LAV Filters - DirectShow Media Splitter and Decoders

[pankov]

Mark_A_W,
sorry to join so late in the discussion about the madFlac problem but could you post a sample, so I can try and confirm/deny if actually is a problem only on your system?

[Andy o]

Nev, DTS-ES discrete has wrong channel mappings on LAV decoder (not Arcsoft). See this test (http://www.megaupload.com/?d=NQ36GF0I).


With my current settings it works like this, on the content i threw at it:

- Stereo, 5.1 and 7.1 gets output as-is (2, 6, 8 channels respectively)
- Mono gets output as Stereo (only 1.1.0.8 of the dll seems to work at all with Mono)
- 6.1 gets converted to 7.1 by doubling back-center to back left/right.

With 6.1 on a 7.1 system there's one thing... L and R side surrounds get leaked to L and R back surr. respectively. Center back surround plays in both back surrounds as you say. It's not a big problem, but if you're actually running a 6.1 system, you'll see your L side and R side leaked into the same Back Surround speaker.


EDIT: The DTS-ES discrete issue with LAV audio only happens with WASAPI exclusive, where the decoder is outputting actually 6.1, and making Windows output 6.1 as well. If Windows outputs 7.1 (not exclusive mode) then mapping is OK.

[SamuriHL]

Just because it can be done doesn't mean the resulting file is valid. Having this kind of track in a MKV is *NOT* a valid file. MKV does *NOT* allow for interleaving two audio streams into one mkv stream. mkvmerge cannot do it, and Mosu specifically denied the feature some time back because it would go against any MKV design, and it is basically the reference muxer.

Besides, i don't get the advantage of that hack. You will not be able to play the AC3 stream with most players, if even with any player. Some tools might even get confused and break on that stream (like LAV Audio). So, why?

The original reason I was looking into it was when i first started doing mkv's. I wanted to be able to quickly convert back to BD without having to rerip the disc. Tsmuxer won't touch a non-interleaved THD track. Hence the interest in it. I wasn't suggesting it SHOULD be done, only that it could be. I personally just use makemkv to make them and no longer worry about the BD format.

[nevcairiel]

Nev, DTS-ES discrete has wrong channel mappings on LAV decoder (not Arcsoft). See this test (http://www.megaupload.com/?d=NQ36GF0I).

With that you mean when ffmpeg is doing the decoding, the "dca" decoder, right?

It could just be that outputting 6.1 through WASAPI is not supported. Once PCM leaves through HDMI or S/PDIF, the channel layout is gone. The layout is *only* used by the DirectSound mixer to know which channel is what when upmixing. Of course its also used when mapping PCM to analog speaker outputs. But HDMI does not carry speaker assignments.

I guess thats why the ArcSoft decoder upmixes to 7.1
If playing with the ArcSoft decoder, is that surround leak into the back channels also present?

[Mark_A_W]

Mark_A_W,
sorry to join so late in the discussion about the madFlac problem but could you post a sample, so I can try and confirm/deny if actually is a problem only on your system?


Thanks.

Firstly, there almost certainly isn't a madFLAC problem, or LAV Filter problem, just something funny with my system.

Any bluray sourced MKV file with FLAC audio (created with eac3to and mkvmerge) will show the problem for me.

I just can't get madFLAC and the LAV Splitter to go together.

If you can test any file, that would be appreciated. If you don't have an MKV file, I'll find a link to a sample.

Mark

[joeydrunk]

Using an external player from XBMC works, but its not perfect. The integration is never really 100% smooth.

DSPlayer does not support madVR, and i'm not going back. :p
Also, its dead. :)

It's pretty much smooth, its not perfect but its a lot better than just using a player to watch all your movies. Also I know its dead bit I'm using a pre Eden build compiled with dsplayer to bitstream seamlessly in xbmc because Im having trouble with the playercorefactory.xml file in my htpc. Just saying. Is jriver the only actual media center then where you can use your filters on an internal player, why not use that then? Why no madvr in dsplayer?

[Andy o]

With that you mean when ffmpeg is doing the decoding, the "dca" decoder, right?

