AC3Filter prerelease version (ac3/dts/lpcm support)

[Valex]

http://ac3filter.sourceforge.net/download/ac3filter_1_01a_rc1.exe

Whats new:
* AC3/DTS/LPCM/MPEG Audio layer I/II decoding support
* AVI/AC3 & AVI/DTS support
* Bass redirection crossover frequency
* AC3/DTS/MPEG Audio SPDIF passthrough
* Real-time AC3 encoding for multichannel SPDIF output
* full MatrixMixer replacement (can be used with any other audio stream too)
...
(equalizer was removed but it will be back)

Just want to test it before release...

[P0l1m0rph1c]

Great news!

Thanks Valex!

[timeismoney]

Wonderful!

So long's waiting...

PS: I found if I want WMP play a dts audio file, I still need a source like Gabest's dtsac3source.ax

[AJ Bertelson]

How will this work for analog out sound cards?

[avih]

Good news Valex. Why don't u add this file to the sourceforge page? You can add it under a 'development' branch or something similar.

cheers.

[Sycho]

is it possible to have a 7.1 output from AC3filter. I mean there's lots of new 6.1 and 7.1 soundcards, and different ways of upmixing to get the new surround back channels

[The Belgain]

Great work! ac3 audio encoding will be really nice for anyone who doesn't have an nforce2 board. What kind of spec machine is needed for that to run in realtime? What's the quality like? Any testing?

I would give it a go out of interest, but I have analog speakers, so no need for it.

[Monamona]

Great wrok!!

Well,
Could someone tell how to enable real-time AC3 encoding?

[Solo]

Thanky You Valex.

I can finally play DTS audio on my PC. Even my PowerDVD (which is supposed to include DTS support) never played it. Looks promising. Unfortunately I don't have DTS support on my amp to test the DTS pass through. But still, great stuff :D

[Blight]

Looks like audio-encoding is freezing up on people and S/PDIF output doesn't always work properly. Here's more info:
http://www.avsforum.com/avs-vb/showthread.php?threadid=403818

[alexnoe]

AVI/DTS support :-)

[MentholMoose]

Originally posted by Blight
Looks like audio-encoding is freezing up on people and S/PDIF output doesn't always work properly. Here's more info:
http://www.avsforum.com/avs-vb/showthread.php?threadid=403818

I reported on the avsforum thread that I couldn't get passthrough to work, but I've since found that I can get it to work sometimes. I'm testing with GraphEdit because the behavior is intermittent with Zoom Player (WMV Pro v.4.00 beta pre RC1) and Media Player Classic (6.4.8.1).

From my observations in GraphEdit, I think the setting "Main > Options > SPDIF" is not being loaded when AC3Filter initially starts.

If I launch GraphEdit with the SPDIF option already enabled and create a graph with AC3Filter decoding an AC3 stream, GraphEdit will not perform as expected: nothing happens if you click play, and if you click stop GraphEdit will hang for 10 seconds and give this error message: "The graph was unable to complete pause within 10 seconds. Press Retry to wait another 10 seconds for completion or Cancel to attempt to stop the graph". This will happen regardless of what "System > SPDIF passthrough > AC3" is set to.

However, if then I access AC3Filter properties (from within GraphEdit, not the Start menu), disable and re-enable the SPDIF option, then press play, the graph will play as expected: passthrough will work fine if it is enabled; if passthrough is disabled, AC3 Encode will work but drops out after an undetermined amount of time, even with low (<10%) CPU usage.

Also, if I load GraphEdit with the SPDIF option disabled, then enable it after starting GraphEdit, it will function as expected.

This strategy does not seem to work in Zoom Player at all, and it works a little bit in Media Player Classic (I've done very little testing with either, though).

I've tried this with a variety of sources (vob, ac3, mkv+ac3, avi+ac3) and filters (various splitters) and the behavior seems consistent, so other filters don't seem to have an effect on this.

Some info on my PC:
Win2K SP4, 2.2GHz AMD Barton, nforce2 motherboard (note: AC3 encoding is disabled in soundstorm), 512MB RAM.

