Realtime AC-3 encoder?

[HarryAngel]

I am not sure if this is the right forum for this, but I was wondering if anyone knows if there is something like a working encoder which can encode AC-3 in real time so that the AC-3 stream can then be sent via SPDIF to an external receiver.

I found a direct show filter which does exactly that at http://oss.netfarm.it/ac3encode.php. I even got it to work somehow (i.e. my receiver is receiving 5.1 DD sound from an avi with stereo MP3 sound, but the sound is stuttering and the filter crashes every two minutes).

It seems development of this filter is halted but it would be great if someone else did something similar or continued with this project. It seems for people (like me) who have their multi-speaker systems connected via an external receiver that can only decode DD (and DTS), this would be the only way to really take advantage of these new multi-channel capable formats like OGG Vorbis or AAC.

[daphy]

I found a direct show filter which does exactly that at http://oss.netfarm.it/ac3encode.php. I even got it to work somehow (i.e. my receiver is receiving 5.1 DD sound from an avi with stereo MP3 sound, but the sound is stuttering and the filter crashes every two minutes).

Hi,

I don´t understand the sens of this transcoding:

You have a MP3 2.0 at let´s say 160Kb/s
and you want transcode it in realtime into a 5.1 DD (let´s at 448Kb/s)???

In my opinon this wouldn´t work because of the missing channels; you´ll get something like 5.1 but with definitfly NO surround effect.
The crashing might depend on the missing support of MP3 -> the author of this filter have confirmed that (´till now) only 16bit WAV PCM input is possible.

I think better results are possible if you use your MP3 analoge input and one of your DSP program which every dolbydigital amp offers.

CYA daphy

[HarryAngel]

Originally posted by daphy
Hi,

I don´t understand the sens of this transcoding:

You have a MP3 2.0 at let´s say 160Kb/s
and you want transcode it in realtime into a 5.1 DD (let´s at 448Kb/s)???

Well, if I look into this forum and see to what length people are going to transcode stereo to surround I guess there is a need for that. In my opinion, it's only really useful when you have a (non-AC3) multichannel source which you want to listen to in true surround without transcoding it first. That's why I mentioned multi-channel OGG and AAC, which I would like to try, but can't at the moment.

In my opinon this wouldn´t work because of the missing channels; you´ll get something like 5.1 but with definitfly NO surround effect.
The crashing might depend on the missing support of MP3 -> the author of this filter have confirmed that (´till now) only 16bit WAV PCM input is possible.

To my knowledge (and I am not an expert) any directshow MP3 decoder outputs PCM. I only used the MP3-AVI for quick testing and I forgot to mention that in order to to get a multichannel PCM I used matrixmixer to upmix the stereo source, but I assume that doesn't matter as long as you use any filter that outputs 16bit PCM.

So the filter setup looked like this: MP3 Decoder -> Matrixmixer (output 6 channel PCM) -> ac3encode (output AC-3 5.1) -> AC-3 Filter (tried it with Cyberlink and Intervideo; output to SPDIF). Everything seemed to work fine, the AC-3 Filter confirmed that it is receiving an AC-3 5.1 stream and seemed to output it correctly to SPDIF. And I defintively got true surround sound (e.g. muting one channel in matrixmixer actually resulted in no sound in the respective speaker). The sound was stuttering (and with it the video, so it seems it was still in sync) but one could still understand the words and recognize the music.

I think better results are possible if you use your MP3 analoge input and one of your DSP program which every dolbydigital amp offers.

Yes, that might be true for MP3s but certainly not for real multichannel content encoded in anything other than AC-3. And even for stereo sources it would be nice to have all the flexibility and the tools a PC has to offer for DSP and not only rely on the receiver.

[Saiya-Jin]

I have searched the forums and either I didn't find it or I missed (if I did I truly apologise).

I'm trying to play 5.1 audio through SPDIF (optical out) from my Hercules Fortissimo III to my receiver. Ok the thing is Dolby Digital and possibly DTS can be passed through to the receiver without any problem. However in this case I'm trying to play an HE-AAC soundtrack in 5.1 and it's obvious that my receiver won't decode it, so I was wondering if there is a way of passing the decoded stream to my receiver. It does play, but I think it's being downmixed to 2 channels.
My receiver is a Pioneer VSX-D811S if it matters.
If there is no way of playing digitally (which sucks because that's the whole point of getting an expesive optical cable) I could connect my analog outputs to the 5.1 analog input on the receiver, but I really would like to avoid that because, first, I would have to buy more cables (the pc is very close to the receiver at least), the inputs are already being used by my DVD player (for DVD-Audio, gotta hate those stupid restrictions) and analog kinda degrades the quality and stuff. If it's absolutely is the only way, I would like to know if there is some sort of adapter that enables me to connect both the sound card and the dvd audio player at the same time.

So I would greatly appreciate any help, with filters or whatever that would make 5.1 audio playback to my receiver possible. ;)

Cheers,

Saiya-Jin

[FlimsyFeet]

I don't think it's possible, unless your card can do real-time Dolby Digital encoding; it sounds like it can only do DD pass-through.

