NEW CODEC! 128 kbps and less for 5.1 surround encoding for MPEG4 (Xvid/Divx) movies

[3dsnar]

Dear All,

I am happy to annonce that we have
just finished a huge 5.1 surround sound
compression research project.


Software tools for encoding movies
with full 5.1 surround,
and
for real time decoding are available now.

www.aud-x.com
THE TECHNOLOGY IS FREE

I hope that you will
find it useful.
All comments are welcome :-)

Peace
3d

[Nic]

Perhaps you could post a little more about how the codec works?

You mention MP3 compatibility (on the website). Could you give us more details on how this works?
(i.e. is this a Multichannel MP3 implementation?)

You also state that the projects uses code from AC3Enc and LAME. You realise that these are LGPL projects and can only be linked to in a certain way? The rules are quite strict...(such as releasing object files of your project when not linking dynamically to the libraries as in Aud-Xs case, etc)

Thanks for any insight,
-Nic

[3dsnar]

> Hi, thanx for interest.

[QUOTE=Nic]Perhaps you could post a little more about how the codec works?

You mention MP3 compatibility (on the website). Could you give us more
details on how this works?
(i.e. is this a Multichannel MP3 implementation?)
> This is not not a Multichannel MP3 implementation.
> Some details related to the codec are available on the web page
> (whitepaper, etc).
> In general some additional information is stored in the mp3 stream,
> however the information is not related to mp3 encoding.
> It is in this way similar to mp3surround, where the multichannel panorama
> is reconstructed based on merged to stereo (or mono) multichannel stream
> and some phase information on how to reconstruct the full 5.1 sound during
> the decoding process are placed between mp3 frames. Frauenhofer stores energy related spectral
> information, which does not allow phase to be properly reconstructed.
> And phase is very important in surround perception... Thus our side
> stream (stored "between" mp3 frames) contains energy + phase info.

You also state that the projects uses code from AC3Enc and LAME. You realise that these are LGPL projects and can only be linked to in a certain way? The rules are quite strict...(such as releasing object files of your project when not linking dynamically to the libraries as in Aud-Xs case, etc)
> I must talk to the software guy who did the implementation.
> But thank you for suggestion! I will make sure that everything is legal.

[Nic]

Thanks :)

Don't know how I missed the whitepaper ( http://www.aud-x.com/images/stories/downloads/aud-x_whitepaper.pdf ). I did look for something of that nature but didn't spot it.

Good luck with your project :)

-Nic

[Sirber]

That would be cool in BeSweet too, so GUI developers could use itin their own app.

[3dsnar]

We will release some DLLs and info on how to use'em,
necessary for BeSweet and other apps.

I hope the project will gain interest among users, but also
among developers. If people really interested in including our codec
will contact us, we will provide the necessary tools :)

[Sirber]

Well, I wouldn't mind including your codec in RealAnime 4 (via beSweet) :D

[3dsnar]

ok, ok, ok.
Please give us couple of weeks to recover.
The last 2 months was crazy ;)
I will post some info when we will publish
the DLLsm

[3dsnar]

BTW. If anyone can help in any way
to popularize the software, that would be ofcourse great! :D
(among users, movie coders, etc)
Any ideas or suggestions?
Please write your thoughts.

[scharfis_brain]

besides the graphs of the computated difference of the Audio compressors, are there representative results from listening (ABX) tests?

[3dsnar]

Unfortunately not. Such tests are time consuming and expensive,
thus I hope that users will test the codec by their own.

[Sirber]

failed to register audxdshow.ax: error -2147024770

on XP SP2, AMD64 3000+, 1GB DDR400 DC.

[Sirber]

Wouldn't your codec be better using HE-AAC instead of MP3?

[3dsnar]

Yeah. Bugs are possible... This is
first release. Please try to restart your computer
and install the codec again. If this will not help,
please let me know.
----------------------
AAC issue:
Well there are problems with AAC and AVI file container...
Although we might do that in the future, because the
technology can be based on any audio compression method
(including also Ogg Vorbis)
This are matters for the future :)

[Sirber]

Sorry I can't restart yet, I'm in the middle of an encoding :) I will let you know later.

[tebasuna51]

Yeah. Bugs are possible... This is
first release. Please try to restart your computer
and install the codec again. If this will not help,
please let me know.
Restarted and reinstalling and same error.

[SeeMoreDigital]

Wouldn't your codec be better using HE-AAC instead of MP3?Agreed ;)

We could really do with some more 6Ch capable AAC-HE (and AAC-LC) encoders. Nero has been the driving force for far too long!


Cheers

[Sirber]

imagine... a 32kbps HE-AAC v2 + PS using their technology :D

5.1 at 32kbps :D

[bobcat56458]

Just thought I’d make mention that the Xvid\DivX 5.1 audio encoding using Aud-X is not limited to DVD’s you have ripped. It also works with as an example a DV home movie that you’ve made a DD5.1 soundtrack for using ursmatl’s V.I stereo to 5.1 VST plugin for Plogue Bidule, and a DD5.1 encoder. This also works with a captured MPEG2 file you've made a DD5.1 soundtrack for. Thanks 3dsnar for your useful tool\codec.

[falcon2000eg]

imagine... a 32kbps HE-AAC v2 + PS using their technology :D

5.1 at 32kbps :D:eek:
You are greedy;)
anyway it is a great idea,i will thier codec at 80kbps and see.

[Sirber]

Does someone got the DS filter work?

[3dsnar]

For those who have problems with registering the codec (DS decoder,
to be more precise). Please check if you can find the file: audxdshow.ax
in windows\system32
If yes, please try to register it manually with for example free DirectShowFilterManager software.
If the audxdshow.ax is not present in system32 at all, please download it
(attached to this post). I am sorry for problems and thank you for patience.
BTW. Did anyone try to encode movie with VirtualDubMod and Aud-X?

[guada 2]

@3Dsnar :)

That is very interesting.
But i noted that it was possible to make (MP3 HQ 192 Khz 48KHZ 5.1) in your guide, whereas you tell in your post that: "New codec! 128 K and less for 5.1"

It's true or no?
Aud-x 1.0 would it be better than the mp3SX 5.1 from Fruehaufer?
Can I use it with a lossless codec?

Thank you to clarify.
Bye. ;)

[3dsnar]

Well, I did not want to complicate the title of the new thread ;)
But in fact Aud-X offers 4 levels of encoding quality, mp3 backwards compatible.

1) STRQ - streaming quality -> 80 kbps (mp3 mono backwards compatible)
2) STDQ - (recommended) standard quality -> 128 kbps (mp3 mono backwards compatible)
3) HGHQ - high quality -> 192 kbps (mp3 stereo, Dolby Pro Logic II backwards compatible)
4) SPBQ - superb quality -> 192 kbps (mp3 mono backwards compatible)
--------------------------------------------
Please read more what is the difference between two 192 kbps options, etc.
In general (although I am not to objective here), our 128 kbps is significantly
better than Frauenhofer 192 kbps! Please read the whitepaper and listen to it by your own.
Our both 192 kbps are far more superior to mp3surround (also 192 kbps), and in fact when listened to the sound produced by HQHQ and SPBQ with 5.1 speakers, the decoded signal cannot be distinguished from the original sound
(based on our informal subjective experiments).
But I think that everyone should make a judgement and subjective evaluation individually :)

[3dsnar]

What do you mean by "use it with a lossless codec"?
Lossless codec for video compression?
Ofcourse, it can be used with any video codec, which can be accessed via
VirutalDubMod, or any other software supporting 6 channels for audio. The software must utilize ACM filters, because Aud-X encoder is such a filter :)
(beside the executable, which also allows to compress with Aud-X, but only separate 6 channel PCM wave files)

[CruNcher]

Has this reached the Hydrogenaudio Community yet ?
3dsnar you should really go there you find the real audio experts over there :)
they also can judge the codec and take it into the next ABX test http://www.hydrogenaudio.org/

[3dsnar]

I will do that. Thank you.

[dimzon]

Wouldn't your codec be better using HE-AAC instead of MP3?
Ogg Vorbis !!!

[bond]

imagine... a 32kbps HE-AAC v2 + PS using their technology :D

5.1 at 32kbps :Dsirber, there is no need to use proprieatary technology for this
mpeg is working itself on a standardised solution for "fake surround info" for aac

[dimzon]

sirber, there is no need to use proprieatary technology for this
mpeg is working itself on a standardised solution for "fake surround info" for aac
AFAIK "Mpeg Surround" standart is out!

[bond]

AFAIK "Mpeg Surround" standart is out!any link (dont confuse it with the proprieatary fhg mp3 surround...)?
if yes i wonder why nero hasnt announced them implementing this already :D

[dimzon]

any link (dont confuse it with the proprieatary fhg mp3 surround...)?
if yes i wonder why nero hasnt announced them implementing this already :D
:scared:
http://codingtechnologies.com/products/mpgsrnd.htm
http://www.hydrogenaudio.org/forums/lofiversion/index.php/t36355.html
http://forum.doom9.org/showthread.php?p=700616

[Sirber]

From CT: Combining the new MPEG Surround technology with aacPlus provides high audio quality at bit rates in the range of 64 - 96 kbps for 5.1 multichannel audio signals. This further gain of coding efficiency by a factor of 2-3 extends the existing aacPlus family of codecs towards an even more powerful standard suitable for all kinds of application scenarios from mobile music to home theatre in a forward and backward compatible way.

[bond]

:scared:
http://codingtechnologies.com/products/mpgsrnd.htm
http://www.hydrogenaudio.org/forums/lofiversion/index.php/t36355.html
http://forum.doom9.org/showthread.php?p=700616yes, but this doesnt mean it was standardised for aac, like parametric stereo (eg he-aac v3 or so :D )

[Morpheus_xx]

I'd like to thank you for this great codec!

It's the first (non-AC3) audio format, that my external DD-receiver recognizes as Dolby Digital!

Is this a "feature" of the compression format or more of the decoder?
If so, is there a AAC-decoder that also sends "AC3 format"-flag when decoding 5.1?

[scharfis_brain]

@Morpheus_xx: it is not a AC3-like format. it reencodes the decoded 5.1 audio stream during playback to AC3 and sends it via SPDIF to your Receiver.

@3dsnar: I am not able to append AC3filter to the filterchain, in order to downmik the decoded 5.1 to DPL2, even if I force AC3Filter within MPC.

[SeeMoreDigital]

@Morpheus_xx: it is not a AC3-like format. it reencodes the decoded 5.1 audio stream during playback to AC3 and sends it via SPDIF to your Receiver. Agreed....

I actually use FFdshow's audio filters to transcode 6Ch AAC-LC/HE, 6Ch WMA Pro, 6Ch Vorbis, 6Ch WAV and a few others to 6Ch AC3.... on-the-fly.

So it can be done ;)

[Morpheus_xx]

I actually use FFdshow's audio filters to transcode 6Ch AAC-LC/HE, 6Ch WMA Pro, 6Ch Vorbis, 6Ch WAV and a few others to 6Ch AC3.... on-the-fly.

So it can be done ;)

this is new to me and very interesting! can you explain it a bit more detailed hove to "on the fly transcode aac to ac3"?

[SeeMoreDigital]

this is new to me and very interesting! can you explain it a bit more detailed hove to "on the fly transcode aac to ac3"?This post should help get you started: -

http://forum.doom9.org/showthread.php?p=718812#post718812

And then there's this one, in the same thread: -

http://forum.doom9.org/showthread.php?p=718978#post718978


Cheers

[3dsnar]

Please use our front end, if you would like to transcode
Aud-X to AC3 or 6 channel wave (for tests).
--
Yes, we transcode the Aud-X stream to AC3 in real-time in our DSfilter
(so it goes: File -> Aud-X -> 6 ch PCM -> AC3 -> SPDiF)
I hope that this feature is useful (it should be especially for inexperienced users).
Please check the DSfilter structure in our whitepaper @ www.aud-x.com

[shon3i]

I can't register ax file. I try regsvr32 audxdshow.ax but i recivided message "LoadLibrary falied to register file is missing", but i find this ax file into windows\system32 dir

[3dsnar]

Please try to download DirectShowFilterManager.exe
And use it to register the filter. If it does not work,
please let me know.

[shon3i]

link to DirectShowFilterManager.exe

[3dsnar]

http://www.softella.com/dsfm/download/9CB9749C/DSFMgr.zip

[shon3i]

Thanks i try later

[3dsnar]

It seems that LoadLibrary
is missing rather than audxdshow.ax.
Therefore DSFMgr will not help (the same error)!
Maybe someone has an idea what to do:scared:

[Sirber]

3dsnar, Does it load on your side?

[Kurtnoise]

DShow filter works fine for me...[win XP Pro SP2]

Question: why your ACM codec doesn't appear in GraphEdit as Audio compressors ?

[SeeMoreDigital]

Question: why your ACM codec doesn't appear in GraphEdit as Audio compressors ?Did you manage to find it showing up anywhere else?


Cheers

[3dsnar]

Yes. Our ACM encoder can be accessed via
the modified Virtual Dub Mod,
The VDM_Aud-X_enabled is available from our web page:
(please check download subpage @ www.aud-x.com)
--
I do not know why it does not appear as audio compressor...
I will have to talk to the guy who wrapped my algorithms
in DS and ACM. He should return from Israel at the end of February :(
--
The filter registering problem is (I think) related to some versions
of windows XP, or maybe it is samewhat related to the firewall...
Honestly, I have no idea (I am not that much into Windows).
If anyone has any ideas, please help.
(BTW. I have tested our installer on plenty of computers,
and the reported problem seems to appear pretty rearly...
since I have never experienced such problems)

[Rockaria]

Beautiful work! Whether it is discrete or not, if the surrounds reconstruct with full(or resonable) fidelity, it's a strong substitution.