It could just be that outputting 6.1 through WASAPI is not supported. Once PCM leaves through HDMI or S/PDIF, the channel layout is gone. The layout is *only* used by the DirectSound mixer to know which channel is what when upmixing. Of course its also used when mapping PCM to analog speaker outputs. But HDMI does not carry speaker assignments.Yeah, I meant with the dca decoder. So then HDMI only carries the number of channels, and the receiver has to know what to do with them? I'm not sure what you meant exactly by 6.1 not supported through WASAPI, but my receiver is getting 6.1 LPCM (with surround back center lit up) with that decoder. Is that what you meant?

I guess thats why the ArcSoft decoder upmixes to 7.1
If playing with the ArcSoft decoder, is that surround leak into the back channels also present?Actually I meant that it happens with the Arcsoft decoder. I suspect it's actually the DTS spec for it, when mapping 6.1 to 7.1 systems, and not a bug. My receiver does it with DTS-HD MA 6.1, but not with DTS-ES. The only difference is that Arcsoft does it with DTS-ES and DTS-MA 6.1. You can tell this with that sample as well.

[nevcairiel]

Yes, HDMI can only know the number of channels, any other metadata cannot be carried over the link. So what exactly do you mean with wrong channel mappings?

The only thing possible is that the channels are ordered wrong.

[Andy o]

When playing with WASAPI exclusive, the ATI 5770 outputs 6.1 channels PCM with the dca decoder, including the discrete rear center. L, C and R are fine, but then SR goes to SB, SB to SL and SL to SR, then LFE is OK too. The Nvidia doesn't output 6.1 via WASAPI?

When bitstreaming though, the receiver also gets the 6.1 channels, and then channels are OK.

[nevcairiel]

The default PCM layout for 6.1 would be like this:

L R C LFE BC SL SR

From what you're describing, your receiver expects

L R C LFE SL SR BC (which is actually L R C LFE BL BR BC)

I will test on mine, and see if its the same.
Can you test other 6.1 sources, if you have any?

PS:
MS defines the channels like this:

http://msdn.microsoft.com/en-us/library/ff537083(v=vs.85).aspx

The order of the flags also defines the order in the bitstream (thats why its BL BR BC, and not SL SR BC)

Edit:
My receiver does not seem to like 6.1 at all, it claims to receive Stereo, and only the two front channels play. I cannot say if thats a Onkyo thing, or maybe NVIDIA. Bitstreaming works, however, and properly doubles the back center into the back speakers.
It would be great if you could test other 6.1 sources, so i can switch the matrix around.

[Andy o]

If bitstreaming works OK it's probably not the file that has incorrect mapping, right? As far as I can tell the DTS-HD 6.1 content also carries a DTS-ES discrete core, but not the 7.1 content. I think I got one DTS-HD MA 6.1 movie, I'll see if I can tell the channel switches without a discrete test.

It would be great if you could test other 6.1 sources, so i can switch the matrix around.
But when not using WASAPI exclusive and setting Windows to 7.1, mappings are correct. If you switch the matrix, won't this be affected?

Just in case, it's not that I'm complaining, cause I have to use Arcsoft anyway for DTS, but just thought you should know. I'll keep testing, lemme know if there's anything in particular you'd like me to test.

[nevcairiel]

It wouldn't be affected as it would also change the channel mask, so the mixer knows.

[Andy o]

Once PCM leaves through HDMI or S/PDIF, the channel layout is gone.

Hmm so it just dawned on me what this means. I've done some more reading, and I think the DTS-HD MA "2.0" with surround "flag" that some Pioneer receivers like mine are having trouble decoding, don't have a "flag" at all and the mono surround is not matrixed. I think they're 3.0 channels. So, I also know that there are DTS-HD MA 3.0 movies (http://forum.blu-ray.com/blu-ray-movies-north-america/158996-list-bds-dts-hd-ma-1-0-2-0-3-0-4-0-mixes.html), whose 3rd channel is center front. So, then if the decoding was done in the PC, HDMI would not be able to tell the receiver if that 3rd channel is center or mono surround. I know that my receiver when bitstreaming detects mono surround, from the movie I'm having trouble with (Delicatessen), and also detects center properly (from You Will Meet a Tall Dark Stranger). The way the Arcsoft decoder deals with the latter, is to put the 3.0 in a 5.1 PCM stream. With the former, it does not decode the mono surround at all, and just sends 2.0 out.