Maybe someone else has knows how?

[KpeX]

AAC 5.1 over SPDIF is currently not possible unless your receiver supports it (very few) and your directshow filter is capable of passing it (none currently).

As you have found, the only way to play 5.1 AAC over an optical connection is to pass the decoded PCM data to your receiver, which is only two channel.

Analog is currently the only way to have true AAC 5.1 sound. The only other possibility would be to encode to 2.0 AAC with a Dolby Pro Logic downmix. hope this helps,

[daphy]

The only other possibility would be to encode to 2.0 AAC with a Dolby Pro Logic downmix. hope this helps,

-> transcoding into 5.1 AC3 using the optical out-/input might be possible, too :rolleyes:

[Saiya-Jin]

I see. But will it be possible then with a new directshow filter? Are they working on it? If so, that would be awesome.
And about the analog, is it a bad idea to put an adaptor so I can plug my sound card along with my dvd player in the same set of inputs (I only have one :rolleyes: )?

Thanks for the replies.

EDIT:

As you have found, the only way to play 5.1 AAC over an optical connection is to pass the decoded PCM data to your receiver, which is only two channel

So does that mean that optical connection is only capable of passing PCM in 2 channels? That bites :(

[KpeX]

Originally posted by Saiya-Jin
I see. But will it be possible then with a new directshow filter? Are they working on it? If so, that would be awesome. Correct, but only if your hardware device supports it, which very few do. If you really have a receiver that can decode AAC, I suppose you could ask the CoreAAC developers to look into it. Originally posted by Saiya-Jin
And about the analog, is it a bad idea to put an adaptor so I can plug my sound card along with my dvd player in the same set of inputs (I only have one :rolleyes: )? As long as you aren't planning on using them at the same time, I see no problem with this. But to me it'd be pain to turn off the PC every time I wanted to use my DVD player. You should be able to find a switcher box that could switch between the outputs for fairly cheap, if all else fails, you could rig it up with three stereo line-in switchers.Originally posted by Saiya-Jin
So does that mean that optical connection is only capable of passing PCM in 2 channels? That bites :( Correct.

[daphy]

Originally posted by Saiya-Jin
So does that mean that optical connection is only capable of passing PCM in 2 channels? That bites

Correct.

:D I managed to mux (http://www-user.tu-chemnitz.de/~noe/Video-Zeug/AVIMux%20GUI/index-eng.html) a 5.1/5.0 DTS-WAV (PCM) into a AVI - this might solve your problem, because if your amp understands PCM (maybe limited to 44.1, can´t remember whether 48 is also supported) it also could interpret this PCM as DTS-WAV!!! (My KISS DP450 + Yamaha A2 did it)
You can try to do this also with a DD-WAV ;)

result -> PCM (5.1) in AVI

please give us some feed back

CYA Daphy

[Saiya-Jin]

My amp actually can handle 24/96 sound so that's covered.
It sounds great what you did buy the thing is PCM is huge and there's no way I could save that to a CD, it would be bigger than the movie itself, hehe.

Is it too cpu intensive to convert aac to dts or dolby digital in real time, so that it could be passed through with a possible future DS filter? Just a crazy idea hehe.

[KpeX]

Originally posted by Saiya-Jin
Is it too cpu intensive to convert aac to dts or dolby digital in real time, so that it could be passed through with a possible future DS filter? Just a crazy idea hehe. It's quite possible, some soundcards do it already via hardware encoding. Although the DShow filter encoder idea is definitely viable. There are open source AC3 encoders (FFMPEG) and S/PDIF filters (AC3filter) available.

[Saiya-Jin]

Originally posted by KpeX
It's quite possible, some soundcards do it already via hardware encoding. Although the DShow filter encoder idea is definitely viable. There are open source AC3 encoders (FFMPEG) and S/PDIF filters (AC3filter) available.

Really?! That sounds interesting. I'm sure my sound card won't do it, cos it's rather cheap, although pretty good. You think that they could implement the real-time encoding in a DS filter or something like that?
But still, I'm not so sure that my current pc can handle it, it's only a 1.4Ghz P4, one of the first models, 256MB of RAM, and there will be the video decoding going on as well, so it might not be up to the task. Well, anyway it would really be nice if they could implement that in the filter. ;)

[KpeX]

It's very possible. The problem really would be quality loss, because you'd be decoding a lossy AAC stream, and reencoding into lossy AC3, so the quality would be far from perfect.

The other problem would be finding a developer to code it. Valex (AC3Filter author) could probably do it, if he had the time and the desire to code such a filter.

I think the way to go would be to make a standalone filter similar to MatrixMixer that could take any 5.1 audio output by some other filter such as CoreAAC or CoreVorbis, encode to AC3 and output SPDIF on the fly.

[Saiya-Jin]

Originally posted by KpeX
It's very possible. The problem really would be quality loss, because you'd be decoding a lossy AAC stream, and reencoding into lossy AC3, so the quality would be far from perfect.