The DS filter works fine for me with WinXp Sp1. Not yet reviewed the white paper nor performed the encoding & listening tests, but looking at the Aud-X Program Files folder, it appears the aud-x.exe gui encoder is using the audx.exe cli encoder backend with no explicit parameter explanations.

Can we get any cli parameter option help? It will be useful for many existing encoders...

[shon3i]

It seems that LoadLibrary
is missing rather than audxdshow.ax.
Therefore DSFMgr will not help (the same error)!
Maybe someone has an idea what to do:scared:
I have same message. I using XP SP2. I tryed at my friend an i get same message. I don't know why.

[3dsnar]

I am suggesting to use VirtualDubMod_Aud-X_enabled
for encoding films :)
because this is the most convenient way to combine it with Xvid/DivX movies.
For separate audio files, ofcourse th exe frontend is the best choice.
---
For those who want to utilize our exe in their own apps, here are the command line complete commands (for encoding from 6 channel PCM to Aud-X. Please note that the channel order is of importance)
---
audx.exe STRQ FileNameWithExtention
audx.exe STDQ FileNameWithExtention
audx.exe HGHQ FileNameWithExtention
audx.exe SPBQ FileNameWithExtention
---
DECODING / TRANSCODING TO AC3
audx.exe PCM AudxFileNameWithExtention
audx.exe A128 AudxFileNameWithExtention
audx.exe A192 AudxFileNameWithExtention
audx.exe A256 AudxFileNameWithExtention
audx.exe A384 AudxFileNameWithExtention
audx.exe A512 AudxFileNameWithExtention
audx.exe A640 AudxFileNameWithExtention

[3dsnar]

I will try to find out how to overcome the problem with filter registration
and will report the solution ASAP.
I am sorry for this bug:confused:

[Rockaria]

Thanks for the reply. I apparantly missed one crucial issue.:)
Does the audx.exe encoding support stdin(pipe) as the input wav source?
In other words, can I replace the 'FileNameWithExtention' with '-' ?

Based on my test, it does not seem to support the pipe(linked from many different audio format decoders & frame servers) yet.

[3dsnar]

unfortunately not. The file name is necessary.

[Teegedeck]

I say, the whitepaper reads a bit like the technology is not very different from MP3-encoded Dolby-Prologic, only with some additional channel-info as 'pointers' of sorts. Is that so? Or do you use actual spatial encoding the way it will be implemented in HE AAC?

If the surround info is hidden in a channel by phase-shifting before it is encoded to MP3 that gives me an uneasy feeling. You know that the psycho-acoustic component of MP3-encoders (You use LAME?) doesn't react well to such tricks? Simply because MP3 isn't exactly meant to preserve such non-standard information.

[3dsnar]

Thanks for your thoughts.
The signal is merged to stereo/mono (identical precedure is in HE AAC and mp3surround)
and encoded with LAME. We did not change anything in LAME, i.e. we did not manipulate in the psychacoustic model in the mp3 encoder. Before merging the channels, some information describing how to 'unmerge' the signal are separately computed. And this is the algorithmic solution that we developed. Mp3surround and HE-AAC has the same approach (i.e. providing information for 'unmerging' and algorithm to carry out such operation).
As far as I am concerned these two technologies are focused on variations of binaural cue coding.
---
We used a completely different approach got preserving the spatial info.
The difference is that we put emphasis on phase distortions of the restored 5.1 sound (it happens after 'unmerging' the mp3 main stream to PCM and has nothing to do with LAME code).
----
Phase is extremely important for movie sound tracks!
Therefore we have mostly tested and tuned our codec on movie sound tracks mainly because of one very important reason
(I started to play with multichannel music,
but I decided to focus on movies only).
Very often in movie sound tracks channels contain the same sound components, but of shifted phase (phase shifts vary for different frequencies).
For example in movies like "Fifth Element" nice ambient spatial feeling at the begining was achieved based on phase shifts of some spectral components.
PHASE REALLY MATTERS FOR THIS MASTERPIECE OF 5.1 SOUND,
and as well for other movies with good surround audio.
-----
This makes coding spatial information EXTREMLY difficult, because
merging channels ends up in loosing the phase relations.
So after reconstruction, even when you reproduce energy relations between channels, phase gets completely distorted (I mean the phase shifts of spectral components between channels are lost).
I tried to show it in my whitepaper when I did the objective test
(I know this is not an appropriate way to compare codecs and subjective tests are much better, but this was the only thing that we could produce at that time).
And it shows phase and energy erros of our codec, AC3 for comparable bitrates and mp3surround.
Mp3surround is really poor in preserving phase.
So when you watch movies like "Fifth Element" on high quality
5.1 home theatre system, really alot of the spatial feeling is gone...
And we did everything to avoid this!
----
Interestingly the phase problem is not so important
for multichannel music (in opposite to multichannel movie sound tracks),
because the channels mostly differ in energy (if they have some common components).
Also, in subjective tests ONLY this type of audio (i.e. 5.1 music)
was used for evaluation of the encoders.
(HE-AAC and mp3surround)
(check below links with whitepapers)
http://www.chiariglione.org/mpeg/working_documents/mpeg-d/sac/RM0-listening-tests.zip
http://www.iis.fraunhofer.de/amm/download/flyer/dl.html?f=introduction_to_mp3surround.pdf
===
To summarize, our codec is dedicated to movie 5.1 (surround) sound tracks.
It would be interesting to see a subjective comparison
of Aud-X, mp3surround and HE AAC...
Maybe someone objective (some organization specializing in such tests, not the codecs scientists)
will perfm it...
IMO subjective tests for movie sound tracks and 5.1 music should be carried out separately,
since both differ in terms of production (utilizing phase shifts for certain spatial effects is used only in movies)

[Teegedeck]

So the approach is very similar to actual BCC, and closely related to MP3-surround if I understood that correctly. :) AFAIK encoding phase differences also is part of BCC.

Does the encoded signal of aud-x of also carry information about the channels' level difference, time difference, and coherence?

[3dsnar]

The phase info must be very limited in mp3surround (please see our comparison tests in our whitepaper),
http://www.aud-x.com/images/stories/downloads/aud-x_whitepaper.pdf
because the reconstructed 5.1 sound has very distorted phase.
Since it was US who did perform the comparison (in our whitepaper), and we are not very objective here ;)
you can use our front end to decode Aud-X stream (and for example winamp
to decode mp3surround) and perform the tests on your own (listening on 5.1 speakers set and objective evaluation in Matlab, or something like that)
---
Yes, ofcourse, Aud-X carries energy information related to various frequencies. Energy based error (also presented in the whitepaper) is also
significantly lower, than in case of mp3surround.
It should be stressed that we used very difficult movie sound track fragment,
for the test (fragment of Luc Besson's "Fifth Element"),
and maybe for 5.1 music the comparison would result in similar performance...I do not know.
Our goal was to provide a solution dedicated to 5.1 audio in movies :D

[dimzon]

Thanks for your thoughts.
The signal is merged to stereo/mono (identical precedure is in HE AAC and mp3surround)
and encoded with LAME.
LAME is LGPL software! Does You pay for it? Otherwise, please provide source code!

[3dsnar]

We did not modify LAME...and as you noticed it is LGPL.
Aud-X code is something separate
(I hope I explained it clearly in the previous posts).
Our software is a freeware

[dimzon]

We did not modify LAME...and as you noticed it is LGPL.
Aud-X code is something separate
(I hope I explained it clearly in the previous posts).
Our software is a freeware

1) Does it mean you can use any other lossy encoder instead LAME? Maybe You can provide some SDK to be able to connect Your algorytm to any encoder (read: Ogg Vorbis).

2) Can You provide full algorytm/bitstream specification (follow RED LINE from my signature to understand why I want it)

[3dsnar]

These subjects were exploited in this thread.
Please refer to the previous posts.

[dimzon]

These subjects were exploited in this thread.
Please refer to the previous posts.
I have re-read all thread again and dont found Your answers on
Maybe You can provide some SDK to be able to connect Your algorytm to any encoder (read: Ogg Vorbis).
and
Can You provide full algorytm/bitstream specification
You wrote:
BTW. If anyone can help in any way
to popularize the software, that would be ofcourse great!
(among users, movie coders, etc)
Any ideas or suggestions?
And I'm trying to help You! I understand why does you chose mp3 as carrier bitstream but I really believe Ogg Vorbis - based impementation woud be a really revolution!

[3dsnar]

Yes you are right. Cobining our technology with Ogg Vorbis would be great!
We really want to try it and talk to guys from Vorbis (anyone here?)!
Honestly, I am really exhausted by this project,
thus working on the next stage will be possible after couple of months
(I need to take care of my wife and kids, and finish my PhD ya know ;) )
But seriously. We are exactly going to do as you suggest (if Vorbis group is interested). I will keep everyone informed!

[shon3i]

Did you find solution for ax file

[3dsnar]

I am sorry, not yet.
I am going to work on this over this weekend.
Also, after the weekend I will update the installer,
so now using Aud-X will not exclude using FFDshow.

[shon3i]

OK Thanks

[shon3i]

You ain't gonna believe this. I'm managed to start the codec by deleting it first, and then, after restarting the computer a couple of times, I've reinstalled it and it worked. Conclusion: I think HE-AAC is better than this one. Why does DirectShow Filter overrides mp3 stream?

[kotrtim]

well, instead of movie, I tried to ecode audio CD with Aud-X. Aud-X 128 kbps sounded worst than QT LC-AAC 80 kbps!!!!
Why don't just drop Aud-X project, no point keep using mp3, since AAC is made by MPEG to replace mp3
Vorbis is superior to mp3, Besides, I can't figure out how to decode the aud-X wav file with a mp3 decoder.....
It's better to just merge/joint force Aud-X (Free) project with Vorbis (OSS).
Create an enhanced Vorbis "Vorbis 2" which have SBR, PS, "fake surround", optimized channel coupling for surround....isn't that cool? We must discourage people from using avi, Vorbis doesn't work properly in avi...
Encourage people to use MKV instead......

[Rockaria]

I find no point of mentiong the quality on 2 ch transcoding except the (dynamic) 5.1ch upsampling (from 2ch CD) through the DS filter or aud-x ac3 encoding which any aac solutions do not have.

It is basically a technology pseudo embeding & reproducing the multi channel stream into an existing popular format and looks a lot enhanced than the original mp3 surround solution from FH.
Of cource, if the vorbis encoder can keep the auxilary spatial information better than the Lame encoder, the vorbis can be considered a better carrier.

I am listening an original DTS, aud-x shrinked mp3 and aud-x encoded ac3 5.1ch music. No noticible difference found so far compared to the AAC HE VBR except a bit lower volume level possibly because of the different DS Filter decodings. Maybe I need some spectral image comparison tests on before & after images. The only trouble is this machine is not setup for such tests yet.

[vinouz]

3dsnar:
I didn't see any post on HydrogenAudio.org ? It's the place to get heavily tested/commented/advised for sound encoding.

Seems impressive. I'd like to test your 'Superb Quality' settings with an ogg q4 mono at the core.

And your internet quality on an ogg q0 mono core too...

kotrtim:
As for testing 128kbps Aud-x vs 80kbps LC/AAC, I find the comparison useless : probably Aud-x is using mp3 around 80-96kbps at the core, and surely it is easily beaten by LC-AAC on that segment. No wonder.

3dsnar again:
So, interesting proposal, but : what amount of bitrate does your scheme add (In the four different modes. At min, average, and max.) ?

I'm glad a new codec is developped, and I'm also glad it seems you have solid grounds on which to continue. Finally the presentation, the restricted mode choice and the application domain seem well thought (IMHO). And I wish such an initiative a good luck.

If it appears promising, I might as well mention it in my A/V compression courses.

[3dsnar]

Thank you all for tests an all the opinions.
Please note. What we were trying to do, is to
work on preserving the spatial feeling of 5.1 sound. Thus if our core mp3
stream sounds worst than core AAC (due to better sound quality of AAC than LAME mp3) it is a different story, and beyond the scope of our work.
--
This scientific experiment was meant to preserve the phase relations in movie sound tracks for low bitrates.
Therefore it is important to test it with a 5.1 home theatre system (or 5.1 system set) against
mp3surround or HE-AAC.
And I think that users of such equipment will appreciate our results.
--
As I explained previously, the reason to chose mp3 as a carrier was that
we wanted the Aud-X stream to be compatible with AVI. AVI is crap...but very popular. That is all.
--
It is very important to remember, when using our executable front-end for encoding Aud-X, to feed the encoder with a proper channel order signal!
Please refer to the information in HELP.
--
Please look carefully at Hydrogenaudio, because
I have submited alot of posts there, and the discussion
was interesting :)

[3dsnar]

For those who tested Aud-X on music.
1) Encoding stereo music converted to 5.1 makes on sense for two reasons.
The decoder contains pseudosurround algorithm, thus when you play a
stereo track, by chosing 5.1 output or SPDiF (for external amplifier), the
algorithm will be applied, and there is no need to encode such tracks
with Aud-X.
------
2) It is not dedicated to 5.1 music. Please try to find a nice movie sound track, which sounds really good on your 5.1 sound system and perform the tests.
------
Making the comparison you a stereo system (with headphones), or listening to each stream with headphones makes no sense! As I mentioned, we focused on recreating phase relations of various channels, which is important only on 5.1 speaker set. And this is the only reasonable comparison environment.