So, that's one advantage of bitstreaming, I guess. Content with same number of channels, but different configuration.

[nevcairiel]

Yeah, with PCM you need to wrap it into a bigger container. I'm working on extending that support now, so that such a file might be properly wrapped into 5.1. Of course when using that mode, you would lose the ability to use full upmixing capabilities of your decoder (however it handles that), as i would already send some extra PCM channels with just silence.

[Andy o]

I'm pretty sure that Neo:6 could not be enabled with that 3.0 LCR stream, so it's probably the same. You can use it with 2.0 and I think 5.1 content, but nothing weird like 3.0.

[DrNein]

It's pretty much smooth, its not perfect but its a lot better than just using a player to watch all your movies. Also I know its dead bit I'm using a pre Eden build compiled with dsplayer to bitstream seamlessly in xbmc because Im having trouble with the playercorefactory.xml file in my htpc.

There is only a brief flash (desktop and/or wallpaper if applicable) between the end of MPC-HC playback and resumption of XBMC browser. If blank or the same backround image then it is even less noticeable.

I run XBMC with the switch "-p" for portable so the user data is in a subfolder of the application and therefore the playcorefactory.xml for an external player. See below and edit monitor number and file types as necessary.

C:\Program Files (x86)\XBMC\portable_data\userdata\playercorefactory.xml

<playercorefactory>
<players>
<player name="MPC-HC" type="ExternalPlayer" audio="false" video="true">
<filename>C:\Program Files (x86)\MPC-HC\mpc-hc.exe</filename>
<args>"{1}" /monitor 2 /fullscreen /close</args>
<hidexbmc>false</hidexbmc>
<hideconsole>false</hideconsole>
<warpcursor>none</warpcursor>
</player>
</players>
<rules action="prepend">
<rule filetypes="avi|flv|mkv|mov|mp4|m2ts|ts|vob|wmv" player="MPC-HC"/>
</rules>
</playercorefactory>

Note: if internet streaming is desired then do not add certain filetypes (since MPC-HC is incapable of streaming) and rather leave them to XBMC internal player. These include at least flv, mp4, and wmv.


MPC-HC + madVR + LAV + ArcSoft or CyberLink + XBMC = :)

:thanks: to gabest, madshi, nevcairiel, and other developers.

[Plutotype]

Hi folks, Nev,
Need a help to explain some things which are related to audio time shift parameter in MPC-HC. Recently, I have noticed that I need to set almost 200ms audio delay for all of my videos which are being played back on my SONY KDL-40EX500 to have (near) perfect lipsync. When I play the same movies on my PC Monitor ( DELL U2311H ), audio time shift is not needed. So I have used input input lag tool to determine the approximate input lag of the SONY LCD TV. As reference I have used the DELL LCD, which has according to following test 0-30ms input lag ( 10.6ms in average ). http://www.tftcentral.co.uk/reviews/content/dell_u2311h.htm#input_lag
Using high shutter speed camera I have managed to capture the input lag of the SONY LCD TV - 90ms http://imageshack.us/f/813/inputlagtestdell2311hvs.png/
As I can not have input lag tool "always on top" and madVR does not allow to have the fullscreen exclusive mode on two separate displays at once, I have played the same video in windowed mode and run the input lag tool on top ( duplicated desktop ).
So basically. I should set at least 100ms as time shift in MPC-HC and lipsync issue shouldnt occur. I have tested 10 different films where lot of dialogues were present. After further perception, I decided to increase the audio time shift ( audio delay ) for playback on the SONY TV to around 200ms to get ( near ) perfect lipsync. I dont use any video image processing at the SONY´s picture settings beside some colour/B/W calibration. Im not in a game mode, but using the general SONY scene mode. No frame interpolation.
LAVaudio 0.26.
madVR 0.61 with working autochange to 23/24 - 1080p and absolute smooth playback.

