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View Full Version : Can't read audio DTS- Pls help ! (long read ahead)


Skeul
22nd March 2005, 13:01
Hello everyone.

I have big problem...
I am posting post I wrote from an other forum. If anyone could be help I would really appreciate.

Thks.

Alex


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Hello everyone,

I am going crazy, I really hope your guys will be able to help me.
Believe me before asking your help and creating a new topic in this great forum I have looked quite a long time to find a solution to my problem with no real luck.

Let me explain everything...

First of all, the hardware & software I own:

- Sony S1 laptop with dvd player
- M-Audio Sonica Usb sound card (plugged onto the laptop)
- DD & DTS Amp (Thomson) connected to the Sonica via SPDIF
- and of course a 5.1 loudspeaker kit.
- WinDVD 4 & 6
- PowerDVD 5
- VLC
- Foobar2000 & foo_dts.dll + asio plugin installed
- ZoomPlayer with tons of codecs/filters (Ac3Filter 1.05, Cinesmaster, Ogg etc...)
- DTSParser

Now what I am trying to do:

Listen to DTS-Audio files in DTS mode of course and not stereo.
I want to listen to them directly from the hard drive, even If I have to make an image with nero and load it with daemon tool. But I don't want to have to burn a real cd-r.

Here is what I acheived to do:

I can read with ZoomPlayer, WinDVD and PowerDvd any dvd with a DTS track or a DD track no pb. The amp is getting the right signal, the DTS sign is diplayed and the world is wonderfull.
I can also read AC3 track from divx files using AC3filter on ZoomPlayer with good AC3 signal to the amp.
With should actually be proving that my sound card + spdif + amp is working fine decoding numeric audio signal (DTS & AC3).

I can read any DTS-Wav files and DTS true files with either Foobar2000 and/or VLC but only in stereo mode... I cannot get those two player to activate the spdif output. For foobar I tried what have read: using kernel streaming, installing ASIO plugin but everytime it says: could not create device blablablaalbb crap.

So I tried the burning system... I downloaded for test purpose different DTS-wav files (from that swedish radio and album for p2p like The eagles).
I first burned files to real cds, and I had exactly the same result as creating an NRG image of the disk, therefore I stopped wasting CDs and used NRG image with nero.
By burning the DTS-wav I got 3 differents behavior reading the cd with WinDVD, but in all case the dectected signal on the amp was DTS (I am getting closer...):
- The most common: only a weird pulsating sound jumping to one speaker to another and the LFE signal blinking on the amp.
- For one file (downloaded from the swedish radio): the same weird pulsating sound + a very ugly version of the original signal I could distinguish people speaking and stuff.
- And only for one file (The demo called the run, still from that swedish radio): A perfectly working DTS signal, no distortion everything was perfect ! It just makes me want more than anything be able to listen to DTS-Audio...

I start wandering what the difference could be between those files ... I checked everyone of them and it seems that they are all: 5 channels (I guess no LFE) at 44.1KHZ and with 1411bitrates. So I really don't understand why some (actually only 1) are working and some aren't.

I pushed my investigtion further... I read some stuff about some soundcards resempling the 44.1Khz signal to 48Khz which destroy the sound. But could it be that since 1 files is perfectly working and another one seem close to it ?
Does anyone knows if the M-Audio Sonica is resempling (like SoundBlaster) ? I guess it shouldn't be the case since it seems that the other M-Audio don't. And M-Audio should be quality no ?

Then I read that PowerDVD could read directly DTS encoded files (*.dts) so I got myself a version of it and I used DTSParser and/or Wav2Dts to convert my dts waves to true DTS. I did it with the files I just tested. I played them in powerdvd. At first I was literaly jumping on the room, I was so happy to see the DTS sign displaying on the amp. But I got even sader few second later when I started to hear the sound. The pitch has raised, the rythme was quicker and the voices sound more girly. Then I remember reading some stuff about DTSParser altering the DTS quality and pitch. But the weird thing is that is seems to me but I am not quite sure that when I converted the signal of the one that was already working the pitch hadn't change... Very bizarre because the file property was the same as the other one.

I am really starting to loose faith... Would I ever be able to read those file ? The best lead I am following rigth now is to find a way to activate the spdif output with foobar2000 (I am not even trying with VLC because I am using for quite a long time now I never could achieve that...). The encouraging things with foobar is that it can read both files type: wav-dts and true dts. I can hear the difference even in stereo mode with foobar with the same source the dts-wav is good (pitch, voice etc.) and the other one I created with DTSParser from the source has a higher pitch and all. So I am hoping that if I can get that spdif out on foobar I will be able to read directly the dts-wav file (juste need to add a .dts extension) and pass the signal to the amp for a DTS decoding this time.