[3dsnar]

For those who performed the tests:
Please provide links to the tested sound (so I will be able
to focus on potential problems for future releases)
1) Aud-X signal
2) Competitive signal of similar bitrate
3) Original signal (used for encoding).
-----
Thank you.

[3dsnar]

The core bitrates that we use
1) STRQ 80 kbps -> base mp3 is 56 kbps mono (24 kbps for spatial info)
2) STDQ 128 kbps -> base mp3 is 64 kbps mono (64 kbps for spatial info)
3) HGHQ 192 kbps -> base mp3 is 128 kbps stereo (64 kbps for spatial info)
4) SPBQ 192 kbps -> base mp3 is mono 64 kbps (128 kbps for spatial info)
---------------------------------------------------------------------
So from above you can see that in fact SPBQ has no better base sound quality than STDQ. We decided that 64 kbps LAME 3.96.1 is good enough, and the rest of kbps should be used to recreate the 5.1 panorama and spatial feeling. This and only this was an objective of all the research that we carried out. Improving base psychoacoustic encoder was beyond the scope of our work (any did not modify LAME at all)

The emphasis was put on the recreation of phase relations (i.e. spatial feeling). Please
check out our whitepaper to see how SPBQ performs (in terms of phase and energy error) with reference to AC3 192 kbps and mp3surround 192 kbps.
http://www.aud-x.com/images/stories/downloads/aud-x_whitepaper.pdf

[tebasuna51]

About HGHQ 192 kbps option.

Decoded with a mp3 decoder I get a stereo output but with two equal channel (FL = FR), then is not a real stereo output. If this is always true, I think the users must know this issue (important for standalone players).

Decoded with Aud-X decoder, is there any reason to use HGHQ instead SPBQ quality?

[3dsnar]

No, this should be a normal stereo (otherwise we would not make this option
available at all):
To make sure, I just tested it and it is 100% ok.
So:
1) Maybe the supported (for encoding) stream order was not correct?
2) Maybe by mistake you used SPBQ instead of HGHQ?

[tebasuna51]

Sorry, is only for a 'special' test input.

In other test I obtain a correct stereo output with:
FL' = 0.4xFL + 0.2xC + 0.4xSL
FR' = 0.4xFR + 0.2xC + 0.4xSR

[3dsnar]

If you are so deep in the test :)
please check how much better is:
1) front vs rear channels crosstalk for HGHQ than in case of mp3surround
2) front vs rear channels crosstalk for SPBQ than in case of mp3surround
(i.e. in Aud-X there is no front vs rear channels crosstalk for SPBQ)

[Rockaria]

While performing the listening tests prior to any possible professional ABX tests, I found some usability problems making it hard to compare to other format dsfilters play. Overriding the dsfilter merits and filter orders in the players(I tested with MPC, gomplayer, bsplayer and zplayer) will allow choosing the dsfilter for the (aud-x) mp3/ac3 play without (un)installations. Also the new installation of the aud-x dsfilter does not show the illegal memory address error on normal mp3 upmix play any more.

Below are the check points I've tried to figure out while reviewing & testing for this new promising FREE codec.

<aud-x dsfilter decoder + alpha>
mp3(normal) -> decode + upmix/48k resample + apu rendering/ac3 encode & spdif passthrough
mp3s(aud-x)-> decode + apu rendering/ac3 encode & spdif passthrough
ac3 -> (decode/resample) + apu rendering/(ac3 encode) & spdif passthrough

1. indeed a complete set but with no DRC or gain control? cannot connect to FFDShow or AC3Filter to utilize the DRC or other DSPs
-> provide the DRC/gain/EQ control or allow connecting to other filters, the volume level is critical in perceived sound quality for normal ears.
2. cannot be used for AVISynth DirectShowSource frame source, looks too strongly tied to the sound (out) device.
3. what kind of an algorithm is used for the mp3 upmix, martrix. DPL or DLP II? can users choose or set the parameters?
4. what else formats can it decode besides the mp3, mp3s(aud-x) & ac3?

<aud-x cli encoder>
1. restricted to the wav file sources. when the duration is too long, i.e. over 1hr, the temporary disk usage would be a big problem.
-> allow std-in stream for the wav source for tons of existing cli decoders for transcoding as I said before
2. why the bit rate is lmited to 128k &192k CBR? even the nero AAC HE VBR streaming quality is around 33% bigger than aud-x.
-> allow VBR for non-avi environment
-> rescale(upsize) the bitrates on the quality classes(internet, streaming...) in real quality grades, complicated signals require bigger size for the same perceived quality.
3. would the lossless(FLAC) or hybrid(wav pack) formats be better as a container for the base & spatial data? If the signal is compressed & extracted through the lossy format, any loss in the spatial(& base) data would be critical in the reconstruction of the channels. It would be interesting to see how much space can be saved by using the lossless format, if any, for multi channel encoding.

<aud-x acm codec for the specific vdubmod only?>
1. as is pointed out, it does not appear anywhere else except the vdubmod audio stream codec selection.
2. also with the help of many mux tools, the movie transcoding does not need to perform the a/v transcoding together, making the standalone encoder more usable.(it worries me repetedly emphasized for the movie soundtrack use..)

If set up in the identical decoding(playing) environment except the decode function of the dsfilter, it would be an easy & resonable job to perform the ear-comparison without any biases(volume, DRC, EQ and many hidden other factors).

Regards.

[3dsnar]

Thank you for all the really valuable suggestion.
We will work on future improvements.
-----
For the subjective tests, please use our executable front-end
(to decode the Aud-X stream to PCM).
-----
Cheers, 3d

[Rockaria]

I have some result to share as for the listening test performed on aud-x mp3 and aud-x decoded wav compared to original dts-decoded wav.
Initially, I tested on my digital system(AN7(soundstorm) connected to ONKYO TX-L5) and an analog headphone(5.1 Kinyo in stereo dolby surround encode mode), which resulted in fairly good quality considering the low bit rate of SPBQ(192kbps CBR).

But when I decided to use the Kinyo in 5.1ch mode today, I finally could find the crosstalks all across the channels, which I could manage to minimize by setting the 'invert phase' for rear and center/lfe channels. HGHQ was better than SPBQ in my case incrosstalk-wise.
But the digital is gone bad, the sound space has changed/reversed/flattened with the invert phases!

There was no crosstalks for the original wav. Maybe a special care for the analog surrounds phase handling is reuired.

Regards.

[3dsnar]

OK. Thanks for the tests!
We has still some things to do (such as incorporating Prologic II)
etc. So your suggestions are very valuable for us.

> There was no crosstalks for the original wav.
> Maybe a special care for the analog surrounds phase handling is reuired.
Indeed! This also requires additional work (i.e. sounds encoded with Prologic).
Therefore we mus incorporate this technology in Aud-X.
We hope to do it before the next release.
--

[3dsnar]

BTW. I forgot to ask.

What should be the library format to incorporate Aud-X with BeSweet (and BeLight)?

[dimzon]

BTW. I forgot to ask.

What should be the library format to incorporate Aud-X with BeSweet (and BeLight)?
For BeHappy integration I need command line console encoder wich can handle data from stdin

[3dsnar]

OK, thanx for info

[Rockaria]

ac3 -> (decode/resample) + apu rendering/(ac3 encode) & spdif passthrough
...
4. what else formats can it decode besides the mp3, mp3s(aud-x) & ac3?
..
But when I decided to use the Kinyo in 5.1ch mode today, I finally could find the crosstalks all across the channels, which I could manage to minimize by setting the 'invert phase' for rear and center/lfe channels. HGHQ was better than SPBQ in my case incrosstalk-wise.
But the digital is gone bad, the sound space has changed/reversed/flattened with the invert phases!
It does not (seem) to play the ac3 anymore(sort of a conflict between the filters?). Instead, I want to mention it can also play 5.1 wav.
Recently, just before the aud-x test,I changed my test machine from au13 to An7, both soundstorm, which seems to have made the DAC & analog response change on the phase. It sounds better now with the invert phase checked in the nvmixer (hidden) option regardless of the decoder/player with the analog speaker.
But the analog crosstalks only happened with the aud-x transcoded & played mp3/wav/ac3 when the phase invert are unchecked on this machine.

Also I want to mention the rear encode & decode routine of the DPL II would be useful for the reference.

[3dsnar]

Rockaria,
thanx for the test.
Yes. I have disabled AC3, since people were complaining that Aud-X
replaces AC3filter. So to avoid such a conflict, I've decided not to decode AC3.
--------
The phase experiments that you've performed are interesting. However they may depend on the signal (phase relations in current signal). In general there is a channel cross talk for certain Aud-X encoding options (for example for STDQ = 128 kbps). For SPBQ and HGHQ it is significantly limited.
But the most important thing is that phase relations between channels are preserved (i.e. the crosstalk concerns the spectrum elements, for which phase and spatial perception are not linked with each other).
And in general this is what Aud-X is all about -> preserving phase relations, which are responsible for the spatial feeling.
---------
It is also advised to use original AC3 (or PCM generated from AC3), rather than DPLII decodec multichannel signal, because the problem with DPLII
is that it forces specific phase relations in the surround panorama.
Thus when you convert DPLII to Aud-X, you will not gain increase of quality, but rather more crosstalk (the original DPLII phase distortion will be preserved, since Aud-X tries to keep the inputs phase).
---------
I do not know if you have noticed. But when you feed a stereo sound to our decoder,
and choose 5.1 output, pseudosurround effect is performed. I am curious of your
thoughts regarding this effect :)
Please listen to it in 5.1 speaker environment.

[Rockaria]

OK. Thanks for the tests!
We has still some things to do (such as incorporating Prologic II)
etc. ..
> There was no crosstalks for the original wav.
> Maybe a special care for the analog surrounds phase handling is reuired.
Indeed! This also requires additional work (i.e. sounds encoded with Prologic).
Therefore we mus incorporate this technology in Aud-X.
We hope to do it before the next release.
..
The phase experiments that you've performed are interesting. However they may depend on the signal (phase relations in current signal).
--
(5.1 Kinyo in stereo dolby surround encode mode)
Well, maybe you misunderstood my initial test about the crosstalks with the knyo 5.1ch analog headphone.
The soundstorm has a DSP in the nvmixer encoding/downmixing 5.1ch signal to 2ch DS/DPL for analog speakers/headphone. I tested the kinyo in this 2ch DPL mode then original 5.1 mode. The latter produced the crosstalks as I mentioned before and disappeared when I set the invert phase on center & rear.
The aud-x encoded then decoded 5.1ch wav generated the crosstalk when played with aud-x while none with other players/dsfilters. Also the digital connection has no crosstalks at all.
So it cannot be concluded it's a signal(encoded) problem alone.

Also now I am curious why you mentioned the DPL II when I reported the crosstalks between the channels. I thought that it( reference) would be only useful to eliminate the crosstalks in the signal(encoding) and decoding(aud-x) happening on certain DAC & analog speakers.


Thus when you convert DPLII to Aud-X, you will not gain increase of quality, but rather more crosstalk (the original DPLII phase distortion will be preserved, since Aud-X tries to keep the inputs phase).
...
And in general this is what Aud-X is all about -> preserving phase relations, which are responsible for the spatial feeling
I never think of myself transcoding the DPL II content to aud-x as well as mp3 to aac. The scenario itself already contains some big bias.
When the aud-x embed the surround data with the phase shift technology then seperates the channels with aud-x filter, how much channel seperation is reserved other than the abstract 'sprtial feeling'? Is there anything artificial sound generation related?

[3dsnar]

OK, indeed I misunderstood what you meant

> I never think of myself transcoding the DPL II content to aud-x as well as mp3
> to aac. The scenario itself already contains some big bias.
> When the aud-x embed the surround data with the phase shift technology
> then seperates the channels with aud-x filter, how much channel seperation is
> reserved other than the abstract 'sprtial feeling'? Is there anything artificial
> sound generation related?

I am not sure if I understand your question correctly.
In general Aud-X is not about phase shifting, or other phase manipulations and sound decomposition.
So maybe I will be more specific (I hope this is not redundant to what I have already written).
---
In general for humans there are several aspects which influence perception of sound direction. Energy of the sound and phase of the sound.

In more details, perception of phase and energy is frequency dependent and perceived in non-linear frequency scale (barks).
- Also perception of energy changes (dynamics) is frequency dependent and nonlinear.
- Perception of phase is related to tonality of certain sound components
(tonality can be represented by something called Unpredictability Measure -> this parameter is used in Aud-X) and frequency dependent as well (for example you cannot determine LFE direction, but you can very precisely of a 2 kHz tone)
--
So the more tonal are the sound components, their phase matters more (but with reference to certain frequency range).
BTW. Tonality descriptor is also relevant in modeling instantaneous masking phenomenon,
but this is a different story.

Anyway, considering above allows to generate a map of energy and phase spectral components and decide which are more significant in terms of perceiving surround panorama, hence which should be considered as more important in generating the 5.1 sound reconstruction scenario (from mono or stereophonic base stream).

The more bitrate is available, the better reconstruction can be performed.
------
I hope (but I am not sure) this answers your last question.