Yea, I kinda figured that....

I suppose it's better for me to just use the analog outputs.
The quality shouldn't be too bad, right?
Anyway, thanks a lot for the replies, everyone. :D

[Stux]

Originally posted by KpeX
It's very possible. The problem really would be quality loss, because you'd be decoding a lossy AAC stream, and reencoding into lossy AC3, so the quality would be far from perfect.


Well, the quality loss might not be that bad... after all you can use the maximum ac3 bitrate, would a 192kbps HE-AAC 5.1 soundtrack re-encoded into AC3 at 768kbps (or whatever) really have that much loss?

[wtbreen]

As an answer on the orriginal question:
this week I bought a SPDIF cart (costs: € 14,--, that is aproximatly $ 17,--) and a coax-cable (5 meters, € 19,--).
The cart is from Epox as is my motherboard. I wanted to listen to at least ac3-tracks out of my pc trough my receiver, because I make them by myself (see the guide Eye of Horus wrote). Before I burn my projects on cd or dvd, I want to listen the result (could spare me a lot of cd's / dvd's).
I play the tracks with windvd platinum and as output select SPDIF.

Ac3-tracks are no problem: my receiver accepts the stream and shows the DD-symbol. It sounds as you would expect: 5.1.
DTS-tracks (in wav-format) are a problem: hissing trough the speakers.

I think the problem here is that ac3-tracks are in 48 KHz and dtswav-tracks in 44.1 KHz. I haven't yet tried to convert dts-tracks to 48 KHz in surcode. Maybe that could work.

It is strange though, because dts-tracks burned on a cd and played with my dvd-player via my receiver are also in 44.1 KHz. Here the receiver gives the DTS-symbol as I want it.
My dvd-player has NO decoder; my receiver does.
44.1 KHz is no problem for my receiver, but out of the pc it doesn't work. It has to be the SPDIF-card what can't handle 44.1 KHz.

I hope this helps a bit.

One warning: you cannot use every card on every pc (motherboard). It is possible that + en - are different between the card and the motherboard. I burned a cable from an Asus-spdif card and was verry pleased that there was no other dammage :readguid:

[Umma]

I think the problem here is that ac3-tracks are in 48 KHz and dtswav-tracks in 44.1 KHz. I haven't yet tried to convert dts-tracks to 48 KHz in surcode. Maybe that could work.

It is strange though, because dts-tracks burned on a cd and played with my dvd-player via my receiver are also in 44.1 KHz. Here the receiver gives the DTS-symbol as I want it.
My dvd-player has NO decoder; my receiver does.
44.1 KHz is no problem for my receiver, but out of the pc it doesn't work. It has to be the SPDIF-card what can't handle 44.1 KHz.

If you already have a dts wav in 44.1, you will not be able to change it to 48. Re-encoding the original source (the mono wav files)- IF the source is 48 - with Surcode DVD DTS encoder would work, but that is it.

It is my understanding that the sound card, like a DVD player, must be capable of passing on the 5.1 dts signal for the receiver to decode it. The M-Audio Revolution card, I have read (AND intend to purchase), will do this. The Audigy (sp?) card upsamples the 44.1 to 48 and messes up 44.1 dts cd...I think: someone correct me if I'm wrong, please...when someone tries to play it back. I read that the M-Audio and Logitech Z680 speakers work well together, and the sound is not upsampled to 48. I intend to get the both of them "soon."

I would think, from reading your posts, that the AC3 5.1 that you burned are on DVD. Right? I've never tried to burn a 44.1 AC3 onto DVD, but I know with dts it's a no-go unless the music is in 48 KHz.

Someone please correct me if I'm wrong.

[wtbreen]

Sorry for the misunderstanding: I didn't mean to convert 44.1 to 48, but when I make new dts-files I will go for the 48 KHz-option in surcode right away.
I'm curious if that will work.

Untill now I only burned dts-wavs on CD, not on DVD. They were all in 44.1. It is possible to burn ac3-files on CD, but you have to use a program like sonics dvd architect together with Nero and make a mini-DVD. That also works. The ac3-files are in 48 KHz.

An add-on spdif-card is, in my opinion, not realy a sound-card; it only transports the bits, without any encoding or decoding to another place, in my case a receiver.

[gircobain]

Maybe some directshow genius aka gabest could take a look at it and make it work properly ;)
It would be awesome to be able to transcode 5.1 AAC and output it through SPDIF

[KpeX]

Wow, I must have missed this thread the first time around.Originally posted by gircobain
Maybe some directshow genius aka gabest could take a look at it and make it work properly ;)
It would be awesome to be able to transcode 5.1 AAC and output it through SPDIF Agreed, this would be great for 5.1 AAC. I can connect a 5.1 AAC stream with this filter through GraphEdit but the output is extremely poor as mentioned above.

[gircobain]

Yes I guess that's because ac3encode is based on ffmpeg, which is known to provide bad quality ac3 encoding.