I know it was quite long to read... But if your reading this your are almost at the end .
Please help me !! I am beging... Joking appart I would really appreciated any advise, tips, or whatsoever.

Alex tired of typing on the keyboard.

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Hi, Episkeul:

Some things to check and try:

1. Make absolutely sure that you have the latest drivers for your M-Audio sound card ... Even if you already have them, download them again and reinstall them. (When installing other apps like WinDVD/PowerDVD, etc., certain driver files and/or codecs may be replaced by versions that your sound card doesn't agree with)

2. When playing a dts.wav file in VLC: If the file does not contain all 6 channels it is possible that VLC does not choose the correct output mode automatically, therefore you must change the playback mode AFTER playback begins. To do this, click [Audio]+[Audio Device]+[5.1] for local playback (or [3/2]) (or A/52 over S/PDIF for streaming S/PDIF to another device) ... Also, check these settings (after playback begins): [Settings]+[Preferences]+[Audio] ... place a check in the [Advanced options] box at the bottom-right ... [Audio Output Module] = Default ... [Use the S/PDIF audio output when available] = Checked ... (scroll down) [Audio filters] = Blank ... [Channel Mixer] = DTS Coherent Acoustics audio decoder ... Finally, under the [Stream Output] preferences, make sure [Enable audio stream output] IS checked.

Another thing to try is changing the VLC volume to FULL ON (all the way up) ... Don't forget to keep your receiver's audio volume turned down pretty far while you try all of this to avoid the loud white-noise from damaging your equipment.

Post back your results, and good luck to you!

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Hello thank for the advice... But still no luck !
I tryed everything you said but I could get VLC to activate the spdif out.
Even with a regular DVD I can't get it to work with VLC (the spdif) whereas it works perfectly fine with PowerDVD, WinDVD or even with ZoomPlayer properly configured. The only option I could find was the "A/52 over S/PDIF for streaming S/PDIF to another device", I don't see where you get that from ?
Anyway thanks for trying to help.
Anyone elese as any advice ? I ready to try almost everything... even if I have to dance on one leg to make this thing work !!

Seriously, anyone knows if my M-Audio Sonica (usb sound card) is resempling from 44.1Khz to 48Khz (like sound blaster) ?

This week end I downloaded an other demo dts-wav file, I burned it right aways with nero and It worked... The DTS signed appeared on my amp but the weird thing is that the pitch was accelerated, juste like when I was reading modified dts file (with DTSParser) with power dvd... I tried to read the file with foobar and vlc, and when in stereo mode i got a correct pitch and the voice was therefore not girly anymore...

I just don't get it. Why some files are working without doing anything special, why some are working with a wrong pitch ? And why some file don't work at all ? Could this be the resempling problem ? I am afraid so, but if it is the case why on some file and not on some other ?

Any help would be greatly appreciated...

Alex

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I am still stuck at the same point... I am getting desesperate but I am still trying everything that I can think of.
The closest I got for reading dts audio is by converting my dts-wavs to full/true dts and read them with powerdvd (the only player so far who can read dts and pass it through the spdif to the amp, because I still can get vlc nor foobar to activate that &*^$]@"# digital output !). But the problem is that the files seems to be no read well, I have a speed/pitch problem.
Let's talk numbers:
I have a sample files (encoded at 44.1 Khz) which normaly last 5 minutes and 53 seconds (that is what's telling me VLC and foobar), I transformed it to a full DTS using BeSplit 0.9b6 and indeed it stills last 5 minutes and 53 sec (trusting VLC and foobar). Ok now the weird thing is when I play that file (the full dts one) into power dvd whis the spdif output activated it last 5 minutes and 24 sec !!! And if you solve the equation 353 / 324 * 44.1 = 48 Khz !!!!! d**n it the file is read in 48 Khz .... How is that possible ? Ok let's forget that for a moment, now I deactivate the spdif output inside power dvd and set it to regular stereo, and has you can guess the file length has changed !! But not has I expected, in fact the length is not 5 min 53, but a weird number 5 min 40 sec... ! I can't see the mathematical logic of that... So I start thinking that it just maybe powerdvd nutsting around. Just a non sense hypothesis from me, since power dvd usually reads DTS encode at 48Khz from dvd sound track maybe it asumes that all dts should be read at the 48Khz ? I just don't know so I tried burning that file with nero I read it using Windvd in audio mode with spdif activated, but has I expected I got those usual weird static sound, but even the sound was messy the file length was the expected one: 5 minutes and 53 sec as it should always be !!!
Everything is terribly confusing for me... Power dvd reads the files but with a wrong pitch/speed which changes the files length and Win Dvd which seems to have no problem reading the rigth length of the file can't get a good sound out of it. And at last the two player who also read the correct file length can't activated the spdif output on my M-audio Sonica.
Hellllllllllp me please...