====================

I do not know what is the Kinyo device, how it works and therefore I do not understand why it produces different results than a normal speaker set.

Our informal subjective tests performed on 5.1 speaker set indicated that people were not able to distinguish STDQ Aud-X sound from 448 kbps AC3.
This does not refer to situation where single channel streams are reproduced separately as monophonic, or in pairs - stereophonic. In such a case there is a difference between Aud-X STDQ and AC3 448 kbps.
We have also ran some objective experiments, which we included in the whitepaper.

Ofcourse I will be happy to answer more questions.

[Rockaria]

I really had a hard time to establish a connection between the crosstalks(fact) and your DPL II.:D
Just regard the kinyo as normal 5.1ch analog speaker. I am also informed the crosstalks from other users. So it's not a matter of specfic speaker model to streighten it. Rather it seems to be a matter of the codec & analog speaker environment overall.

I've heard something similar with the dolby headphone, but because of too deep professional norms considering the written data and my lack of hobby time, I am relying on some future free time and research.

Thanks.

[SeeMoreDigital]

Is anybody able to provide some samples please?


Cheers

[3dsnar]

Why do I get confused all the time? ;)

I am not sure if this was what you desired, but:
Are you interested in Aud-X examples?
If yes, here is the instruction how to generate them

1) Obtain an input AC3 file for tests
2) Decode it to PCM with BeSweet
3) Feed the PCM to Aud-X executable encoder
--
If you want to convert Aud-X stram back to PCM:
1) Feed the file to the executable decoder
2) Choose PCM as output format
3) Wait for results
-----
You can also transcode Aud-X stram directly to AC3, if you wish to do so.
=======================================================
All this is very simple, but in case of questions I'd be happy to help.

[scharfis_brain]

@3dsnar: I have a idea for ultimative compatibility to standard MP3 (or other codecs) decoding:
- while Aud-X encoding downmix the 5.1 signal into two channels using a DPL2 matrix
- then build a 5.1 separation bitstream according to this downmixing.

this will allow users without the Aud-X-Decoder to receive a completely transparent DPL2-Stereo, while Aud-X users will enjoy the advantages of much better channel separation.

[3dsnar]

Sharfis, thanks for your thought.
Maybe this is a good idea.
I must investigate this issue.
Probably encoding with DPL2 results in stereo stream which is not necessary a linear downmix (with certain weights applied to each channel), but rather a nonlinearly (phase and amplitude) distorted downmix.
---
If my speculation is correct, than marrying Aud-X and DPLII is not possible.
And I am afraid my intuition leads me to the right direction... :(
---
But your post gives me an idea :)
To include real time DPLII transcoder, which would
transcode 5.1 stream (obtained after decoding Aud-X) to
DPLII for those who want to hook the computer up to the external DPLII decoders.
--
We have already implemented AC3 transcoding for external Dolby Digital compatible
home theatre systems, but actually I did not think
of DPLII compatible equipment...

Do you think it is worth considering such an option in our DS filter?

[scharfis_brain]

DPL and DPL2 are linear downmixes. BUT the sourround channels are encoded with less left/right separation
They have a total phase of 180° between Lt and Rt. The only differnence are their weights.

I do not see the (technical) difference between your downmixing:

Lt= L+C+Ls+LFE
Rt= R+C+Rs+LFE

to DPL2 downmixing:

Lt= L + C + 0.8Ls - 0.5Rs + LFE
Rt= R + C - 0.5Ls + 0.8Rs + LFE
(the weights aren't totally correct here. look at the dolby specs, please)

the result is the mentioned 180° phase difference to restore the surround channels with an additional 180° phase rotation at the decoders side.

maybe it is more easy for you to stay for a DPL1 endoding like this:

S=Ls+Rs
Lt=L+C+S+LFE
Rt=R+C-S+LFE

and then use a 2-step bitstream channel separation:
1st step: separate L, C, R and the mono surround
2nd step: separate Ls and Rs from the mono surround

I hope this could be implemented into Aud-X endoding because it would help it being much more widespread cause it then contains standard DPL(2) which ca be played back at every equipment without Aud-X-decoder without loosing the surround info.

[3dsnar]

I see.
So the only problem is LFE, which is separately
resynthesized in Aud-X. But this is not important.

So Actually I could generate a new encoding option (DPLII compatible).
It would have to be 192 kbps.

[scharfis_brain]

actually, Dolby recommends not to include the LFE into downmixing.
Anyway, I like it having downmixed :)

WOW. It'd be great to have fully DPL compliant encoding :)

btw.: it would be good, if the decoder can downmix to DPL2 itself, too.

[3dsnar]

Alright. Thanx for your ideas :)

[3dsnar]

Scharfis. I will try to change Aud-X HGHQ into DPLII compatible, during this weekend,
because I think that your idea is GREAT!
----
What weight do you suggest for LFE?
(After few thoughts, I know how I can also mix it in and separate it nearly perfectly in case of Aud-X decoding)

[Rockaria]

As you see, AC3filter has both DS/DPL and DPL II encoder(speaker set), ideally can be attached at the aud-x out pin(5.1ch PCM).

However, having these DPL encoder inside of the aud-x dsfilter would make a good comparison & option.

[bobcat56458]

I second the Dolby ProLogic II addition, also there have been posts here on this forum asking for a way to do this in real time. I’m speaking of having a stereo TV tuner card, or line input on the PC, output Dolby Prologic II to the speakers, this would be a great additional feature. I would also like to see a way to temporarily disable, or unregister, the audxdshow.ax filter\codec because with my P4 1.8GHz processor when I have Aud-x running, and play a uncompressed, or DV AVI video file it sometimes makes the video run not so smoothly. Also when I’m editing stereo audio\video I want to hear the sound output in stereo, not 5.1 surround. I like the 5.1 Built-in Pseudo Surround module for playback of my MP3 files though. I have an Audigy I soundcard connected to a 5.1 amp by way of its analogue inputs\outputs. Thanks for your continued development of this fine application\codec.

[3dsnar]

Bobcat,
thanx for this suggestion. This is exactly what I mean.
A realt time DPLII encoder (similarly to already embedded AC3 encoder).

[Rockaria]

This post is moved from other thread because of going too deep on the aud-x only issue. Any aud-x issue will be discussed here.

We seem to have got some clearer data than the white paper in terms of the codec mechanism :
1. FV : the core technology of the aud-x codec which decides the fidelity
2. mp3 : asynchronous storage of core & meta(FV) data, affects the core(base) quality

Supposing upper scenario as the base architecture, there might be some more inquiries like below :
ENCODING:
1) Obtaining energy and phase relations between channels for future reconstruction from the merged signal (performed by Aud-X). Such information is stored in a feature vector - FV - (for each frame such a vector is produced)
FV : a meta storage for energy & phase relations between the channels on each frame
What is the frame size(nSamples) ? depending on the quality criteria?
2) Merging all channels
Core-Stream : 1ch or 2ch to be encoded in the mp3 body
3) Encoding with LAME
Encodes core only...
4) Inserting the Aud-X feature vector to mp3 stream (there is a space which can be utilized for any kind of information)
Where exactly : header, body or trailer?
Using own mp3 file access routine?
How much space is available in the meta data area for Lame & mp3?
Also isn't it(meta space) dependant on the core codec?

DECODING:
1) Decoding mp3 frame to obtain base audio stream
2) Getting Aud-X FV and reproducing the complete 5.1
Is it possible for streaming purpose when you get the FV set after the stream(body)?
What if the meta data area is altered by any tag programs or truncations ?

My impression of the asynchronous stream access(core & meta) is that it has some potential problems.(streaming, truncation...)
Like WavPack's hybrid format, a seperate meta file(if not exists, it decodes as lossy) can be a good idea for the synchronous & safe access.

Meanwhile, the core blackbox(FV manipulation)'s spatial image abstraction(relative to the core?) into energy-phase sets sounds fantastic but surely is beyond my touch, and now agreed totally different from the phase-shift technology.

Many things I described could be wrong. But that is my best analysis based on the given data, requiring some clarifications, for whoever is interested in aud-x.

[Rockaria]

OK. This is not my area of expertise. But I roughly know the answers to your questions (frankly, my area of expertiese are all the phase and energy manipulations, which are not so interesting for you ;) )
Our software expert is not very easy to reach...until March.
-
But I will try to answer, based on what I know.
If you have a let say 128 kbps mp3 stream, it is possible to change it
into for example 192 kbps, by manipulating some header information. And this makes some room free for additional information (FVs in our case).
So available space is not a problem, since it is possible to control the average kbps available.
---
There are no problems with streaming. This is just a matter of a delay to get
the main audio and FVs related to that. In practice, when you reproduce Aud-X sound, it is in fact streamed from the disk to the DirectShow filter. And jumping from one point of the stream to the other is possible...

---

As far as I remember, the mechanizm of insterting the FVs is also used by mp3surround and was used earlier by mp3pro.
So actually knowing this, we started to investigate this issue.
I guess when you read carefully MPEG Layer3 Bitstream Syntax andDecoding.doc you can find all the answers to your questions there
(as our colleague did).
If you do not have this document, I can send it to you.
---
@3dsnar, you can delete the content from other thread.
Thanks for the clarification on the aud-x architecture.
Maybe the major success key seems to lie on the PV technology, looks like more suitted for the game 3d effect area, IMO.

[3dsnar]

OK. Deleted. I will report
DPLII related changes in our codec.
Thanks for all the valuable suggestions.

[3dsnar]

Aud-X v.1.1 Dolby Pro Logic II compatible is available for downloading.
Please read more in a separate thread devoted to the new release.

[bond]

@3dsnar: I have a idea for ultimative compatibility to standard MP3 (or other codecs) decoding:
- while Aud-X encoding downmix the 5.1 signal into two channels using a DPL2 matrix
- then build a 5.1 separation bitstream according to this downmixing.hm if compatiblity to mpeg is a goal, why not use the mpeg surround standard right away? :confused:

3dsnar, would it be hard to make your encoder compatible to this?

[3dsnar]

Well, now you can decode the stream by a DPLII decoder,
which is not ofcourse as good as a digital decoding methods.
--
But it is still better than stereo, when you use an external DVD player and DPLII decoder.

If you can use a PC with 5.1 speakers, or hook it up to an external
Dolby Digital decoder, it is better to use our PC based DirectShow decoder,
since it producess surround sound of much better quality than DPLII analog decoder.

===

Do you mean mp3surround?

- it is not DPLII compatible (AFAIK),
- it has poor channel separation quality, with significant phase errors.
- it is NOT FREE,
- It is not VirtualDubMod compatible,
- There is no directshow filter (to decode it) available
- It does not have transcoding and decoding tools available.

[bond]

Do you mean mp3surround?no, i dont mean mp3surround, which is imho just another proprieatary surround method

i mean the official surround standard mpeg released
http://codingtechnologies.com/products/mpgsrnd.htm
http://www.hydrogenaudio.org/forums/index.php?showtopic=36355
http://forum.doom9.org/showthread.php?p=699471#post699471

[3dsnar]

Aud-X is mp3 backwards compatible.
It can be used with AVI and is CBR.
With encoding bitrates similar to AAC 5.1
(i.e. 192, 128 and 80 kbps).
----
Now it is also DPLII compatible.
And this is useful for external home theatre equipment.
----
So as you can see it might be an interesting alternative,
to mpeg-4 5.1 format.

[bond]

so you cant/dont want to make your surround implementation compatible with the official mpeg surround standard?

why not?

[3dsnar]

I have nothing against the standard, and we would like
to see in the future if our spatial information encoding could be used with
mpeg-4, as an extention to Aud-X (so it will cover both mp3 and mp4).
-------
W tried to make something compatible to mp3 and AVI
due to their huge popularity. That is all.
And it turned out to be comparable (in terms of quality and bitrate)
to existing mpeg-4 based surround encoding technologies.

[Gabriel_Bouvigne]

Aud-X is mp3 backwards compatible.
It can be used with AVI and is CBR.
With encoding bitrates similar to AAC 5.1
(i.e. 192, 128 and 80 kbps).

I do not really understand how you can add ancillary data to mp3 and still have a cbr stream, without:
*modifying Lame's bit allocation
or
*waste a lot of space by setting the original mp3 cbr stream 1 cbr step higher.

As an example, PlusV was also inserting ancillary data into the mp3 stream from Lame. But to do this they had a frame repacker, and the result was abr, not cbr.

[bond]

I have nothing against the standard, and we would like
to see in the future if our spatial information encoding could be used with
mpeg-4, as an extention to Aud-X (so it will cover both mp3 and mp4).
-------
W tried to make something compatible to mp3 and AVI
due to their huge popularity. That is all.
And it turned out to be comparable (in terms of quality and bitrate)
to existing mpeg-4 based surround encoding technologies.hm, i think you are misunderstanding the "mpeg surround" standard! its not only for mpeg-4, it also works with mp3!

[3dsnar]

I understand that you are talking about Binaural Cue Coding?

[3dsnar]

OK. Yes, you are referring to BCC (based on which mpeg surround extention
has been created).
Well, I actually think that this approach has several limitations related to phase preservation. Therefore I carried out my own research on reconstructing the surround panorama (which resulted in Aud-X concept).
---

[Rockaria]

Well, if it is backward-compatible with existing core codecs, shouldn't the SAC standard also be forward-compatible(pluggable) with the future spatial codecs(with own ancillary & core data), as far as they meet the requirements?