1. Can the SONY TV add such terrible input lag of around 200ms?
2. Why multiple tests showed difference between U2311H and the SONY LCD TV only 90ms, but from my perspective I had to set almost around 200ms?
3. Does madvr add something specific what increases for example the rendering times when in fullscreen exclusive mode?
4. Does the input lag goes up with bigger and bigger LCD TVs ( like 46, 52, 60 etc)?
5. Would it be possible to have different audio time shift settings for each display device in MPC-HC :-) ?

Thanks for any input.

Pluto

[nevcairiel]

Somehow i dont think this belongs in this thread. :D

I do have the same issue with my TV however, it needs around ~200ms if delay, except it only needs it at 24p, on 50Hz or 60Hz, its fine. It must do something special in that mode that causes, but i couldn't figure it out yet. I also have any image processing features disabled.
Briefly i thought maybe the delay is in number of frames, as ~5 frames delay at 24p is roughly 205ms, but 5 frames at 50Hz is only 100ms, and maybe i just don't notice the 100ms anymore? But i still think i would notice 100ms, thats still alot of delay..

[Plutotype]

Thanks nevcairiel for your input, sorry for asking here this stuff. Im glad that I´m not the only one that has sensible eyes/ears and needs to add 200ms of audio delay..(-:

Anyway, if you will manage later to add downmixing capabilities for LAVaudio, pls dont forget to add audio delay feature there. As some users have 2 or 3 different display devices it would be nice to have the possibility to set separate audio delay for each display device like madshi has done for the refresh rate autochanger in madVR. But this is just an idea.

The downmixing feature is much needed as with integration of Arcsoft HD audio decoder, user can not change the default number of channels outputted by Arcsoft ( the win.ini hack ). If 7.1 is inputed,7.1 goes out. This would help us to have 7.1 channels downmixed into 5.1 setups.

[nevcairiel]

The Audio decoder will definitely not go around trying to figure out which display devices you have. :P

Not sure about actual downmixing yet. Upmixing is so much easier, you know!
I actually found some article about proper mixing of channels the other day, maybe i'll make 0.27 the big audio processing version.
Already have to work on alot of this stuff for perfect DTS-HD decoding.

Spent the better part of the afternoon (and evening) writing a parser for all kinds of DTS headers .. Core, XCh, XXCh, HD, HD-XCh, HD-XXCh .. the guy who designed the HD header deserves to be shot or something, so freaking complicated. Glad i don't care about the other HD headers, i only want the info about channels and speaker layout.

[Plutotype]

One small add-on ( HD -audio ) and so many details to be fixed afterwards..but i like your approach..

I have another request for correct channel number identification by the LAVsplitter. For example 7.1 audio stream is being reported correctly as 8 channel in Reclock, but as 5.1 ( 6channel ) at the LAVsplitter. For average users like me, it can be confusing..(-:
Last time I have asked about 96k audio streams files reported by LAVsplitter as 48k, you wrote that this is not possible for now. Maybe you will manage to fix this and the correct number of channels reported by LAVsplitter in the future too. LAVaudio is reporting this correctly...superb.

[nevcairiel]

Unlikely.
ffmpeg does all that, and hacking the parsing in ffmpeg just for a cosmetic issue, nah.

[Markolc81]

This LAVsplitter is amazing. Combined with the Arcsoft HD Audio Decoder, I'm getting fully decoded LPCM through Reclock to my AVR and it's awesome when it's lipsynced! However I'm having a problem with my MKVs that contain TrueHD. I don't get any sound when there's a TrueHD track. It won't bitstream properly, and also if I let LAV Audio decode it, it's just a series of pops and hisses. AC3, DTS, DTS-HDMA and FLAC all bitstream or decode perfect with LAV Audio, but no go when it comes to TrueHD. Specifically I tried the rips of Batman Begins BD and Star Trek (2009) BD. Both discs were ripped with Mike's Another EAC3to-gui. Also both these movies bitstreamed fine when I had my ffdshow configured for bitstreaming. Any thoughts?

