Additions and comments on "GUIDE LIST: Stereo-to-surround Conversion.." [Archive]

Shayne

30th October 2004, 03:28

So with a list like that a little grouping of whats worth trying for what type of material is def required (grouping according to material). And no way in my wilds dreams am i going to try each one of these. So i would suggest order to these groups 1 2 3 in order of best results (your liking). This is a good start for some debate.

N i would like to know what happen to Eye of Horus guide sticky?

http://forum.doom9.org/showthread.php?threadid=60137&perpage=20&pagenumber=37

Fairly lame if i do say so myself.

I think a public response to why doom9.org has taken this type of knowledge out of peoples reach is required.

Maybe it can be added to doom9.net guide list.

Peace

KpeX

30th October 2004, 15:05

EoH removed his own guide, it was his choice to do so. I was forced to remove it from sticky status as it is useless to have a sticky thread without a guide.

ursamtl

30th October 2004, 15:53

Hi Shayne,

I understand that no one has time to try all the methods listed but rating them could be problematic. Where will these ratings come from? No matter how it's done, someone is going to feel it's unfair. The moderators have asked me to categorize the methods, and that's what I've been doing. It's not complete yet by any means, but the grouping is coming along. You can check the guide list's List History section for a bit of an explanation.

As for the EoH guide sticky, as kpex stated, the Eye of Horus guide was removed by Eye of Horus himself on October 12.

Since then, I've been updating the guide list almost daily with new information as it becomes available, which you can see listed in the update notices. Daphy has put together a couple of excellent animated visual guides and I've written a generic guide thread that applies to all Plogue Bidule-based methods. Several forum members are putting a lot of work into their surround methods. SurroundBoy has contributed a nice guide for Motown-style music, and joshbm has been constantly working to perfect his DPLII guide. These developments will continue as forum members continue to contribute valuable information and build up our surround knowledgebase. I look forward to any contributions you may be able to make.

Regards,
Steve.

Shayne

30th October 2004, 16:18

Originally posted by ursamtl

...... I understand that no one has time to try all the methods listed but rating them could be problematic. ...........

Regards,
Steve.

I don't know Doom9 has been doing it with video codec for sometime. You need to start it somehow (your preference) and from there who knows scientific tests could develop. But to put this huge list here as a sticky is not of too much help.

With regard to EoH guide if daphy and EOH are around i have this pdf'ed and we could link to daphys' ftp if EOH doesn't mined?? And then we can put it high on the list.

ursamtl

30th October 2004, 17:51

Originally posted by Shayne
I don't know Doom9 has been doing it with video codec for sometime. You need to start it somehow (your preference) and from there who knows scientific tests could develop. But to put this huge list here as a sticky is not of too much help.

With regard to EoH guide if daphy and EOH are around i have this pdf'ed and we could link to daphys' ftp if EOH doesn't mined?? And then we can put it high on the list.

As I understand it, Daphy and @ndy removed the PDF version of the thread from Doom9, because EoH requested that the guide be removed as he considered it his "intellectual property."

As for the ratings system, I'd certainly be willing to consider some sort of ratings, but part of the difficulty is because different audio sources produce varying results, depending on the method. Perhaps a better system would be to simply find a way of listing projects that provided good results using a specific method. We could start a thread dedicated just to results and then I could link to specific posts from within the guide list. What do you think?

Steve.

Shayne

30th October 2004, 18:04

Originally posted by ursamtl
[i] ..... As for the ratings system, I'd certainly be willing to consider some sort of ratings, but part of the difficulty is because different audio sources produce varying results ....

Steve.

Please reread the first post?

Grouped in terms of audio source number in terms of best results is i think what i said?

And this rerouting a post in this fashion is borderline ridiculous.

ursamtl

30th October 2004, 20:34

Originally posted by Shayne
Please reread the first post?

Grouped in terms of audio source number in terms of best results is i think what i said?Ok, but again since rating the quality of a method is fairly subjective, it's hard to assign ratings that can be considered fair. Perhaps a better way would be for people to report on specifics of their experiences, such as I did with some test results (http://forum.doom9.org/report.php?s=&postid=536560#post536560).Originally posted by Shayne
And this rerouting a post in this fashion is borderline ridiculous.Sorry you feel that way. Forum moderators, not only here but everywhere, often transfer messages or whole threads for a variety of reasons. I've discussed this with the moderators and they agree that the list is meant to be a resource and that discussion of it is better if it's placed in this separate thread. That is why they split off Daphy's message at the beginning of this thread, thereby creating the discussion thread. As well, at both the beginning and end of the guide list, this thread is mentioned as the place to discuss the thread.

Regards,
Steve.

trooper11

6th December 2004, 05:59

I applaud you for going to all the trouble to organize the varied guides and processes for 5.1 conversions. For a while I was trying to work alot with the first big guide that EOH helped put together, and it was great.

i did have a question though, or maybe its a suggestion lol.

Would it be possible to organize guides by what type of music the results work best for? For instance, my source will be alot of cd soundtracks, from anime and gaming types. Of course they all dont have the same sound, so I guess thats too broad of a genre. I guess Im just trying to save myself some time trying them all since i have alot of cds to do. If guides could be organized in order toof type like rock, classical, etc, that would be a help.

Also, a rating of complexity might be nice too, such as guides that are easier to impliment then others.

ursamtl

6th December 2004, 14:20

This is definitely a good idea, it's just a matter of how we implement it. It would require a lot of feedback from the people using the different methods. They could report back on their findings for specific projects. I'm certainly willing to include some sort of info in the guide list or to organize it in such a way, but I'd need the info from those doing all the different conversions because firstly for the results to be seen as credible and even somewhat objective, they have to come from a variety of sources, and secondly, I simply don't have the time to try all the different methods on different types of music. I just purchased a new 5.1 soundcard and speaker system. I'm spending a lot of time right now playing with that and exploring all its potential. Once I get that all sorted out, I'll be doing a lot of new surround experimentation.

There is also the issue of message length. I tried editing the guide list a few days ago and when I went to add another entry, the system told me I'd reached the message length limit. So, I'm going to have to split it up a bit.

Saving presets and plugin settings is another idea I've had floating around on my to-do list for quite some time. This could hold a lot of potential, at least for Plogue and Cubase/Nuendo-based methods. If someone came up with a consistently good result for certain types of music, he or she could save the settings in a file that would be placed somewhere so that others could download the settings. This is definitely something I plan to implement with V.I by distributing am fxb settings file with it.

So thanks for your idea, I'll definitely think about it. You might help by sharing your experiences with the different conversions, and hopefully others will follow suit.

Steve.

trooper11

7th December 2004, 07:55

ok great, as i do more testing ill definitely report my findings and help organize things with you as much as i can, but of course im still rather new to the whole process lol, so it may take some time.

ursamtl

7th December 2004, 13:52

Great! Thanks for your offer. Don't worry about the time it takes. We're not working under any deadlines here.

oelewapper

5th May 2006, 23:17

An addition for general Reading and Information :

This forum contains more than 30 different guides about converting stereo to surround. A lot of the methods there give almost professional results and most of them use freeware only. It contains guides, not found here or anywhere else !