The DPL II incorporation surely has some benefits than AC3 encoding-passthrough if the fidelity is not a big issue. But the bandwith & channels between the PC and external digital decoder shouldn't be of any constraint on DPL II & AC3 stream.
. The s/w AC3 encoding seems to have more than 500ms of delay, of course depending on the CPU, compared to the 32ms(soundstorm DICE) of h/w encoder, which asks for the a/v syncronization setting in the players or DSFilters.
. The external AC3 decoders are known to have some delay(as much as 1000ms) depending on the models.

As seen with existing established DSFilters such as FFDShow or AC3Filter, all the 'plus alpha' in the aud-x(plus lots of other DSPs) are already provided if it simply allows the filter chains(instead of the redundancy).
What is questioned and to be verified/implemented with aud-x(FV technology) seems to be the quality & standardization for general purpose audio codec use(not specific to movie or game use), not the 'plus alpha' at the moment, IMHO.:)

[3dsnar]

I also started thinking of a proper DPLII decoder in our DSfilter.
I.e. since it is unknow how the signal is in fact decoded, I will have
to be a bit creative... I know how to separate the center channel
in the frequency domain (by analyzing energy, phase relations
and differences between imaginary and real parts of the channels).
I also have a rough idea how to create a stereophonic subwoofer stream,
so
I think that I will include the decoder in our DSfilter within
next couple of weeks.

[tebasuna51]

I also started thinking of a proper DPLII decoder in our DSfilter.
...
I also have a rough idea how to create a stereophonic subwoofer stream
Stereophonic subwoofer stream?
I think a proper DPLII decoder must be a upmix to 5.0 (without LFE), because:
1) Dolby recommendations.
2) Don't mistake LFE channel with subwoofer.
3) Low frequencies don't need be directional.
4) The hardware multichannel audio player always redirect the low frequencies no matter in which channel are present.
5) This redirection must be a user choice. For instance, I have a 5.1 audio equipment but only 5.0 speakers, then I redirect all low frequencies to front speakers (with enough capability to play low frequencies). The normal choice is redirect to subwoofer.

Then, for a DPLII upmix, I think the LFE channel is unnecessary.

[3dsnar]

oops.
I am sorry. what I wrote is rubbish...
I meant stereophonic surround channels,
and I wrote subwoofer by mistake.

---
Regarding the LFE in a DPLII downmix.
Well, I decided to include it, because people seem
to like it. I also did some tests with and without it
with my DPLII standalone decoder, and I think that
downmixing the LFE sounds better...

But I guess this is a matter of taste.

[JnZ]

Hi 3dsnar,
your codec looks very well. So I decide to try it. I'm encoding now 137min long 6ch WAVe and waiting for result.

BTW: I found small error in cmd encoder. Instead percentage progress it shows only 1.#J% ;) This happend only with this long wavefile...

Thx for Aud-X.

[tebasuna51]

Regarding the LFE in a DPLII downmix.
Well, I decided to include it, because people seem
to like it.
I have not problem with include the LFE channel in DPLII downmix (disregarding Dolby recommendations), there are a info and can be preserved. Only in the upmix I think is unnecessary because the criterium to select the signal is only the frequency range, and this job is always maked by the hardware audio player.

[3dsnar]

JnZ:
Thank you :)
I hope you know that you can use our release of VirtualDubMod to
encode the complete movie with Aud-X (you can find the release of VDM
on our web site).
------------------
So you take the input movie with AC3 inside (AVI, VOB, etc) and directly
(using the VDM) you generate Xvid video (or whatever you use) with mixed in Aud-X surround
sound.

=============

Tebasuna51:
100% agreed

[JnZ]

My 137min Terminator 2 test WAV is OK. So give it to my frend to test it on 5.1 speakers (I have only stereo :( ).
Coding via VDM Aud-X is OK.

But today I test small M$ test 6ch WAVe, and results are very interesting.
(http://download.microsoft.com/download/winmediatech40/Utility/1.0/W98NT42KMeXP/EN-US/6channel.exe)
This WAVe must be converted to real WAVe, because it have strange ID.

Audio have little higher pitch, than original (test yourself). I test it on 128, and both 192kbps profiles. And subwoofer have less expressive.

[Rockaria]

I also started thinking of a proper DPLII decoder in our DSfilter.
I thought that the aud-x core(FV-technology) was a lot improved than the 10 year old technology as for the quality and bit efficiency?
IIRC, the DPL II requires the same amount of the bit rate on each channel but just volume-reduced(check the matrices) and shift-phase(+-90 degree, rear) merged in the two fronts. There's no abstraction layer to reduce the size, making it(encoding/decoding) straightforward and sounds with equal fidelity on each channels if decoded correctly(yeah I know center!).

Any recent: advanced Spatial Audio codecs are expected to sound(reconstruct) a lot better than DPL II, given the same amount of the bit rate in total with its bit-efficiency and improved algorithm. It made me believe the incorporation of the DPL II with aud-x should be at the end of the aud-x decoder(speaker set or mixer DSP) like other approaches. It's because of less restriction of the bit rate between the external decoder, also because not all the receivers have the mp3 decoder equipped.

Another established DSF FFDShow also has lots of useful mixer options including the DPL II which, I tested, seperates the channels with good quality when applied the correct DPL II matrices. Maybe aud-x is gonna include all the FFDShow features...

Also any combination is mathematically possible for each encoder & decoder set, but requires a lot of art...
The pair DPL II encode->FV-decode makes little sense as well as the pair FV-encode->DPL II-decode.
But the chain FV-encode->FV-decode->DPL II-encode(mpeg/PCM)->external DPL II-decode is expected to have some advantages, provided & proved the FV-technology is more efficient & effective than DPL II in the quality.

Good Luck.

[3dsnar]

JnZ, the wave is the older type wave format.
The pitch change is caused by the fact that originally it is 48000 Hz.

So you just changed the header and also the information of the sampling rate
was changed to 44100 Hz.

The pitch change also results in less expressive LFE reproduction.

---
You can play the wave (produced by the Aud-X frontend) with FFDshow for example.
Or maybe you can manage in the conversion that you performed
to preserve the 48 kHz sampling
frequency.

[3dsnar]

I thought that the aud-x core(FV-technology) was a lot improved than the 10 year old technology as for the quality and bit efficiency?
IIRC, the DPL II requires the same amount of the bit rate on each channel but just volume-reduced(check the matrices) and shift-phase(+-90 degree, rear) merged in the two fronts. There's no abstraction layer to reduce the size, making it(encoding/decoding) straightforward and sounds with equal fidelity on each channels if decoded correctly(yeah I know center!).

Yeah. I know all of this. I have read some interesting discussions on Doom9
regarding this subject :)
http://forum.doom9.org/showthread.php?s=&threadid=57988

So now HGHQ contains a proper DPLII information.
I did a test (connected my PC to my home theatre system via SPDIF).
So when I reproduce the HGHQ stream and route it via SPDIF with Aud-X DirectShow filter, I can hear surround produced by Aud-X. When I played the sound with DS incompatible player, the stereo PCM was sent via SPDiF which resulted in DPLII decoding (my amplituner switches to DPLII automatically if the incoming stream is PCM).
The surround was also decoded...but the difference is significant...
And it is true that DPLII is not much comparable to digital encoding techniques.


Any recent: advanced Spatial Audio codecs are expected to sound(reconstruct) a lot better than DPL II, given the same amount of the bit rate in total with its bit-efficiency and improved algorithm. It made me believe the incorporation of the DPL II with aud-x should be at the end of the aud-x decoder(speaker set or mixer DSP) like other approaches. It's because of less restriction of the bit rate between the external decoder, also because not all the receivers have the mp3 decoder equipped.

Well, in fact Aud-X does not use the DPLII at all. But since the downmixing
is a linear channels merging (which I've learned during discussions on this forum - I thought that DPLII downmixing has some nonlinear DSP operations) I realized that this should not affect the decoding concept, but the way the spatial information is gathered must be modified. So I did the modifications during this weekend and this resulted in the latest release of our codec.

Another established DSF FFDShow also has lots of useful mixer options including the DPL II which, I tested, seperates the channels with good quality when applied the correct DPL II matrices. Maybe aud-x is gonna include all the FFDShow features...

I tried to make Aud-X compatible with FFDshow. So you can decode
the DPLII stream to 6 channels and it can be passed through Aud-X
(if you have both codecs installed).
Morover, when you are playing a stereophonic file (lets say mp3) the signal is decoded by FFDshow and passed to Aud-X, where it can be converted with the pseudosurround module and sent via SPDIF to the external amplifier.
etc...

Also any combination is mathematically possible for each encoder & decoder set, but requires a lot of art...
The pair DPL II encode->FV-decode makes little sense as well as the pair FV-encode->DPL II-decode.
But the chain FV-encode->FV-decode->DPL II-encode(mpeg)->external DPL II-decode is expected to have some advantages, provided & proved the FV-technology is more efficient & effective than DPL II in the quality.

Yeah, as I mentioned the DPLII matrix downmixing is a separate story.
I did it only to make the HGHQ stream more flexible in terms of applications,
i.e. so you can play it on an external DVD connected to the home theatre amplifier and have some sort of surround (the one that DPLII offers ;) )
But if you can use a PC, a proper surround, resembling the original AC3 is reproduced with the Aud-X decoder. To see the difference, please perform a test which I described in the upper part of this post.

[Rockaria]

Let me point out some contraries.

1. Your aud-x involves too many (contrary) stuffs : What do you mean by aud-x exactly. I believe you must exclude everything else besides the core FV technology, the quality of the Spatial Audio Codec we are questioning.

So now HGHQ contains a proper DPLII information.
Well, in fact Aud-X does not use the DPLII at all

2. possibly you can attach the aud-x at the end of other decoders such as ffdshow.
But have you ever tried to attach the ffdshow at the end of the aud-x at all, to utilize other excellent & proven DSPs?

I tried to make Aud-X compatible with FFDshow. So you can decode
the DPLII stream to 6 channels and it can be passed through Aud-X
(if you have both codecs installed)....

3. the discussion is two years old and actually nothing new as for the +-90 degree phase shift. : some say delay others say analog encoder... looks like identifying a 3d object reflected on the photo, grayed.

But since the downmixing
is a linear channels merging (which I've learned during discussions on this forum - I thought that DPLII downmixing has some nonlinear DSP operations) I realized that this should not affect the decoding concept, ...

4. the DPL II decoder can actually upmix any streams on 2ch(forced) to create any types of illusion. what matters is the reconstruction quality onto the originally DPL II downmixed contents from 5.1ch. If you want the compatibility, you must not add any ancillary data for later decodings.
But if you want to because of the better decoding, you are actually creating a new spatial audio codec combining the DPL II & FV.

I realized that this should not affect the decoding concept, but the way the spatial information is gathered must be modified. So I did the modifications during this weekend and this resulted in the latest release of our codec.

I have no trouble in my setup to evaluate any new codec. The only trouble I have experienced is that it's moving continiously before grasping(evaluating) anything... So I'll just have to wait and see. Thanks. :)

[edit]
OK. thanks for disclosing your philosophy clearly, I'll respect that.

[3dsnar]

By Aud-X
I mean the compression technology based on reconstructing the full 5.1 panorama from the spatial information (stored in the Feature Vectors) and the merged base audio signal. The carrier of the audio main stream is mp3 audio, while the spatial info is stored in the ancillary data of the mp3 stream.

However our DirectShow decoder is much more than the Aud-X stream decoder.
It also contains resampling algorithm, PseudoSurround algorithm, AC3encoder and DPLII downmix module.

Indeed, you can attach Aud-X only at the end of the filter chain.
However, since you can pass-through any incoming signal, there is no problem with it. I.e. you can apply any DSP algorithm of FFDshow or other filter (e.g. AC3filter). And the resulting stream will be fed to Aud-X and passed through it without any changes to the rendering device.
So I think there is no conflict.
This was the most reasonable compromise that I could have come up with...
That is why it is done in such a way.

The ancillary data, where the Aud-X spatial information is stored, is not "visible" for a standard (including hardware) mp3 decoder. And there is no compatibility problem. Simply, when you play the Aud-X mp3 stream, the main audio data will be reproduced. In case of HGHQ the audio data is stereophonic and downmixed from original 5.1 in a DPLII manner.
When the accillary data is accessed by the Aud-X decoder, it is used to reconstruct the 5.1 stream. The way of downmixing, however, is unimportant (assuming that any nonlinear processes were not involved). In other words, the created downmix does not have to be DPLII compatible, because it is not relevant for the decoding algorithm...

The only significant change in terms of the Aud-X spatial encoding algorithm was done once. During the last weekend, to consider the DPLII downmixing weights and procedure :)
Other upgrades are related only to functionality of the DirectShow decoder
(they are not a part of the coding/decoding methodology).

[elenhil]

Another established DSF FFDShow also has lots of useful mixer options including the DPL II which, I tested, seperates the channels with good quality when applied the correct DPL II matrices.