[nevcairiel]

Sounds like you're not using LAV Splitter, rather some other splitter like Haali or the MPC-HC splitter. TrueHD will not work with those.

[Markolc81]

I do have Lav Splitter listed in my external filters list, and I have it set to preferred. I see that LAV Splitter is being used when playing the MKV by right clicking and looking in the filters being used. I'm going to get Graphstudio installed for a better picture of what's going on. Thanks

[Sebastiii]

The Audio decoder will definitely not go around trying to figure out which display devices you have. :P

Not sure about actual downmixing yet. Upmixing is so much easier, you know!
I actually found some article about proper mixing of channels the other day, maybe i'll make 0.27 the big audio processing version.
Already have to work on alot of this stuff for perfect DTS-HD decoding.

Spent the better part of the afternoon (and evening) writing a parser for all kinds of DTS headers .. Core, XCh, XXCh, HD, HD-XCh, HD-XXCh .. the guy who designed the HD header deserves to be shot or something, so freaking complicated. Glad i don't care about the other HD headers, i only want the info about channels and speaker layout.

:) good luck :) it's amazing stuff again :P

[boyumeow]

Thanks for .26!

BTW, any chance to teach LAVF audio decoder RealMedia COOK format?

If my setup is not wrong, RealMedia COOK format is supported, but not RealMedia video. Thanks.

[Markolc81]

Sounds like you're not using LAV Splitter, rather some other splitter like Haali or the MPC-HC splitter. TrueHD will not work with those.

You were absolutely right. fter setting up Graphstudio I learned that I forgot to uncheck Matroska from the internal filters list of MPC-HC. Thanks Nevcairiel!

Keep up the good work!

BTW, do you take paypal donations for your efforts?

[kennae]

Hi. Thanks for your great filters!

Only feature I really need for my whole PC/HTPC setup is profiles. I need to have two separate profiles for headphones and for AVR.
I have great USB DAC and good headphones, and use spdif/coax out to my AVR receiver when i'm not using the headphones.

With AVR I can use all fancy HD-codecs and stuff, but with the USB DAC it only supports 48KHz 24bit stereo.

So what I need is an easy way to change every audio related setting from one to another several times a day, and it's pain in the ass right now.

I've used ffdshow for audio and there you have profiles but it isnt so fluent and would like to skip the part.
I also need AC3Filter so I can set the channels right but I would love to be able to ditch it and just use your filter and reclock.

The setup I have now is all latest: MPC-HC - FFDSHOW (just the video right now) - LAV Filters - MadVR - Relock.
If I want to change from headphones to AVR I need to change settings in reclock and in LAV audio, and it's not that fast when you have to open a file first.

I can live with my setup since this is what I have been using for years but it would be lovely to have it change profiles/settings by wich device is in use.

[Mark_A_W]

Sounds kinda why I use Zoom Player for movies on the projector (full blown custom filter setup, madVR, LAV, Reclock, Room Correction), but MPC-HC (internal filters and VMR9) for general video/tv on the preview 24" monitor.

And WMP for background music on the "PC speakers" (focal bookshelves), but JMRC for music on the HT system (it has the room correction set up via a VST filter).


Sometimes you want it to "just work" and sometimes you want perfection....

[Andy o]

Why do you have to change settings in LAV? Also, what USB interface do you have? Have you tried using WASAPI exclusive? I'm using a Xonar DX for Dolby Headphone, and exclusive mode takes care of channel mapping. I get correct 5.1 mapping and I don't have to change anything but devices.

[mindbomb]

lets say you have a dts-hd file, and you only have dts checked in the bitstreaming options, but you have the arcsoft dll in the lav audio folder.

what happens? Do you bitstream dts core, or do you decode dts-hd?

I can imagine people using spdif prefer the former, while people using hdmi without PAP prefer the latter.

[Andy o]

You bitstream the core. If you're using S/PDIF though, you might want to use lossless decoding and then have ReClock convert to AC3 so you can take advantage of ReClock.