Elektra999

12th September 2006, 15:57

Hi, Steve

I am testing guides it of Wave Lab 5.1, but me do not I clarify on what one must do in the step 6.

He seems me stupendous this he guides, but me do not I clarify in the step 6.

Thanks..
Elektra999

ursamtl

12th September 2006, 23:03

Look on the left side of the track. You will see a silver box with the number "1" in it. When click on the box, it becomes a menu. From that menu, choose "Clone Track." This makes a copy of the track underneat it. Now do the same thing again to create a third track. This will give you three stereo tracks, to make up your six channels.

Regards,
Steve.

Elektra999

15th September 2006, 19:14

Hi, Steve
al to select, mode 6 channels, a message seems me of ASIO DRIVERS, I go al menu Options Preferences and does not appear ASIO Drivers

http://img181.imageshack.us/img181/1061/wavelabqy3.jpg

Thanks
Elektra999

ursamtl

15th September 2006, 19:45

You could try installing ASIO4ALL, a free ASIO driver that sits on top of your WDM driver.

Elektra999

15th September 2006, 20:10

Thanks by the rapidity of the contestacion.
This it solved.
Good guides Steve

http://img201.imageshack.us/img201/8100/wavelab2fk5.jpg

Regards
Elektra666

Elektra999

20th September 2006, 16:57

I have been still 5.1 the whole guide of WaveLab to the letter, with the Suite of the V.I - LF LR, CLFE, SL SR.
OBSERVATIONS AND QUESTIONS:

After ending the whole procedure with Wave Lab, I have listened to channel for channel.

1 - The channels SL SR., scarcely are voices heard, with Plogue Bidule if the voices are heard, for the channels SL and SR., because?
2 - have I noticed the a bit lower serious ones of the LFE and the a bit more heightened sharp ones, not if the culprits will be my ears that old men do to themselves???? Jajaja;-)
3 - The Classic Master Limiter, have I it fitted to 0dB, this one exactly well?
4 - serious possible, that you were making a plug-in of the channel CS (central back) of the VI.I?

I have enjoyed very much the guide of the WaveLab, you are Steve A GENIUS, I have liked the results very much, I am very amazed of well that works you the sound.

Thanks Steve
Elektra999

ursamtl

22nd September 2006, 02:09

Hi Elektra,

Once more thank you for your feedback. I'm glad you're enjoying your experiments with the V.I Suite.

I don't know why you get voices in the surrounds with Plogue but not in Wavelab. Perhaps it is the source material. Sometimes voices will appear in the surrounds depending on how the source is recorded.

Yes, 0dB is a good setting for the Classic Master Limiter. At this level, it is just preventing overload and not compressing anything more than necessary.

Steve.

raquete

22nd September 2006, 04:46

Elektra999

22nd September 2006, 10:53

Hi, Raquete and Steve.

Raquete
I do not understand very well his question Raquete it, had to, to my English evil, but, I have a Marantz SR5500 (they can connect even the 7.1) I have 6.1 Infinty Beta20 connected to the Marantz. Central The channel back, I it have connected to the Marantz in the peg "SBL", this way I listen to 6.1.

I expect to have said something to him of interest.

Regards.

Steve
I remain experienced by WaveLab.
Thank you for everything, a greeting Steve.

ursamtl

22nd September 2006, 13:04

hi Steve,Elektra and all.

Steve,
tomorrow i will get one new receiver/decoder Sony 6.1 Dolby (ex II)....(oops i forgot the model).
don't mind the model now,is not important for my question...

i only want to know about the central back channel.
i can left the decoder "create" it (have this option in the receiver)but can i use like "oops" to this channel?
can you give one hint? (i still need to find any AC-3 6.1 encoder)

thanks!

I haven't really experimented much with 6.1 other than to create a separate VI.I plugin when some users requested it. The V.I surrounds are based on a modified Ambisonic layout that contained a rear center channel, which I simply mixed 50-50 into the sL and sR of the 5.1 version. So I ran it out separately to create the 6.1 version.

You could try generating the channel with the decoder and see how it sounds. As for the "oops" method. I've seen that name lately but it's really nothing new. As far as I understand it, the method is s the same old Hafler method that's been around since the 70's. It consists of taking the difference signal between the two channels as the surround feed. This is at the very core of the original Dolby Surround, Pro Logic I & II, the Y signal in Ambisonics, crosstalk cancellation, etc. Electronically, you can get the channel by hooking a speaker to the two positive outputs of a stereo amplifier or on a pro mixer by switching to M-S mode and taking just the S signal, or mixing the left and right channel together with one channel inverted. In software, subtract one channel from the other, use an M-S plugin and take the S output, or invert one of the two stereo channels and mix them together.

Regards,
Steve.

raquete

22nd September 2006, 21:11

I haven't really experimented much with 6.1 other than to create a separate VI.I plugin when some users requested it. The V.I surrounds are based on a modified Ambisonic layout that contained a rear center channel, which I simply mixed 50-50 into the sL and sR of the 5.1 version. So I ran it out separately to create the 6.1 version.i remember that you did a separate VI.I plugin now,i will find the post.
oops is undercover old Hafler method in 70's! :-)
in 74 i did this "dangerous" experience in my amplifier (lol)....well,works and i still have this ampl. working.

thanks Steve,very clear. ...thankyou too Elektra.

ursamtl

22nd September 2006, 21:27

oops is undercover old Hafler method in 70's! :-)
in 74 i did this "dangerous" experience in my amplifier (lol)....well,works and i still have this ampl. working.

Yes I first discovered it in an article in one of those old "101 Electronic Projects" magazines. I had a couple of 12" "full-range" speakers I sort of inherited so I built enclosures for them then bought a Radio Shack volume control and wired both of them up for "surround." When I was at university, I always had people visiting my dorm room to hear the system. I also wired the front in-dash speaker in my old car to a volume control and used it as a Hafler speaker and then the two big 6x9 speakers in the rear as the main ones. It sounded great! The trick was always to adjust the speakers until they just become apparent and then back off the volume just a bit. You could always reach a point where individual speakers cease to be apparent and you're just surrounded with a field of sound.

Actually when I got my first Dolby Pro Logic receiver, I didn't like it as well as the old tried and true Halfer speakers so I hooked them up too and had a 7-speaker setup!

Steve.

raquete

22nd September 2006, 22:12

comments (off topic..but in topic) :
i have seen lots of technicians/university audio students searching for money.great part don't have taste for music/audio.persons that really can discover or create something,are people that really like of what are doing.
i was an technician because i like music and i can feel that you was in university for taste,not for money.
now tell me: how many technicians/enginners "money searchers" can do something like VI. and send as gift to the community?
they only "copy", do little modifications and "paste" but don't create.(if you know what i mean)
i did crazy things with eletronic like start my car using a single calculator as remote control in 79. was completely useless (lol) because i was outside the car. (lol)
the good taste and experimentations like you did a long time takes to creativity. ;)
is like compare wine with water and lossless with mp3(lol).
when i was young i did lots of tube amplifiers,speakers and trust me,they still are working fine(well..more or less,new tubes are too rare here).
end of off topic(or little off topic)

regards Steve

Elektra999

25th September 2006, 14:57

Hi, Steve, Raquete and to all.

