DTSWAV DirectShow Filter released

[hypercube]

beware of your DTS source. As I said, DTS/WAV files must
start with the DTS/WAV sync word. If not, you have to skip
the partial audio frame until the next one.

when you rip the DTS stream from Laserdisc, you don't know
if your record start precisely at frame start.

I can publish a tiny tool to skip bad bytes from the begining
of DTS/WAV file.

are you interested ?
:D

[fritzie]

very interested !
:D

[Maxiuca]

Originally posted by hypercube
beware of your DTS source. As I said, DTS/WAV files must
start with the DTS/WAV sync word. If not, you have to skip
the partial audio frame until the next one.

when you rip the DTS stream from Laserdisc, you don't know
if your record start precisely at frame start.

I can publish a tiny tool to skip bad bytes from the begining
of DTS/WAV file.
:D

But besplit already does that... It cuts the first and the last partial frame... But DTSSource still does crash. I use your DTSWAVSource...

[hypercube]

you are sure of that ?

hi ho ! DSPGuru are you there ? :p

[DIggedy]

Hi Maxiuca thanks for the answers...

I'm currently working on it, but there are still many unanswered questions.... First thing that is a real problem is that the sound loses some of it's "surroundness" after being resampled and reencoded... These are probably only problems of encoding setting, but I'm not quite sure.


I'm guessing actually that the resampling would play a big part in that... The quality of the resample would be a huge factor in retaining the original fidelity. Once you have your 6 x 44.1khz wav files I would recommend using Cool Edit or Sound Forge to apply a high quality resample, it will take alot more processing time but the results would be worth it.
It's a shame there isn't a way of just tricking the authoring program into accepting a 44.1khz DTS stream to avoid resampling & reencoding altogether.

Question: when you reencode are you encoding back into full-rate DTS? Half rate DTS would have a loss in quality in comparison.

[Maxiuca]

Originally posted by hypercube
you are sure of that ?

I'm preatty sure. I've taken the lms.wav that you've been working on and I've removed manually the first and the last truncated frames. Then I converted it with both BeSplit and your WAV2DTS. The result had in both cases the same file size and was playable via DTSWAVSource but not via DTSSource.

I also converted the unaltered lms.wav (so with bad first and last frame) with besplit and the result had also the same file size like the two previous one and was playable in DTSWAVSource, not in DTSSource.

So I think that DTSSource has a bug...

--
Maxiuca

[DIggedy]

By the way thanks for the advice on the soundcard Maxiuca... I've got one on the way and it's only $50 delivered to my door! Can't wait to start my LD ripping!
PS. you have a PM.

[specise_8472]

Originally posted by Maxiuca

I also converted the unaltered lms.wav (so with bad first and last frame) with besplit and the result had also the same file size like the two previous one and was playable in DTSWAVSource, not in DTSSource.


--
Maxiuca

Remember that DTS wave is like other wave files - Byte Swapped!
I would expect a file that played in DTSWAVsource not to be able to be played in DTSSource.
Even the sync bytes are swapped.

[hypercube]

I my opinion, the DTSSource bug is related to...seeking ! :D
my filter does not implement seeking. it send DTS data
directly to the output without timestamp calculation.
May be DTSSource use the bitrate to compute timestamp.

You can use my DTS Parser to see the bitrate setting of your file:
it should be "open", may be DTSSource does not support "open"
mode. You can also see that frame size is the same than Surcode
files: 3584 bytes. This mean, the bitrate from laserdisc is
1411.2, like DTS audio CD, but bitrate value is "open", not "1411.2".
I can patch my DTSParser to force 1411.2 when bitrate is "open".
fixed DTS stream should be playable with DTSSource...

[fritzie]

I also concluded that the problem must be with dtssource.ax

Hypercube: If you could make a version that forces bitrate to 1411.2 I would be very happy. If it's not to much hassle of course.

[fritzie]

Oh I forgot:
The best solution (for me) would be to implement 44.1 input in HT3, then I wouldn't need any other tool. :D

Is it very difficult to implement that ?

[hypercube]

it is not so difficult. I plan to do this may be next week.

If you could make a version that forces bitrate to 1411.2 I would be very happy
I will try. the only problem could CRC calculation.
I have only one or two bits to change in the stream. :D

[cjaar]

I was trying to play this file
http://www.sr.se/laddahem/MultiKanal/Dts/BILJAKTEN_LOW_DTS_010925.zip
but i was not playable in windvd/pdvd.