[Valex]

@MentholMoose
If you can spend some time for several testings, please write to my email...

[MentholMoose]

PM sent... I couldn't find your email addy, sorry.

[pogo stick]

Ofigetitelno! :D
So it's more like ffdshow of audio now!
Are you intersted in AAC? It's becoming more and more popular.
And Ahead is about release an improved version, as rumors say.

[Valex]

@pogo stick
>Are you intersted in AAC?

Planning (and OGG too)... Moreover it is no troubles to add new decoders anymore.

[Valex]

@sycho
> is it possible to have a 7.1 output from AC3filter.

Generally it is possible, but it will require some work. Plus I can hardly imagine upmix equations for 7 channels. With using of HRTF techniques and spectral filtering it is possible to feed any number of channels but I'm still too far from all this things...

[Valex]

@The Belgain
> What kind of spec machine is needed for that to run in realtime?
Encoding consumes about the same processor time as decoding.

> What's the quality like?
I got FFMPEG as basis for encoder (hardly rewrited and optimized) so quality is the same. Fixed bitrate used for output - now it is 448kbps (just for testing), release will use 640kbps. Of course, quality is worse than of specialied encoders. Bitrate factor is about 0.6, i.e. 640kbps quality of FFMPEG encoder ~= 384kbps quality of standalone encoder (it is theoretical value; it was no special quality tests).

[Valex]

@Monamona
> Could someone tell how to enable real-time AC3 encoding?

AC3 encoding is used when SPDIF output is specified and passthrough mode for selected format is disabled. Current SPDIF output mode you can see near SPDIF checkbox (in parenthesis).

But note that encoding may not be possible because some sound cards cannot transmit 44100Hz encoded audio, so if you have such source it will be encoded.

[pogo stick]

Originally posted by Valex
Planning (and OGG too)... Moreover it is no troubles to add new decoders anymore.
Great!
You are real WIZARD! :)

[yidaki]

I love AC3filter, it's like the FFDshow of surround sound :)

[Nippur]

Thank you!!!! I will try it! Waiting for the final version!!!

[athos]

Originally posted by Valex
But note that encoding may not be possible because some sound cards cannot transmit 44100Hz encoded audio, so if you have such source it will be encoded.
Perhaps you could include a sample rate conversion function in future versions, based on SSRC (http://shibatch.sourceforge.net/download/ssrc-1.29.zip)?
It would be good also for analog output, as the quality is better than for example SoundBlaster's hardware conversion (IMO).

Anyway, good work on the ac3filter, I always wanted a realtime ac3 encoder, even considered buying an nforce2 platform because of this.

[obieobieobie]

Great job Valex! I love Ac3filter.

Is the new ac3filter supposed to decode mp3 streams in avi-files?

Using Zoomplayer 3.20 I get both the standard Fraunhofer MP3 decoder in WinXP and the new ac3filter showing up.

Edit: By deactivating PCM decoding in the ac3filter system config, it stopped showing up, but it showed up when I disabled both the MPEG audio options.

[KpeX]

Excellent work valex :) I finally had time to test and the new features are working flawlessly on my box, AC3 encoding on the fly uses surprisingly low CPU and audio sync stays right on. I agree with athos that a resampling to 48Khz feature would be great for upmixing music on the fly. Cheers,

[E-Male]

I wanna second (third?) the resampling idea

resampling to 48khz (if source is not 48khz yet) and encoding to 5.1 ac3 at 640k on the fly would rock

that will help the new multichannel audio codecs a lot

looking forward a lot to the final release version

E-Male

[Tuning]

Thanks valex,

but i don't understand why ac3 filter is used while decoding AAC?:confused:

[yidaki]

ac3filter works as matrixmixer aswell, that's why it pops up while decoding other sound formats.
it disperses a 2.0 stream like a stereo mp3 to all your speakers(or atleast 5.1)

it would seem ac3filter is a litte buggy when using it together with reclock though, audio disappears for 3-5seconds once every 15min or so (even more often when used together with spdif, but i dont really need spdif when i'm using ac3filter so it's ok)

[SeeMoreDigital]

Yep this app is truly awesome!