[KpeX]

Originally posted by gircobain
Yes I guess that's because ac3encode is based on ffmpeg, which is known to provide bad quality ac3 encoding. I'd have to disagree, BeSweet encoded AC3 streams (using ffmpeg) sound much better than the output from this filter. Encoded at a high bitrate and played in real time, I think it could work.

[E-Male]

a real time ac3 encoder filter would be really great

it would finally make new 5.1 formats like ogg and aac a real option, i'd love that


also it could find usage in non movie relateded subjects, like games (duke3d in 5.1, would be cool IMO)

[brute]

I need such things, too!

Imagine what you can do, you could encode a multichannel Ogg, which is really small compared with AC3, or you can play games with EAX and the signal is encoded into AC3, so only 1 cable is needed to connect the PC to your receiver and you still get surround gaming.


And there's really what we're searching for on the market, and since some years now.

The nVidia Nforce Soundstorm hardware encoder. That's the only realtime AC3 encoder worldwide. That thing can all we need, but therefore we need to buy an nforce board :(


bah, my english is so bad, I'm sorry for this shit :(

[Haaan]

I have developped a very basic DirectShow filter which does AC3 data -> AC3 SPDIF output AND PCM 6channel data -> AC3 SPDIF output.
It's based on the ffmpeg avcodec dll for the ac3 encoding (and AC3Filter's sources helped me a lot to know how to "communicate" with SPDIF).

PM me if you're interesting, you just must know that it's still very buggy (it's a 7 days old project) and there is not yet any UI.

[Haaan]

I've developped something, just see my post here (http://forum.doom9.org/showthread.php?s=&threadid=69206#post448231).

It's based on ffmpeg and encodes in 448Kbps, but everything can be changed with your experts advices ;)

[gircobain]

I must say I am very interested in trying it out

PM sent :D

[wtbreen]

That sounds very hopefull.
PM is coming!

[gircobain]

Wow i just tried it out on my htpc

I played a matroska file with 5.1 aac audio and the receiver does get a nice dolby digital signal

Too bad my htpc is too slow (p3 866) to decode aac and encode to ac3 at same time without losing sync with video (I have an athlon 1700 laying around, just need a new mobo, guess that's reason good enough to make the investment :D )

Unfortunately the only file with 5.1 aac sound i have at hand is a small 2 min sample, guess i'll be doing a few dvd backups to test it out thoroughly

Hopefully i'll post more info later this week

Keep on the good work, this is all I need to switch to matroska + aac once for all :D

[KpeX]

Very nice work, channel mapping is spot on with MP4 and OGG ( MP4 transcoded with FAAC through FB2k, Ogg transcoded through besweet ). (Test file here (http://www.tfm.ro/ac3/download/test_ac3.rar)) @ Haaan any idea why higher bitrates crash the filter?

Quality is quite nice, and decoding LC AAC + encoding AC3 + decoding 704*320 mpeg-4 video only takes about 15-40% CPU on a 2.0 Ghz P4. This should be very useful.

[ChristianHJW]

Originally posted by Haaan
I've developped something, just see my post here (http://forum.doom9.org/showthread.php?s=&threadid=69206#post448231).

It's based on ffmpeg and encodes in 448Kbps, but everything can be changed with your experts advices ;)

This is something everybody has been waiting for for a long time now, and will definitely help AAC 5.1 and Vorbis 5.1 to get more widely used.

However, i recommend to increase output bitrate to 640 kbps, the maximum allowed in the AC3 standard, as file size is of no interest here ( we just send it over SPDIF to the receiver, it doesnt get stored anywhere ). Hopefully FFMPEG has such an option already, if not then we should have a short look at the target bitrate definition in the code, shouldnt be too big of a problem i guess ....

[Haaan]

Originally posted by ChristianHJW
This is something everybody has been waiting for for a long time now, and will definitely help AAC 5.1 and Vorbis 5.1 to get more widely used.

However, i recommend to increase output bitrate to 640 kbps, the maximum allowed in the AC3 standard, as file size is of no interest here ( we just send it over SPDIF to the receiver, it doesnt get stored anywhere ). Hopefully FFMPEG has such an option already, if not then we should have a short look at the target bitrate definition in the code, shouldnt be too big of a problem i guess ....

It's done in the next version (actually, you'll have the choice lucky boys)

[Haaan]

Originally posted by KpeX
Haaan any idea why higher bitrates crash the filter?


If your talking of AC3 bitrates, it's just because ffmpeg doesn't support higher bitrates than 640 and I tried 768 when I say that it crashes

[clima]

/you a un nouvel ami

[KpeX]

Originally posted by Haaan
If your talking of AC3 bitrates, it's just because ffmpeg doesn't support higher bitrates than 640 and I tried 768 when I say that it crashes I see. Well an option for bitrate might not be a bad idea for future versions. Customizable between maybe 384-640. Keep up the good work, cheers,

Edit: Saw your post on this matter in the other thread. Merging the threads now for clarity.