I converted to dts with azidts, but the resulting file didnt play in windvd and pdvd said unknown formate. Tried wav2dts but no sound!!!!
the above clip is form a swedish radio(web site) i think its a car chasing...

can any1 help how do i play this file(i dont hv standalone dts decoder)

PS: how do i setuo graph edit??? I read this thead form the begning, and follwed but i couldnt write nor play... i hv all the ds filters installed, plz let me the graph sequence to play and write. When i drop the dtswav file in GE i see matrix-mixer and morgan steam switches added 2wice???? and no output

thanks
cjaar

[hypercube]

try my DTS parser (see my website)

[cjaar]

hupercube... ya i triend even that, but no luck, when i open the dtswav or dts file if says done... and i dont see any details about frmaes, bits etc....
i tried with other working dts file, am able see the details etc..

thanks
cjaar

[hypercube]

BILJAKTEN_LOW_010925.wav is perfectly played on my PC. :D
use my DTSParser,

check "Rebuild Stream"
click on "DTS File" button, then select
"DTS audio files (*.dtswav, *.wav)", do not select
"DTS audio files (*.dts)" !!

then wait...
BILJAKTEN_LOW_010925.wav.fixed.dts will be created and can be
played by graphedit with DTSSource and Intervideo Audio Decoder.

BILJAKTEN_LOW_010925.wav is perfect. there is no error in this
stream !
I don't understant how you cannot play it...may be a trial
expiration...
;)

[cjaar]

@hypercube...

i tried what u instructed, but no luck... i am unable to play the dtswav file and the fixed wav file :( ,in fact the fixed file was of size 0kb. my machine is xp-sp1 dx9. But i tried ddwav file and converted to ac3 with besplit and it worked fine.

cjaar

[hypercube]

may be the PATH of your file is big... and my code
is buggy. ;)

[sebus]

Fascinating read 11 years later!

Sorry to bring back this ancient thread, but just got round to capturing some DTS from LDs (using Reaper with ASIO drivers) & the resulting file (http://www.sendspace.com/file/oocdxkDTS) plays back perfectly fine via external amp using ie foobar2000 & SPDIFER plugin (http://www.ga.cba.pl/spdif_w7.html) or Reaper

AC3filter does not work.

Using AC3 filter, DTS light will flash on Amp once when it encounters single entry

7F FE 80 01 in the file

I could not find any way to process this file (besplit 0.9 b8 can not deal with this wave), so the resulting .dts can be muxed back to mkv OR be recorded as DTS CD

Surely by now somebody must know the answer?

Thanks

sebus

[hypercube]

indeed, all of this is far away for me now. anyway, I will take a look on your file, may be I will found something.
My DTSParser v2 is not able to read it actually... don't know what's wrong with it.

[sebus]

That would be wonderful! Thanks

sebus

[hypercube]

according to this: http://wiki.multimedia.cx/index.php?title=DTS

your file should contain one of these headers
7F FE 80 01 raw big-endian
FE 7F 01 80 raw little-endian
1F FF E8 00 07 Fx 14-bit big-endian
FF 1F 00 E8 Fx 07 14-bit little-endian

this is not the case. I think the way you capture is bad, and could introduce a displacement in byte words. for example, you capture in the middle of a word boundary. so everything is fucked up, BUT a hardware receiver working at bit level, should read the data, because he don't care about bytes.
most DTS parsers search the headers at bytes boundaries, not at bit level. This is certainly why none of them can read your file.
A solution could be to build a parser with the ability to search a dts header anywhere at a bit level. I don't have the time to do this. So, you should try to change the way you capture.
As far as I remember, LaserDisc DTS have already captured properly here, respecting bytes boundaries. And my DTSParser was able to read it.

[sebus]

The file does contains single 1F FF E8 00 07 at offset 2B8E6E (that is when the DTS light comes on while using AC3filter)

OK, I can only try a different capture card to see if it does anything different (but finding USB one really is a challenge, as there is not info around of what does actually work)

There are bits & pieces around on different forums, I beleieve one mentioned was EMU 0404

With Phonic, there is not much else I can think of that I could do

sebus

[Brazil2]

@Hypercube:

Your site is not working anymore
http://www.downforeveryoneorjustme.com/http://hypercube.is.dreaming.org
http://www.downforeveryoneorjustme.com/hypercube.is.dreaming.org

so where can we get this DirectShow filter ?

[hypercube]

it is 7F FE 80 01 at 2B8E6E.
anyway, this kind of header should appear many times.
here there is only one occurrence. that's certainly a
"false positive".

[hypercube]

my site is here: http://hypercube-softwares.dyndns.org/
but I don't support this dev anymore.

there is a mirror here:
http://forum.doom9.org/attachment.php?attachmentid=7972&d=1199902229

[sebus]

it is 7F FE 80 01 at 2B8E6E.
anyway, this kind of header should appear many times.
here there is only one occurrence. that's certainly a
"false positive".