And it arrived less than two hours after I asked about such an tool here (http://forum.doom9.org/showthread.php?s=&postid=497184#post497184). What a stroke of luck!


Many thanks

[rjamorim]

Originally posted by Valex
* AC3/DTS/LPCM/MPEG Audio layer I/II decoding support


Just wondering... why no layer III decoding?

(Please don't hate me if it's there and Valex forgot to mention... I didn't download it... waiting for a stable version ;) )

Also, don't you think it's about time you change the filter's name?

[DSPguru]

Great news!
Originally posted by rjamorim
Also, don't you think it's about time you change the filter's name? good point, mate :)

[SeeMoreDigital]

Originally posted by rjamorim
Also, don't you think it's about time you change the filter's name? I was just thinking that myself. It's obviously much more than an AC3 (Dolby Digital) DSdec filter now!

Personally I would really love it if it could process AAC LC and HE streams too, then it would become the ultimate S/PDIF tool!


Cheers

[obieobieobie]

I like the name AC3filter. It's not good marketing to change name now. ;)

[SeeMoreDigital]

Originally posted by obieobieobie
I like the name AC3filter. It's not good marketing to change name now. ;) I understand what you mean. The thing is this is a brand new tool which can do way more than decode AC3 (aka Dolby Digital) streams.


Cheers

[DKDIB]

Valex wrote:
> * AC3/DTS/LPCM/MPEG Audio layer I/II decoding support
> * AVI/AC3 & AVI/DTS support

Great news!!! :D:D:D
:respect:



Tuning wrote:
> but i don't understand why ac3 filter is used while decoding AAC?:confused:

I agree.
IMHO it's not a good idea to make a single DS filter that decodes too many formats, because the source is quite difficult to manage (bugfixes, new features, optimizations, ...) and some users may prefer an other decoder for some format.



E-Male wrote:
> resampling to 48khz (if source is not 48khz yet) and encoding to 5.1
> ac3 at 640k on the fly would rock

Sycho wrote:
> is it possible to have a 7.1 output from AC3filter.

IMVHO they're a couple of very interesting features (especially the 1st one).



Time is money wrote:
> I found if I want WMP play a dts audio file, I still need a source
> like Gabest's dtsac3source.ax

Arimicio for this info! ^__^

[rjamorim]

Originally posted by DKDIB
I agree.
IMHO it's not a good idea to make a single DS filter that decodes too many formats, because the source is quite difficult to manage (bugfixes, new features, optimizations, ...) and some users may prefer an other decoder for some format.

It shouldn't be difficult yo manage if each format has a good library.

The MAD library is stable, the Vorbis library is rock stable, the FAAD2 library has been stable for the last few months (although it tends to change a little whenever support for a new AAC profile is added), a52lib also seems to be stable.

If the format has a good library, and it gets updated, you just need to drop it in the place of the older library, and recompile. There should be no problems.

BTW: ffdshow is a single DS filter that decodes too many formats, and it's a great filter neverthless :)

About users wanting another decoder for a specific format: Valex should do just like Milan, and let the user decide which formats you want associated with the filter and which you don't.

Regards;

Roberto.

[Mitchjs]

WOW!

valex its great!!

Im having a few issues... not 100% sure what they are

mostly DTS passthou... I couldnt get a .DTS file to pass
its 44.1k sampling... that if it passed would play tune...
and if it passed and got Kmixer screwed id get noise, but
nothing passed...

also i think it caused the playback program to hang

could you make a option to disable all AC3 encoding...

I have a Nforce board and want the encode to happen via hardware

basicly, i enable SPDIF and check PASS DTS & AC3
but i want my pcm, passed un encoded, so my HW can do it


Valex, again wonderfull code

...I prob didnt have to buy the nforce2 board :)
I have a few wma 5.1 audios and needed it out spdif
for my decoder to do...

Mitch

[Shayne]

Hi has anyone got spdif passthru of 48 Hz dts to play? and if so what settings are being used?