Alex

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Congratulations !!! I warned you it was a really long read :) Anyways thank so for at leat caring enough to read through the end. But if you have a solution I would be greatefull for ever !

Alex

Skeul
24th March 2005, 11:44
Up.

Please someone help me !

Rockaria
29th March 2005, 10:36
Hi Skeul,

Amazed with several things you've brought on. But mine will be very short.

How did you set the zoom player to play the DTS CD? Was it through AC3 Filter? I could only play dts, no dtswav with it.

VLC can play dtswav too(no need to change the wav extention to dts).
I believe it will play the original DTSWAV without problems(cause it's very generous on the formats)

Check if you can decode the original dtswavs to 5.1 analog and spdif.
If you doubt some of the dtswav file formats for the spdif out but works for analog 5.1 out, transcoding to wav might help for further experiments.
VLC can outstream to a wav file. Play it using AC3 filter which has the soft DD enclde to spdif. If the m-audio wants 48K for the spdif out, upsample(ssrc, foobar..ffdshow with graphedit) the wavs and try again.

It seems to me the original dtswavs have some problems in the formats. Converting them to dts with other tools might have broken them further.

Skeul
29th March 2005, 11:00
Hello Rockaria,
First of all, thanks for answering.
Actually I did not get ZoomPlayer to play DTS CD. But using graphedit I managed to play DTSWAV using Ac3Filter in analogic (wouldn't use the spdif output) and also DTS using Ac3Filter still in analogic.
The problem is that my m-audio sonica has only 1 stereo jack out and an spdif, and I am not even using the analogic stereo out.
Therefore when I am decoding a DTS analogicly it only gives my stereo out, which is not what I want of course...
I tried VLC, and it works perfectly to decode DTS and DTS WAV, but I just can't activate the Spdif with VLC, only analogical stereo...
I noticed a weird thing using Graphedit... Like I said in my big post, the only I can have the DTS sound over Spdif is by using PowerDVD but I goes faster that it should... So I recreated the filter graph of powerdvd in graphedit I got exactly the same results, and what I saw makes me think that it is powerdvd who is messing around with the frequency. When I created the graph I can see the input at 44.1Khz, then I plug the PowerDVD Audio Decoder and configure It properly, the I plug my Audio Render: M-Audio Sonica. When I got the property setting it says: Spdif in use and the freq: 44.1Khz which is exactly what would one expect. And the weirdly when I start the playbakc the freq jumps from 44.1Khz to 48Khz !!!
So my guess is that power dvd mess up the playback... And getting more and more sure of that because I saw posts from other forum from guys who said the sonica wasn't resempling at all.
But anyway I would like to test what you said, resempling the file, could you explain to me how exactly, because I am sorry I did not quite understand all the step to get there...
About the files I have I think there are good, I downloaded a lot of DTS-WAV files... And all gives the same behavior (except only when I uses WinDVD when the files are burned).
I also got a version of TheaterTek with audio pack, when I am trying to read eather a DTSWAV or DTS I get no sound at all...
Alex.

daphy
29th March 2005, 12:50
Then I read that PowerDVD could read directly DTS encoded files (*.dts) so I got myself a version of it and I used DTSParser and/or Wav2Dts to convert my dts waves to true DTS. I did it with the files I just tested. I played them in powerdvd. At first I was literaly jumping on the room, I was so happy to see the DTS sign displaying on the amp. But I got even sader few second later when I started to hear the sound. The pitch has raised, the rythme was quicker and the voices sound more girly. Then I remember reading some stuff about DTSParser altering the DTS quality and pitch. But the weird thing is that is seems to me but I am not quite sure that when I converted the signal of the one that was already working the pitch hadn't change... Very bizarre because the file property was the same as the other one.
this cannot work - you have to reencode the dts stream, not only changing the file header -> see many guides around here.

try mediaplayerclassic (latest) it can play DTS (->I haven´t tested DTS-WAV), the only thing you have to change is the direct output option -> SPDIF in settings, uncheck also the flag ~> downsample to 44.1KHz...

Skeul
29th March 2005, 13:28
this cannot work - you have to reencode the dts stream, not only changing the file header -> see many guides around here.