It did not play a single DPL-encoded stream of mine. Were they all encoded with incorrect matrices? Who cares? PowerDVD's DS filter did decode them properly, so should have any other. Which disqualifies FFSShow as a good free DS DPL decoder. And PowerDVD's not freeware (not to mention that it is poorly suited for things like playing avi's with more than one audio stream)

[3dsnar]

It did not play a single DPL-encoded stream of mine. Were they all encoded with incorrect matrices? Who cares? PowerDVD's DS filter did decode them properly, so should have any other. Which disqualifies FFSShow as a good free DS DPL decoder. And PowerDVD's not freeware (not to mention that it is poorly suited for things like playing avi's with more than one audio stream)

OK. If you are trying to play Aud-X stream, this means that it disqualifies FFDshow.
However, it makes no sens to decode such stream by DPLII direct show decoder - i.e. it is much better to use Aud-X decoder to obtain the 5.1 signal from Aud-X encoded audio (it is superior to DPLII decoded multichannel signal).
---
Normally for all other streams (not Aud-X) FFDshow and Aud-X DSfilter are connected in a filter chain. And the outcoming signal from FFDshow can be passed through Aud-X DSfilter without causing any problems.
---
Also, you can disable Aud-X by chosing in FFDshow output settings
"direct sound" output.
================
So normally any mp3 (excluding Aud-X stream), AC3, AAC, etc. can be played with all other filters.
================
Indeed, multiple audio strams cause problems. This bug will be fixed ASAP.

[Rockaria]

Who cares? ... Which disqualifies FFSShow as a good free DS DPL decoder. ..Perhaps me neither.. And it could be either yourself or FFDShow.
Were you able to find any DPL (II) decoders in the FFDShow mixer options?
I could find ENCODERS(mixers) only. Actually there is a 'dolby decode' on the top of the DSP lines. Try that one to DECODE.

[elenhil]

OK. If you are trying to play Aud-X stream, this means that it disqualifies FFDshow.
I did not refer to Aud-X streams specifically. I meant regular DPLII software-encoded (using BeSweet, etc.) streams used in 2ch AC3 DVD backup.

Say, will the new Aud-X DS filter decode DPL in mp3s only?

[elenhil]

Perhaps me neither.. And it could be either yourself or FFDShow.
Were you able to find any DPL (II) decoders in the FFDShow mixer options?
I could find ENCODERS(mixers) only. Actually there is a 'dolby decode' on the top of the DSP lines. Try that one to DECODE.
That's what I did. No separation at all.

BTW, neither does AC3Filter (with DPL/DPLII output presets) decode DPL(II) streams.

[Rockaria]

No, you are misunderstanding.
AC3Filter does not have any DPL (II) decoders(explicitly), it only has encoders as speaker set to be decoded on your external receiver.
Likewise, FFDShow has these encoders as mixer options.

Also if you are using any valid DPL II decoder, it will produce 5.1ch as far as the input stream is valid 2ch stereo(whether DPL II encoded or not), unless the input stream is messed up.

[elenhil]

No, you are misunderstanding.
AC3Filter does not have any DPL (II) decoders(explicitly), it only has encoders as speaker set to be decoded on your external receiver.
Likewise, FFDShow has these encoders as mixer options.

I see. Thanks. So, I'd like to see a universal software DPL _decoder_ capable of processing other formats besides ac3 or mp3 (it is OGG Vorbis that I have in mind, of course).

Also if you are using any valid DPL II decoder, it will produce 5.1ch as far as the input stream is valid 2ch stereo(whether DPL II encoded or not), unless the input stream is messed up.
But without proper positioning, of course.

[Rockaria]

That's what I said by illusion.:)

You can try the ffdshow 'dolby decode' dsp.
Any formats(mp3,aac,vorbis...) decoded to 2ch PCM will be DPL(II) decoded(or upmixed) again if it is checked. not sure if it does DPL or DPL II though.

[elenhil]

You can try the ffdshow 'dolby decode' dsp.
Any formats(mp3,aac,vorbis...) decoded to 2ch PCM will be DPL(II) decoded(or upmixed) again if it is checked. not sure if it does DPL or DPL II though.
I have tried activating 'dolby decode' in ffdshow DS audio filter. Nothing even close to correct DPL decoding happened. Are you omitting some other crucial functions I must turn on to make ffdshow decode DPL?

[Rockaria]

Are you omitting some other crucial functions I must turn on to make ffdshow decode DPL?Instead, I guess you must turn off the mixer DSP(probably DPL II turned on?). What a side effect!:cool:

[Amnon82]

http://img443.imageshack.us/img443/3010/untitled39jr.png

Download XviD 5.1 Aud-X 128 Sample (19.44 MB | 2.28 Min) (http://www.megaupload.com/de/?d=XZB5X0N4)
Download Aud-X-Codec (http://www.aud-x.com)

Setup your system like this:

http://img442.imageshack.us/img442/7044/untitled45jv.png

[elenhil]

Instead, I guess you must turn off the mixer DSP(probably DPL II turned on?). What a side effect!:cool:
It was turned off.

[Rockaria]

I tested it again.;)
Uncheck the mixer itself, where is only(I believe) place to downmix the decoded(upmixed) multichannel by 'dolby decode'.

[3dsnar]

Amnon82, thanx for your example and using Aud-X
:thanks:

[Rockaria]

Finally, peace(well..). Must have been improved recently especially on mp3 HGHQ decoding.(I haven't done much of the quality test on the latest version)

I've not encoded the 128k mp3 myself yet but the movie sounds O.K.
But the SPBQ mp3(encoded before) sounded weird and aud-x decoded wav(also before) still has crosstalks on analog(disappeared by invert phase also).
Definitely need a total reencoding/test job but my impression is O.K.(busy...)

Now when the two-way compatibility(between the dsfilters) is ever enabled, I think I can perform some personal comparison tests on mp3 & m4a at the same decoding condition(DRC, DPL II, volume...).

[Summary] based on aud-x dsf & standalone compared to the original 6ch wav
STDQ.mp3 : artifacts
HGHQ.mp3 : clean, thin tone, wider seperation(artificial?), no crosstalk
SPBQ.mp3 : thicker than HGHQ but still thin tone, unbalanced, no crosstalk
HGHQ.wav, SPBQ.wav : all crosstalks on analog, removed by invert phase.
Done.

[3dsnar]

Rockaria, thanx for all your time and tests
:thanks:

Interestingly, IMHO SPBQ has better quality than HGHQ.

BTW. Artifact may be caused if the input channel order is incorrect in the 6 channel PCM.
(please check that out).

I do not know if you noticed, but the crosstalks happens mostly to phase-chaotic components (noisy components), while tonal ones are separated well. This results in preserving the spatial feeling, when listened to the decoded sound in 5.1 speaker environment environment (due to preserved phase of the tonal components). In the multichannel invironment it is (shoule be) impossible to hear the crosstalks.
I think subjective tests with headphones are not to much suited...for the purpose of 5.1 encoders evaluation ;)

[Rockaria]

fact is fact. If you want to see only what you'd like to see..why you listen.

What I've noticed is that aud-x only has made the crosstalks so far on this headphone.

Thanks for providing all this experiments.

[Sagittaire]

The core bitrates that we use
1) STRQ 80 kbps -> base mp3 is 56 kbps mono (24 kbps for spatial info)
2) STDQ 128 kbps -> base mp3 is 64 kbps mono (64 kbps for spatial info)
3) HGHQ 192 kbps -> base mp3 is 128 kbps stereo (64 kbps for spatial info)
4) SPBQ 192 kbps -> base mp3 is mono 64 kbps (128 kbps for spatial info)

Perhaps that VBR quality mode could be better ... if possible

VBR mode with lame
Quality mode VN -> base mp3 is VN lame join stereo (VN too for spatial info)
and
Quality mode VN -> base mp3 is VN lame mono (VN too for spatial info)

[Cyberace]

Cool stuff, but why not release the full source code and license it under standard GPL or LGPL?, (if not the full codec with the encoder then how about just the decoder), that way open sourced players that are licensed under GPL/LGPL can also incooperate the decoder to be able to play Aud-X encoded audio files and soundtracks, ...example of such players are MPlayer, XINE, VideoLan-Client (VLC), and the FFmpeg/FFdshow codec-suit.

[3dsnar]

I don't know about the source code, but I know that Aud-X DLL
encoder and decoder will be released probably before the end of this month
(for incorporating it in third party software). And ofcourse it will be free.
-----------------------------------------------------------------------
Maybe Aud-X owners will decide to release the source code as well...
I do not know...

[3dsnar]

Perhaps that VBR quality mode could be better ... if possible

VBR mode with lame
Quality mode VN -> base mp3 is VN lame join stereo (VN too for spatial info)
and
Quality mode VN -> base mp3 is VN lame mono (VN too for spatial in
Thanx for suggestions.
Yeah. But we would rather focus on extending the spatial information processing
with mpeg-4 audio, or ogg vorbis. But this is for the future, which will depend on the performance of the CBR mp3 based release. CBR was chosen for one particular reason. AVI (crap) compatibility ;)

[Gabriel_Bouvigne]

The core bitrates that we use
1) STRQ 80 kbps -> base mp3 is 56 kbps mono (24 kbps for spatial info)
2) STDQ 128 kbps -> base mp3 is 64 kbps mono (64 kbps for spatial info)
3) HGHQ 192 kbps -> base mp3 is 128 kbps stereo (64 kbps for spatial info)
4) SPBQ 192 kbps -> base mp3 is mono 64 kbps (128 kbps for spatial info)
OMG! Up to 128kbps of parametric information!
Isn't it a little too much?
SPBQ is especially odd to me, as the mp3 base layer is only 64kbps for a total of 192kbps for a 5.1 reconstructed stream.
Mp3@64kbps mono should provide comparable quality to mp3@96kbps stereo. If you give the same bitrate to front and rear (which would be a little odd), it means that @192kbps you could have a 4.0 stream with the same quality as the SPBQ base channel, but without any parametric info.
If you reduce a little the bitrate of the rear channels down to 80kbps, we would have a 176kbps stream for 4 full channels. Don't you think that the results could be better by adding 16kbps of parametric info to a 176kbps 4 full channels encode than by using 128kbps of parametric info to a single 64kbps channel?

[3dsnar]

Gabriel, thanx for ideas and suggestions.
This may seem a little odd, but reconstructing the total panorama
is a balance between the perceptual encoder settings, and ability to reconstruct phase in all channels.

Therefore so much bitrate has to be used for the encoding the spatial information.
(this is also the major difference between our approach, and the one applied in mp3surround
and AAC 5.1).

Ofcourse everything is a matter of subjective taste. After hundreds hours of listening tests (all in 5.1 speakers set in a studio) the balance between
the main audio stream quality and spatial perception was established.
Such parameters tuning are very, very time consuming and decision is also not easy. We have tested several configurations on approx. 30 people of various
listening experiance, and based on their feelings the decision was made.

Probably the listeners preferences (resulted in high bitrate for spatial info),
were probably caused by the fact, that the chosen test material was only movie soundtracks
with very reach 5.1 panorama.
And creatomg codec tuned for movies was our primary target.

---

But you have a point here. A ~160 kbps bitrate would fill a gap between
128 and 192 kbps. And therefore might be useful.
This is something for the future, which we will definitely consider.

[elenhil]

I tested it again.;)
Uncheck the mixer itself, where is only(I believe) place to downmix the decoded(upmixed) multichannel by 'dolby decode'.
It has never been checked in my configuration. Ever.

[elenhil]

3dsnar, will the new Aud-X DirectShow filter support other formats besides mp3?

[3dsnar]

In fact our filter supported also AC3, but I have disabled this.
Also, the mp3 compatibility is limited, to allow for FFDshow and AC3filter
usage with our DSfilter.

Very soon, we want to add a control panel of our DSfilter,
which will enable to choose formats decoded by the filter,
and it will also allow for disabling the direct show module.

I hope to have this ready in February.

But maybe you have some specific problem
and maybe I will be able to sort it out?
Please let me know

[elenhil]

Very soon, we want to add a control panel of our DSfilter,
which will enable to choose formats decoded by the filter,

The question is whether it will support OGG Vorbis (and thus decode a DPL-encoded ogg file)?

[3dsnar]

Why not.
So far, we have implemented mp3 and AC3 decoding,
so decoging Ogg would require a bit of extra work.
Therefore I cannot specify when exactly Ogg compatible
direct show could be released, but it will be considered for 100%.

[Rockaria]

It has never been checked in my configuration. Ever.OK... it seems not be either my or FFDShow's fault.
The 'dolby decoder' does the DPL II decoding(to 6ch) no problem here(but with somewhat too much center).
Anyway, read the Audio-FAQ which is the best help you can find here.

What I can do is just helping each other if we can and want.. voluntary.
So if you don't call me again (ever), this thread is none of my interest.

[3dsnar]

We have just lounched Aud-X forum
www.aud-x.com/forum

So everyone who has qustions, suggestions,
requests, bug reports, etc. related to the project

is welcome to share her/his opinion :)

[sigma_sigma]

very good project. generally a movie DVDRIP on 2 cd's with ac3 sound (6ch) has a good video quality but with your codec, 5.1 sound will be smaller and video percent for video will be bigger and this results a better quality.
Greta job|!

[3dsnar]

sigma_sigma,
thanx for interest in our project.

