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Old 31st October 2011, 00:09   #10521  |  Link
6233638
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Quote:
Originally Posted by THX-UltraII View Post
I want to try ReClock. Can someone post correct settings for LAV Audio decoder and ReClock settings (all three tabs) to get audio decoding done by HTPC?
I don't remember there being any configuration necessary with LAV Audio other than dropping your Arcsoft "dtsdecoderdll.dll" in the folder along with it.

These are my ReClock settings:


(note: you're probably best not to touch the pre-buffer and latency options, it requires a much faster CPU to avoid audio problems)

If you can upsample the audio, you should turn the sampling rate/format up. (with WASAPI output, you will receive an error if it is not supported) Don't go above 96kHz.

Make sure it is set as the default audio renderer in your player.
The first time you load a video with ReClock, you should also load up its preferences from the filter list, make sure speed is set to "Auto (best)", enable PAL SpeedDown, and Lock the settings.

Under no circumstances should you enable V-Sync correction or "Slave reference clock to audio".

Quote:
Originally Posted by pacemaker1000 View Post
correct me if i am wrong but the only way Reclock can work(and i used it for years before i got a 24 capable display) is to have the audio in decoded form so it can sync it to the video. If you want your audio sent out untouched in bitstream form then this is a no go?
There is no difference between bitstreamed HD Audio, and HD Audio decoded by LAV Filters.

It all ends up as LPCM either way, it's just that you have your PC doing the conversion (where you know it is being handled correctly) rather than your amplifier. (which may or may not)

If you then have reclock processing things, it will not be "bit-exact" as it has to do some manipulation of the audio stream to work its magic.

This should be imperceptible though, especially if you are upsampling the audio, and better than the alternative of watching the entire film at the wrong speed & pitch for the sake of being "bit-perfect."

If you believe that you can hear a degradation of audio quality when using ReClock, then you must hear a much bigger change from playing back films at the correct speed.
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Originally Posted by pacemaker1000 View Post
ideally we would all set our outputs to 24Hz and speed up the 23 stuff and slow down the 25 stuff
Yes, this is what ReClock does.

Quote:
Originally Posted by Qaq View Post
Music? Hmmm, I can only remember one music 24p video in my collection - some Bon Jovi live concert. All the others are 29i or 25i and they are already 60p and 50p after de-interlacing.
I actually meant music on the soundtrack of a film, rather than a concert Blu-ray or whatever.

Quote:
Originally Posted by Qaq View Post
IMO, 24 is just not enough for smoothness, 24/60 has pulldown judder and 24>25/50 is ... perfect.
I agree that 24p is not smooth, but the better fix is motion interpolation in my opinion. Sony's motionflow is the best in my opinion, as it generally keeps natural film-like motion, rather than looking "sped-up" but makes panning easier on the eyes.
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Old 31st October 2011, 00:13   #10522  |  Link
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Quote:
Originally Posted by pacemaker1000 View Post
...setting bitstream out renders it useless....
Well, it depends. If you have Reclock set to "accept bitstream", then:

Reclock will drop/repeat bitstream packets to keep sync if "disable media speed correction" is not checked.

got it?
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Old 31st October 2011, 00:13   #10523  |  Link
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Originally Posted by pacemaker1000 View Post
so you would have to decode the audio from, say dts hd, input it to reclock so it could sync it with the video clock then re-encode it back to dts hd or possible lpcm and send it out over hdmi without loosing quality
Um, HUH?? Decoding any encoded audio gives you LPCM. That includes AC3/DD+/TrueHD/DTS/DTS-HD MA/DTS Express/etc. Once it's in LPCM, ReClock is able to work with it by manipulating the playback rate. It's not "re-encoding it" or anything like that. More like resampling it. There is *NO* quality lost by decoding. Resampling is arguable, but, I'm quite certain you'd not be able to tell the difference in a double blind test. Sending the decoded LPCM out to the receiver over HDMI gives you the *EXACT* same quality as the encoded audio. After all, you have to decode it somewhere. Bitstream vs bit exact....there's no difference other than where the audio is decoded.
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Old 31st October 2011, 00:58   #10524  |  Link
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I still can't believe I returned an onkyo 805 for this thing, everyone except for me loves the "HK sound"
I am an unabashed HK hater after dealing with a 254 for the 8 most frustrating hours of my life, but those details are at the HK thread at AVS.