What do you mean by that ? I did not changed the header... I removed the wav header around the dts data to get pure DTS using DTSParser and BeSplit. Those "pure" DTS file can be read from many players: powerdvd, foobar2K, vlc etc. But with powerdvd the pitch is bad because the output is done at 48Khz and with foobar2k and Vlc I cannot get the spdif to work.
With foobar2k I use kernel streaming, it works fine for any file (mp3, classic wav...) but for any dts or dtswav it give me an error: Cannot create KS device or something...

But thanks for the tips, I will try media player classic. I will give you a feeback tonight when I get home.

Alex

daphy
29th March 2005, 13:43
Those "pure" DTS file can be read from many players: powerdvd, foobar2K, vlc etc.
sure but toooooooooooo fast :(

Skeul
29th March 2005, 13:50
Ok, but it reads it to fast only when the spdif is activated... Because when it is in analogic mode (decoding by the software) it plays in stereo :( but at the rigth speed :) How come that is ?
I don't understand.
You were talking about re encoding the dts-wav to a real dts file. Please could you point me a tutorial please ? Because I thought I was doing the right thing... :(

Alex

daphy
29th March 2005, 14:55
You were talking about re encoding the dts-wav to a real dts file. Please could you point me a tutorial please ? Because I thought I was doing the right thing...
the main steps could be found here (http://forum.doom9.org/showthread.php?s=&postid=610377)
the rest at the beginning of this thread :rolleyes:
the most of the needed apps are on needfulthings (http://www.needfulthings.webhop.org) -> see BeSweet Installer

Skeul
29th March 2005, 16:01
Thank you a lot Daphy for taking time to answer.
I am sorry if I am rather slow, but I am really new to that audio encoding/decoding stuff. I don't know all the subtilities...
But I read carefully the thread you gave me, a learned some new things but I still have few questions about it:
- It seems to me that all the guys in this thread have no problem reading dts audio, which is unfortunalty not my case... I can't even read a dts-audio using windvd .. it gives me only weird statics excepts for 2 specifics case (one is the firs file therun from http://www.sr.se/multikanal/index.stm).
- Second, I did not noticed any change in the file length/speed while reading them in analogic with vlc (normal because it reads dts-wav) but even with pure dts... I always used bsplit 0.9b6, what would the 0.9b7 change (for me ?)
- And last I don't really understand the point of getting the dts out of the original dts-wav to decode it and to get 6 mono waves files upsampling them and then at last reencoding them into a dts wav file... That must be degrating the sound ?
- When you do that dts-wav => dts => 6 mon wavs => upsample from 44.1 to 48 => reencode dts-wav , I have tried this chain once, I got to reencode the final dts with surcode Dts pro .. but every time it warned me that the origin wav files were in 48Khz and that it expected 44.1Khz .. and the result file was at 44.1 Khz again .. I was not able to read it correctly just like the other ... so what is the point of that ? I don't understand. Can surecode dts can encore 48Khz dts ? I could not find any option that would allow that...

So basically your telling me that my only solution to get DTS sound through my pc is to reencode all my 44.1Khz dts-wav files to 48Khz dts-wav ? Or is it something else ? Because I don't really mater to convert my DTS-wav to dts to read them, but I don't want to loose quality...

Thks, I hope I am not bothering you too much.

Alex

Rockaria
29th March 2005, 19:04
Originally posted by Skeul
I tried VLC, and it works perfectly to decode DTS and DTS WAV, but I just can't activate the Spdif with VLC, only analogical stereo...

Of course, the analog stereo will be only useful for you for the verification of right playback(speed).
If the spdif out has the issue with VLC, it sounds to me(very loud) m-audio wants 48K(or still the m-audio driver issue).

I noticed a weird thing using Graphedit... Like I said in my big post, the only I can have the DTS sound over Spdif is by using PowerDVD but I goes faster that it should... So I recreated the filter graph of powerdvd in graphedit I got exactly the same results, and what I saw makes me think that it is powerdvd who is messing around with the frequency. When I created the graph I can see the input at 44.1Khz, then I plug the PowerDVD Audio Decoder and configure It properly, the I plug my Audio Render: M-Audio Sonica. When I got the property setting it says: Spdif in use and the freq: 44.1Khz which is exactly what would one expect. And the weirdly when I start the playbakc the freq jumps from 44.1Khz to 48Khz !!!
So my guess is that power dvd mess up the playback... And getting more and more sure of that because I saw posts from other forum from guys who said the sonica wasn't resempling at all.

Power DVD might not be checking the input format strictly.
Also there can be a compatibility issue with the original dtswav. The core is surely 44.1K 6ch. But the wrapped format is regarded as 48K stereo by some converters and players...
Also if you look at the headers in hexa format, some has the longer format(not welcomed to me).
My feeling is that m-audio allows only 48.K for spdif out, requiring your own verification.