Please note,
you may find very useful and user friendly guides
for encoding, decoding and transcoding on our web site:
http://www.aud-x.com/images/stories/downloads/aud-x_dvd_backup_guide_(full).pdf
and also
http://www.aud-x.com/index.php?option=com_remository&Itemid=34&func=download&filecatid=2
http://www.aud-x.com/index.php?option=com_remository&Itemid=34&func=download&filecatid=10

[3dsnar]

Short info:
Aud-X MP3 5.1 support added to FFDshow
http://bittekeinspam.googlepages.com/
:thanks:

[shon3i]

Nice, Thank You

[buzzqw]

@3dsnar

Could be possible to implement stdin support ?

this is very useful with bepipe/behappy encoding

thanks

BHH

[3dsnar]

@3dsnar

Could be possible to implement stdin support ?

this is very useful with bepipe/behappy encoding

thanks

BHH
This is already done... :)
By Dimzon. Please check this out:
http://www.gotdotnet.com/workspaces/workspace.aspx?id=1bb59ddf-901b-43a5-bd54-b0999e8e223e

[buzzqw]

:) :thanks:

i missed the link !

BHH

[3dsnar]

I have requested some interface changes regarding Aud-X usage
in FFDshow.
I think this should be more straight forward and stable.
Please take a look at this post
and share your thoughts.
http://forum.doom9.org/showthread.php?t=98600&page=60
:thanks:

[raquete]

hy all!
:stupid: but i:readguid: and can't find answers!
excuse my ignorante,newby and maybe stup question:
can i encode and play mp3 5.1(my preference is vbr) in my standalone without video?
i only want the sound.

thanks !
:o

[3dsnar]

You can encode sound with our HGHQ encoding option
(192 kbps). This is Dolby Pro Logic II compatible, so you will
get some sort of surround sound on standalone DVD (assuming the DPLII decoder will be deployed).
When decoding this sound on PC, Aud-X auxiliary information contained in the stream will be used for upmixing to 5.1
(this results in much better, than DPLII quality).
So this approach is doulbe-commpatible (Aud-X MP3 5.1 and DPLII)
--
Alternatively, you can create a DPLII downmix and encode it as normal mp3 VBR, and also you will be able to get a sort of surround on standalone DVD player.
But it will not be compatible with Aud-X.

[raquete]

wonderful and clear explanations 3d

thank you so much!
:)

[TFM_TheMask]

How do I know if the avi that I have contains Aud-X audio or just normal MP3 audio?? Is there a way to check this.

When I open an Aud-X file it always shows MP3 audio with only 1 channel (mono).

[Pomyk]

Please fix the bugs first.. :|

[3dsnar]

How do I know if the avi that I have contains Aud-X audio or just normal MP3 audio?? Is there a way to check this.
When I open an Aud-X file it always shows MP3 audio with only 1 channel (mono).
1) You have to use a direct show compatible player (e.g. MediaPlayerClassic, BSplayer etc. ).
2) Than, make sure that in CONFIG tab
[start->programs->Aud-X->Aud-X.exe] of our decoder
Aud-X/MP3 output is enabled.
3) Play the avi, and Aud-X icon shoud appear in the tray. Double click on it, and you will acces direct show filter window.
4) at the bottom of the DSfilter window it should be indicated what type of stream is currently decoded. This appears only during playback.
---------------------------------
In FFDshow it is not possible to determine that (no appropriate description is displayed).
Please request this feature through SourceForge:
http://sourceforge.net/tracker/index.php?func=detail&aid=1485338&group_id=53761&atid=471492
Cheers, 3d

[3dsnar]

Please fix the bugs first.. :|
What bugs do you have in mind?
Actually the latest versions are very stable.
However, if there are any problems that you would like to report,
please follow this link:
http://www.aud-x.com/forum/viewforum.php?f=2
:thanks:

[TFM_TheMask]

1) You have to use a direct show compatible player (e.g. MediaPlayerClassic, BSplayer etc. ).
2) Than, make sure that in CONFIG tab
[start->programs->Aud-X->Aud-X.exe] of our decoder
Aud-X/MP3 output is enabled.
3) Play the avi, and Aud-X icon shoud appear in the tray. Double click on it, and you will acces direct show filter window.
4) at the bottom of the DSfilter window it should be indicated what type of stream is currently decoded. This appears only during playback.
---------------------------------
In FFDshow it is not possible to determine that (no appropriate description is displayed).
Please request this feature through SourceForge:
http://sourceforge.net/tracker/index.php?func=detail&aid=1485338&group_id=53761&atid=471492
Cheers, 3d


Can't you build in something that it can be detected before playing, like with mediainfo or with AviInfo.

[3dsnar]

Well, it is backwards mp3 compatible, therefore it appears as a regular MP3. However there is a 'Aud-X1.0' string in the audio stream.
Based on this an application can recognize that it is dealing with Aud-X.
--
When you use our DLL, it returns an appropriate information,
indicating whether this is or isn't the Aud-X stream.
You can download the SDK here:
http://www.aud-x.com/index.php?option=com_remository&Itemid=34&func=download&filecatid=11

BTW. I have PM-ed you. Please chack your mailbox.

[Pomyk]

What bugs do you have in mind?
Actually the latest versions are very stable.
However, if there are any problems that you would like to report,
please follow this link:
http://www.aud-x.com/forum/viewforum.php?f=2
:thanks:

It's crashing on some normal mp3's. I reported it some time ago and sent a sample.

[3dsnar]

IMHO this bug is fixed. Please do the following:
1) make sure that you are using v. 1.21 (help tab)
if not
2) if not, please uninstall the current version
3) manually delete audxdshow.ax from c:\windows\system32 folder
4) install the new version

If the problem persists,
please let me know.
Thank you.

[3dsnar]

Big kudos to Jacquers
for (unofficial) AVI.NET
support for Aud-X MP3 5.1
http://forum.doom9.org/showthread.php?p=826587#post826587

[Pomyk]

The problem is still in v1.21. I think reported it after 1.21 was released so how could it be fixed? And I would really like to use this codec :/

edit: it crashes on every 192kb 48kHz mp3 here.

[3dsnar]

Please send us some example (problematic) file once more.
helpATaud-x.com
Thanks for the bug report.

[dimzon]

3dsnar
Any news about licencing problem (MPG123 still GPL)

[3dsnar]

So far we have provided information
(as u suggested) regarding this issue:
http://www.aud-x.com/content/view/5/27/
Cheers, 3d

[dimzon]

So far we have provided information
(as u suggested) regarding this issue:
http://www.aud-x.com/content/view/5/27/
Cheers, 3d
Please, dont get me wrong, but this is only short-term solution.
You must get explitit permissions from MPG123 author OR make your code open source

http://lame.sourceforge.net/license.txt

Can I use LAME in my commercial program?

Yes, you can, under the restrictions of the LGPL. The easiest
way to do this is to:

1. Link to LAME as separate library (libmp3lame.a on unix or
lame_enc.dll on windows)

2. Fully acknowledge that you are using LAME, and give a link
to our web site, www.mp3dev.org

3. If you make modifications to LAME, you *must* release these
these modifications back to the LAME project, under the LGPL.


*** IMPORTANT NOTE ***

The decoding functions provided in LAME use the mpglib decoding engine which
is under the GPL. They may not be used by any program not released under the
GPL unless you obtain such permission from the MPG123 project (www.mpg123.de).

PS. Can You provide me link to Aud-X Logo?

[dimzon]

BeHappy extension to support Aud-X 5.1
http://img105.imageshack.us/img105/6730/untitled7ir.png

[3dsnar]

Hi Dimzon,
yes, the explanation is a temporary solution.
---------
Looks good. Thanks for utilizing the graphics.
Cheers, 3d.

[Rockaria]

I come to think the spatial info(ancillary data) is imbeded in the 'phase-out' area, controlling the frequency band anyhow, based on the FV-LAME-DPL II technology.

The ffdshow imbeding is an unexpected implementation for me, especially on a deviation(?) version, losing some functionalities except the core codec.
I didn't test the quality of the core codec because there was no mention of any improvement.

Now some usability issue :

<the ffdshow imbeded codec>
. the mp3 decoding works well, and supposedly the aud-x also(I didn't test)
. the ffdshow version has lost the 'check DTS signal in wav' feature.(reinstalled the formal version)

<the audx.exe cli encoder>
. perfectly 'no help', in a hope to integrate to foobar2k cli encoder front end.

<the aud-x dsfilter>
. insistantly sticks to the sound-out device only...
. not tested the crosstalk issue.

The hrtf feature is a good plan especially for LARGE headphone users.

Good Luck.

[3dsnar]

I come to think the spatial info(ancillary data) is imbeded in the 'phase-out' area, controlling the frequency band anyhow, based on the FV-LAME-DPL II technology.

The ffdshow imbeding is an unexpected implementation for me, especially on a deviation(?) version, losing some functionalities except the core codec.
I didn't test the quality of the core codec because there was no mention of any improvement.

Now some usability issue :

<the ffdshow imbeded codec>
. the mp3 decoding works well, and supposedly the aud-x also(I didn't test)
. the ffdshow version has lost the 'check DTS signal in wav' feature.(reinstalled the formal version)

<the audx.exe cli encoder>
. perfectly 'no help', in a hope to integrate to foobar2k cli encoder front end.

<the aud-x dsfilter>
. insistantly sticks to the sound-out device only...
. not tested the crosstalk issue.

The hrtf feature is a good plan especially for LARGE headphone users.

Good Luck.
Hi Rockaria,
thanks for your comments.
It is not similar to DPLII, because the way the signals are being mixed is not used for creating 5.1 sound. In other words, there is not information hidden in the signal, which is later used for decoding, but only in the ancillary data. Moreover I have found out that people think that this is matrix (like DPLII) encoding. This is incorrect, because information describing other channels (beside the ones that are audible when played with standard mp3 player) is also present there.

Please address the FFDshow issues to FFDshow developers:
http://sourceforge.net/tracker/?group_id=53761&atid=471492

And audx cli issues to Dimzon:
http://www.gotdotnet.com/workspaces/workspace.aspx?id=1bb59ddf-901b-43a5-bd54-b0999e8e223e

---
Indeed the DSfilter still has the limitation that you mentioned.
When the HRTF thing will be implemented, I will report it here.

:thanks:

[Rockaria]

Still got a bit puzzled, not that much impacting my life yet though.;)

The audx.exe issue, I have probably figured out the parameters somewhere in this thread before, is I believe your issue for any users who might be interested in some other cli frontend integration....

[3dsnar]

Still got a bit puzzled, not that much impacting my life yet though.;)

The audx.exe issue, I have probably figured out the parameters somewhere in this thread before, is I believe your issue for any users who might be interested in some other cli frontend integration....
Uhm, indeed. We will support all software toolz using aud-x on our web site and provide necessary information related to them (e.g. how to use CLI, etc).
Cheers, 3d.

[Rockaria]

Good, that sounds reasonable.

..but only in the ancillary data. But it still does not answer where itself is located.
Moreover I have found out that people think that this is matrix (like DPLII) encoding. This is incorrect, because information describing other channels (beside the ones that are audible when played with standard mp3 player) is also present there.
AFAIK, I didn't start mentioning the DPL II as part of the solution to the crosstalks, fortunately.;)

Hum! but again, I conclude it deserves 'not disclosable', indeed.. (http://www.answers.com/indeed)

[sillKotscha]

Hi 3dsnar,

just a little bugreport concerning the Aud-X DirectShow decoder...

if you choose "send surround sound through spdif" and you choose 'movie mode' as output then it is not remembered until the next start... it switches automatically to 'music mode'

and a little question... is it possible to implement some kind of sound detection? because if you choose "send surround sound through spdif" your sound will get upmixed to dpl II even if you have standard stereo - and that sounds not always good :)

maybe just like in ffdshow: you have the mixer option 'same as input' - so, if dpl II streams are available they will get decoded via dpl II if not only standard stereo decoding is used

thynk you - Sill

[3dsnar]

Hi Sill,
If you chose: "Send surround sound through spdif"
all streams will always be converted to 5.1 (i.e. mono or stereo).
When you want to play a stereo sound, the 'Pseudosurround' mode
should work fine (instead of DPL2 upmix)

However, if you do not want to do any channel conversion,
you should select a standard stereo output, and force your soundcard to send the sound (as PCM) through spdif output.

:thanks: 4 the bug report.

[Rockaria]

<the audx.exe cli encoder>
. perfectly 'no help', in a hope to integrate to foobar2k cli encoder front end.


Uhm, indeed. We will support all software toolz using aud-x on our web site and provide necessary information related to them (e.g. how to use CLI, etc).
Cheers, 3d.Sorry, but it's not that I wanted to hear!
Imbeding the help in the encoder is the widely accepted simple way.

[3dsnar]

Sorry, but it's not that I wanted to hear!
Imbeding the help in the encoder is the widely accepted simple way.
Oh, I see.
U are right
----------
Dimzon, are you there?
Could you please add some help option in the STDIN CLI
that you've prepared?
Please let me know.
----------
We will add help to audx.exe (used by our frontend) as well.
Thanks, 3d

[Rockaria]

Hi Sill,
If you chose: "Send surround sound through spdif"
all streams will always be converted to 5.1 (i.e. mono or stereo).
When you want to play a stereo sound, the 'Pseudosurround' mode
should work fine (instead of DPL2 upmix)

However, if you do not want to do any channel conversion,
you should select a standard stereo output, and force your soundcard to send the sound (as PCM) through spdif output.

:thanks: 4 the bug report.
If I am not incorrect, he made two inquires :
1. can it save the last setting to be reused for the next launches ?
2. can the output mode set automatic depening on the system speaker setting detection ?

Another sorry but I believe none of them answered?

[3dsnar]

I am sorry, I forgot to anwer :scared:

Interestingly, in my computer all the settings are saved.
Is your sound card 5.1? or stereo with SPDIF?
---
Automatic is not possible, because many soundcards support
stereo, 5.1, spdif, etc. Which should be chosen by default?