[E-Male]

Haaan's filter works exelent on my system
i can now play a matroska file containing xvid and 5.1 ogg with tcmp getting 5.1 on my amp

[Haaan]

by the way http://acidbao.free.fr/isoft/dlpage.php?prog=DolbyOutDS

[bond]

Originally posted by E-Male
file containing xvid and 5.1 ogg5.1 ogg vorbis doesnt make any sense as vorbis doesnt offer channel coupling
before using vorbis for multichannel better stay with ac3

[E-Male]

ok, first i have to ask: what is "channel coupling"?

but even thought i might not knwo alltechnical details i kmow what i just tested
and 5.1 ogg worked fine at bitrates below ac3

[Ana]

This something I have dreamed. Thank you! I love'd to test it but there seems to be little problem. The ffdmpeg site is down so I can't find the necessary avcodec.dll. Any way to get it?

[E-Male]

just download the filter
the needed dll is included in the archive

[Haaan]

Originally posted by Haaan
by the way http://acidbao.free.fr/isoft/dlpage.php?prog=DolbyOutDS

Updated:
* You can swap channel orders (you have the three same choices as in Besweet, the default one works fine with an ogg 5.1 encoded with the default besweet channel order)
* You can mute any channel you want (to test one by one)
* You can redirect any channel to Left and mute the other to check the channel order manually.
* You can choose AC3 bitrate (256-640)
* You can enable or disable priority on AC3 streams and 6ch-PCM streams.

[KpeX]

@Haaan

Nice work, testing now.

@E-Male

Channel coupling (http://www.audiocoding.com/wiki/index.php?page=channel+coupling) is the sharing of information between channels to increase encoder efficiency. In stereo this is called joint stereo. Since this has not been correctly implemented in Vorbis yet, vorbis is very inefficient in six channel mode, and a bitrate as high or higher than AC3 would be required to achieve the same quality. That is why bond recommended you not transcode AC3 to OGG.

[gircobain]

It's a bit late at this side of the pond, but I'll be testing the new version tomorrow.

Meanwhile, may I suggest adding a config entry point to the filter, in the same fashion as used in VobSub, AC3Filter, MatrixMixer, etc, so one would be able to invoke the configuration dialog by running rundll32 DolbyOutDS.ax,config for instance?

[Haaan]

Originally posted by gircobain
It's a bit late at this side of the pond, but I'll be testing the new version tomorrow.

Meanwhile, may I suggest adding a config entry point to the filter, in the same fashion as used in VobSub, AC3Filter, MatrixMixer, etc, so one would be able to invoke the configuration dialog by running rundll32 DolbyOutDS.ax,config for instance?

Okay, i will :)

[daphy]

@ Umma
The M-Audio Revolution card, I have read (AND intend to purchase), will do this.

you bought this card? Is it a good one?
I´ve read many reviews about this card - hardware dts/ac3 decoding, 7.1 support, circle surround support ... sounds good :rolleyes:
but no SPIF input - this is not that what I expect from such card :(

CYA Daphy

[KpeX]

@Haaan

An idea for future versions: What about an option to resample (preferably using SSRC) to 48Khz for sound cards that only allow 48Khz output over S/PDIF. This could be useful especially since the Nero AAC encoder likes 44.1 khz or lower sampling rate for most input.

[E-Male]

so ac3 got teh advantage of channel coupling
but does this have such a great effect that ogg's vbr advantage is gone??

[bond]

Originally posted by E-Male
but does this have such a great effect that ogg's vbr advantage is gone??the most important thing you have to realise is that you will NEVER get better quality than the source with reencoding! meaning when reencoding from ac3 to any other vbr able format there is no such thing as a "vbr advantage"

the only advantage with reencoding is that you can have ~ the same/little (hopefully not hearable) worse quality at a lower bitrate

as vorbis doesnt really offer that at a reasonable lower bitrate it doesnt really make sense to use it for multichannel

if you want reasonable lower bitrates with ~ the same quality you definitely should go with he-aac!

[Haaan]

Originally posted by KpeX
@Haaan

An idea for future versions: What about an option to resample (preferably using SSRC) to 48Khz for sound cards that only allow 48Khz output over S/PDIF. This could be useful especially since the Nero AAC encoder likes 44.1 khz or lower sampling rate for most input.

Where can I get information on SSRC api ? Is it not juste for 48Khz -> 44.1Khz

[KpeX]

Actually, SSRC can be used for almost any resampling source and destination. Regarding the API I don't have much information on the subject. Here (http://shibatch.sourceforge.net) is the homepage, DSPguru uses the SSRC dll in BeSweet, he might have more information on the subject. Also there is a Winamp plugin based on SSRC (http://members.lycos.co.uk/bhafool1/rarities/out_wave_ssrc.cab) with included source that may be helpful. Unfortunately I don't have enough programming knowledge to give you any more SSRC information.

Edit: VirtualVCR (http://virtualvcr.sourceforge.net) has a directshow audio resampler included. Although I don't know what resampler it uses, the source (http://cvs.sourceforge.net/viewcvs.py/virtualvcr/2.6.9/AudioResample/) may be useful for DirectShow interface.

