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Old 17th August 2011, 01:56   #11181  |  Link
ramicio
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Any time I get movies with 6.1 DTS-HD tracks the resulting file is much longer than what it's supposed to be, and the sound is bad/wrong. This is with the Arcsoft decoder.
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Old 17th August 2011, 07:13   #11182  |  Link
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Quote:
Originally Posted by ramicio View Post
Any time I get movies with 6.1 DTS-HD tracks the resulting file is much longer than what it's supposed to be, and the sound is bad/wrong. This is with the Arcsoft decoder.
You need dtsdecoder.dll version 1.1.0.0 to properly decode 6.1. However 1.1.0.0 does NOT work correctly with 7.1 (* on non-standard channel mapping only), for that you need version 1.1.0.8. When you switch versions you will need to re-register ASAudioHD.ax by entering this into the command prompt:

Code:
regsvr32.exe ASAudioHD.ax
Both versions work with regular 5.1, and to my knowledge there is no need for any other versions.

Last edited by nibus; 17th August 2011 at 12:00.
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Old 17th August 2011, 09:30   #11183  |  Link
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Quote:
Originally Posted by nibus View Post
... However 1.1.0.0 does NOT work correctly with 7.1, for that you need version 1.1.0.8.
Please can you explain for what don't work 1.1.0.0 with 7.1, works fine for me.
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Old 17th August 2011, 11:02   #11184  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Please can you explain for what don't work 1.1.0.0 with 7.1, works fine for me.
I think I read earlier in this thread that the channel mapping was off... I'll see if I can dig the specific post up.

edit: Here it is-

http://forum.doom9.org/showthread.ph...84#post1469584

Quote:
Originally Posted by TDiTP_ View Post
Yes, we need switch in this situation.
In case of non-standart scheme decoder 1.1.0.0 give garbage in some channels. So, i tried other two versions and they are decoded properly. If you have DTS-HD M.A.S. you can try yourself. If not, try decode test sample of DTS-HD 7.1 (scheme 2) and compare results.
AFAIK, there are only two differences between 1.1.0.0 and 1.1.0.8:
- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standart 7.1
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standart 7.1
Both decode DTS(-HD) 1.0 correctly, unlike 1.1.0.7.
Only for non-standard channel mapping.

Last edited by nibus; 17th August 2011 at 11:23.
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Old 17th August 2011, 13:52   #11185  |  Link
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I know we're not supposed to ask where to get these filters, but I'm going to just ask rhetorically...where are we supposed to find all these random older versions of this filter? Is there one version that works perfectly with every possible layout? Someone really needs to get on the ball and get DTS-HD decoding into libavcodec.
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Old 17th August 2011, 14:07   #11186  |  Link
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1.1.0.0 work also with non-standard channel mapping without garbage.
Like don't exist channels Ls-Rs in WAV make a remux to BL-BR correct to my opinion.
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Old 17th August 2011, 15:33   #11187  |  Link
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It's almost getting to be more worth the time to just fake-header a DTS-HD file and let the DTS suite decode the files. I really find that having to switch filter versions to be inexcusable (on Arcsoft's part, not madshi, he makes awesome stuff). I have no idea how I would make a 6.1 wav, though. There is no documentation on channel masks for wavavimux.
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Old 17th August 2011, 18:14   #11188  |  Link
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Quote:
Originally Posted by ramicio View Post
It's almost getting to be more worth the time to just fake-header a DTS-HD file and let the DTS suite decode the files. I really find that having to switch filter versions to be inexcusable (on Arcsoft's part, not madshi, he makes awesome stuff).
I agree. However I do not have the entire recipe for adding a new header (AND a new footer) to a "naked" DTS-HD file — would you mind sharing yours?

Quote:
I have no idea how I would make a 6.1 wav, though. There is no documentation on channel masks for wavavimux.
Channel Mask for "normal" 6.1 audio = 319.

Code:
CHANNEL NAME --- Decimal Value
ŻŻŻŻŻŻŻŻŻŻŻŻ     ŻŻŻŻŻŻŻŻŻŻŻŻŻ

FRONT_LEFT             1
FRONT_RIGHT            2
FRONT_CENTER           4
LOW_FREQUENCY          8
BACK_LEFT              16
BACK_RIGHT             32

FRONT_LEFT_OF_CENTER   64
FRONT_RIGHT_OF_CENTER  128

BACK_CENTER            256
SIDE_LEFT              512
SIDE_RIGHT             1024

*************************************

TOP_CENTER             2048
TOP_FRONT_LEFT         4096
TOP_FRONT_CENTER       8192
TOP_FRONT_RIGHT        16384
TOP_BACK_LEFT          32768
TOP_BACK_CENTER        65536
TOP_BACK_RIGHT         131072
RESERVED               262144
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Old 17th August 2011, 18:48   #11189  |  Link
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I make a mono wav that contains silence, make it substantially longer than the track we're trying to work with (we'll call this "the useless source") I need to decode, encode a track with the exact specs as the useless source, copy the first 140 bytes in a hex editor, not the last character and what its pattern of character looks like before that character, in the useless source, cut up to the last character of our nice header, and copy INTO the beginning of the useless source. It's very time consuming, but since it's a DTS tool, and all things most likely come from the tool, there can't be anything it can't handle.

