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Reclock's Resampler version 0.1.7 ICL11

very nice, thanks! g sounds better than f, f still has that thinner sound I don't like...one day you'll tell me what it's all about :D

and comparing b against g, I'd go for g :)

when she says "She was one of", the "Sh" sound is full of artifacts in 96KHz...they're still there in 192, but a lot less.

I've done a lot of comparisons, running this sentence in loop in KMP...basically:
*untouched: this sentence has a lot of "depth" like if Beatrix were talking in the same room where I stand(I use LT1364 op-amps that give uber-lively vocals)
*48KHz: the trebles are completely mushed and the sound is like a 192kbit MP3
*96Khz: it's better but still sibilating
*192KHz: it's much better, still sibilating(as any resampling increases distortion)...it loses some of the "room" ambience of the studio cabin where that sentence was recorded, but it's the best I've heard so far :agree:

still makes me crave for 47.952Hz w/ no resampling....too bad ogo chose libsamplerate over libavcodec(that seems to sound clearer and less sibilating) :eek:

and my board runs an AK4396 that does 128X oversampling up to 192Khz, but for instance the PCM1792A only does 64X above 96KHz...so YMMV.
 
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I reckon you guys need a thread just for this purpose.


It sure as hell aint bit-exact with no resampling ;)
 
Yes, I agree, this discussion is completelly off-topic.

leeperry, maybe it's better you to create a new thread for this discussion, and let's hope that James could remove some of the posts here and put them in the new thread...
 
leeperry said:
we've always been discussing resampling in this thread, I dunno what's up w/ you ppl.
Click to expand...
Just this little thing (thread title):
"Bit exact audio in Reclock without any resampling";)
Please create a new thread, about oversampling, that will also attract more people. Who would think that oversampling is being discussed in a thread about reclock without any resampling?:D
Let's hope James can move the posts from here to there...
 
It seems the new thread does not appear, so, after all these posts, one more will not make it worse...

Here is a decription of the several resamplers:
resampler2x_b: same as regular resampler, but resamples to 2x the source sampling rate
resampler2x_c: resamples to 2x the source sampling rate, but the code added to decrease the volume in case of overload was removed
resampler2x_d: same as 2x_b but with a lower volume, to guaranty no overloads
resampler2x_e: same as 2x_c but with a lower volume, to guaranty no overloads
resampler2x_f: same as regular resampler, but resamples to 4x the source sampling rate
resampler2x_g: resamples to 4x the source sampling rate, but the code added to decrease the volume in case of overload was removed

So, leeperry, is strange that at 2x you prefer the 2x_b compared to the 2x_c and then at 4x you prefer the 4x_g to the 4x_f.

What was my idea?
When in a digital signal the maximum available value is reached, we cannot know if we are in the presence of an overload or not. It could simply be the highest value of the signal, or it could also be a value below that. In the first case, there is no problem, no distortion is added, but in the second case, don't.
When we oversample a signal like in the second case, some of the new samples could be outside of the valid range. There, if we disable the code that corrects that by lowering the volume proportionally, the distortion would be higher, and this would affect particularlly the higher frequencies. So, when leeperry first said that he prefered the 2x_b, it was possible that it could be caused by an overload. Later, when he prefered the 4x_g, it showed that cannot be an overload problem (it was a bit strange that in a dialog existed any overload, but who knows?...)
In any case, I have found one more good reason to add the possibility of oversampling to reclock: help to avoid the overloads that could exist in the original signal.;)
If you want to read more about this subject, see here, or google about "loudness war".

I hope I was clear enough, but I only have learned this in the past two days, so it's all still very new to me...
 
yesgrey3 said:
It seems the new thread does not appear, so, after all these posts, one more will not make it worse...