[E-Male]

first of all: i'm not stupid, i know that reencoding doesn't make quality better (sorry, but i had to say that)
i just compared ac3 and ogg

i'll look into he-aac
how's the channel order of that format?

[E-Male]

ok, made a short test with aac
sounds good
i'll go into more detailed tests later

[SeeMoreDigital]

Originally posted by E-Male
....i'll look into he-aac
how's the channel order of that format? AAC audio sounds very good, when compared to AC3.

However, if you decide to use Nero's AAC encoder in 'HE' mode, the maximum 'HE' setting is 96kbps. Anything above 96kbps will be encoded in 'LC' mode.

If anybody knows any different, please let me know!

Cheers

[bond]

aac has the following channel order:
C, L, R, SL, SR, LFE

vorbis:
L, C, R, SL, SR, LFE

Originally posted by SeeMoreDigital
However, if you decide to use Nero's AAC encoder in 'HE' mode, the maximum 'HE' setting is 96kbps. Anything above 96kbps will be encoded in 'LC' modefor stereo these bitrates are correct, for multichannel it can go up of course

[Haaan]

Originally posted by bond
aac has the following channel order:
C, L, R, SL, SR, LFE

vorbis:
L, C, R, SL, SR, LFE

for stereo these bitrates are correct, for multichannel it can go up of course

Actually OggDS seems to remap the channel order (in output) to the channel order defined in WAVEFORMATEXTENSIBLE : FL, FR, C, LFE, SL, SR.
Maybe the AAC DirectShow decoder does the same (I haven't tested)

[KpeX]

In my tests the default channel order in Dolby Out DS maps the channels correctly for Vorbis and AAC.

[Haaan]

So it means that thoses filters respect the MS-defined standard. And my channel mapping option is useless as the mapping is always the same.

[SeeMoreDigital]

Originally posted by bond
....for stereo these bitrates are correct, for multichannel it can go up of course Hmm that's strange!

Because if I feed a 6Ch .WAV or 6Ch .AC3 stream into foobar2000 (or a 6Ch .WAV into dbpowerAMP). And select Nero HE above 96kbps, it reverts to LC.

Both MP4UI and Nero's ShowTime Player confirm this!

I thought it was a little strange. Maybe it's because I only have a 'trial version' of Nero.

If this is the case, I'm surprised I'm able to use Nero's 6Ch encoder at all, because I can't when using Recode2.

Any confirmation would be much appreciated.

Cheers

[bond]

So it means that thoses filters respect the MS-defined standard. And my channel mapping option is useless as the mapping is always the same.as the dshow decoders output uncompressed audio, i assume that they all output the same channel order, tough i am not sure

Originally posted by SeeMoreDigital
Because if I feed a 6Ch .WAV or 6Ch .AC3 stream into foobar2000 (or a 6Ch .WAV into dbpowerAMP). And select Nero HE above 96kbps, it reverts to LC.yes of course, he-aac is also not usable for too high bitrates with multichannel, as it isnt usable above 96kbps for stereo
its ok to use normal lc-aac with too high bitrates

[KpeX]

@SMD the displayed bitrates are for stereo, I don't think Nero adjusts the bitrates based on the number of channels input. But if you use the VBR presets you should be able to get a 5.1 HE-AAC file well above 96 kb/s.

[SeeMoreDigital]

I've just tried again to encode a 6Ch .WAV using Nero's HE 'variable bitrate' Extreme-High and Transcoding-Ultra settings.

And it keeps reverting to LC....

...bummer!

[bond]

once again SMD, he-aac is only usable up to the streaming preset, for both stereo and 5.1
it doesnt make sense to use he-aac with high bitrates

[SeeMoreDigital]

Originally posted by bond
once again SMD, he-aac is only usable up to the streaming preset, for both stereo and 5.1
it doesnt make sense to use he-aac with high bitrates I thought this was indeed the case. But gave it another go on strength of what KpeX said!

Cheers

[KpeX]

My (communication) mistake, I meant that when using a low enough preset HE-AAC will be used.

[SeeMoreDigital]

Originally posted by KpeX
My (communication) mistake, I meant that when using a low enough preset HE-AAC will be used. Uuuwww! I thought I was loosing it -again.

Good to have a clarification, none the less.

