo_dts (http://www.saunalahti.fi/cse/foobar2000.php) to decode dts files. Yesssss :) One of the good point for this player is we can use directly the encoder and some plugins/dsp. That's why the transcode processing is reduced to one step instead of two like dtsdec. I'm testing dts to aac/mp3/vorbis (6 channels/2channels) and all seems to work very well...and all it's free ;)

Feel free to test it. :D

[daphy]

ehy

this is really great news - something to play for the weekend :p
What´s about DTSWAV - does this work, too :rolleyes:

[Kurtnoise]

Hi daphy,

Originally posted by daphy
What´s about DTSWAV - does this work, too :rolleyes:
Mmmh it seems to work but only with 44,1 KHz files. I tested with some 48 KHz samples but dtswav crashes.


And I've forgotten to say one thing. There is an other free way : the lastest AC3Filter to decode to wav with graphedit. Then we can use BeSweet of course...But once again there are 2 steps. ;)



I'll post some examples + screenshots this week-end.

[daphy]

Mmmh it seems to work but only with 44,1 KHz files. I tested with some 48 KHz samples but dtswav crashes.
Am I wrong, but DTSWAV should always be 44.1KHz?

[Kurtnoise]

Yes..you're right. :p It's my fault...most of my dts samples come from my DVDs. And these files have a 48 Khz Sample Rate. It was just for testing...;)

[daphy]

It was just for testing...

me? :D :D :D ;)

[Kurtnoise]

Achhh nein...:D


Like I said, here are some examples for transcoding dts files with foobar2000. First, grab the latest fb2k release here (http://www.saunalahti.fi/cse/foobar2000.php). Just one note before to begin : don't forget to check Optional Components during installation otherwhise you can't make these steps correctly.


DTS to AAC :

For information, fb2k_special version provides natively the FAAC support.


Load the player and add your dts file with Playlist --> Add file Menu (or drag it & drop it directly into the playlist).
Then go to Foobar2000 --> Preferences --> DiskWriter.
In the Output Directory, check "Same as source file directory".
In the Output Presets, search FAAC : MP4(LC),quality 100...from the list and click on "Edit" button to modify or change the preset. There you can choose quality/bitrate Mode...

http://kurtnoise.free.fr/misc/fb2k_dts_to_aac.gif

Then if you want to keep the 6_channels for the output file, go directly to number 8. ...else if you want to downmix (i.e 6 channels to 2 channels) then read the next items.
In the Processing part, check "Use DSP" and click on "go to DSP Settings" button.
There, you can choose one or more DSP. By example to make a downmix, you must put the "Convert 5.1 to Stereo" on the left.
If all it's ok, then ckick on "Save all" and "Close". To launch the process : a right click on the file following by Convert --> Run Conversion. The select file writer appears. So, click on OK and wait. ;)



DTS to HE-AAC :

For High-Efficiency AAC, we need to install the last Nero Burning ROM (http://www.nero.com). There is of course a special plugin for fb2k (http://www.saunalahti.fi/cse/foobar2000/foo_nero.zip) to access of the Nero decoder/encoder.


Put foo_nero in components folder + copy and paste aacenc32.dll & aac.dll into the foobar2000 directory. These libraries are in Program Files/Commom Files/Ahead/AudioPlugins directory.
Load the player and add your dts file with Playlist --> Add file Menu or drag it & drop it directly into the playlist.
Then go to Foobar2000 --> Preferences --> DiskWriter.
In the Output Directory, check "Same as source file directory".
In the Output Presets, click on "Add New" button and choose Nero Encoder from the scrolling list. If it doesn't appears then you have missing something.
The Nero Encoder window pops up :

http://kurtnoise.free.fr/misc/fb2k_dts_to_heaac_2.gif

[1] : select Nero Digital Audio (HE-AAC) (.mp4).
[2] : check on "Configuration" button to choose a preset. You have some infos here (http://www.doom9.org/oagmachine.htm) to set the different parameters.
[3] : check this case to use .m4a extension. But it's not mandatory. Click on OK to validate.
If you want to transcode 6_channels to 2_channels, then check the "Use DSP" button from the Processing Part in DiskWriter window and click on "go to DSP settings".
On the right, we can see some DSP. To activate one or more, just push it(them) on the left side. By example, the "Convert 5.1 to Stereo" to have 2 channels in output.
If all it's ok, then click on "Save all" and "Close". To launch the process : a right click on the file following by Convert --> Run Conversion. The select file writer appears. So, click on OK and wait. ;)



DTS to MP3 :

For MP3, we need to grab a command line encoder. I use Lame (http://www.rarewares.org/mp3.html).