Here is a decription of the several resamplers:
resampler2x_b: same as regular resampler, but resamples to 2x the source sampling rate
resampler2x_c: resamples to 2x the source sampling rate, but the code added to decrease the volume in case of overload was removed
resampler2x_d: same as 2x_b but with a lower volume, to guaranty no overloads
resampler2x_e: same as 2x_c but with a lower volume, to guaranty no overloads
resampler2x_f: same as regular resampler, but resamples to 4x the source sampling rate
resampler2x_g: resamples to 4x the source sampling rate, but the code added to decrease the volume in case of overload was removed

So, leeperry, is strange that at 2x you prefer the 2x_b compared to the 2x_c and then at 4x you prefer the 4x_g to the 4x_f.

What was my idea?
When in a digital signal the maximum available value is reached, we cannot know if we are in the presence of an overload or not. It could simply be the highest value of the signal, or it could also be a value below that. In the first case, there is no problem, no distortion is added, but in the second case, don't.
When we oversample a signal like in the second case, some of the new samples could be outside of the valid range. There, if we disable the code that corrects that by lowering the volume proportionally, the distortion would be higher, and this would affect particularlly the higher frequencies. So, when leeperry first said that he prefered the 2x_b, it was possible that it could be caused by an overload. Later, when he prefered the 4x_g, it showed that cannot be an overload problem (it was a bit strange that in a dialog existed any overload, but who knows?...)
In any case, I have found one more good reason to add the possibility of oversampling to reclock: help to avoid the overloads that could exist in the original signal.;)
If you want to read more about this subject, see here, or google about "loudness war".

I hope I was clear enough, but I only have learned this in the past two days, so it's all still very new to me...
Click to expand...
well, my comparisons were honest...but placebo is very real.

anyway I've just watched a 23.976fps@96Hz movie w/ the 4X "g" oversampler and mVR....well, let's just say that my whining might have come to an end :D

lossy DTS sounds bad(avoid libdts, it's terrible), resampling kills the SQ...but I think this is as good as it's gonna get, as you can't quite get smooth playback at the right speed w/ untouched audio(on a board that runs an AK4396 w/ swappable op-amps ;))

mVR is amazingly smooth in 96Hz on my CRT in KMP, not a single dropped frame, the stereo phase was perfect, and the trebles very clear..and your resampling in 192KHz in KS was flawless too, no glitches whatsoever!

Kazuya said he had some random dropped frames in 50Hz so I'll try some movies in 48Hz on my DLP pj in the next coming days....but in 96Hz mVR 0.11 is totally flawless, such a relief to drop HR and its stinky jitter(that completely put me off watching movies :mad:)

BTW, The Mothman Prophecies is a great little movie.

PS: when James will have added oversampling, maybe I can try again...
 
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leeperry said:
well, my comparisons were honest...but placebo is very real.
Click to expand...
I never doubted that, but yes, it could be a little placebo effect.;)

leeperry said:
avoid libdts, it's terrible
Click to expand...
So, what do you suggest? And for AC3?

leeperry said:
but I think this is as good as it's gonna get, as you can't quite get smooth playback at the right speed w/ untouched audio(on a board that runs an AK4396 w/ swappable op-amps ;))
Click to expand...
You can always buy a RME FF800, open it, and swap the op-amps.:D
 
well, I like the "g" resampler in 192KHz, the phase and the trebles are as good as it's gonna get :)

for AC3, I like liba52 in ffdshow...it decodes in 32float and the AC3 specs are well known.

the problem w/ libdts is that it's been reversed engineered and never updated due to DTS legal threats I think? it sounds terrible....the dialogs are hissy/distorted, ugly!

libavcodec sounds a heck better, too bad it's 16int only(DTS is 18/19bit natively)...I personally got a hook for Sonic 4.2, but you can't completely disable DRC.