Now I see the scheme for the channel mask, it's just sums! I never saw the chart you posted though, so thank you!

Last edited by ramicio; 17th August 2011 at 19:13.
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Old 19th August 2011, 05:00   #11190  |  Link
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Would there ever be any chance of taking multiple mono WAV files in to create a single multi-channel file? I can't find any decent and sure-fire way to do this with any other utility. WavAVImux to graphedit with wavdest, you are limited to a 4 GB output. Avisynth and VirtualDub, save WAV, can break the 4 GB barrier, but what about weird channel mappings such as 6.1? The there's wavewizard which I can't even begin to figure out, and have no idea about size limitations or channel mapping.
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Old 19th August 2011, 05:56   #11191  |  Link
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^ wavi.exe:

Code:
WAVI v1.06F - (c) 2k7 Tamas Kurucsai, with lots of help from tebasuna51
Licensed under the terms of the GNU General Public License.

This utility extracts the first uncompressed PCM audio track
from an AVI file and saves it to a WAV file. This is not quite
useful for most AVI files since they usually contain some kind
of compressed audio, but it can come handy when it's needed
to save an audio track from AviSynth.

Usage: WAVI <avi-file> [ <wav-file> [ /R | /X | /M <mask> ] ]

If <avi-file> is a valid AVI file which contains PCM audio,
the audio track will be written to <wav-file> as a WAV file.
If '-' is passed as <wav-file>, the WAV file will be written
to the standard output.
If <wav-file> is not given, only information will be printed
about the audio track.

On success, the exit code will be 0 and the first line printed
to the standard error will look like the following:

Found PCM audio: <c> channels, <r> Hz, <b> bits, <l> seconds.

where <c> is the number of audio channels, <r> is the sampling
rate, <b> is the number of bits per sample and <l> is the length
of the track in seconds.

If the audio track contains floating-point samples, the next line
printed to the standard error will be:
Audio track contains floating-point samples.

If an error occurs, the exit code will be 1 and some useful
error message will be printed to the standard error.

WAV files larger than 4 GB may be created. However,
such WAV files are non-standard and
may not be handled correctly by some players and encoders.
A warning will be printed to the standard error when such a WAV file is created.

WAVI accepts the following options:

/R - Write a raw file of samples without the WAV header.

/X - Write an extended WAV header containing the default
     channel mask for multi-channel audio.

/M <mask> - Write an extended WAV header containing the
     specified channel mask for multi-channel audio.

The default channel masks are:

Mask  Chan. MS channels                Description
----  ----- -------------------------  ----------------
   4   1    FC                         Mono
   3   2    FL FR                      Stereo
 259   3    FL FR BC                   First Surround
  51   4    FL FR BL BR                Quadro
  55   5    FL FR FC BL BR             like Dpl II (without LFE)
  63   6    FL FR FC LF BL BR          Standard Surround
 319   7    FL FR FC LF BL BR BC       With back center
 255   8    FL FR FC LF BL BR FLC FRC  With front center left/right

Some other common channel masks:

Mask  Chan. MS channels                Description
----  ----- -------------------------  ----------------
   7   3    FL FR FC
 263   4    FL FR FC BC                like Dpl I
 271   5    FL FR FC BC LF
  59   5    FL FR BL BR LF
Quote:
but what about weird channel mappings such as 6.1?
Why do you think 6.1 is "weird"? IMHO, the only "weird" audio channels are the infamous/Sonyc Front-Left-Of-Center and Front-Right-Of-Center.

Quote:
The there's wavewizard which I can't even begin to figure out, and have no idea about size limitations or channel mapping.
Me too hates wavewizard, you are not alone
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Old 19th August 2011, 09:53   #11192  |  Link
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WaveWizard works fine for me but don't fill the MaskChannel field (always 0) then you need some other util like:

Code:
wavfix
Usage:
 WavFix <input.wav> [output.wav] [-ignorelength] [-o #] [-m #] [-i/f #] [-c #] [-s #]

Where:
 <input.wav>  wav or w64 up to 8 channels, int or float, 8-16-24-32-64 bits,
              PCM/WAVE_FORMAT_EXTENSIBLE/W64, any extra chunks, allowed > 4GB.
              To use STDIN use - as input filename.

 [output.wav] If not present, input.wav is used suffixed with: _fix
              The header can be selected, see -o parameter.

