LAV Filters - DirectShow Media Splitter and Decoders

[Portioli]

may i ask you sth?
if i have LAV FILTERS in wmp & w7mc(using codec tweak tool) the only way to have external subs is using ffdhow?

[Thunderbolt8]

Originally Posted by ney2x

DirectVobSub v2.40.3237 still broken. I thought someone fixed it but still a no go...

OT, but +1.
has already been like that with .2995 (and maybe also before).

[Rectal Prolapse]

Hello! I would like to confirm if LAV Audio + ARCSoft decoder will properly decode DTS-HD Master Audio. Will it do this losslessly or will it only decode the DTS core? I tried searching the thread but could not confirm.

Thanks!

[SamuriHL]

Lossless.

[Portioli]

the best way to have lossless DTS-HD MA with the exact same sample size & sample rate happens with lav audio and the dtsdecoderdll.dll using reclock in wasapi exclusive

[Gleb Egorych]

Portioli, your screenshot shows that you don't have lossless and exact sample rate because ReClock definitely resamples on your screenshot. It sends bit exact resampled audio not source audio.

[Gleb Egorych]

I tested 5 different versions of dtsdecoderdll.dll along with LAV Audio, here are results.

Versions tested: 1.1.0.0, .1, .5, .7, .8.

Samples tested:
1) Lossy DTS 2.0, 2.1, 5.1, 6.0, 6.1 configs, 16/48, 24/48 and 24/96 bitdepth/sample rate;
2) DTS-MA 1.0, 2.0, 2.1, 5.1, 6.1, 7.1 configs including 7.1 "strange setup" config (eac3to term), 16/48, 24/48, 24/96 variants.
My results differ from results that some people posted in eac3to thread, I guess there may be some eac3to specific problems.

lossy DTS
1) 1.1.0.0 and 1.1.0.1 always decode lossy DTS as 24 bit, and their decoding results differ from each other.
2) 1.1.0.5 and up decode in proper bitdepth.
3) Versions 1.1.0.5 and up decode lossy DTS identically.
4) 1.1.0.1 and 1.1.0.5 decode 24 bit lossy DTS identically.
5) Decoding of 16 bit lossy DTS was changed from 1.1.0.0 to 1.1.0.1 and from 1.1.0.1 to 1.1.0.5.
6) 1.1.0.7 and 1.1.0.8 decode 6.0 without back center channel.

The conclusion for lossy: I assume the best version for lossy DTS is 1.1.0.5. It decodes all configurations I've tested, in proper bitdepth, and decoding algorithm didn't change since this version (except 6.0 bug in .7 and .8).

DTS-MA
All versions output identical streams except mono case. Mono decoding was broken in 1.1.0.5 and partially fixed in 1.1.0.8 (outputs as stereo). Note that MA 7.1 16/48 "strange setup" is decoded as 24/48 by all versions.

The conclusion for lossless: the best versions are 1.1.0.0 and 1.1.0.1.

Of course, to be sure that decoding is done properly some "reference proper" decoder is needed. I don't have it.

EDIT: Forgot to mention about 2.1 bug. None of the versions can decode LFE channel in 2.1 files.

[nevcairiel]

"Lossy" codecs do not have a "proper" bitdepth, in a perfect world it would output floating point data as the ffmpeg decoder does.
Because the process is lossy, there won't be an "accurate" result, different decoders produce different results (most likely), and most decoders also decode to floating point .. ArcSoft is most likely just cutting/rounding to the bitdepth of the source material.

Unless you mean that its outputting 24-bit samples which only have 16bit filled? (Which i've seen before as well - but i may be able to influence that from the outside actually)

Regarding the "reference proper", over in the eac3to thread, some people compared output to the nero and sonic decoders, and apparently only the newer version of the ArcSoft decoder match the output of those two (which is otherwise identical) - on lossless DTS-HD MA only, of course, comparing bit-exactness of lossy is pointless.

[Portioli]

Dts hd - ma is a codec. As long as we have a non hd audio capable receiver we cannot bitstream.
So our goal is the player ( pc+ audio filter ) to decode this codec and send multi-Chanel PCM to our receiver if it has hdmi ( or analog)
if these PCM tracks have the same sample size and same samplerate why this process is not lossless as long as PCM Is set : wasapi exclusive & PCM output's sampling rate is set same as input.
I am talking only for lossless and not bit perfect.
What's the difference of using wasapi for flac files?

[e-t172]

Resampling with Reclock is not lossless. It is very close to being lossless (so close the difference is probably inaudible) but it is not lossless.

If you want lossless bit-perfect sound you need to disable Reclock's time correction ("slave" option). But, obviously, the downside is you'll probably have some frame drops/repeats if you do that. So it's a compromise. Most people prefer to let Reclock resample because the effects on the audio are negligeable, whereas a frame drop/repeat can be quite noticeable.

Also, according to some people, speeding up and resampling from 23.97 fps to 24 fps is, in fact, the Right Thing to Do.