Cheers you guys

[E-Male]

ogg vs aac – 5.1 audio quality test
(using the realtime ac3 encoder for spdif output))

source: the fast and the furiouse ce rc1 chapter3 english ac3 5.1

converted to:
a)AAC
using besweetgui 5.1/old, high quality, ltp, at multiple vbr presets, with and without pns
b)OGG
using besweetgui 5.1, nomal bitrate 192

then i took the ogg file and the 2 aac files that came closest in size/bitrate:
ogg, 192kbps: 192kbps 6.64MB
aac, internet :: medium, with pns: 204kbps 7.04MB
aac, tape :: lowest, without pns: 187kbps 6.47MB

result:
the aac with pns sounds horrible, you notice the compression all the time
the aac without pns with with lower bitrate sounded better, but still the compressio is audible
the ogg file sounds fine, the comprssion is not audible and the audio is detailed

conclusion:
ogg is my clear winner

ok, everyone now please give your thoughts

[bond]

e-male

1) interesting results, you are the first one with these ;)
2) at that low bitrate you should definitely encode with he-aac, its there for a reason
3) it would be nice if you could make a so called ABX test and post the results, cause only that way you can get scientifc results with subjective audio quality comparisons

[SeeMoreDigital]

Originally posted by bond
...at that low bitrate you should definitely encode with he-aac, its there for a reason... Agreed!

But sadly as you can only encode 6Ch AAC HE audio at 96kbps, it does not always sound that brilliant!

Well that's what I've found.

Shame really

[bond]

SMD,
KpeX already wrote that the 96kbps are only mentioned for stereo encodings, multichannel encodes will of course use higher bitrates even with he-aac

[E-Male]

well, if my results are that unlikely to be correct, i can only assume that i've done something wrong at the aac encoding
could someone tell me whats the best way to do it, maybe point me to a guide?

or could it be a decoding problem? or related with this filter i use for spdif output?

oh, and how do i do a ABX test, i mean i can't do much more than listen, can I?

thx in advance

[SeeMoreDigital]

Yep, but am I right in assuming that E-Male wants to generate 5.1 (6Ch) audio streams?

If this is the case, why did you bother mentioning AAC HE?


Cheers dude ;)

[bond]

Originally posted by E-Male
well, if my results are that unlikely to be correct, i can only assume that i've done something wrong at the aac encoding
could someone tell me whats the best way to do it, maybe point me to a guide?hm unpacking latest besweet+gui, copying the needed nero dlls in the besweet folder, loading the source ac3, ticking 5.1 (new) and show conflig dialog, pushing ac3 -> mp4, choosing he-aac and streaming preset, ok that should be it :D

or could it be a decoding problem? or related with this filter i use for spdif output?well it can be a decoding problem if you used he-aac and used the .aac file, cause than often the high efficiency part isnt decoded (ie with coreaac) which results in really bad quality

oh, and how do i do a ABX test, i mean i can't do much more than listen, can I?you can do much more than only listening ;)
for an ABX test you need
this utility (http://ff123.net/abchr/abchr.html), i think you will also find a guide on ff123's site on how to use it
maybe for the first time its hard to use it but if you want to get really reliable results (better than "i listened to two different movies, one using codec A and the other codec B, with my sister screaming in the background during i was on the train on my back way home"-results) there will not be a way around it :D

Yep, but am I right in assuming that E-Male wants to generate 5.1 (6Ch) audio streams?yes

If this is the case, why did you bother mentioning AAC HE?because multichannel is THE area where he-aac can show best what it is able to do ;)

[E-Male]

i used "5.1 old" (i think because new didn't work), i'll try with new again
i didn't see an option called "ha-aac"

so if the file is named .mp4 it should be decoded correctly?

i'll try this program, but it think a randomized playlist will do the job, if i don't look at the order and take notes while listening

[bond]

Originally posted by E-Male
i used "5.1 old" (i think because new didn't work), i'll try with new again5.1 old will result in the wrong channel order with new nero dlls
make sure you have the latest dlls for best quality and possible bugfixes

i didn't see an option called "ha-aacits in the nero config dialog, where you choose the encoding settings

so if the file is named .mp4 it should be decoded correctly?the output is .mp4 by default
old besweet guis outputted a bugged .aac file, make sure you use the latest gui version

[Ana]

The Nero aac encoder downsamples all files to 44.1 khz, but movie soundtracks are 48 khz. So when encode 5.1 ac3 to 5.1 aac nero makes it always 44.1, but with vorbis 48 khz is kept. This might be one reason why aac souns worse. Another problem with 44.1 is that atleast my soundcard doesn't pass it properly to amplifier (via s/pdif).

But you can make 48khz files with nero encoder using foobar. Guide: http://forum.doom9.org/showthread.php?s=&threadid=67746

[bond]

from what the nero developers said you do not hear a difference between 44.1 and 48khz anyways (well thats a reason why it is downsampled too)

[SeeMoreDigital]

After some more fiddling about with Nero AAC HE, I have finally managed to generate a 6Ch encode higher than the max 96kbps CBR setting.

As KpeX mentioned it had to be done using VBR, Nero's 'Streaming-Medium' setting to be precise. And I ended up with around 170kbps.

Anyway, if anyone is interested, I've posted a short 'Channel Mapping' sample here (http://82.2.167.24/Uploaded_Files/Doom9_Forum_files/6Ch AAC HE VBR (170kbps).zip).

Cheers

[E-Male]

maybe i missed dlls
exactly what files do i have to copy from nero to besweet?

[bond]

Aac.dll, aacenc32.dll, NeroIPP.dll

Anyway, if anyone is interested, I've posted a short 'Channel Mapping' sample here.:)
edit: funny, in matrixmixer the rear channels are always reported as sound being available in all channels! how that?