[nevcairiel]

lets say you have a dts-hd file, and you only have dts checked in the bitstreaming options, but you have the arcsoft dll in the lav audio folder.

what happens? Do you bitstream dts core, or do you decode dts-hd?

I can imagine people using spdif prefer the former, while people using hdmi without PAP prefer the latter.

If DTS bitstreaming is active, it'll always bitstream all DTS formats, no other checks are done.

[Budtz]

Hey guys. Im using ffdshow audio right now to decode audio. Any reason to switch to LAV audio? Is there a quality difference between them? And what about settings like "jitter control" and "resampeling to 48khz" that ffdshow has. Will LAV audio have those or maybe its just not needed with LAV.

[nevcairiel]

The "jitter control" is stupid and not needed. An option that kinda replaces it is the "Auto A/V sync correction" or how i named it, but its really alot different.

Resampling is not supported yet.

[Dogway]

Is there a way to delay audio? Im not sure if it has to do with MPC or LAV audio, but just asking.
Also how can I be sure audio is being bitstreamed? Does it depend on Graphic Card capabilities?

[nevcairiel]

Currently, there is no option in LAV Audio to delay the audio. I'll implement that option in the future, however.

If you enable bitstreaming in LAV Audio, it'll always bitstream. If you get audio, and LAV Audio is still being used, then its bitstreaming.
Yes, it depends on your HDMI connection, usually supplied by the graphics card. Of course also depends on your receiver, if it understands that specific audio format.

[Gleb Egorych]

Im using ffdshow audio right now to decode audio. Any reason to switch to LAV audio? Is there a quality difference between them?
Depends on your usage profile. Some formats are better supported in ffdshow, some are better supoorted in LAV Audio. ffdshow also has a lot of postprocessing features.

[ney2x]

@nevcairiel - can you make a tray icon for LAV CUVID and LAV Filters? I mainly used WMP12 for movies and music, I just want to make sure if they're being used when playing audio/videos. Thanks.

[Sven75]

@nev - did you find some time to look at the lavsplitter + PowerPoint issues I posted on page 130 (#2597)?

I tested version 0.26 and the behaviour is still the same in both cases: flv not working in PowerPoint 2003/2007 and the mpeg file not working at all. In graphstudio everything is working fine.

When using MPC-FLV-Splitter for the flv and Elecard Demuxer for the mpeg both files are fine in PowerPoint, too.

FYI, we are about to finish our first big congress (in Hamburg, btw) with LAV on our computers and it's doing a great job so far. So thanks a lot again!

[nevcairiel]

@nev - did you find some time to look at the lavsplitter + PowerPoint issues I posted on page 130 (#2597)?

Not yet, i've been busy working on new shiny features, its always so much more tempting.... :)
I'll keep it in mind and look at it when i get a chance.

[Sven75]

No prob :)

It is just the icing on the cake (and me being so curious to know as to what is really going wrong with PowerPoint).

[joeydrunk]

There is only a brief flash (desktop and/or wallpaper if applicable) between the end of MPC-HC playback and resumption of XBMC browser. If blank or the same backround image then it is even less noticeable.

I run XBMC with the switch "-p" for portable so the user data is in a subfolder of the application and therefore the playcorefactory.xml for an external player. See below and edit monitor number and file types as necessary.

C:\Program Files (x86)\XBMC\portable_data\userdata\playercorefactory.xml

<playercorefactory>
<players>
<player name="MPC-HC" type="ExternalPlayer" audio="false" video="true">
<filename>C:\Program Files (x86)\MPC-HC\mpc-hc.exe</filename>
<args>"{1}" /monitor 2 /fullscreen /close</args>
<hidexbmc>false</hidexbmc>
<hideconsole>false</hideconsole>
<warpcursor>none</warpcursor>
</player>
</players>
<rules action="prepend">
<rule filetypes="avi|flv|mkv|mov|mp4|m2ts|ts|vob|wmv" player="MPC-HC"/>
</rules>
</playercorefactory>


MPC-HC + madVR + LAV + ArcSoft or CyberLink + XBMC = :)

:thanks: to gabest, madshi, nevcairiel, and other developers.