Steve Already because (I create) with WaveLab, in the channels SL SR. scarcely they appreciate the voices:

1 - Correct me if I am wrong Steve, when the audio converse with V.I in Plogue, if the Mode Movie is activated, there is activated the Mode Movie of all the channels? Included the backs?

2 - In WaveLab the plug-in SL SR. of the V.I cannot activate the Mode Movie and the voices are not estimated for this motive, I am in the certain thing?

3 - Always I activate the Mode Movie in Plogue, and now I have realized, that with WaveLab I can activate the Mode Movie of the plug-in CLFE that is in the one that I am interested, since I like to listen the voice with mas to clarity.

Thank you very much
Elektra666

Elektra999

25th September 2006, 18:11

Allow me Steve, if it is possible, I him can send two audios exactly equal of, the channels SL SR., but one is worn out with WaveLab and other one with Plogue.

Regards
Elektra999

raquete

25th September 2006, 19:14

...WaveLab and other one with Ploguei'm around but receive my excuses Elektra.i don't use this programs,better wait for Steve a little more.

regards

ursamtl

25th September 2006, 23:40

1 - Correct me if I am wrong Steve, when the audio converse with V.I in Plogue, if the Mode Movie is activated, there is activated the Mode Movie of all the channels? Included the backs?

No Movie Mode does not affect the surround channels. It simply removes some of the frequencies in the range of the human voice from the front left and right channels and puts them in the Center channel. This is why the Movie Mode switch does appear in the fLfR and CLFE pluginsm but not the sLsR plugins. These three 2-channel plugins duplicate their equivalent channels in the full V.I 5.1 plugin.

Regards,
Steve.

Elektra999

26th September 2006, 16:11

Steve excuses, I have just reported an audio in Plogue and other one in WaveLab and they are heard equal...
Not because? But on having done one day the conversion in Plogue there remained for the channels SL SR. a few voices that were listened very much.

I sit it Steve, forgive me for being weighed jajaja, sometimes with so much sound I finish madman ;)

ursamtl

27th September 2006, 23:16

Hehe. No problem, Elektra. I'm glad it worked for you.

raquete

28th September 2006, 00:50

I haven't really experimented much with 6.1 other than to create a separate VI.I plugin when some users requested it.
i remember that you did a separate VI.I plugin now,i will find the post.
Steve,i can't find the post,can you help?

thanks

Elektra999

28th September 2006, 12:39

Do they refer to this post?

http://forum.doom9.org/showthread.php?t=113590&highlight=VST+Plogue

Elektra999

28th September 2006, 15:36

I have a doubt:

Can channel create CS (central back) with WaveLab 5?
Does some way exist to create it?

Thanks
Elektra999

raquete

28th September 2006, 21:25

Do they refer to this post?

http://forum.doom9.org/showthread.php?t=113590&highlight=VST+Plogue

Elektra,
Steve send the "Suite VI.I 6.1" for you by mail? ! ?

send me a private message with your email and I'd be happy to put one together for you http://forum.doom9.org/showthread.php?p=856353#post856353

thanks!

ursamtl

29th September 2006, 13:05

I don't think it would work in Wavelab. You would need to create a 3-channel file and I don't believe that Wavelab supports this.

Elektra999

1st October 2006, 16:50

AUDIO FOR MUSIC

FL FR, Mode Movie - OFF and Master Classic Limiter to 0dB

Believe that the dynamic range is listened better in OFF

http://img441.imageshack.us/img441/7870/flfrea1.jpg

http://img513.imageshack.us/img513/8067/classichu8.jpg

CLFE, Mode Movie - ON and and Master Classic Limiter to 0dB

Will Lift the voice and the instruments will not be heard so much by the Central Front channel

http://img441.imageshack.us/img441/69/clfeqs1.jpg

SL SR, and Master Classic Limiter to 0dB

http://img426.imageshack.us/img426/7272/slsrux7.jpg

It is not necessary to do anything

The plug-in V.I in Plogue, if the Mode Movie is activated, activates the Mode Movie of the channels FL FR.
These methods of stereo to surround 5.1 with the Suite of the V.I are better that I have tried, the quality that V.I offers is, EXCELLENT!!!

Elektra999

1st October 2006, 17:42

What I work with WaveLab 5.1, also it is possible to do in Plogue Bidule.

Steve, I hope, that it comments that ii seems to him...

I am been charmed with of being in this forum:)

http://img351.imageshack.us/img351/5476/suitevinp7.jpg

Layout

raquete

1st October 2006, 22:59

Elektra,
using Master Classic Limiter to 0dB you don't hear(or feel) the sound like "tube warmth" that lose the whole dynamic of the music?

ursamtl

2nd October 2006, 13:34

At 0dB, The Classic Limiter functions as a brickwall limiter. This basically means that it doesn't affect any sound below 0dB but any peaks that would try to pass 0dB (and be cut off because digital audio can't go higher than 0dB) would be reduced in level to just reach 0dB.

This is a "last-resort" safety measure to prevent digital clipping. Some research I've been doing lately indicates that for complete preservation of the audio information, it's best to stay about -3dB below the 0 level because intersample estimating of levels can go above 0dB, and also because some electronic devices start distorting before the signal actually reaches the clipping point.

This would mean keeping the source levels from getting too loud. It requires a bit of experimentation and level monitoring or analysis before doing the actual upmix.

Elektra999

2nd October 2006, 16:07

Steve it to explained very well, I it had not explained better je je.

Steve,
who looks like to him the connections in Plogue and my commentary on the Mode Movie of the plug-in FL FR?

Regards
Elektra999

Certainly, I am working the channels SL SR, adding a small lack of coordination of milliseconds, to see, if I can do the channel CS Discrete, not Matrix.

raquete

2nd October 2006, 17:41

It requires a bit of experimentation and level monitoring or analysis before doing the actual upmix.you're right and i always do that for each track(i only encode musics,not movies)

Some research I've been doing lately indicates that for complete preservation of the audio information, it's best to stay about -3dB below the 0 level because intersample estimating of levels can go above 0dB, and also because some electronic devices start distorting before the signal actually reaches the clipping point. - 3dB is my "limit", i agree and you too.this is why i had asked about Master Classic Limiter to 0dB, for me,at odB sounds like warm tubes next to distortions.volume is for amplifiers,not for the source and with -3dB don't need to prevent digital clipping or i'm missing something in your explanations?

Elektra999

5th October 2006, 18:38

Hello
I finish talking, my original audio CD 16/44100 PCM WAV, (Evanescence - The Open Door) in DTS 5,1 with WaveLab and the Suite of the V.I, sounds of Wonder :):)

Elektra999

9th October 2006, 17:09

Hi, Steve.