I will say though, as a general rule, any decent piece of (especially digital) audio electronics should be audibly transparent (and that's what it should strive for, unless it's an EQ or something like that). So when one hears "the XXXX sound" by audiophiles, it's either false (which in this case I think it is), or it's true and the equipment is not transparent, and thus lower quality.

Of curse this excepts speakers and mics and stuff like that, don't know if those can be considered "electronics".
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Last edited by Andy o; 31st October 2011 at 01:01.
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Old 31st October 2011, 00:59   #10525  |  Link
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i love to be proven wrong when it means i can go back to using Reclock

the only reason i changed my ati 4650 was to get hd audio out to my new hdmi amp. si i may have wasted my time as i think the old one could do just that be it via a dvi dongle

so if i remeber i use the setting in TMT to output lpcm?

in mpc do i use lav splitter?

thanks guys always learning
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Old 31st October 2011, 01:05   #10526  |  Link
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You use LAV Audio. With LAV Splitter ideally, yes. In the LAV Filters directory, copy the dtsdecoderdll.dll from your TMT install. In the LAV Audio properties, have it set to decode all audio. Use ReClock as the audio renderer. Done. You did not need to buy a new card just for audio. The software decoders have been highly stable for quite a while now. TMT's is the best for DTS formats. LAV Audio works great for the Dolby codecs. Bitstreaming isn't necessary.

(To clarify when I say LAV Audio is the best for Dolby and TMT is the best for DTS, LAV Audio is using TMT's decoder for DTS and its own decoder for Dolby in that setup)
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Old 31st October 2011, 01:15   #10527  |  Link
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If it's just for audio, AMD has the edge cause of the lack of the "silent stream bug". No HW deinterlacing with madVR though, but madshi said we should wait a bit before buying an NV card just for that

If we could do decent deinterlacing with AMD, I'd think for madVR it would be no contest. AMD doesn't have the glitches problems, and continuous, instantaneous audio are a win. Plus, generally more accurate refresh rates, for most.
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Old 31st October 2011, 01:19   #10528  |  Link
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Yea, I can't wait to see what madshi and/or Nev have in store for us on deinterlacing.
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Old 31st October 2011, 01:36   #10529  |  Link
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Originally Posted by jmelan View Post
yeah works fine with my AVR (hdmi 1.3) and HDTV (hdmi 1.4) to allow for bitstreaming audio, unaltered video, and frame packed 3D playback. i am using a sony HDTV (xbr hx909 series), not a computer monitor -not sure if that would make a difference

i have tried running the cables both ways, does not make a difference, both work, right now i have the DVI>HDMI cable connected to the TV and the HDMI cable to the AVR.

check out this chart posted by renethx over at avsforum, the relevant line is "HD audio via HDMI 1.3 AVR + Frame packing (HDMI + DVI)"

only works with current nvidia and intel at the moment i believe
Okay, so do you extend the desktops then or?

Might be tempted to try this, right now my video is also going through my AVR along with the audio.
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Old 31st October 2011, 05:41   #10530  |  Link
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yes, the desktop is extended, i believe that is the default
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Old 31st October 2011, 09:40   #10531  |  Link
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Quote:
Originally Posted by SamuriHL View Post
You use LAV Audio. With LAV Splitter ideally, yes. In the LAV Filters directory, copy the dtsdecoderdll.dll from your TMT install. In the LAV Audio properties, have it set to decode all audio. Use ReClock as the audio renderer. Done. You did not need to buy a new card just for audio. The software decoders have been highly stable for quite a while now. TMT's is the best for DTS formats. LAV Audio works great for the Dolby codecs. Bitstreaming isn't necessary.

(To clarify when I say LAV Audio is the best for Dolby and TMT is the best for DTS, LAV Audio is using TMT's decoder for DTS and its own decoder for Dolby in that setup)
thanks a lot

one question though, copy the dtsdecoderdll.dll to where?
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Old 31st October 2011, 10:33   #10532  |  Link
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Originally Posted by pacemaker1000 View Post
thanks a lot

one question though, copy the dtsdecoderdll.dll to where?
To where you put LAVAudio.ax in.
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Old 31st October 2011, 11:29   #10533  |  Link
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Maybe stupid question, but I throw it anyway
So, is lavaudio better than ffdshow? Right now, I am using ffdshow (liba52 and libdts) because I can use the "night mode" for watching movies at night (you got it ). At daytime I just donīt check the "volume" tab, and I get normal sound... So, is LAVaudio better for that "normal sound"? Or is libavcodec better than liba52 and libdts?