[E-Male]

{voice=teletubby}Oh Oh{/voice}:scared:
i think i only copied *aac*.*, so i missed one dll
i'll do another (better) test next week

[SeeMoreDigital]

Originally posted by bond
...funny, in matrixmixer the rear channels are always reported as sound being available in all channels! how that? I thought it might interest you!

I made it available for download because it's a good example to show that audio can be accurately steered to a given channel, even when keeping all the channels open.

It would have been quite confusing if the video element had not been included. :confused:

Cheers

[ZeB]

wooot, nice!

I requested something like this back in 2002 ( I'm getting old :/ ).
http://forum.doom9.org/showthread.php?s=&postid=82427#post82427

thx Haaan!!

[E-Male]

ok, got the he-option now
i'll encode the full movie (the fast and the furiouse, as above) overnight in he-aac and ogg

[E-Male]

ok, i'm back, after my raid crashed it took some time to get everything work again

i encoded a sample with he-aac (using besweet and nero dlls)
when playing the resulting file in graphedit using the ac3 encoder filter and spdif output the file played to fast and high pitched
i gues it's a samplerate problem, because the he-aac file is 44.1 (22.05?)

the questions is where what filter causes the problem

or is there something i did wrong??

EDIT: update
i muxed both ogg and he-acc with the xvid video into a matroska file
ogg playes fine in sync with 5.1
aac playes fine in sync in tcmp with matrix mixer
but when i play the mkv with aac in graphedit through spdif audio again is too fast and high pitch and so also not in sync with the video

and before someone asks: my amp can handle 44.1 ac3

i assume either the ace encoder filter can't detect/handle 44.1 properly or it doesn't get the information that the audio is 44.1

EDIT2:

i experimented with the stretch feature of matroska
no success

[gircobain]

Make sure you are using latest BeSweet and Nero dll's.
I believe some old ones are known to be buggy when it comes to aac encoding.

[E-Male]

newest besweet beta. newest besweet gui and dlls from newest nero
all put in a clean directory
i daubt that this is the root of evil

[E-Male]

tested 2 6ch-waves, one 44.1 the other 48khz
and only the 48khz one played fine
the 44.1khz file played to fast (and high)
so i'm now quiet sure that DolbyOutDS can't handle 44.1khz input
so Haan maybe that something for you to look into
for me it's the only bug/missing feature i can name

[E-Male]

just managed to encode a 48khz mp4 with foobar2000 and nero
it works
will compare tomnorrow

[fjarle]

Ok. I don't know jack shit about the programming skills required to do what that DolbyOutDS does, so, I might be asking for a lot here..

Got my Audigy2 a year and a half back... without really looking into the pros and cons of the card. I've found that it'll produce nice quality sound, which is really what I wanted with it in the first place. But. Its drivers are a hellish landscape of one AC3 decoder filling in for the other just when you thought you'd finally managed to sort out SPDIF passthrough just right. After a while though, I managed to get SPDIF to work just great, even without the DolbyOutDS passthrough function. As for other digital surround formats I haven't had the need to test them yet, allthough I'm sure DolbyOutDS will come in handy when those needs arrive.

Still.. I've got four cables between my Audigy2 and Yamaha surround-receiver. 3 MonsterCables for analog output from the card, and one el-cheapo generic minijack->phono cable to get to the spdif-output.

So... what I'd like..
I'd like DolbyOutDS to be "wrapped around" all of my analog outputs... -or at least what will eventually be analog output- when it's on the card. As Audigy2 has no less than 3 spdifs (1 for f. l/r, 1 for l/r s. and one for uhm... front center and rear center. ) I'm guessing everything is PCM bitstream in there at one point or another.

Results being : I can remove my analog cables 100% and play for instance Max Payne2 with full EAX relayed via AC3/spdif to my receiver, and all other surround-sound-fooling-around on my computer will also be output as AC3.

Is this possible at all? As said, I don't know anything about how much work lies behind all of this, so... please don't shoot me :)

- By the way. Why would anything output surround by use of 3*spdif's? I'd need high end... urhm... that'd be WEIRD-end equipment to get one box to decode it all... wouldn't I?

- One more sidenote - There wouldn't happen to allready be a tool out there that'll make use of directshow filters and output all sound through your selection of filters? This'd the trick.... Now, if I could only find it...

[Mitchjs]

I cant get a graph built with DolbyOutDS

has anyone got it to work?

my example is a 6ch .wav file that i want to convert to ac3

i figured as a test

i could send the DolbyOutDS-xformout pin to a file...

but i cant connect anything to it, not even the audio renderer

help
thanks
mitch

[Jeroi]

Where this filter can be downloaded nowdays? Haans links do not work anymore. Is there any other solutions to encode realtime aac into ac3?

[nautilus7]

i don't about that, but nowdays everybody uses ac3filter for realtime ac3 encoding. You can get it separate or with ffdshow.