I have a problem with Steinberg Cubase SX, only it recognizes 2 income of audio and 2 audio exits.

My card of sound is incorporated in the motherboard.

http://img50.imageshack.us/img50/9305/dibujoaf0.jpg

Thanks
Elektra999

ursamtl

9th October 2006, 17:19

I'm not sure if you'll be able to get this working with the built-in soundcard. You might try installing the ASIO4ALL driver available for free from http://www.asio4all.com/. This might provide a way for it to work.

Good luck,
Steve.

Elektra999

9th October 2006, 17:45

Steve, you are a genius really :)
I have been long time, trying to form Cubase SX and there was no form to make it work in 5.1, and you in a minute he solves to the problem jajajaja

Thanks you very much

Elektra999

http://img230.imageshack.us/img230/5709/dibujoof5.jpg

raquete

9th October 2006, 17:59

off topic
Elektra,
can you host only one track from the CD 16/44100 PCM WAV, (Evanescence - The Open Door)?
i (still) don't use the tools you're using and i want to do 5.1 and host for you again,then...you download,compare and tell me if sounds good!
can you?
regards!

Elektra999

24th October 2006, 18:32

Hi, Steve.
I believe that it is possible to make 6.1 with WaveLab v5.

To install AsioDrivers.
Menu / Tools / Preferences / Audio Card / Playback/Record, to select AsioDrivers.
If we select / Connections, we will see the channels and the order, OK.

http://img74.imageshack.us/img74/7447/asiodriversee7.jpg

01. Menú / Edit / Audio Properties, 32bits float, OK.
02. Menú / Eliminate DC offset, OK.
03. Menú / Edit / Create Audio Montage from Wave
04. Whole Range, OK.
05. Clone 4 tracks “stereo”. Menu / Edit / Mode / 8 Channels.

http://img324.imageshack.us/img324/7459/61canalesrj0.jpg

http://img60.imageshack.us/img60/23/014pistasmh4.jpg

http://img60.imageshack.us/img60/5149/024pistasav5.jpg

06. Track 1, Effect FL FR. Track 2, Effect CLFE. Track 3, Effect SL SR (channel ES Matrix). Track 4, Effect SL SR .
07. Channel 1 – FL. Channel 2 FR.

http://img261.imageshack.us/img261/4319/01fe4.jpg

08. Channel 3 – Center. Channel 4 – LFE

http://img317.imageshack.us/img317/5369/02tz2.jpg

09. Channel 5 – Surround (ES Matrix). Channel 5 – Surround (ES Matrix).

http://img410.imageshack.us/img410/1848/03rt7.jpg

10. Channel 6 – SL. Channel 7 – SR.

http://img108.imageshack.us/img108/3554/04op3.jpg

Steve. It seems to me that, is mixed well, channel (Central back) with WaveLab?

Thank you for his Suite V.I:)

Frank

Elektra999

24th October 2006, 19:07

11. Render, save Multimono.

http://img442.imageshack.us/img442/6532/guardarij7.jpg

WAV 1 - CHANNEL FL
WAV 2 - CHANNEL FR
WAV 3 - CHANNEL CENTER
WAV 4 - CHANNEL LFE
WAV 5 - CHANNEL ES MATRIX?
WAV 6 - CHANNEL SL
WAV 7 - CHANNEL SR
WAV 8 - DELETE

Elektra999

10th November 2006, 16:27

Hi, Steve and to all.
I believe that the 6,1 obtained with WaveLab 5, have come out well.
I have a question, so that 32bits float is applied?

Thanks
Frank

ursamtl

13th November 2006, 14:16

Elektra,

Sorry but I do not understand your question. If it's about bit depth, the best advice I can give is to always work in the highest bit depth you have available, preferably 32 bits, and then reduce to 24 bits for DVD or 16 bits for CD. Whenever reducing, be sure to use dither. In the past I had read that dithering when reducing from 32 to 24 bits is unnecessary, but recently I read some that to maintain the maximum resolution, always dither when reducing bit depth. However, avoid doing this until the finishing your project.

Elektra999

14th November 2006, 11:58

I,m sorry Steve
My question was:

01.In order that to apply them 32bits?
02. In an audio of 24/96000 is it necessary to apply them 32bits to codify 5.1?

I am glad to see him Steve :)
Thanks

ursamtl

14th November 2006, 13:54

1. Convert everything to 32 bits for processing.
2. Create the 5.1 mix.
3. Convert the individual files to 24 bits with dither. If this is not practical, you can do without dither when moving from 32 to 24 bits but you should never convert to 16 bits without dither as the results (quantization distortion) could be audible in quiet passages or in the surrounds.

Elektra999

15th November 2006, 16:31

Thank you very much Steve.

I believe that with its guide WaveLab 5.1, it is possible to add 3 effect, the dithering UV22 HR to every track stereo. From 32/96000 to 16/44100.

Regards! :)

Elektra999

11th December 2006, 13:07

ursamtl

11th December 2006, 13:55

Hi Steve
Do you know some VST plug-in Surround Panner of 6/7 channels that it works in Plogue Bidule? Though of payment.

I am waiting for the update of the DTS Master Audio, I will have to think of doing since 7.1

6.1 ES MATRIX I have managed it to do, as I commented in this post mas above .

Thanks Steve
Frank

Hi Frank,

You could try looking at http://acousmodules.free.fr/. This is a French site but there is also an English section. There are quite a few interesting multichannel panning plugins available.

Elektra999

11th December 2006, 16:32

Thanks Steve
I have been experimenting with the plug-in SpatMass 66. It is is more or less what was searching.

http://img216.imageshack.us/img216/2210/dibujo2nc0.jpg

I believe that the connections go this way:

http://img299.imageshack.us/img299/1948/dibujo3oz2.jpg
CS - SR - FR - LFE - CENTER - FL - SL

Audio Fle Recorder

WAV 0 - FL
WAV 1 - FR
WAV 2 - CENTER
WAV 3 - LFE
WAV 4 - SL
WAV 5 - SR
WAV 6 - CS
WAV 7 - DELETE

Regards :)

Elektra999

18th December 2006, 18:27

Hi Steve.
I have been following step by step, its guide of Nuendo. I have to say, that it seems to me fantastic the guide and her work. But I have two questions:

01. The Surround Panner I have fit it thus and not if it is well?

http://img92.imageshack.us/img92/4002/dibujoms3.jpg

02. In Output 5,1, I do not have the OCtoMaxx2-0, but the Pick Masters. The connections I have made them, but they seem to me thus
http://img283.imageshack.us/img283/9725/dibujo2zi2.jpg

I have used the Steniberg Cubase SX 3, and the mixture in 6 channels I believe that to come out well.

Thaks Steve
Frank

ursamtl

19th December 2006, 00:23

Your peakmaster is only affecting the front channels. If you only have two channels to limit, the most important ones in this process are the rears because the surround generation algorithm tends to unmask peaks and transients in the material that are normally cancelled out in the regular 2-channel stereo.