But anyway I would like to test what you said, resempling the file, could you explain to me how exactly, because I am sorry I did not quite understand all the step to get there...
About the files I have I think there are good, I downloaded a lot of DTS-WAV files... And all gives the same behavior (except only when I uses WinDVD when the files are burned).

foobar2k can do it in a step(also ffdshow audio filter dynamically can do it if the DTSWAVSource works in graphedit).
foobar2k requires the file extention to be dts for the dtswav(rename it).
In the preference->playback->DSP Manager, activate a resampler.
click the resampler you activated in the DSPManager>filters and set it to 48000.
Open the renamed source and test it using the KS(if it works with you).
If you want to save to a 6ch float wav, stop the play,right-click the title and choose convert->settings, choose output preset to any wav format and be sure to check the processing->use DSP(where you set the resampling), fill other parameters as you want, save all, close, right click again and choose convert.

This thread (http://forum.doom9.org/showthread.php?s=&threadid=84915) contains more information for transcoding with foobar in a step.

Skeul
30th March 2005, 08:32
Rockaria,

Ok, two more question...

First, what exactly am I trying to convert the dtswav (44.1) to ?
- pure dts (44.1 or 48) ?
- dtswav (48) ?
- any other stereo format ?

Second, what I really don't understand at all... When I transform my dtswav to dts using the last besplit (0.9b7 2nd) I got some file which sound perfect in analogic (rigth speed etc.) but not in optical ? That is what I don't understand, why is it different ? I would understand if both playback were messed up, but it is not the case...

So the final solution to my problem, be able to read my dts-wav file in one way or another on my dts amp to by converting them ? Is that right ? But why should that be a problem only for me ? And not for everyone reading dts-wav file ? Is it because the resampling thing of the sonica ? Which actually if I beleive other guys from other forum the sonica is not doing any resampling on the spdif....

Alex

Rockaria
30th March 2005, 10:08
First, what exactly am I trying to convert the dtswav (44.1) to ?
- pure dts (44.1 or 48) ?
- dtswav (48) ?
- any other stereo format ?

48K 16bit fixed(original) or 32bit float wav(for upsampling and encoding).
Play them using AC3Filter Enabled player and check the spdif passthrough option.
The ac3filter will encode the wave to AC3 dynamically and redirect it to spdif in 48K AC3 format.
If the M-Audio allows the encoded AC3 to be sent to the receiver decoder, you are getting the solution.


Second, what I really don't understand at all... When I transform my dtswav to dts using the last besplit (0.9b7 2nd) I got some file which sound perfect in analogic (rigth speed etc.) but not in optical ? That is what I don't understand, why is it different ? I would understand if both playback were messed up, but it is not the case...

I have also experienced the besplit incorrectly stripping some dtswavs. That's why I am suggesting you to use VLC/Foobar2K to decode to 6ch wav.
If some DTS files play correctly but not through the spdif, I believe its because of the m-audio sonica which seems to be picky on the sample rate.
That's why I suggest you to upsample to 48K wav which can be soft-dd-encoded to 48K AC3 to be passed to the spdif out before encoding to any compact formats.
If encoded in AC3/DTS format, AC3Filter will direct it to the spdif. Otherwise it will be encoded to AC3 to be passed through.


So the final solution to my problem, be able to read my dts-wav file in one way or another on my dts amp to by converting them ?
not directly(correction : foobar2k upsampled stream cannot be directed to spdif without encoding to ac3 dynamically),
ffdshow audio filter and ac3filter have the AC3 dynamic encoding function.


Is that right ? But why should that be a problem only for me ? And not for everyone reading dts-wav file ? Is it because the resampling thing of the sonica ? Which actually if I beleive other guys from other forum the sonica is not doing any resampling on the spdif....

I've tested them. I am using nForce soundstorm which accept only 48K digital streams to my receiver. I think everything is quite similar except that you read somewhere m-audio sonica not up-sampling. But who knows until tested and verified by oneself.

I think you need to test it to narrow the suspects not sacrificing the original quality too much.
Upsampling and encoding will decrease the fidelity(but not in decoding to wav).