 [-ignorelength] If present the length in wav header is ignored, useful for wav
                 > 2/4 GB. Problem: extrachunks at end of file treated as data

 [-o #]          Output header. 0=RAW, 1=WAV(Default), 2=RF64, 3=W64

 [-m #]          MaskChannel. Force output WAVE_FORMAT_EXTENSIBLE. 0 to default

If any of the next parameters are present the input is considered RAW:
 [-i #]          BitsPerSample Integer. Default 16. Valid 8, 16, 24, 32

 [-f #]          BitsPerSample Float. Valid 32 or 64.

 [-c #]          NumChannels. Default 2. Valid only 1 to 8.

 [-s #]          SampleRate. Default 48000 Hz. Any value is allowed.
Attached Files
File Type: 7z WavFix.7z (4.5 KB, 168 views)
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Old 19th August 2011, 14:55   #11193  |  Link
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Wavi doesn't work for me once an AVI file is created with wavavimux, even for ones that are under 2 GB. It used to, now it doesn't. It simply says there is audio in there, but it's not PCM. I don't understand how to use wavewizard, and I don't understand what that wavfix thing would even do. I just don't get how more people, or anyone at all has asked about eac3to being able to accept multiple mono wavs as input.
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Old 19th August 2011, 18:23   #11194  |  Link
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Quote:
Originally Posted by ramicio View Post
Wavi doesn't work for me once an AVI file is created with wavavimux, even for ones that are under 2 GB. It used to, now it doesn't. It simply says there is audio in there, but it's not PCM.
Thanks for useful the info --- I wasn't aware of that, because, until now, I had used wavi.exe only for .AVS files. Anyway, here is the workaround:

Code:
<intermed.avs>
DirectShowSource("audio-only.avi")
</intermed.avs>

wavi intermed.avs final.wav -M ###
Still, I don't know what you intend to do by creating a real multichannel .WAV greater than 4GB Applications like NeroAacEnc, WavPack, Aften, whatever, will happily encode from the stdin, and commercial compressors, by default, work with mono .WAVs

Last edited by Midzuki; 19th August 2011 at 19:56. Reason: grammar; overdue correction;; argh;;;
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Old 19th August 2011, 18:27   #11195  |  Link
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Because, as previously stated, I have problems with certain channel schemes of DTS-HD Master, and I'm not going to try to hunt down the internet for specific versions of filters that I'll never find. So I must use the DTS suite to decode these weird ones into mono wavs. My end goal is a 5.1 FLAC. So I need to feed eac3to a multichannel WAV to downmix into 5.1 channels.
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Old 19th August 2011, 19:18   #11196  |  Link
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Ah, I see, at last

Hmmm, IMHO there really is no good reason why eac3to doesn't support .AVI as input.

It's a shame that neither sox, nor wavi, can output .W64 or .RF64

Last edited by Midzuki; 20th August 2011 at 00:23.
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Old 19th August 2011, 19:34   #11197  |  Link
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That would be a nice step forward if it supported AVI as an input. But a lot of things still have trouble with opening an AVI with only audio.
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Old 19th August 2011, 21:18   #11198  |  Link
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Quote:
Originally Posted by ramicio View Post
... So I must use the DTS suite to decode these weird ones into mono wavs. My end goal is a 5.1 FLAC. So I need to feed eac3to a multichannel WAV to downmix into 5.1 channels.
eac3to can downmix the 7.1 L,R,C,LFE,Lss,Rss,Lsr,Rsr DTS channels to 5.1, but not 7.1 L,R,C,LFE,Ls,Rs,Lsr,Rsr.

How want you downmix the Ls,Rs,Lsr,Rsr to only 2 Surround channels? Please put the coefficients X1, X2 in:

SL = X1 x Ls + X2 x Lss
SR = X1 x Rs + X2 x Rss

I can make an avs file with your mix and can be used by BeHappy (or wavi) to obtain the 5.1 flac
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Old 19th August 2011, 21:26   #11199  |  Link
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I'm fine with the way eac3to handles 7.1 stuff. Never had a problem. It's these pesky 6.1 I come across now and then. I like eac3to because it detects clipping. Would it be possible for avisynth to turn out 6.1?
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Old 20th August 2011, 03:19   #11200  |  Link
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Does ffdshow decode 6.1 DTS-HD? If so you could open the DTS file via DirectShow, use ffdshow's mixer matrix to mix it to stereo, 5.1ch, or any way you like. From there, any encoder which will encode via DirectShow should be able to do the job.

I sometimes use ffdshow to mix to stereo (for those times when the audio track starts off stereo then switches to 5.1ch, and foobar's internal 5.1ch mixer has a hissy fit), while using foobar2000 to encode to MP3. Assuming ffdshow decodes 6.1ch DTS, I can't think of a reason why you couldn't use ffdshow to mix it to 5.1, then use foobar2000 to encode the output directly to flac.

Or, if you don't want to use ffdshow's mixer, you could leave it disabled and get foobar200 to spit out a multichannel wave file.

You'd need the foobar DirectShow decoder plugin.
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