Pookie

19th December 2006, 12:13

I'm probably not looking in the right area - wondering if there is a recommended free VST host for this guide. Isn't Bidule no longer free? Perhaps Kristal Audio Engine instead?

Elektra999

19th December 2006, 12:51

Where I can find, the plug-in Octomaxx or PeakMaster of 6 channels?

Thanks

ursamtl

19th December 2006, 14:08

I'm probably not looking in the right area - wondering if there is a recommended free VST host for this guide. Isn't Bidule no longer free? Perhaps Kristal Audio Engine instead?

Kristal is a nice program but only does two channels at a time and only in real time. There is a guide for using it with the V.I 2-channel plugins that a guy on videohelp.com wrote (I listed it in the surround guide sticky at the top of this forum).

Plogue released a new demo on November 13. This will work until mid-February 2007. You can also try this: GUIDE: Converting stereo to 5.1 surround for FREE (http://forum.doom9.org/showthread.php?t=105684)

Regards,
Steve.

ursamtl

19th December 2006, 14:10

I believe Octomax came with Nuendo 2. I don't know if there is a 6-channel version of Peakmaster. You could set up three 2-channel limiters in the bottom three effects slots in Nuendo. Just move the connections until you line them up the way you want.

Elektra999

19th December 2006, 19:11

Thanks Steve and Pookie

I believe that you say to me, that she connects therefore the PickMaster, is thus?

http://img272.imageshack.us/img272/7388/dibujojt4.jpg

It completes question: the order the channels with the PeakMaster 2 channels, is thus?
WAV 1 - FL
WAV 2 - FR
WAV 3 - CENTER
WAV 4 - LFE
WAV 5 - SL or SR.?
WAV 6 - SR or SL?

Thanks you very much

Pookie

19th December 2006, 21:00

Thanks for the info, Steve :)

jordisound

25th February 2007, 21:12

Hi, i've a question about VI base method guides. I want to use VI plug-in with my audio editor. May I select audio 32-bits or 16-bits?

ursamtl

26th February 2007, 14:16

Hi jordi,

Yes, V.I will work with whatever bit depth you set for your audio editor. However, it's a general rule for audio editing that you should always work in 32 bits and then convert down to 16 bits as the last step and add dithering when you do the conversion. By the way, if you're doing the sound for DVD, you can reduce to 24-bit depth instead of 16. The quality is better than 16 and the DVD spec fully supports 24-bit audio up to a 96kHz sampling rate.

Regards,
Steve.

Elektra999

26th February 2007, 15:20

Hello
Steve, ¿if audio the original one, is to 16bits/48000. .hz don't mention it serves, to talk to 24bits/96000…hz?

If audio the original one is, to 24bits/96000 ¡sounds of wonder! and z sounds better than to 16/48000…

ursamtl

26th February 2007, 16:08

Franck,

If I understand correctly, you're basically asking if there's any benefit to upsampling to 24/96 if the source is 16/48. I'd say yes in my experience it helps. First of all, audio apps with VST plugins end up converting the data to 32-bit resolution anyway so the bit depth is automatically increase (and must be dithered if reduced, even to 24-bit according to pros such as Bob Katz). As for upsampling to 96kHz, I've been reading some material lately about the effect of filtering on data. Audio data at 44.1 and even 48kHz can suffer from filter rippling effects in the highs, whereas 96kHz audio moves the filtering up far beyond the limits of human hearing, so it can help. I know from experience with my home CD/DVD player (A Yamaha DVDS657), its CD upsampling function tends to make high frequency transients such as cymbals, guitar strums, etc., sound much smoother to my ears.

The only time I'd stick at 48kHz is if I were working with a slower PC or had very limited storage resources. But again, the golden rule is to never, never, never work at a 16-bit resolution. If you need the final file to be at that resolution reduce as the very last step and with dither. Even a level change after dithering can result in data loss.

jordisound

28th February 2007, 17:42

What happens if the source is mono? I've extracted audio from a AC3mono DVD. I convert mono wav to stereo wav (both channels are identical) to apply VI plug-in. Does VI plug-in works OK?

ursamtl

28th February 2007, 18:24

No, a mono source won't work because V.I doesn't synthesize fake ambience. It simply extracts the ambience already present in stereo signals using modified ambisonic/M-S principles. Just duplicating the mono file on two channels won't give you the difference information required to create the side channel in M-S or the Y channel in ambisonics.

I've had some success with mono material by running it through a convolution plugin first with some carefully chosen reverb impulses loaded. You might also try the freeware reverb plugins Ambience or Glaceverb. There are also some good stereo simulation plugins available. Elevayta's Wide Boy is a good one (it's almost free, just a 90-cent US bandwidth fee). I'd stay away from some stereo simulator plugins such as Voxengo's Stereo Touch as it's designed for smooth keyboard-pad style sounds (according to their web site). Anything with sharp transients such as percussion sounds really terrible through this type of simulator. The free mda plugins have a stereo simulator plugin that's not bad. you might try it.

Anyway, good luck.
Steve.

ursamtl

4th March 2007, 15:54

Hi folks,

I frequently get private messages with questions about using V.I. I don't mind answering them (and I think my track record on answering is ok), but I think it would be better if everyone asked their questions in the public forum in this thread. It would help others if you'd ask the questions in the forum instead of privately. Like most forums, Doom9 is a community where we all improve our hobby by sharing knowledge. In addition, I (or whoever else you message privately) may not have the answer you're looking for or perhaps I'm out of town, etc.). Someone else may be able to help you.

One question I get a lot is when someone takes a file that was previously compressed (usually 2-channel AC3, MP2, MP3) and tries to create a surround file using V.I or one of the other methods. After converting, artifacts or weird sounds seem to appear that didn't seem to be there in the stereo file.

I my experience, it's always tricky to try creating surround out of files that were compressed (AC3, MP2, MP3, etc.). V.I works by extracting ambience that was already in the track. So does Dolby Pro Logic-based methods, Ambisonics, Quadraphonic methods, etc. The compression algorithms used by these formats (AC3, MP2, MP3) all mess the ambience in some way or another. You don't really hear this much when the files are in stereo the algorithms were designed so that the stereo files mask these artifacts. As soon as you try to extract the ambience, you unmask the artifacts.

So whenever possible, start with an uncompressed source, such as the 2-channel LPCM from a movie, or a wave file.

If you do have a compressed file, I suggest redigitizing it. This means, recording the analog output of the stereo file back into your PC. This way, the playback equipment recreates a version of the soundwave and may reduce the artifacts problem.

So remember, share your ideas. I don't mind receiving private messages, but I often think it's too bad we didn't share the problems and solutions in public where more people could benefit.