Put Lame.exe in foobar2000 directory.
Load the player and add your dts file with Playlist --> Add file Menu.
Then go to Foobar2000 --> Preferences --> DiskWriter.
In the Output Directory, check "Same as source file directory".
In the Output Presets, search Command Line Encoder : Lame --alt-preset standard from the list and click on "Edit" button to modify the preset. fb2k has already different presets for Lame. So, just look around the list.
In the Processing part, check "Use DSP" and click on "go to DSP Settings" button.
There, you can choose one or more DSP. By example to make a downmix, you must put the "Convert 5.1 to Stereo" on the left.
If all it's ok, then ckick on "Save all" and "Close". To launch the process : a right click on the file following by Convert --> Run Conversion. The select file writer appears. So, click on OK and wait. ;)



DTS to Vorbis :

For information, fb2k provides natively a vorbis_enc plugin.

First, load the player and add your dts file with Playlist --> Add file Menu.
Then go to Foobar2000 --> Preferences --> DiskWriter.
In the Output Directory, check "Same as source file directory".
In the Output Presets, search Ogg Vorbis : -q 5.00 from the list and click on "Edit" button to modify the q-value with the slice bar following your desires :

http://kurtnoise.free.fr/misc/fb2k_dts_to_vorbis.gif

Then if you want to keep the 6_channels for the output file, go directly to number 8. ...else if you want to downmix (i.e 6 channels to 2 channels) then read the next items.
In the Processing part, check "Use DSP" and click on "go to DSP Settings" button.
There, you can choose one or more DSP. By example to make a downmix, you must put the "Convert 5.1 to Stereo" on the left.
If all it's ok, then ckick on "Save all" and "Close". To launch the process : a right click on the file following by Convert --> Run Conversion. The select file writer appears. So, click on OK and wait. ;)


That's all...:)

[Dark-Cracker]

why nobody try to made a working plugin for besweet using libdts this could avoid a lot of headache when u want to convert your DTS file.

[KpeX]

@Kurtnoise

Thanks for posting that :). I hope you don't mind that I added a link to the DTS FAQ.

[Kurtnoise]

No problem...;)

@DC : you're the man for that...:D

[tormento]

Great work! :)

Is there a way to transcode in a simple way from DTS to AC3?

Thanks again... :D

[Kurtnoise]

The easiest way comes with BeSweet, but you must realize that the ac3enc from BeSweet isn't very good...In addition, the only one free ac3_encoder seems to be ffmpeg. But still buggy...

[drpaulng]

Any way to do it in "one step"?

[Kurtnoise]

Besure (http://doom9.org/BeSure.html) enables the trick. ;)

[drpaulng]

Originally posted by Kurtnoise13
Besure (http://doom9.org/BeSure.html) enables the trick. ;)

Thank you.
Are you going to add this feature for your GUI program?
DVD dts ripped -(transcode the bit rate/sampling rate)-> CD dts format (or reverse the process).

[Kurtnoise]

Nope...but I would like (I don't know when exactly) to create a special tool for dts transcoding. :D

[hans-jürgen]

Originally posted by Kurtnoise13
DTS to AAC :
For information, fb2k_special version provides natively the FAAC support. That's right, foo_faac.dll is the output plugin created by Case. By the way, nice guide! :) I'll add it to the Audiocoding.com Wiki page for DTS (http://www.audiocoding.com/modules/wiki/?page=DTS) today.

In the Output Presets, search FAAC : MP4(LC),quality 100...from the list and click on "Edit" button to modify or change the preset. There you can choose quality/bitrate Mode...

http://kurtnoise.free.fr/UV/format_dts_section_4_transcode_aac.gif
It would be important to mention that foo_faac.dll still uses the old bitrate setting "per channel" for the ABR mode, so 192 kbps for one channel wouldn't make much sense and wouldn't work either, because FAAC still has an internal maximum limit at ~75 kbps per channel when using ABR. That's why I would rather recommend using the VBR mode at a quality of 100% or higher, because DTS is supposed to deliver a higher sound quality than AC-3, so storing a DTS->AAC transcoding on the HDD or on a DVD/CD should try to match that higher sound quality somehow which is not possible at much lower bitrates with the AAC LC profile. But if you have to use smaller file sizes/bitrates with FAAC, you should read this thread (http://forum.doom9.org/showthread.php?s=&threadid=74708) again.