I do not know... If I rip (with EAC) original DTS CD

Diana Krall - Love Scenes (DTS)[/URL]

(just one track for simplicity)

I get 7F FE 80 01 in this wav only ONCE.
Yet, it is recognized as DTS by Mediainfo and it does play in VLC & in fb2k with DTS decoder [URL="http://www.foobar2000.org/components/view/foo_input_dts"]foo_input_dts (http://www.audioholics.com/music-reviews/diana-krall-love-scenes-dts) , but could not get it to play with MPC or GE with DTSWAVSource & AC3filter (some settings somewhere missing)

Reaper will report it as:


Length: 4:41.026
Sample rate: 44100
Channels: 2
Bits/sample: 16 (int)
Total samples: 12,393,276
Blockalign: 4
Datablock start: 44


sebus

[Brazil2]

my site is here: http://hypercube-softwares.dyndns.org/
but I don't support this dev anymore.

there is a mirror here:
http://forum.doom9.org/attachment.php?attachmentid=7972&d=1199902229
Thanks but it looks like DTS Parser is not a DirectShow filter at all and I can't find the filter on your site.

[sebus]

Found one around (attached)

sebus

[hypercube]

oops sorry, my mistake...

[Brazil2]

Found one around (attached)
Nice, but please could you upload it to http://www.tinyupload.com or http://www.zippyshare.com ?
Because it usually takes several days before attachments are approved on this forum.

[sebus]

Sure, here (http://s000.tinyupload.com/index.php?file_id=02502689692536383047) it is.

If you get it working, let me know which way

[Brazil2]

Thanks for the upload :)

But I can't get it to work, not even with DTS-WAV files which are playing fine with VLC and foobar2000, not even on my XP computer trying several different combinations of filters and decoders within GraphStudio :(

[sebus]

So it is exactly same as here then...:(:(

But default (without any tweaking) I have in GraphStudio:

.dtswav (via quartz.dll) --> wave parser --> AC3filter --> Default sound device

DTSWAVSource does not get used at all

After installing interVideo.Audio.Decoder DS filter version 10.0.4.251 from "interVideo.Audio.Decoder.v10.0.4.251.exe"

I can do GE like this:

DTSWAVSource --> interVideo.Audio.Decoder --> Default sound device

but get nothing (no sound

What can actually connect to DTSWAVSource OUT pin? (apart from wave IV audio or AC3 filter?)

[tebasuna51]

But I can't get it to work, not even with DTS-WAV files which are playing fine with VLC and foobar2000, not even on my XP computer trying several different combinations of filters and decoders within GraphStudio :(

I can't understand for what you need to play dtswav files.

These files must be used only to burn CD's, and you must play these CD's in a CD/DVD/BD player not inside PC's.

To play in PC's or standalone players, like Xtreamer and others, you can use regular DTS files without problems to play and with less size (14/16) than dtswav's.

[sebus]

No need, just fun really (at least for me)

sebus

[Brazil2]

Same here, no need, just curiosity and the wish to hear music rather than noise in DirectShow players.

[sebus]

Sorry to bring back this ancient thread, but just got round to capturing some DTS from LDs (using Reaper with ASIO drivers) & the resulting file (http://www.sendspace.com/file/oocdxkDTS) plays back perfectly fine via external amp using ie foobar2000 & SPDIFER plugin (http://www.ga.cba.pl/spdif_w7.html) or Reaper

AC3filter does not work.

Using AC3 filter, DTS light will flash on Amp once when it encounters single entry

7F FE 80 01 in the file

I could not find any way to process this file (besplit 0.9 b8 can not deal with this wave), so the resulting .dts can be muxed back to mkv OR be recorded as DTS CD

Surely by now somebody must know the answer?


Eventually after week of reading ALL around about it I got a hit.

The person that noticed the "obvious" was Jerome Martinez (https://sourceforge.net/u/zenitram/) - MediaInfo author!

The channels were reversed (Left and Right channels inverted). No idea what caused it (hardware - LD, digital out mod to the LD years ago, hardware - Digitrack, software - usb-audio.de drivers, software - Reaper).

But once the channels order was reversed in Audacity (easiest) the it becomes proper dtswav (http://www.sendspace.com/file/4pofxc)

And besplit works perfectly with it producing DTSDigitalSound-16bit-reversed_channels_besweet_WORKING.dts (http://www.sendspace.com/file/1xr2wv)

And also wav2dts makes a MESS out of it!

So issue SOLVED!

sebus