I'd advise AC3Filter(in 32float) if you don't need to make autoload profiles in ffdshow, as it's broken at this point...so I can't use it :bang:

hehehe, it sounds a bit pricey to open a brand new FF800 to swap the op-amps, I have the feeling that these "pro" brands ask for a premium because the market is willing to pay extra for lotsa exotic I/O and very powerful routing in the drivers...but the hardware part is nothing to write home about..most of the RME cards use JRC4580, which is very nice for driving headphones...but lacks the transparency of the LT1364 on vocals, or the audiophile mids/natural soundstage of the OPA2132 ;)

I will prolly try to replace my OPA2132P buffer by a Burson op-amp, these are just out of this world I was told..
 
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leeperry said:
I'd advise AC3Filter if you don't need to make autoload profiles in ffdshow
Click to expand...
I don't, so I will try AC3Filter... Is it as good as liba52 for AC3?

leeperry said:
but the hardware part is nothing to write home about.
Click to expand...
That's not true. Maybe the opamps are not very high quality, but the dacs are and the clock circuits too. They have jitter < 1ns, and this should be way more significant for the sound quality than the opamps... ;)
Since my FF400 is already out of warranty, maybe one day I will be crazy enough to open it and swap the opamps.:D
 
leeperry said:
the problem w/ DTS is that they are not.
Click to expand...
Have you ever tryed converting a dts soundtrack to flac using eac3to? That way you can decode using official decoders.;)
leeperry said:
yes ok, that's true...they have good clocks indeed, very nice for S/PDIF or AES/EBU transmissions.
Click to expand...
And also for the digital to analog conversion... The clock is one of the most important things for you to be able to recover the original analog sound. The samples need to be in the right place in time, not before, nor after.;)
 
well, I had the problem w/ DTS 96/24...the Arcsoft decoder can decode it in a DS player, but it always reverts to stereo output by default...it's most likely embedded in the IPersistStream interface, but no media players cares about this :rolleyes:

yeah true! but it all depends on the DSP jitter itself, the Asus STX>ST have a different clock...but their crappy CMI8788 DSP has a 750ps jitter spec, so any clock improvement is totally pointless :doh:
 
DTS is so '90s.

DTS-HD MA is where it is now ;)


Who cares about subtle (inaudible...) differences in a lossy compression method....
 
yesgrey3 said:
I would say more like an ear exam...:)
Click to expand...
hehe, my otologist hides when he sees me :D
Mark_A_W said:
Who cares about subtle (inaudible...) differences in a lossy compression method....
Click to expand...
very audible yes, libdts in ffdshow is unbearable on dialogs :disagree:
yesgrey3 said:
I've looked into it but AC3Filter is also based on libdts.
Click to expand...
O RLY? I couldn't find the info...got a link? anyway AC3Filter is buggy for DTS apparently : http://forum.doom9.org/showpost.php?p=1334298&postcount=17

anyway, it's funny how everything has a role to play in PC audio...I just ditched my BeQuiet 500W for a Corsair 400W, because it gave the lowest ripple measurements on both a french specialized site and anandtech...and the SQ is just miles better!

http://translate.google.com/transla..._CX_400_Watts.html&sl=fr&tl=en&hl=en&ie=UTF-8

wider soundstage, clearer trebles, better stereo imaging...we're basically listening to the 220V current, so better transform it to 12/5/3.3V w/ the lowest ripple.

some guys on head-fi have made stabilized PSU's for their Asus STX, they get 0.1mV ripple....man, the SQ must really be as good as they say...too bad Asus is the only company making soundcards w/ a separate molex plug, and that their C-Media drivers suck monkeys ***** :/
 
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Mark_A_W said:
DTS is so '90s.
DTS-HD MA is where it is now ;)
Click to expand...
Yes, but sometimes only dts soundtracks are available...
Maybe I will decode them to flac with eac3to.;)

leeperry said:
O RLY? I couldn't find the info...got a link?
Click to expand...
No need. Just go to the AC3Filter About tab and see the "Thanks to" group.

leeperry said:
anyway, it's funny how everything has a role to play in PC audio...
Click to expand...
Another point to my FF400, with its own power supply...:)
 
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