[dbone1026]

OK, just did a clean reinstall of W7 on my PC. Installed LAV splitter, I can't remember, how do I get into the options for LAV? I have the LAV Splitter set up on MPC HC as my external filter (preferred), just can't remember how I used to get into the LAV settings (i.e. to enable forced subs)

[SamuriHL]

LOL. Run a video, right click go to filters, and you should be able to get to the properties there.

[dbone1026]

Originally Posted by SamuriHL

LOL. Run a video, right click go to filters, and you should be able to get to the properties there.

yeah that is what i thought. i will have to try again, my head has been in the clouds of late!

[Andy o]

If upsampling makes audio not "lossless", then the audio is never lossless, because the DACs upsample anyway. You need to define what you're "losing" here. Lossless codecs remain without "loss" until the decoding part. Then people usually have no qualms in messing with it via EQ, DSP, room correction, etc. If you don't then your speakers and room will inevitably mess with it for the worse. I see ReClock as just one more of those processing steps. You're not losing sound quality, so does it matter? You're gaining smooth playback (if you have Intel, I would say ReClock is essential), it's not like you're just doing it for kicks.

[robpdotcom]

Originally Posted by dbone1026

yeah that is what i thought. i will have to try again, my head has been in the clouds of late!

You can also double click LAVSplitter in the MPC-HC external filters list.

[SamuriHL]

Originally Posted by robpdotcom

You can also double click LAVSplitter in the MPC-HC external filters list.

Hey quiet you! I was making him suffer a little.

[Snowknight26]

One of the most recent versions of the splitter has introduced a large delay when opening files (up to 5 seconds). I ran Process Monitor and saw that the splitter was reading the first 20MB or so of the file (in 4kB chunks!) before a graph was rendered (then started back from byte 0 after it was rendered). Switching to MPC-HC's internal (TS) splitter removed the delay.

The issue seems to be most prevalent with MPEG-TS files.
Process Monitor logs: http://stfcc.org/misc/mpchc_splitters.zip

[nevcairiel]

The underlying IO system should do read-buffering, therefor the read-size is irrelevant. And yes, on many files its actually required to read this much to find all the codec parameters. Stupid MPEG-TS doesn't define them properly.
If it takes 5s to read 20MB on your drive, you should get a faster drive.

Anyway, off to vacationing!

[Gleb Egorych]

Originally Posted by nevcairiel

"Lossy" codecs do not have a "proper" bitdepth, in a perfect world it would output floating point data as the ffmpeg decoder does.
Because the process is lossy, there won't be an "accurate" result, different decoders produce different results (most likely), and most decoders also decode to floating point .. ArcSoft is most likely just cutting/rounding to the bitdepth of the source material.

I understand that, bitdepth in DTS header is clearly source wavs bitdepth. Anyway generally you need to convert output bitdepth to integer, you can do in decoder or you may rely on windows mixer or ReClock. If it's done in a decoder and done properly it's OK for me. The only question is whether this is done properly in ArcSoft decoder

Originally Posted by nevcairiel

Unless you mean that its outputting 24-bit samples which only have 16bit filled? (Which i've seen before as well - but i may be able to influence that from the outside actually)

I don't know whether high-order bits are zero, I do know that 1.1.0.0 and 1.1.0.1 output different 24 bit streams and it looks suspicious in my opinion.

Originally Posted by nevcairiel

Regarding the "reference proper", over in the eac3to thread, some people compared output to the nero and sonic decoders, and apparently only the newer version of the ArcSoft decoder match the output of those two (which is otherwise identical) - on lossless DTS-HD MA only, of course, comparing bit-exactness of lossy is pointless.

AFAIK there is DTS-HD StreamPlayer which can be considered as a reference decoder. It's interesting how it decodes lossy and lossless tracks. Unfortunately I don't have it.

About eac3to: people in that thread discussed that MA 7.1 16/48 "strange setup" is not properly decoded with 1.1.0.0, and 1.1.0.8 is needed. Such track was found on "Black Hawk Down" blu-ray. I do have this blu-ray and I reproduced the problem, with 1.1.0.0 there is clearly audible noise. But it's eac3to problem, not ArcSoft! Noise appears only when decoding is done using eac3to, no noise when using 1.1.0.0 through LAV Audio. I dumped outputs of LAV Audio+1.1.0.0 and LAV Audio+1.1.0.8 and they were identical.

[madshi]

Originally Posted by Gleb Egorych

About eac3to: people in that thread discussed that MA 7.1 16/48 "strange setup" is not properly decoded with 1.1.0.0, and 1.1.0.8 is needed. Such track was found on "Black Hawk Down" blu-ray. I do have this blu-ray and I reproduced the problem, with 1.1.0.0 there is clearly audible noise. But it's eac3to problem, not ArcSoft! Noise appears only when decoding is done using eac3to, no noise when using 1.1.0.0 through LAV Audio. I dumped outputs of LAV Audio+1.1.0.0 and LAV Audio+1.1.0.8 and they were identical.

Can you upload a sample of the MA 7.1 strange setup track where that noise in audible with eac3to?