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31st October 2011, 00:09 | #10521 | Link | ||||
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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:
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. Quote:
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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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31st October 2011, 00:13 | #10523 | Link |
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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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31st October 2011, 00:58 | #10524 | Link | |
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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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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 |
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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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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31st October 2011, 01:36 | #10529 | Link | |
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Might be tempted to try this, right now my video is also going through my AVR along with the audio.
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31st October 2011, 09:40 | #10531 | Link | |
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one question though, copy the dtsdecoderdll.dll to where? |
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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... |
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. |
31st October 2011, 11:34 | #10535 | Link | |
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31st October 2011, 11:38 | #10536 | Link |
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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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31st October 2011, 11:56 | #10537 | 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.
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31st October 2011, 12:30 | #10538 | Link | |
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31st October 2011, 12:59 | #10540 | Link | |
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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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Tags |
direct compute, dithering, error diffusion, madvr, ngu, nnedi3, quality, renderer, scaling, uhd upscaling, upsampling |
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