Regards,
Steve.

jordisound

15th March 2007, 10:40

it's a general rule for audio editing that you should always work in 32 bits and then convert down to 16 bits as the last step and add dithering when you do the conversion. By the way, if you're doing the sound for DVD, you can reduce to 24-bit depth instead of 16. The quality is better than 16 and the DVD spec fully supports 24-bit audio up to a 96kHz sampling rate.
Hi Steve,
I've a question about what you said.
Using Aften to encode from the fRfL, CLFE anf sLsR wav (32bits 48000hz), the ac3 final is 16bits 4800hz. If Aften converts 32 to 16 while encoding, maybe is not necessary convert the wavs to 16-bits as you recommended at last step. isn't necessary?

tebasuna51

15th March 2007, 12:22

Hi Steve,
I've a question about what you said.
Using Aften to encode from the fRfL, CLFE anf sLsR wav (32bits 48000hz), the ac3 final is 16bits 4800hz. If Aften converts 32 to 16 while encoding, maybe is not necessary convert the wavs to 16-bits as you recommended at last step. isn't necessary?

AFAIK Steve speech about LPCM for 24 bit 96 KHz.

Ac3, internally, don't have different sample size like wav and the input wav can be with any sample size. The first thing Aften do is convert any sample size to 32 float, is not necessary any previous reconversion.

"...the ac3 final is 16bits 4800hz."?

For what do you think this? Maybe is your decoder, try with NicAc3Source() in AviSynth, the output is always 32 bit float. Use SoundOut or last BaHappyMod to see.

raquete

15th March 2007, 14:00

@ tebasuna
BaHappyMod? i don't know.where to find? (in behappy thread? )

For what do you think this?hummm..the same here tebasuna.
loading one wave(6 interleaved) 48k-32bit in EncWavtoAC3 give me the ac3 final is 48k-16bit.

ursamtl

15th March 2007, 14:04

I strongly recommend reducing bit depth to 16-bit resolution with dither (dither is ESSENTIAL when going to 16-bit) before encoding. Use triangular dither. Noise shaping might be a good idea but lossy compression may filter out some or all of the dither noise added when using noise shaping.

If Aften does the bit-depth reduction with dithering then yes, it could be done by it. If it's doing it by truncation (chopping off data), then do not let Aften do the reduction, do it yourself beforehand.

By the way, slightly related to this is sample rate conversion. With the availability of equipment that supports higher sampling rates such as 96kHz and 192kHz, converting down to 48kHz (DVD) or 44.1kHz (CD, whether regular audio or surround), requires good conversion. Here's an interesting comparison of sample rate converters I've discovered recently:

http://src.infinitewave.ca/

Looks like Cool Edit Pro/Adobe Audition provides some of the better performance among tools used by a lot of forum users. r8brain isn't as impressive as I expected, and some of the others (Wavelab for example) are a bit disappointing.

raquete

15th March 2007, 14:24

(i'm lucky,i use audition :) )

thanks for the clarifications and the picture Steve!

tebasuna51

15th March 2007, 14:55

If Aften does the bit-depth reduction with dithering then yes, it could be done by it. If it's doing it by truncation (chopping off data), then do not let Aften do the reduction, do it yourself beforehand.
Read my post:
"The first thing Aften do is convert any sample size to 32 float, is not necessary any previous reconversion."

No bit-depth reduction, no truncation.

@Raquete, Jordisound
What software say you an ac3 is 16 bit depth?
Where are the fields in ac3 header about bit depth?

We only can say bit depth accepted by a encoder or supplied by a decoder, internally not exist difference.

@raquete
Yes BeHappyMod is in behappy thread.

raquete

15th March 2007, 15:52

The first thing Aften do is convert any sample size to 32 float, is not necessary any previous reconversionall right,is clever!
What software say you an ac3 is 16 bit depth?in ac3filter playing AC3 (inside mpclassic) but can be my wrong configurations. to tell the true i don't listen the files in pc,i only play in mpclassic to see the infos.
what tool can be used to read the complete AC3 infos?

:thanks:

tebasuna51

15th March 2007, 16:28

in ac3filter playing AC3 (inside mpclassic) but can be my wrong configurations.
In Ac3Filter config Main tab -> Output, you can select the bit depth than this decoder can supply 16/24/32 bit int or 32 float.
what tool can be used to read the complete AC3 infos?

I make a simple header ac3/mp3 reader (http://www.mytempdir.com/1093720) (only first frame showed and statistics).

raquete

15th March 2007, 16:42

great! :)
adjusting ac3 filter give me this results in decoder info:
Input format: PCM32 3/2.1 (5.1) 48000
User format: PCM32 3/2.1 (5.1) 0
Output format: PCM32 3/2.1 (5.1) 48000

Decoding chain:
(PCM32 3/2.1 (5.1) 48000) -> Processor -> (PCM32 3/2.1 (5.1) 48000) -> Dejitter -> (PCM32 3/2.1 (5.1) 48000)

Filters info (in order of processing):

Processor:
(PCM32 3/2.1 (5.1) 48000) -> PCM->Linear converter -> (Linear PCM 3/2.1 (5.1) 48000) -> Input levels -> (Linear PCM 3/2.1 (5.1) 48000) -> Mixer -> (Linear PCM 3/2.1 (5.1) 48000) -> Bass redirection -> (Linear PCM 3/2.1 (5.1) 48000) -> AGC -> (Linear PCM 3/2.1 (5.1) 48000) -> Delay -> (Linear PCM 3/2.1 (5.1) 48000) -> Output levels -> (Linear PCM 3/2.1 (5.1) 48000) -> Linear->PCM converter -> (PCM32 3/2.1 (5.1) 48000)

Dejitter:
-

I make a simple header ac3/mp3mytempdir always give me "The file is temporary unavailable, please try again later."
is possible to host in rapidshare?

thanks so much tebasuna!

jordisound

15th March 2007, 17:54

@tebasuna51: did your application (LeeAudBi) show information about bitdepth on AC3?

tebasuna51

15th March 2007, 18:01

Try here.

Edit: New link and version LeeAudBi03b.7z

Edit: Link dead, attached only exe file

Edit: new version here http://forum.doom9.org/showthread.php?p=1522330#post1522330

jordisound

15th March 2007, 18:13

Well, I've tested again with bepipe and aften in commandline, using a script to merge flfr, clfe and slsr (tebasuna did it and send me)

global OPT_AllowFloatAudio=True
f = WavSource("C:\rip\FLFR.wav")
c = WavSource("C:\rip\CLFE.wav")
s = WavSource("C:\rip\SLSR.wav")
MergeChannels(f, c, s)

and tebasuna was right.
http://img11.imagepile.net/img11/81992aften32.png

@tebasuna51: did your application (LeeAudBi) show information about bitdepth on AC3? I can't find this field.

tebasuna51

15th March 2007, 18:20

@tebasuna51: did your application (LeeAudBi) show information about bitdepth on AC3?
One more time?
Not exist different bitdepth in ac3 the data info is in frequency domain, not in time domain, then the bitdepth is not applicable.