And one more question... Is it better to have 32bit float or 16 LPCM? I am using ReClock and it can work on both streams...
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Old 31st October 2011, 11:33   #10534  |  Link
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I m still wondering which in madVR gives me the correct information. Here are my results:

NVIDIA 1080p custom 23Hz @23,976
- madVR reports 23,97495 in the top of the OSD (so a deviation of 0.00105).
- one frame dropped every 6 minutes
- 22 dropped frames after 120 minutes of playback

The above all seems 'correct', but now

NVIDIA 1080p custom 23Hz @23,978
- madVR reports 23,97795 in the top of the OSD (so a deviation of 0.00195).
- one frame dropped every 18 minutes
- 2 frames dropped after 120 minutes of playback

The above just is not right. With this I mean that some of the information is incorrect. First off all the deviation is BIGGER then with the 23,976nvidia setting but the 'one frame dropped every' information is 'better' then the 23,976nvidia setting.
And the this that is the most strange to me is that I only got 2! dropped frames so this assumes it is almost perfect which does not correspond with the other information.
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Old 31st October 2011, 11:34   #10535  |  Link
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Originally Posted by SamuelMaki View Post
Maybe stupid question, but I throw it anyway
So, is lavaudio better than ffdshow? Right now, I am using ffdshow (liba52 and libdts) because I can use the "night mode" for watching movies at night (you got it ). At daytime I just donīt check the "volume" tab, and I get normal sound... So, is LAVaudio better for that "normal sound"? Or is libavcodec better than liba52 and libdts?

And one more question... Is it better to have 32bit float or 16 LPCM? I am using ReClock and it can work on both streams...
ditch ffsshow and use LAV Audio decoder. ffdshow is old, LAV is the new ffdshow
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Old 31st October 2011, 11:38   #10536  |  Link
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Originally Posted by 6233638 View Post
If you can upsample the audio, you should turn the sampling rate/format up. (with WASAPI output, you will receive an error if it is not supported) Don't go above 96kHz.
I see that this setting is used by most people but why? Isn t it so that this way you ALTER the sound? Why not leave it on same as source? And if you can substantiate why to use it, could you tell me when to use 48 and when 96? (I own a NVIDIA GTX 460)
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Old 31st October 2011, 11:56   #10537  |  Link
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Originally Posted by THX-UltraII View Post
I see that this setting is used by most people but why? Isn t it so that this way you ALTER the sound? Why not leave it on same as source? And if you can substantiate why to use it, could you tell me when to use 48 and when 96? (I own a NVIDIA GTX 460)
Somewhere in the Reclock forum (just before the sampling rate option was implemented in Reclock) it was shown that downsampling from 48 kHz to ~46 kHz (for PAL speeddown) is worse for the sound quality than upsampling from 48 kHz to 96 kHz (or even more). That's why the upsampling options were implemented.
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Old 31st October 2011, 12:30   #10538  |  Link
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Somewhere in the Reclock forum (just before the sampling rate option was implemented in Reclock) it was shown that downsampling from 48 kHz to ~46 kHz (for PAL speeddown) is worse for the sound quality than upsampling from 48 kHz to 96 kHz (or even more). That's why the upsampling options were implemented.
So I just pick the highest possible kHz? Any idea what this would be for my GTX460?
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Old 31st October 2011, 12:33   #10539  |  Link
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These are my ReClock settings:
what does the option 'set matching speaker configuration' do?
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Old 31st October 2011, 12:59   #10540  |  Link
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So I just pick the highest possible kHz? Any idea what this would be for my GTX460?
You could try to find the highest working option. WASAPI output gives an error message if the current one isn't possible at all, but there might also be other problems with too high values. So if you have any weird problems (even something like crashes) try lowering the value.

But I don't think that higher values have any advantage. Just make sure that the value shown at "Clocks correction/Audio clock" is never lower than the one shown at "Media Infos/Audio stream".
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