Skeul
30th March 2005, 11:39
48K 16bit fixed(original) or 32bit float wav(for upsampling and encoding).
Play them using AC3Filter Enabled player and check the spdif passthrough option.
The ac3filter will encode the wave to AC3 dynamically and redirect it to spdif in 48K AC3 format.
If the M-Audio allows the encoded AC3 to be sent to the receiver decoder, you are getting the solution.


ok great, can that be done all dynamicaly ?
What should be the input for ac3filter ? A dts-wav at 48Khz and 16Bit fixed ?
I know how to get those 6 mono wavs and upsamples them to 48Khz... But how do I get them back together to a DTSWAV at 48Khz .. because when I used surcode I could only get to make dtswav at 44.1 Khz.
If the solution works it's great, but It will take me a hell of a long time to convert every dtswav I have... :(


That's why I suggest you to upsample to 48K wav which can be soft-dd-encoded to 48K AC3 to be passed to the spdif out before encoding to any compact formats.

I am sorry I don't understand what it meens... Can you re explain please ?


I've tested them. I am using nForce soundstorm which accept only 48K digital streams to my receiver. I think everything is quite similar except that you read somewhere m-audio sonica not up-sampling. But who knows until tested and verified by oneself.


How can I be sure of that ? How can I test it myself ?

Thks for everything.

Alex

daphy
30th March 2005, 12:50
I start believing it depends on the sonica drivers
From the M-Audio Homepage
Release Notes 1.2.2.0:

Windows 2000,Windows ME,Windows XP

Includes new hotfix that resolves all known issues with SPDIF AC-3 output and video frame dropouts with applications such as Sonic Cineplayer, WinDVD, PowerDVD and others.

update to Sonica 1.2.2.0 (http://www.m-audio.com/index.php?do=support.driverdownload&ID=189)

Rockaria
30th March 2005, 13:12
ok great, can that be done all dynamicaly ?
What should be the input for ac3filter ? A dts-wav at 48Khz and
16Bit fixed ?

No, it's PCM 6ch WAV decoded and upsampled from foobar2K.
The input to foobar is DTSWAV(change the extention to dts)
You can play the PCM WAV to spdif through AC3Filter(passthrough)
You may want to encode the PCM WAV to any compact format if the spdif play is successful.
The decoding does not take long, upsampling a bit, encoding takes somewhat longer.

dividing them to 6 mono wave will be useful for DTS/DTS encoding.
softencode will take the pcm 6ch wav.
for other formats you can use foobar transcode plugins easily( even the whole process in a step).

I am sorry I don't understand what it meens... Can you re explain please ?
You seem to be worrying too much before taking any action:).
It is making the conversation longer and redundant...:p

Have you tested any 48K dts or ac3 spdif on your m-audio yet?
You have just read from somewhere about m-audio no-upsampling without any tests personally, right?
Do it with the method I suggested or playing any DVD movie(mostly 48K) using powerdvd spdif. You will know if it works with 48K DD/DTS.
If it works, you can decode & upsample & encode them to any format you want.

Just do it.

[edit]
Good news daphy. I hope it supports 44K spdif out.

Skeul
30th March 2005, 13:46
First those are the driver I am already using they have not been updated since 2003 !!! We may think the sonica is no longer supported ... :(

Ok so basically what I have to do is to decode the dts-wav into a 6 ch PCM WAV at 48Khz and play it with AC3Filter (I'll probably use graphedit to be sure of what I am doing).
If this plays succefuly I just need to reencode that a DTS or AC3 files (I would rather prefer DTS) but still at 48Khz with the softencode program (that I didn't know unti now :).

And yes I have already tested the 48K dts and ac3 over spdif. I have a lot of dvd containing dts and/or ac3 sound track, and all of them works perfectly using spdif output with powerdvd, windvd, theatertek, zoomplayer but I could never get VLC to activate the spdif output (I have no idea why).


You seem to be worrying too much before taking any action.
It is making the conversation longer and redundant...

I am really sorry for that. Is not so much that I am worried, it is just that I have Internet only at work, and I can test the stuff only at home, so I really want to know exactly what I have to try back home... I have already lost so many days since I am trying all this.

I'll test everything tonight :)
Thank you a lot for all your advices.

Alex

Skeul
30th March 2005, 15:11
Ok, I couldn't wait to try...
I downloaded the last Ac3filter, graphedit, foobar2k, foo_dts and TheRun.wav from http://www.sr.se/multikanal/index.stm.
I played the file and transforme it with foobar2k. I got a new file (still a wav of course but a PCM) containing those 6 channels upsampled to 48k.
Then I used graphedit to build the appropriate graph: Wav Input => WaveParser => Ac3Filter => Default DirectSound Device.
And the file's playback is good but ouf course I don't have any spdif output at work so I am listening in stereo.
But I think so fare it's going good.
I looked a bit on Soft Encode website... It is expensive wow ! I tell me if I am wrong but I can't encode dts ? Only ac3 ?
How can I reencode back to dts but in 48K ? Because I could not find any options to do so with surcode is seems to me that it will only output 44.1k no ?
And when it is reencoded with what should I play the new DTS-WAV 48k file ? Can I play it with DTSWavSource ? Because until now I couldn't get foobar to activate the spdif (ks does'nt work for anything else than basics files such as mp3, it says "ERROR (foo_out_ks) : KS output error: error opening device.")
Shoud I then transcode the DTS-WAV 48k to Pure DTS 48k using BeSplit for exemple ? to be able to read it with power dvd at the rigth speed this time ?