[Bruce Willis]

Nice prog fb2k!

I try to do a dts -> mp3 conversion, but fb2k is giving this error message:

INFO (foo_clienc) : CLI encoder: LAME.EXE
INFO (foo_clienc) : Destination file: file://F:\Video\Project predator\001 PREDATOR DVD2AVI DTS T02 DELAY -5ms.mp3
INFO (foo_clienc) : Source file: file://F:\Video\Project predator\PREDATOR DVD2AVI DTS T02 DELAY -5ms.dts
INFO (foo_clienc) : 48000Hz 24bps 2ch
ERROR (foo_clienc) : Writing to encoder failed
ERROR (foo_clienc) : Encoding failed
ERROR (foo_diskwriter) : Conversion failed.



Conversion to wav is no problem, but trying to convert wav --> mp3 gives a similar error as above..:(

[Kurtnoise]

What's Lame version did you use ? And what preset ?

Could you provide a sample from your dts file please...





BTW, it seems that pictures are disappeared...my fault.:o I'll post new ones tomorrow.

[daphy]

INFO (foo_clienc) : Source file: file://F:\Video\Project predator\PREDATOR DVD2AVI DTS T02 DELAY -5ms.dts
INFO (foo_clienc) : 48000Hz 24bps 2ch
ERROR (foo_clienc) : Writing to encoder failed
general question: does lame support 24bit 48 KHz @ all?
Conversion to wav is no problem, but trying to convert wav --> mp3 gives a similar error as above..
what kind of WAV, float or fixed point, did you created?

[Kurtnoise]

Originally posted by daphy
general question: does lame support 24bit 48 KHz @ all?
Yes...;)

[johnman]

How usable/reliable are the mentioned dts decoding methods?
Is it reliable enough for usage without testing everything afterwards??

I dont know where the foobar dtsdecoding sources come from, but i havent found anything better then some "experimental" but usable code for dts decoding.

[tgc225]

Is there any way to go from DTS -> 6 mono WAV files using Foobar2000?

[daphy]

nope, only a 6WAVs are supported but you can split this file with besweet (beware: take a closer look to the channel mapping before reencoding f.e. to AC3)

[Kurtnoise]

Originally posted by johnman
How usable/reliable are the mentioned dts decoding methods?
Is it reliable enough for usage without testing everything afterwards??

Yes, it is.

Originally posted by johnman
I dont know where the foobar dtsdecoding sources come from, but i havent found anything better then some "experimental" but usable code for dts decoding.
the plugin (foo_dts) used into foobar comes from libdca (http://www.videolan.org/libdca.html). It's still in early development...So, you can have some bugs for sure.

[DSP8000]

Hi :),

Very good guide Kurtnoise13, just a little correction,
for DTS Audio CD's dtswav extension you'll have to rename the wav files to fileaudio.wav.dts and after that you'll be able to use foo_dts and transcode the files.
Also, don't forget to use advanced limiter or soft clipping limiter in the DSP aray because as described before foo_dts is very experimental.

Another way to convert these DTS Audio CD's is if you have Audigy ZS Pro Sound Card.
Burn or mount the cd, play it with Creative Media Source Player, set recording source to "CD Digital" , choose recording format wave stereo, auto sync on, avl on, and you're done.
IMHO this method sounds better than foobar2k ;)

DSP8000

[daphy]

@DSP8000

can you please add a screenshot :thanks:
(maybe if you´re not allowed to add by the moderator you can use http://www.imageshack.us/ ;) )

THX

[DSP8000]

@daphy,

screenshot of what?
The foobar2k conversion or Creative.

DSP8000

[daphy]

of the creative settings :D

THX

[DSP8000]

No probs, i'll do it just give me some time.
(real life priorities :) )

Is there any other way of converting/downmixing 5.0,5.1 DTS CD (dtswav) to 2ch wav?