I say all bitdepth's samples are translated to 32 float before the time -> frequency conversion and the precision is defined by the bitrate.

ursamtl

15th March 2007, 18:40

Yes, but do you not agree that the data has to be converted back into the time domain to be sent to the D/A converter? WinDVD for example, will look at an AC3 stream in a DVD file and identify it as 16-bit/48kHz. Is it taking 32-bit data and truncating it to 16?

raquete

15th March 2007, 22:32

curious...encoder adjusted to 448 and 512:

(File to analyze: H:\ROBERT~1\AC3512~1\01LOVE~2.AC3
SampleRate 0 : 48000 KHz.
BitRate 15 : 448 Kb/s.
...and...
File to analyze: C:\DOCUME~1\ADMINI~1\DESKTOP\01LOVE~1.AC3
SampleRate 0 : 48000 KHz.
BitRate 16 : 512 Kb/s.

why i'm getting this results?

jordisound

15th March 2007, 22:35

Sorry, tebasuna51, i didn't undestand your first answer. Now, I do.
I was using wrong software to reasd the ac3 info, that software show 16bits, when it doesn't exist in AC3!

tebasuna51

15th March 2007, 22:42

Yes, but do you not agree that the data has to be converted back into the time domain to be sent to the D/A converter?
Of course, but this is a decoder problem.
WinDVD for example, will look at an AC3 stream in a DVD file and identify it as 16-bit/48kHz.
Impossible. Nothing in the ac3 stream header say the original sample size was 8/16/24/32 or float.
A good decoder can be instructed to decode from frequency to time using like output 16/24/32 ...
Is it taking 32-bit data and truncating it to 16?
The data in the ac3 stream are exponents and mantissa frequency values and is most precise with more bitrate.

Really I don't understand the problem. We have 3 stereo wav's (flfr, clfe, slsr) 32 bit float, send the data directly to Aften with:
global OPT_AllowFloatAudio=True
f = WavSource("C:\rip\FLFR.wav")
c = WavSource("C:\rip\CLFE.wav")
s = WavSource("C:\rip\SLSR.wav")
MergeChannels(f, c, s)
and verify if is incompatible with DVD or any soft identify this ac3 like 32 bits.

ursamtl

16th March 2007, 13:11

According to the book I'm reading right now, Ken Pohlmann's The Principles of Digital Audio (an excellent "bible" recommended by engineer Bob Katz in one of his articles on www.digido.com), AC3 in principle supports up to 24-bit audio, not 32 bits. He also states that the AC3 decoder outputs linear PCM after the decoding. So you're right, my mention of the data being converted back into the time domain before D/A conversion is a decoder issue. The question here is whether inputting 32-bit audio to Aften will output 32-bit audio to the decoder, which will in turn output 24-bit audio. Pohlmann also mentions a dither flag in the bitstream. What do you know about this? Does Aften support the dither flag?

raquete

16th March 2007, 15:32

AC3 in principle supports up to 24-bit audio, not 32 bits.i read in some Dolby.inc and ATSC docs that AC3 support 16,18,20 and 24 bits,have some data sheets showing that too.
months ago i asked about AC3 24 bit in the Aften thread and why not we don't was using this feature ....no clever answers.
now i have 3 results from tebasuna's LeeAudBi log file using one interleaved 6 channels 32 bit from EncWavtoAC3 encoder:
1- BitRate 15 : 448 Kb/s.
2- BitRate 16 : 512 Kb/s.
3- BitRate 18 : 640 Kb/s.

no one result is 32 bit and it send me to another question:
why 15,16 and 18bit? :confused:

thanks for any clarification.

foxyshadis

16th March 2007, 16:07

Internally it's all infinite bitdepth, because it's all in the frequency domain. It'll be constrained by the filterbank construction and the bitrate, but in theory if you could throw enough bitrate at it you could construct an output that more closely matches a 32-bit input than its 24-bit downsampling. Realistically there probably isn't enough resolution in the bottom filterbanks, that's a problem in mp3 too; maybe eac3 handles it better, like aac or vorbis.

Those three specific bit depths are probably what the codec considers the maximum that it can reliably produce. But then again, you'd probably have to ask the EncWavtoAC3 author.

It would be interesting to renormalize a 24-bit track to 32-bit -140dB and find out whether the codec really is capable of 32-bit, in the absense of other factors.

tebasuna51

16th March 2007, 18:59

now i have 3 results from tebasuna's LeeAudBi log file using one interleaved 6 channels 32 bit from EncWavtoAC3 encoder:
1- BitRate 15 : 448 Kb/s.
2- BitRate 16 : 512 Kb/s.
3- BitRate 18 : 640 Kb/s.

no one result is 32 bit and it send me to another question:
why 15,16 and 18bit?
These values aren't bits per sample, but BitRate codes':
BitRatecod Bit Rate
------------- --------
'00000' ( 0) 32 kbps
'00001' ( 1) 40 kbps
'00010' ( 2) 48 kbps
'00011' ( 3) 56 kbps
'00100' ( 4) 64 kbps
'00101' ( 5) 80 kbps
'00110' ( 6) 96 kbps
'00111' ( 7) 112 kbps
'01000' ( 8) 128 kbps
'01001' ( 9) 160 kbps
'01010' (10) 192 kbps
'01011' (11) 224 kbps
'01100' (12) 256 kbps
'01101' (13) 320 kbps
'01110' (14) 384 kbps
'01111' (15) 448 kbps
'10000' (16) 512 kbps
'10001' (17) 576 kbps
'10010' (18) 640 kbps
the binary data in ac3 header.

From Dolby doc "ATSC Standard: Digital Audio Compression (AC-3)":

"8.2.1.1 Input word length
The AC-3 encoder accepts audio in the form of PCM words. The internal dynamic range of AC3 allows input word lengths of up to 24 bits to be useful."

Maybe this is the origin of the 24 bit depth of ac3 streams, but like foxyshadis say is not inherent to the format (infinite bitdepth), is a feature of the encoder. Aften accept 8/16/20/24/32 bit int and 32/64 bit float wav's.

@ursamtl
Yes, Aften construct the DitherFlag in ac3 streams. But I don't know if this can clarify:

"7.3.4 Dither for Zero Bit Mantissas (bap=0)
The AC-3 decoder uses random noise (dither) values instead of quantized values when the number of bits allocated to a mantissa is zero (bap = 0). The use of the random value is conditional on the value of dithflag. When the value of dithflag is 1, the random noise value is used.
When the value of dithflag is 0, a true zero value is used. There is a dithflag variable for each channel. Dither is applied after the individual channels are extracted from the coupling channel.
In this way, the dither applied to each channel's upper frequencies is uncorrelated."

raquete

17th March 2007, 00:23

These values aren't bits per sample, but BitRate codes'all right,is clever.
if the interleaved wave source is 16,24,or 32 bit,after encode with aften the result is 32 bit.
the ac3filter is showing the right bit of the input!
thanks so much.
:-)

tebasuna51

17th March 2007, 00:52

all right,is clever.
if the interleaved wave source is 16,24,or 32 bit,after encode with aften the result is 32 bit.
Nope. If is a 48 KHz. the result is an ac3 stream with:
Stereo 6 channels
Bit Rate Average bits/sample Average bits/sample
-------- ------------------- -------------------
32 kbps 0.333 0.111
40 kbps 0.416 0.138
48 kbps 0.5 0.166
56 kbps 0.583 0.194
64 kbps 0.666 0.222
80 kbps 0.833 0.277
96 kbps 1 0.333
112 kbps 1.166 0.388
128 kbps 1.333 0.444
160 kbps 1.666 0.555
192 kbps 2 0.666
224 kbps 2.333 0.777
256 kbps 2.666 0.888
320 kbps 3.333 1.111
384 kbps 4 1.333
448 kbps 4.666 1.555
512 kbps 5.333 1.777
576 kbps 6 2
640 kbps 6.666 2.222