Alex

Skeul
30th March 2005, 15:35
I have a last question, how Ac3filter know from the 6ch pcm wav file which channel is which (L, R, C, LS, RS, LFE) ? Is there some info about that is the wav header ? It is just a general wandering...

I think I am getting closer and closer :) Maybe soon I will be able to sleep again :p

Alex

Rockaria
30th March 2005, 19:22
Divide the 48K pcmwav to 6 mono wavs to encode to dts, it accepts 48K mono wavs for 48k dts/dtswav(if 48k dtswav makes sense).
splitter can messup the channel orders. follow each instruction in the forum.

It's abnormal that ac3filter failed to soft-dd-encode-passthrough.
I believe tons of dvds you've played are mostly 48k ac3 and some 48K dts.
Sure it would be worthwhile testing with encoded 48k dts.
Play it with powerdvd(at least it played passthrough).

I had no problem playing the pcm 6ch waves in correct channel orders on all the players. Only some encoders messed up the channels.

Now, I perfectly understand your work environment.
thanks.

Skeul
31st March 2005, 12:10
Ok I am finally getting through this...
I can listing perfeclty to the 6ch wav file at 48K using AC3Filter with SPdif output with no speed pb !!!
But weirdly the the same file at 44k (I made some test) can be read in spdif (I would say normal if it is an issue comming from the sonica) but I analogic it sounds awfull, distorted, very bad quality... Bahhh anyway, I got the 48k version working fine :)
And I got myself a version of SoftEncode to try AC3 encoding, it works fine too exept that the channel order is no right at all.
But like I said before, I actually wanted to keep the dts encoding...
And I find out why I couldn't encode 48K Dts with surcode, it is because I have the Surcode DTS Audio Pro, which only allows 44.1k encoding...
So now I have to find a version of Surcode DTS DVD Pro which as they say on there website always 48K encoding.

But here is few question:

- Can I correct that channel inverting position automatically (and not manually through softencode ?), I saw that there was a corrected version of foo_dts will it correct my pb ?
- When I use foobar to create a 6ch wav file is it significatively better to encode wav to 32Bits floating ? Or too leave the 16Bit fixed ? I would say 32 bits because If i am right DTS are encode at 20Bit ?
- The last thing is since I am willing to reencode DTS, I have to use that Surcode DTS DVD Pro version, but it won't accept 6ch wav... It has to take 6 mono wavs (crap !). So what is the best method of getting those ? Use foobar to get the 6ch wav and they transcode to 6 mono wav or is it good (in terme of quality, time etc.) to use Besweet with the following command line: "D:\Public\BeSweet v1.5b25\BeSweet.exe" -core( -input "D:\Public\DTS-Track-001(dts-wav).wav" -output "D:\Public\temp\DTS-Track-001" -6ch -logfilea "D:\Public\BeSweet.log") -ssrc ( --rate 48000 )

I promess I will very soon stop bothering you :)

Alex

daphy
31st March 2005, 13:43
Hiho,

long answer:D:\Public\BeSweet v1.5b25\BeSweet.exe"

first, you should update BeSweet to b29 (or use my installer from needfulthings), afaik there was an issue with this version with channel mapping stuff :rolleyes:
So now I have to find a version of Surcode DTS DVD Pro which as they say on there website always 48K encoding.
it can do both 44.1 + 48
- Can I correct that channel inverting position automatically (and not manually through softencode ?), I saw that there was a corrected version of foo_dts will it correct my pb ?
AFAIK the channel mapping of DTS is diffrent to AC3, but you can work with mux list ->
DTS->6WAV->6 singleWAV->MUXLIST(with corrected mapping)->AC3(softencode 6WAV input); step 2-4 ´shouldn´t take to much time)
for simply preparing those files for DTS encoding step 4 isn´t needed.
- When I use foobar to create a 6ch wav file is it significatively better to encode wav to 32Bits floating ? Or too leave the 16Bit fixed ? I would say 32 bits because If i am right DTS are encode at 20Bit ?
yep,should work - the question stays what´s the use at this point :confused:
BeSweet.exe" -core( -input "D:\Public\DTS-Track-001(dts-wav).wav" -output "D:\Public\temp\DTS-Track-001" -6ch -logfilea "D:\Public\BeSweet.log") -ssrc ( --rate 48000 )

out of Brother John´s bible ;)

3.1.3 Destination Format Switches

Apart from exceptions (-6chogg, -wavmp3, -ddwav) these switches are only
needed when no encoder section is defined. In most of those cases BeSweet won’t
be used for transcoding, but for decoding or sound processing only.