Thanks

Peace

[pogo stick]

Originally posted by DKDIB
Tuning wrote:
> but i don't understand why ac3 filter is used while decoding AAC?:confused:

I agree.
IMHO it's not a good idea to make a single DS filter that decodes too many formats, because the source is quite difficult to manage (bugfixes, new features, optimizations, ...) and some users may prefer an other decoder for some format.
I may be wrong, but Tuning didn't mean that AAC shouldn't be included in AC3Filter.
AAC decoder is not in yet, but AC3Filter is used after decoding AAC as postprocessor if System -> Use AC3Filter for -> PCM is enabled.
It's something like raw video - all supported in ffdshow.
Please, correct me if I'm mistaken.

[yablo]

Thanks Valex,
I've been looking for the live ac3 encoding feature for some time.

Unfortunately in this version the feature just teases me. It will work for a short amount of time (less than 30sec) before the audio cuts out. Tryed it on two different computers with the same results.

Anyway, I look forward to your next release.:D

[arty]

Valex: just a "THANK YOU" !

how about DTS encoding instead of DD (or together) ?
yes, I know that could be hard without any docs... :D

[yidaki]

No bugs yet :)
(except for the incompability with reclock, but that's not really your department)

Would be cool if there was a sample rate output setting where it automatically chooses the same as the input, so that you wont have to change that everytime you watch another movie.

[Solo]

I battled to get my external amp to recognise my AC3 audio track last night. The SPDIF (disabled) check box confuses me :rolleyes: With old filter I just selected spdif out from drop down menu and all OK. Is spdif then enabled but default ?

Went back to ver70 and it works perfect again.

[Monamona]

Originally posted by Valex
@Monamona
> Could someone tell how to enable real-time AC3 encoding?

AC3 encoding is used when SPDIF output is specified and passthrough mode for selected format is disabled. Current SPDIF output mode you can see near SPDIF checkbox (in parenthesis).

But note that encoding may not be possible because some sound cards cannot transmit 44100Hz encoded audio, so if you have such source it will be encoded.

I connect soundcard and Klipsch GMX-D5.1 with optical cable, but
it still does not recognize dolby digital.(means 'not encoded')
Is there any special way to 'specify SPDIF'?
Always shows 'disabled' in parenthesis.

[MentholMoose]

There are issues with SPDIF on 1.01a RC1. I did post a workaround on the first page, but it might not work 100%.

[Valex]

Fixed (I hope :) ) bug with SPDIF passthrough:
http://ac3filter.sourceforge.net/download/ac3filter_1_01a_rc2.exe

(ac3 encode bug is still alive...)

[Valex]

@obieobieobie
Using Zoomplayer 3.20 I get both the standard Fraunhofer MP3 decoder in WinXP and the new ac3filter showing up.

By deactivating PCM decoding in the ac3filter system config, it stopped showing up, but it showed up when I disabled both the MPEG audio options.

Filter does not decode mp3 but can do post-processing, so both mp3 decoder and filter was included. By disabling PCM support you disable post-processing.

[Valex]

@Tuning
but i don't understand why ac3 filter is used while decoding AAC?

Because it replaces MatrixMixer functionality and can do post-processing for any audio source. To disable this disable 'Use AC3Filter for PCM' option at 'System' page.

[Valex]

@yidaki
it would seem ac3filter is a litte buggy when using it together with reclock though, audio disappears for 3-5seconds once every 15min or so

Could you test it without Reclock? (so is it problem connected with filters compatibility or because of ac3filter only?)

Would be cool if there was a sample rate output setting where it automatically chooses the same as the input

Sample rate? Maybe you mean speakers configuration?

[Valex]

@rjamorim
Just wondering... why no layer III decoding?
Layer 1/2 formats are mandatory for DVD playback. So it is possible that DVD logo uses ac3, menu uses mp2 and video uses lpcm audio track and addons with dts. :). And it is required for decoder to switch between these formats.

Also, don't you think it's about time you change the filter's name?

Why? It is good and well-known 'brand' :)

About users wanting another decoder for a specific format: Valex should do just like Milan, and let the user decide which formats you want associated with the filter and which you don't.

Already done :)