I did coulpe of tests with GraphEdit,Hypercube's DTSWAVsorce,InterVideo WinDVD Decoder,WavDest,FileWriter and in my ears there is big difference between foobar2k and this method.
I'm just looking for acceptable if not the best downmixing method so I can convert some of my DTS CD's to Audio or mp3 disks.
I think that aziddts does this I'm just not sure about the command line process.
Also I'd like to compare the final downmix of the three methods.
I think that aziddts uses different library for decoding the dts stream, but I'm not 100% sure.I remember there was some discussion about azid long time ago and as far as I can remember the last bugfixes solved lots of problems.
The foobar2k method works but it doesn't sound as good as the WinDVD downmix.

DSP8000

[Kurtnoise]

Originally posted by DSP8000
Is there any other way of converting/downmixing 5.0,5.1 DTS CD (dtswav) to 2ch wav?
Maybe with EAC...:)

Originally posted by DSP8000
I think that aziddts uses different library for decoding the dts stream, but I'm not 100% sure.I remember there was some discussion about azid long time ago and as far as I can remember the last bugfixes solved lots of problems.
azidts uses Gabest's DTSSource filter to decode dts files...So, you need to have WinDVD.

[DSP8000]

EAC ? Don't think so :D

I tested the method with azid and WinDVD but for some reason crashes instantly and there is no log file, so I can't find out what is wrong :( .
I think the step with DTSparser to rebuild the dtswav to fixed .dts is giving me some trouble. When I play the "fixed" dts file, sounds faster than the original, maybe that's the problem.
I use the provided command lines in the aziddts readme with no modifications.
Maybe if I can call Hypercube's DTSWAVsource filter in command line the problem will dissapear, but there is no read me for DTSWAVsource filter.
So, the graph will look like this:
DTSWAVsource-->call to WinDVD Decoder with options for manual/auto 2/6ch-->WAVdest-->FileWriter.

Is this possible in command line?

Also i found out on some other forums that the method with foobar2k is NOT reliable, because it's based on VLC DTSDec and DTSDec cannot decode LFE propperly :eek:

This DTS is evil :angry:

DSP8000

[DSP8000]

Update:)

Here's what works for me(slow but good results):

1.Rip the DTS Audio CD with EAC (test and copy files uncompressed) and make image of the CD.
2.Mount the image with DaemonTools or Alchohol 120%.
3.Use any audio capture software that gives you option to record from direct sound, waveout ...
4.Set up WinDVD for 2ch output/send LFE to front l/r
5.Start recording with the app of your choice before you start playing the mounted CD.
6.Set up the app of your choice to record 44.1/16 wav file.
7.Let WinDVD play the whole CD.
8.Split the BIG wave file with app of your choice and export as 44.1/16 waves, DO NOT NORMALIZE OR MODIFY THE ORIGINAL RECORDED WAV IN ANY WAY.
9.Finaly burn as red book CD Audio with Burrrn,BAO,Nero....

To my ears this sounds very good with only one downside, takes lots of time.

DSP8000

[Kurtnoise]

Originally posted by DSP8000
1.Rip the DTS Audio CD with EAC (test and copy files uncompressed) and make image of the CD.
2.Mount the image with DaemonTools or Alchohol 120%.
3.Use any audio capture software that gives you option to record from direct sound, waveout ...
4.Set up WinDVD for 2ch output/send LFE to front l/r
5.Start recording with the app of your choice before you start playing the mounted CD.
6.Set up the app of your choice to record 44.1/16 wav file.
7.Let WinDVD play the whole CD.
8.Split the BIG wave file with app of your choice and export as 44.1/16 waves, DO NOT NORMALIZE OR MODIFY THE ORIGINAL RECORDED WAV IN ANY WAY.
9.Finaly burn as red book CD Audio with Burrrn,BAO,Nero....

Goood...I'll test it asap. Finally, EAC works like a charm, isn't it ? :D

In addition, to your last post : azidts doesn't seems to recognize dtswav files. So, you can't use it to transcode directly.
An another way with directshow filters could be : dtssource --> ac3filter (the last edition which decode the dts files) --> wav dest --> file writer. But I didn't test it...


BTW, I updated my post, on page 1, where you can find the different explanations. (Some pictures were gone.) For DTS to HE-AAC, finally I use the Nero plugin. It works now...but 6 months ago it doesn't. With NAACEnc, we can't have a 6_channels wav input. That's why I prefer the method with foo_nero plugin.