Off topic but what is the purpose if the portable data file?

[Budtz]

Depends on your usage profile. Some formats are better supported in ffdshow, some are better supoorted in LAV Audio. ffdshow also has a lot of postprocessing features.

I play mostly hd mkv-files with dts sound or dolby digital. The resampeling to 48Khz in ffdshow seems like a nice postproccessing feture that should give audio a boost.

Also what excactly does a splitter do? All i know is i cant seem to switch between different subs/audio without one.

[nevcairiel]

HD audio always has at least a 48kHz sample rate already. I fail to see the point. Besides i let ReClock do resampling to 96000Hz before output to the AVR. :)

[madshi]

"shiny"
I think you've watched too much Firefly... :p

[CruNcher]

Hi Andy,

I have great news. I know we've had our differences in the past; however, I was hoping we can call it truce and try to repair our relationship by extending an olive branch... offering a pre-release MpegDemux.ax which sounds like you may be very interested in. This build is only hours old. I was given permission to share this demux for people who needed it. I've been working directly with the developer at sageTV who appears to be responsible for all code changes made to MpegDemux.ax. This demux should hopefully be included in the next sageTV beta.

This pre-release version of sageTV demux is now fullly compatible with Arcsoft Audio Decode HD (full support for DTS-MA/TrueHD decoding without a hitch). In addition to that, it now also supports LAV Audio Decoder's TrueHD. If you have the arcsoft audio decoder package installed, all you have to do it tell sageTV to enable HD audio support and specify a single audio decoder: Arcsoft Audio HD. If you dont have arcsoft, you can at least have TrueHD audio decoding via LAV Audio decoder. My sageTV client has it's internal demuxing off; everything but bluray audio formats (and I think TV).

You might have noticed an earlier post from me today playing back a sample 7.1/96khz m2ts file directly in sageTV.
[/URL]

I got the same result no matter which container I used . I also had the same results with DTS-MA. Of course, it also works just as great with blu-ray discs/iso's as well on on my extender.

Here's the file:
http://hotfile.com/dl/117954203/045f397/MpegDeMux.zip

I've only tested this with TMT3 standalone decoder package (doesn't require to install the entire player just to be able to have access to the audio filter... only a small change in the Windows win.ini file:

1) Open up C:\Windows\win.ini in notepad.
2) Add the following text to the end:
Code:

[HKEY_LOCAL_MACHINE\SOFTWARE\ArcSoft\MPEG Codec\Decoder Setting\]
AudioOutputMode=6
AudioDynamicRange=1

3) Change the numbers to match the settings you prefer.

Code:

AudioDynamicRange:
0: Light
1: Normal
2: Threaded

AudioOutputMode:
0,1,2,5: Stereo
6,13: 5.1 Channels
17: 7.1 Channels
16: S/PDIF

Sincerely,
MKANET

Hui i guess that one doesn't suffer from the Frame accurate seeking issues [url]http://forum.doom9.org/showpost.php?p=1499012&postcount=2642 (http://i67.photobucket.com/albums/h283/mkanet/Screenshot-1509.jpg) with Mainconcepts Decoder @ least i hope so :)

[nevcairiel]

I think you've watched too much Firefly... :p
Too much Firefly? Thats possible? :p

[mkanet]

There seems to be a problem with win.ini hack in Arcsoft standalone. It works perfect if you choose AudioOutputMode: 17 (for 7.1 full bitdepth and bitrate 24/96kz or above). However, if I try to downconvert in win.ini to 5.1 by using AudioOutputMode: 13 (or 6), DTS-MA is limited to 48khz.

I was hoping to let LAV audio to have a universal bitrate/bitdepth/speaker configuration of arcsoft audio HD DTS-MA or other audio formats.. so if I I can set speaker config to 5.1, bit depth to 24bit, and set bitrate to "unlimited". Maybe this is already in the works?

PS: What audio formats will LAV audio decoder allow Arcsoft audio decoder to use? Is it just limited to DTS-MA?

Thanks again,
MKANET