3d

[dimzon]

Dimzon, are you there?
Could you please add some help option in the STDIN CLI
that you've prepared?
Please let me know.

it's already here. Just run enc_AudX_CLI.exe without additional arguments and You will get
##################################################################
Aud-X multichannel encoder CLI frontend
this algorithm has been prepared by: [Aud-X Team www.aud-x.com]
##################################################################
run enc_AudX_CLI.exe --aud-x to view Aud-X license
run enc_AudX_CLI.exe <input.wav> <output.mp3> [options] to encode
options:
--q <quality> - 0 (STRQ 80 kbps), 1 (STDQ 128 kbps), 2 (HGHQ 192 kbps) or 3 (SPBQ 192 kbps), default is 1
--raw <samplerate> - signal input as raw pcm file instead of RIFF WAV
note: to use STDIN use - as input filename

[3dsnar]

OK, :thanks:

[Rockaria]

Hi 3dsnar,
if you choose "send surround sound through spdif" and you choose 'movie mode' as output then it is not remembered until the next start... it switches automatically to 'music mode'
Maybe we need some clarification.
It sounds to me the DPL II 'movie mode' decoding is not effective until you close the player and launch again.
Maybe the dpl II decode mode state variable is not updated dynamically.

[sillKotscha]

If I am not incorrect, he made two inquires :
1. can it save the last setting to be reused for the next launches ?
2. can the output mode set automatic depening on the system speaker setting detection ?

Another sorry but I believe none of them answered?

nono... both are answered :)

just have a look at his "thx 4 the bug report." and also his answer did satisfy me - because he already said that if output via spdif is used every file gets an upmix to 5.1... and what to choose if I don't want that...

so, yes - thx for your answer(s) 3dsnar

cheers Sill

[3dsnar]

Maybe we need some clarification.
It sounds to me the DPL II 'movie mode' decoding is not effective until you close the player and launch again.
Maybe the dpl II decode mode state variable is not updated dynamically.
It is updated automatically. :)

[Rockaria]

if you choose "send surround sound through spdif" and you choose 'movie mode' as output then it is not remembered until the next start... it switches automatically to 'music mode'

nono... both are answered :)

just have a look at his "thx 4 the bug report." and also his answer did satisfy me
Maybe we need some clarification.
It sounds to me the DPL II 'movie mode' decoding is not effective until you close the player and launch again.
Maybe the dpl II decode mode state variable is not updated dynamically.
It is updated automatically. :)
3dsnar, If you agree with it, then I am incorrect, oh well!;)

[3dsnar]

I think I have answered all Sill's questions.
Besides, normally the DSfilter settings should be remembered,
however there always might be some bug.
--
So I think you are incorrect :D
Cheers, 3d

[Rockaria]

..and a little question... is it possible to implement some kind of sound detection? because if you choose "send surround sound through spdif" your sound will get upmixed to dpl II even if you have standard stereo - and that sounds not always good :)
...
ffdshow... 'same as input' .. will get decoded via dpl II
On this part, I indeed misunderstood.
The actual inquiry is that can the decoding/encoding mode set automatically depending on the SOURCES(not by system speaker setting...).

But we still need some clarification :
. the DPL II signal cannot be detected, it's regarded just as stereo, so you have to force it to decode.
. 'upmix to DPL II' is not possible, 'decode dpl II to 6ch' or 'encode to dpl II(2ch)' are possible.

Anyway, it seems to me 'good answers to good questions' beyond my logic.:D

[3dsnar]

On this part, I indeed misunderstood.
But we still need some clarification :
. the DPL II signal cannot be detected, it's regarded just as stereo, so you have to force it to decode.
. 'upmix to DPL II' is not possible, 'decode dpl II to 6ch' or 'encode to dpl II(2ch)' are possible.

Anyway, it seems to me 'good answers to good questions' beyond my logic.:D
Hmm. in both cases you are right.
But I used my interpretation coprocessor (embedded in my brain),
and I was able to figure out what Sill meant, although his questions where not correctly formulated, indeed ;)

[sillKotscha]

I was able to figure out what Sill meant, although his questions where not correctly formulated, indeed ;)

hey, don't pick on me 'cause of my english - I allways think it's near perfect :D

[3dsnar]

your english is nearly perfect :D

[Rockaria]

thx 4 the bug report But still you have to fugure out what has caused the bug. right?

The symptom is plainly not keeping the state during the session. Maybe it is set only during the initialization, which is the frequent mistakes I have seen.;)

[3dsnar]

I think I know what could have caused the bug.
The DSfilter should automatically switch itself to standard
stereo output, when the current setting does not comply with the DirectSound device.
So the answer may be related to this.
This is meant for a situation, where the user choses 5.1 output, while it is not supported.
But maybe there is a bug, and something is wrong...
We will check this out.

[Rockaria]

The DSfilter chennel setting should be independant of the system output device, of course provided no conflicts.

For an example, I use 5.1ch in the FFDShow output when the speaker is set to 2ch in one of my machine.
The system mixer does the plain downmix to 2ch for analog headphone, while I can do everything possible with 5.1 output in the filter chain: dd encode passthrough, 5,.1ch capture, dpl2 encoding...

Maybe then, unnecessarily resetting to stereo during the session in the DSFilter!

[Rockaria]

Another simple Q for a clarification on the audx.exe :

AFAIK, your own cli encoder backend(audx.exe) supported all the encoding features except the stdin input support, with own quality grade scheme.

Any special reason to maintain redundant(except the stdin) versions kept in different places?

[3dsnar]

The DSfilter chennel setting should be independant of the system output device, of course provided no conflicts.

For an example, I use 5.1ch in the FFDShow output when the speaker is set to 2ch in one of my machine.
The system mixer does the plain downmix to 2ch for analog headphone, while I can do everything possible with 5.1 output in the filter chain: dd encode passthrough, 5,.1ch capture, dpl2 encoding...

Maybe then, unnecessarily resetting to stereo during the session in the DSFilter!
The only problem is that when you accidently chose an output option, that is not supported by your device (e.g. you have a stereo only soundcard and you chose 5.1 output and destroy the filter), than in the second initialization, the filter will not occure because the output pins will not comply with your device. So in order to overcome this problem, the filter output should be automatically switched to a supported type of pin (i.e. stereo output) during initialization. Otherwise you would have to switch it back manually with the use of DSFmanager, or something, but this is not so obvious for many users.

[3dsnar]

Another simple Q for a clarification on the audx.exe :

AFAIK, your own cli encoder backend(audx.exe) supported all the encoding features except the stdin input support, with own quality grade scheme.

Any special reason to maintain redundant(except the stdin) versions kept in different places?
Our executable frontend (GUI) uses this exec.
Also BeLigth uses it. So the decision is yours :)

[Rockaria]

The only problem is that when you accidently chose an output option, that is not supported by your device (e.g. you have a stereo only soundcard and you chose 5.1 output and destroy the filter), than in the second initialization, the filter will not occure because the output pins will not comply with your device. So in order to overcome this problem, the filter output should be automatically switched to a supported type of pin (i.e. stereo output) during initialization. Otherwise you would have to switch it back manually with the use of DSFmanager, or something, but this is not so obvious for many users.
That seems to be the exact description of your DSFilter.

Our executable frontend (GUI) uses this exec.
Also BeLigth uses it. So the decision is yours :)
I just realized there's one more option, if I have to choose now.;)

[3dsnar]

Aud-X MP3 5.1 v. 1.22 released:
http://www.aud-x.com/content/view/12/26/

The new version consists of a new virtual surround algorithm,
which enables to reproduce surround panorama with the use of stereo headphones.
Description of the algorithm in a separate thread:
http://forum.doom9.org/showthread.php?t=111889

BTW.
- Changes concern only DSfilter
(No changes in the SDK.)
(No changes in audx.exe command line)



Thanx,
3d

[tebasuna51]

Maybe the link to Aud-X MP3 5.1 v. 1.22 must be:
http://www.aud-x.com/content/view/12/26/

[3dsnar]

Yeah, right, thanx :)
edited.

[Avish]

Hi 3dsnar,

It's crashing MPC when I try to change between 5.1 to Headphone mode or vice versa. MPC just disappears. Is it just me or anybody else xperienced the same?

BTW, Is is possible to add "Normalize" feature to the decoder? I get pretty low volume compare to other decoders.

[3dsnar]

Please give me more info.
What signal did you test?
Did you try other players (e.g. graph edit)?
If the file was AC3, was Aud-X used for decoding, or maybe some other DSfilter preceding Aud-X DSfilter?
Maybe you can send me the clip?
Thank you.

[Avish]

Please give me more info.
What signal did you test?
Did you try other players (e.g. graph edit)?
If the file was AC3, was Aud-X used for decoding, or maybe some other DSfilter preceding Aud-X DSfilter?
Maybe you can send me the clip?
Thank you.Sure, in an hour or so.

But why u ignored my other query?

[3dsnar]

oops, sorry.
Dynamic normalization is... problematic,
bacause it affects amplitude relations in the signal.
However, if you are playing AC3 file, you may use FFDshow
for gain manipulations.

[Avish]

oops, sorry.
Dynamic normalization is... problematic,
bacause it affects amplitude relations in the signal.
However, if you are playing AC3 file, you may use FFDshow
for gain manipulations.OK. Thanks. mp3 output seems to be quiet OK, but ac3 is way low. so I guess I won't be using it for ac3. But I'm OK with mp3's sound now. ;)

About the crash, I tried MPC, GraphEdit & WMP11. MPC seems to crash 8 out of 10 times, but the other two only crashed once or twice. I tried MKVs with x264 + ac3 in it & AVIs containing XviD, DivX & Div3 + mp3s.

I can't provide u any sample, coz I couldn't found any perticular culprit. eg. If it crashed in MPC, it didnt crashed in GraphEdit. So...

Is this much info usefull? :)

[3dsnar]

OK, thanx for the info.
I will do some more tests.

[Avish]

Another thing, Codec tools like GSpot, Sherlock, DShow Filter Manager still shows the product version as 1.0.0.0. Shouldn't it be something like 1.2.2.0? ;)

[3dsnar]

OK, thx :)

[3dsnar]

v1.23 is out

[Kurtnoise]

Hi,

I've a question for you Mr 3d ;) Do you know a SAP which supports AVI files within your mp3 multichannel sound ? If yes, could you point me a link please...If no, is it planned ?

[3dsnar]

Hi Mr Kurtnoise.
What is SAP?

[Kurtnoise]

Stand-alone Players...

[3dsnar]

I see. Now we are not planning to.
(I assume that you mean software players ;) )
But everyone is welcome to embedd our library
in such software (some people mentioned that they're gonna
do it, but nothing was released so far).
Cheers, 3d.

[Kurtnoise]

Note that with this comparison (http://divxtest.surdvd.com/form/divxtest2_view.php), some SAP are able to play mp3 5.1...Dunno, if it concerns your codec.

[edit]After some checking, this is the Fhg surround encoder.

[3dsnar]

OK, thanx for info anyway.

[SeeMoreDigital]

Any new developments?

[squid_80]

Or better yet, how about some source code so you're not violating the GPL? It was promised a long time ago now...

[Adub]

good point squid_80. I am curious to look at some of the source code.

[sillKotscha]

good point squid_80. I am curious to look at some of the source code.

YOU are curious to look at the source code... man, are you even able to read a newspaper let alone a source code of such a complex software??

we are waiting for your optimizations :sly:

[SeeMoreDigital]

Hmmm!

Even though the direct-show decoder offered by Aud-X is very good (with excellent input/output funtionality) I was wondering whether "decoding" support could eventually be added to FFdshow :)


Cheers

[sillKotscha]

Hmmm!

Even though the direct-show decoder offered by Aud-X is very good (with excellent input/output funtionality) I was wondering whether "decoding" support could eventually be added to FFdshow :)


Cheers

it is already done ;)

but unfortunately it all depends on the revision and I can't remember on which revision I saw Aud-X as an option for decoding for the last time...

[SeeMoreDigital]

it is already done ;)

but unfortunately it all depends on the revision and I can't remember on which revision I saw Aud-X as an option for decoding for the last time...Bummer.... I must have missed those meetings :scared:

Just checked my currently installed version and it's not there - But it looks as though there's room for it: -

http://img141.imageshack.us/img141/3919/ffdshowtn5.png

[sillKotscha]

Bummer.... I must have missed those meetings :scared:

I know for sure, that videomixer9 included Aud-X in his ffdshow builds... BUT his last build is dated August 3 (https://sourceforge.net/project/showfiles.php?group_id=173941), 2006 and considered/ marked as unstable...

the last proposed stable version is from December 7 and I must say it is rock solid for me and stable as in terms of stable. Only because of Aud-X I won't go backwards with my ffdshow installation...

I can't say what was the reason to exclude Aud-X from the recent builds but following the ffdshow dev thread no one really missed it. ;)

[SeeMoreDigital]

I can't say what was the reason to exclude Aud-X from the recent builds but following the ffdshow dev thread no one really missed it. ;)Sussed it....

I scanned thru' my external HDD's and found a version of Audxlib.dll (http://82.10.220.174/Uploaded_Files/Doom9_Forum_files/Audxlib.7z). After dropping it into FFdshow's main folder, FFdshow is now able to decode 6Ch MP3's just fine ;)

[clsid]

rev657 aka beta1 contains audx