Is called compression :rolleyes:

After a decoder can decode the ac3 stream to 8/16/24/32/64 or the bitdepth desired.

raquete

17th March 2007, 01:17

Nope. If is a 48 KHz
lol.i'm sorry,i really don't understood what you mean in the last post....
you mean that if the interleaved wave is not 48k can be different? :confused:
i told you that when i load one interleaved(wav 6 channel)48k - 16 to 32 bit in Aften i have always the ac3 in 32 bit.
where i'm lost?

tebasuna51

17th March 2007, 02:25

The average bits per sample is:

BitRate / (SampleRate x NumChannels)

An ac3 5.1 48 KHz 6 channels 448 Kb/s have:
448000 / (48000 x 6) = 1.555

An ac3 5.1 44.1 KHz 6 channels 448 Kb/s have:
448000 / (44100 x 6) = 1.693

You can't say "i have always the ac3 in 32 bit", the ac3 isn't 32 bit, you use a decoder than decode the ac3 in samples of 32 bit.

You can instruct the decoder to output 16 bits samples for the same ac3, is your decision (in the decoder) not a ac3 feature.

raquete

17th March 2007, 05:51

...you use a decoder than decode the ac3 in samples of 32 bit.You can instruct the decoder to output 16 bits samples ...in the pc i understand but how my standalones players and DD decoder manage the files as i can't instruct both?
how the decoders will "know" "what is what" and "who is who" and how much bit to play? (16,32 etc)

foxyshadis

17th March 2007, 10:26

Standalones will probably all use 24-bit for everything, unless they're really cheap and limited to 16-bit.

It would be interesting to renormalize a 24-bit track to 32-bit -140dB and find out whether the codec really is capable of 32-bit, in the absense of other factors.

So I tried this. :p The input wav can be renormalized to sound just like the original, but all I get is a buzzing out of the ac3. At -120dB, it gives something resembling the sound effects to Atari games like Pole Position. :D Oh well. :p

jordisound

22nd March 2007, 00:54

tebasuna51

22nd March 2007, 02:02

Well, I've done a test, with 32-bit waves, using the script to merge 6 wav and using aften like i told you in my last post. the new ac3 was only 30', 1/3 the length of the audio track (1h 30' aprox.).
I've tried again without global OPT_AllowFloatAudio=True in the script (it seems aften works 16bits). The ac3 was ok, the original lenght

You need Aften rev449 or new and a use the aften parameter:
-readtoeof 1

With this parameter Aften ignore the wav header field wrong than say is only 30' and continue encoding until the end of file 1h 30'.

When the AviSynth output is converted from 32 float to 16 bit (without global OPT_AllowFloatAudio=True) the size is half and the new problem limit is at 124 min.

foxyshadis

22nd March 2007, 02:53

tebasuna51

22nd March 2007, 04:21

That parameter should probably be default for any 6 channel or 32-bit input. Or, if you want to get tricky, just calculate whether the input header is pointing to a length of right around 2GB or 4GB, and if so assume the header is broken. That way Aften will work more simply for most people.
I agree, but is not the opinion of aften creator.

You can see this post (http://forum.doom9.org/showthread.php?p=962080#post962080) and related.

jordisound

24th March 2007, 14:33

You need Aften rev449 or new and a use the aften parameter:
-readtoeof 1
Thank you tebasuna, it works.

omega6666

27th October 2007, 16:04

Try here (http://rapidshare.com/files/21180963/LeeAudBi.zip.html).

Tebasuna, I'm looking for your LeeAudiBi. Can I still download that one somewhere?

Thx...

tebasuna51

28th October 2007, 01:53

Tebasuna, I'm looking for your LeeAudiBi. Can I still download that one somewhere?

Not yet finished, and without guaranty, try this new version (http://www.mytempdir.com/2048795) with GUI. Say me if work for you.

Edit: New link and version LeeAudBi03b.7z (http://www.sendspace.com/file/7izmtk)

omega6666

4th November 2007, 23:18

Wow thanks! That showed me more info than I've seen so far! Unfortunately I'm still not sure if this solves my problem described here (http://forum.doom9.org/showthread.php?t=131120). I got this info from the first frame of the problematic dts file;

First frame data:
----------------------------------------------------------
Bytes before header 120867 ; Frame 1 in position 120868
CRC present 0 : Not
Number of PCM Sample Blocks 15 : 512 samples/frame
Primary Frame Byte Size 2012 : 2013 bytes/frame
Audio Channel Arrangement 9 : 5 C + L + R + SL + SR
Core Audio Sampling Frequency 13 : 48 kHz
Transmission Bit Rate 24 : 1536 Kb/s
Embedded Down Mix Enabled 0 : Not
Embedded Dynamic Range Flag 0 : Not
Embedded Time Stamp Flag 0 : Not
Auxiliary Data Flag 0 : Not
Mastered in HDCD format 0 : Not
Extension Audio Descr. Flag 0 : Channel Extension (XCh)
Extended Coding Flag 0 : Not
Audio Sync Word Insert. Flag 1 : Sub-sub-frame
Low Frequency Effects Flag 2 : Present, interpolation factor 64
Predictor History Flag Switch 1 : Yes
Multirate Interpolator Switch 0 : Non-perfect Reconstruction
Encoder Software Revision 7 : Current
Copy History 1 : Definition deliberately omitted.
Source PCM Resolution 6 : 24 bits
Front Sum/Difference Flag 0 : Not
Surrounds Sum/Difference Flag 0 : Not
Dialog Normalization Param. - 0 dB

The original where this was demuxed from was 96/24 for sure. I would like to know if the extension data for the 96 khz is still there, but is maybe flagged wrongly so that my receiver simply won't use the available extension data, or if it is just missing? Here (http://rapidshare.com/files/67478370/dts_samples.rar.html) are 2 dts files (short samples) where there's one that works fine as 96/24 and the other one just plays as 48 khz, even though the original was 96 khz. Can you maybe tell me if there's still extension data there to let my receiver play this one as 96/24?
Oh, and thanks again for your very informative program! I think I'm gonna use that one very often! :thanks:
[edit:] The proper dts file shows this flag; Extension Audio Descr. Flag 2 : Frequency Extension (X96k), and the faulty one shows this; Extension Audio Descr. Flag 0 : Channel Extension (XCh). When I mux the faulty dts track with some black bmp to a VOB file, and change the domain stream attributes of the dts to 24 bits and 96 khz with PgcEdit, and then demux the dts file again, it seems it hasn't changed the Channel Extension flag! How can I do this?