-2ch
Creates a 16 bit stereo wave. Channel order is: FL, FR.

-6ch DTS
Creates six 16 bit mono waves, one for every source channel. If the source lacks
a channel, the corresponding wave will contain silence.
Assumed channel order is: FL, FR, C, LFE, SL, SR.

-6CH AC3
Creates six 16 bit mono waves, one for every source channel. If the source lacks
a channel, the corresponding wave will contain silence.
Assumed channel order is: FL, C, FR, SL, SR, LFE.

-6chfloat DTS
Creates six 32 bit floating point mono waves, one for every source channel. If the
source lacks a channel, the corresponding wave will contain silence.
Assumed channel order is: FL, FR, C, LFE, SL, SR.

-6chwav
Creates a 16 bit multichannel wave.
Channel order of this file is: FL, FR, C, LFE, SL, SR.

please check the mapping and correct me if I am wrong.

Skeul
31st March 2005, 13:56
Ok I got it ! Always use the last version... I should not that :)

But you didn't tell me, what is the best way ? foobar2k or Besweet ?

And the last thing, is there a ways to batch the whole encoding process through dts ? I know how to do that from wavdts to the 6 monowavs... but I have no idea how to automaticaly use surcode to encode back to dts...

I have a lot of dts-wav file, I am already making nightmares to convert them all...

Alex

daphy
31st March 2005, 14:11
And the last thing, is there a ways to batch the whole encoding process through dts ? I know how to do that from wavdts to the 6 monowavs... but I have no idea how to automaticaly use surcode to encode back to dts...
I´ve never tried this but if you use the mux list as input and DTS as output in the BeSweetGUI ... :rolleyes:

Rockaria
31st March 2005, 18:09
For the channel order :

What is certain is that if decoded through the same decoder(the header format), they will have the exact same order for any time/any source.
Softencode projects remember the channel settings, and what is best than other encoder is that you can see and play each channel to verify where it belongs. Once set correctly you can use the project for the same-decoder-pcm-wavs.(once set with dts tuning disk, you can use the project repetedly for the same decoder output)
Another thing worth to remember is the name of the pcm 6ch wav in the project. If you use the same file name, the next launch of the softencode will read & rebuild the peak info automatically and be redy to encode(check the preference not to keep the peak file).
One more thing for your start-from-scratch spirit is softencode also accepts 6 mono and can do batch. you can compare the mono waves to the same version of 6ch pcm wav orders and reflect the order to the besweet mux file.(once set for the same type and file name, you can use it with minimum editing).

For the encode format :

One step deeper decoding for any postprocessing(resampling...) is thought necessary(at least by me).
One latest format to notice is AAC+HE which seems to be best for both playing and keeping for later transcoding to other formats.
What is better is you can perform the entire process(including the transcoding) in a step with foobar(check the link mentioned before) minimizing the quality loss/ch order(with foobar, you can get 48K AAC).
But for the playback, you will have to use AC3Filter until the receiver supports the format.
Thus for the minimum cpu use/quality loss, DTS will be the best format.

The batch process is your hands' favorite.

Good to see your fast/right(?) progress.:p

Rockaria
31st March 2005, 20:06
Originally posted by Skeul
But weirdly the the same file at 44k (I made some test) can be read in spdif (I would say normal if it is an issue comming from the sonica) but I analogic it sounds awfull, distorted, very bad quality...
We seem to have skipped some things. The Sonica might be supporting 44.1K spdif out.
So bestplit drop-trimmed dts might be able to be passed through without soft-dd-encoding(with least cpu time & no quality loss).
The 44.1K 32bit float pcm wav can generate the distortion, check the sonica spec.

Skeul
1st April 2005, 08:51
Ouupsss I meant "can't be read in spdif but it can in stereo with crappy sound"...

Sorry, my mistake.

But anyway I think I will convert to aac 6ch, it plays great using Ac3Filter.
I will configure zoom player to get a proper playback.

Anyway thank you guys for your help.

Alex

ps: for daphy, nevermind about my pm I found what I was looking for.