[daphy]

8.Split the BIG wave file with app of your choice and export as 44.1/16 waves

why?

use EAC in the CUE/WAV mode and mount the cue-file - after finishing your replay-transcoding-methodthe the WAV should have the same length (time/filesize) as the source -> rename your source DTSWAV and give your target WAV this name, mount again the CUE file and you should have the same tracks as the original DTSWAV

(theoretically :rolleyes: )
DO NOT NORMALIZE OR MODIFY THE ORIGINAL RECORDED WAV IN ANY WAY.
:confused: your target is a normal WAV 16/44.1, you can do a overall normalization without any risk, correct me if I´m wrong?!?

[DSP8000]

@Kurtnoise13,

Let me know what is your opinion on the quality with this method:D

An another way with directshow filters could be : dtssource --> ac3filter (the last edition which decode the dts files) --> wav dest --> file writer

Why use dtssource when DTSWAVsorce exists and recognizes the dts stream perfectly:confused:

Tnx. for the update about Nero:)

your target is a normal WAV 16/44.1, you can do a overall normalization without any risk, correct me if I´m wrong?!?

Normalization will modify the overall quality and shape of the original recording, instead if you have to you can use dynamic compression.I'm not an expert on this but this is what some people that work in the pro-audio industry do and recommend.

use EAC in the CUE/WAV mode and mount the cue-file - after finishing your replay-transcoding-methodthe the WAV should have the same length (time/filesize) as the source -> rename your source DTSWAV and give your target WAV this name, mount again the CUE file and you should have the same tracks as the original DTSWAV

Well, there is no option auto sync like in Creative Media Player, so unles you're very quick with the mouse there is no way that you'll be able to get exactly the same lenght of the recorded file as the source.
Plus I'm sort of fan of gapless playback, so the editing after the recording is a must for me and I don't like leading and trailing blocks.

As for aziddts, it's so obvious that works much better than dtsdec.
The advantage with dtsdec is that recognizes dtswav(partialy).

DSP8000

[daphy]

Well, there is no option auto sync like in Creative Media Player, so unles you're very quick with the mouse there is no way that you'll be able to get exactly the same lenght of the recorded file as the source.
some recording apps provide a option to watch the channels and start recording when the reach a predefined peak -> I thought WaveLab works this way :rolleyes:
Normalization is the method I would prever because it changes the peak in the same relation over the complete track -> no change in quality
compressing, I don´t think that´s what you should use to reach fine audio quality :(

[planet1]

@ Kurtnoise13

there was a bug in the vorbis encoder dll of foobar2000 (wrong channel mapping in 5.1 mode).

This was fixed by Case, could you update your guide please:

http://www.saunalahti.fi/cse/foobar2000.php

(BeSweet btw did it right all the time ...)


furthermore that quality slider should be chosen carefully (not too high) in the case of 5.1 (seems it was rather intended for 2.0)

[Kurtnoise]

Ok...thanks to point me out this correction. I'll update my guide asap.


There is also an update concerning foo_dts plugin. ;)

0.1 (2005-03-20): Fixed incorrect handling for files that had no LFE channel, more accurate seeking.

[oddball]

Bumping this.

I am looking for an easy way to convert DTS DVD to DTS CD Audio. I have had no joy with the BeSure method thus far or BeSweet which both rely on Intervideo which does not seem to be working correctly for me.

I used the foobar2k method to convert DTS to 6 channel wav file. I then used wav2wav6 to split into 6 mono files (Is this the correct method?), then re-encode in SurCode.

I am in the encoding phase to DTS CD Audio right now and will test the mapping against the source.

I was a bit worried about lack of normalisation while transcoding in foobar2k. BTW what does wavdest do?

[oddball]

Using the foobar2000 method to convert DTS DVD audio to DTS CD audio results in the following. Success! However...there is a caveat.

The mappings are completely wrong either in the foobar2000 DTS decoder when creating a 5.1 wav file or wav2wav6.exe which splits the 5.1 wav file into 6 mono wav files ready for encoding in something like SurCode.

Which I do not know. However the resulting files are named accordingly with the CORRECT mappings to the right.

front_left.wav -> front_left.wav
front_right.wav -> center.wav
left_surround.wav -> surround_right.wav
right_surround.wav -> subwoofer.wav
center.wav -> front_right.wav
subwoofer.wav -> surround_left.wav

I hope someone finds this useful. :)

[magicclue]

Channel mappings are wrong with windvd filter, see:
http://forum.doom9.org/showthread.php?threadid=70805

[oddball]

I was using foo_dts not WinDVD's decoder :)

[magicclue]

yes you were.

->wav2wav6

uses windvd filter to do 6 wavs out of one multichannel wav

[oddball]

Oh well there ya go. Learn something new every day :)

[Rockaria]

Originally posted by Kurtnoise13

There is also an update concerning foo_dts plugin. ;)
0.1 (2005-03-20): Fixed incorrect handling for files that had no LFE channel, more accurate seeking.
A newbee to the fanastic forum.:D

Is there any fix on the distortion(or skipping) on some dtswav audio?
I found we still have to rename the extention to dts to play.

For upper reasons I am using VLC to decode dts stream to intermediate 32bit float 6ch wav.

Upsampling, normalizing through replaygaining(am I right?) and transcoding with foobar seem to be perfect thanks for the guide.

Foobar wav output also works fine with softencode unless bigger than around 3GB.

Keep on good jobs for these novices...:)

[Kurtnoise]

Originally posted by Rockaria
Is there any fix on the distortion(or skipping) on some dtswav audio?

If you are some problems with dtswav files, try to transcode them into wav. And then you can fix them for sure.

[Rockaria]

Yes, I tried with foobar on both playing and transcoding of some of my problematic DTS CDs(it even did not work with besplit correctly : half the normal size).

The results had some problems.
. some tic sounds in the middle of the playing
. some distorted sound and skipping(maybe the seeking problem?)
. but no SUB problem to me(not sure if sub channel included in the original source)

I hope the seeking correction took care of this.

[gfpk2]

Hi, great thread!

I have some questions,

- In Foobar, the vorbis encoder seems to be an old one, isn't it? "vorbis_vendor = Xiph.Org libVorbis I 20040629" how could we upgrade it?
- I'm now reading an Audio FAQ, where "The 2.0 stream has a better mix for 2 channels than the 5.1 to 2.0 downmix features offered by AC3 decoding tools" I'm riping a Live DVD ( dvd decrypter streaming mode ) with 3 audios, AC3 2 ch 448, AC3 6 ch 448, and DTS 6 ch 768; I want to transcode it to a Vorbis or MPC 2 ch bitrate about 180-200 with foobar, it seems more easy than other tools like besweet and similars. Which stream should I decode? DTS I think but... could be possible that AC3 2.0 is more optimized for this?
- With Vorbis or Mpc, not lame, Would be better downgrade the audio to 44.1 khz or keep the original 48 khz?

Thanks for all and sorry about my english and for this amount of ( probably ) dumb questions. Any other suggestion will be be apreciated, perhaps i must consider the option of transcode it to Vorbis 5.1 channel from AC3 or DTS ch. streams, but i never did it before.


Thanks!

[magicclue]

Originally posted by gfpk2
[B]Hi, great thread!

:thanks: and welcome to the forum.


- In Foobar, the vorbis encoder seems to be an old one, isn't it? "vorbis_vendor = Xiph.Org libVorbis I 20040629" how could we upgrade it?
Yes you can. Just use an ogg vorbis CLI encoder instead!
e.g. from rarewares.org (http://www.rarewares.org)
In the diskwriter tab coose "output presets" select "Commandline Encoder Vorbis -qx" whatever quality. Click on "edit" and put in the path to your updated ogg encoder (oggenc.exe or oggenc2.exe). change the -q settings to whatever meets your needs ;)


.. Live DVD ..with 3 audios, AC3 2 ch 448, AC3 6 ch 448, and DTS 6 ch 768; I want to transcode it to a Vorbis or MPC 2 ch...
Would be better downgrade the audio to 44.1 khz or keep the original 48 khz?

If you want to encode to 2 channels I would grab the 2ch AC3 to transcode as it is professionally downmixed with clear "vocal sound". If you do the downmix it may be that the "surround sound" is too dominant.


..perhaps ... transcode it to Vorbis 5.1 channel from AC3 or DTS...

I would suggest AAC 5.1 as it is more stable.