can ffdshow audio do this dream resample to 192000hz ?

[vola]

if ffdshow can't do that
There are other software?
with 192000hz
:goodpost::goodpost::goodpost::goodpost::goodpost::goodpost::goodpost::goodpost::goodpost::goodpost:
hi

i love ffdshow audio

This improves sound quality
For example, from 48 000 HZ
TO 96 000 HZ

http://img826.imageshack.us/img826/1584/52018169.png

Many sound cards today offer 192000HZ
Why not give the possibility to make UPSCALE TO
192000HZ with audio ?
Through ffdshow audio

ALC 892 supports- 192 000 HZ also
all asus card and more

[Gser]

Hahahahahaaa it doesn't improve quality unless the source is 192000Hz.

[vola]

Hahahahahaaa it doesn't improve quality unless the source is 192000Hz.


How do you say to me it's not good?
Have you tried 192 000 with FFDSHOW?
have you tried 96000 HZ ?

You know how to tune this at all ?

Do you have any home theater system?
Or you use headphones

[Midzuki]

If you do have an audio source @ 192kHz,
then simply DON'T USE ffdshow. :)

[TinTime]

If you've got a burning desire to resample then you could give Reclock a go.

[vola]

If you do have an audio source @ 192kHz,
then simply DON'T USE ffdshow. :)

Are you serious?
The whole idea is sound analysis on 192 000 ON LINE
It does not transform as
WAV TO MP3

This deep sound analysis

[pandy]

simply put 192000 instead 96000 - ffdshow support various sampling rates...

i don't know why You pushing for 192000 - this make no sense for me but it is up to You...

My fault - ffdshow will flag output as 192ksps but not resample the data - this mean that audio will be played twice slower - should be raised as an issue for ffdshow developers

or i was wrong with this and issue is in driver that can't work well with 192 (output from ffdshow is flagged correctly)

[Ghitulescu]

As pandy said, you won't get any quality improvement, quite the opposite (depending on your hearing and the used algorithm).
Candy camera, Crazy animals and so on won't look better just because they are sent HDTV - the same is for audio.

[ramicio]

I have ffdshow resample to 96khz, but I did this with the intention of better quality when it downmixes 5.1 stuff to stereo. I just wish it supported 192khz.

[pandy]

it can be better if Your DAC is better with 96 or 192ksps - but most of the DAC's are bit worse in fact - they are (usually) multibit Delta Sigma converters - they internally resample audio by some constant factor to achieve final conversion frequency - usually few maybe up to few tens MHz - shifting resampling from final stage of DAC to source usually means that real performance can be slightly worse - but of course everything depend from DAC - but don't expect miracles - no data means no data - You can put funny filters, even try to replicate bandwidth - still there is no good data at the input of DAC - maybe if You are "golden ear", you have source with 24 bits, extremely clean power supply after all this maybe You can hear difference but... in real life (ie PC case) You may expect more problems form various noise sources, more problems from jitter or even more problems from software than You can gain for pushing from 48 to 96 or even 192ksps.

24 bits, 48 ksps are more than in enough in normal life - for extremely situation 24 bit, 96 ksps is more than You can hear (especially if a good noise shaping is used) - 192 ksps is interesting but as a substitute to software defined radio or some lab signal generation than for normal listening from PC.

[ramicio]

I find it better to mix audio with a higher sample rate. If ffdshow was nicer one could resample a 5.1 track, mix it to stereo, and then sample it back down. This is going to turn into a damn hydrogenaudio thread i think. 24/48 is way better than CD, and I told them that, and they lashed out at me, so screw that site. Since when is it best to just have the bare minimum, what is so wrong with going better?

[Ghitulescu]

it can be better if Your DAC is better with 96 or 192ksps - but most of the DAC's are bit worse in fact - they are (usually) multibit Delta Sigma converters - they internally resample audio by some constant factor to achieve final conversion frequency - usually few maybe up to few tens MHz - shifting resampling from final stage of DAC to source usually means that real performance can be slightly worse - but of course everything depend from DAC - but don't expect miracles - no data means no data - You can put funny filters, even try to replicate bandwidth - still there is no good data at the input of DAC - maybe if You are "golden ear", you have source with 24 bits, extremely clean power supply after all this maybe You can hear difference but... in real life (ie PC case) You may expect more problems form various noise sources, more problems from jitter or even more problems from software than You can gain for pushing from 48 to 96 or even 192ksps.
Could you direct me to a site, article, spec sheet or similar that describe this negative effect? I was convinced about the opposite.
I find it better to mix audio with a higher sample rate. If ffdshow was nicer one could resample a 5.1 track, mix it to stereo, and then sample it back down. This is going to turn into a damn hydrogenaudio thread i think. 24/48 is way better than CD, and I told them that, and they lashed out at me, so screw that site. Since when is it best to just have the bare minimum, what is so wrong with going better?
One cannot have a better quality by using a better format when the, in this case, audio definition doesn't allow it? A telephone conversation won't gain anything from being converted to 192/24.
Oh, yes, when the material already lies in 192/24 or 284/48 or whatever, well, now we're talking something, but resampling it from say 48 to 192, apart from dithering noise, is useless.
Most audio HW that work internally with 24b actually take the analog input, not the digital one. If they take a digitla one, say 16b, usually they use 20b, 4+4, to avoid the clipping top and bottom during sound manipulation. Or 24b (6+6). In the end the signal is still 16b, and even if the output is 20 or 24b, the resolution of the sound is still 16b.

[ramicio]

You really think music is made entirely in 16/44.1? I doubt it. It's mixed in a higher samplerate and bit depth and THEN converted to RedBook. It is well known that higher sample rates (24/192) are beneficial for mixing sounds and music, even if each channel being mixed is not as high of quality (24/48) as the project.

[MatLz]

Don't kick my *ss if I say nonsensicalhugemistake.
Upsampling the audio samplerate can be done by copying each (or interleaving) audio frame ?
That won't give more "quality", right ?
But by interpoling, let say the avg between 2 consecutive frames....can that give better "quality" ?
Is that a kind "audio dithering" ?
I repeat, don't kick the newbie ! :D

[Midzuki]

Upsampling the audio samplerate can be done by copying each (or interleaving) audio frame ?

Do you mean, let's say, going from 48kHz to 96kHz by simply duplicating every sample? Yes, it's possible, and it's lossless too. :devil:

That won't give more "quality", right ?

Of course it will not.

But by interpoling, let say the avg between 2 consecutive frames....can that give better "quality" ?

"Better" than what ??? :)

Is that a kind "audio dithering" ?

No.

[ramicio]

For instance, if you have a wav file ripped from a CD and upconvert it to 24/192, there will be no benefit whatsoever, and probably even loss because when you dither the bits there is minor quantization noise. I use minor because you will never hear it. The point I am trying to make is, say you have a DTS-HD Master extraced to wavs, or a 5.1 DVD-A track. What I do is open the 6 mono tracks (24/48) in Goldwave, and mix them into a 24/192 file, and then downsample to 96 khz. I suppose even 24/48 after all is mixed is fine, but I have the hard drive space, so I can afford the extra room it takes up and I don't care about ABX.

[Ghitulescu]

You really think music is made entirely in 16/44.1? I doubt it. It's mixed in a higher samplerate and bit depth and THEN converted to RedBook. It is well known that higher sample rates (24/192) are beneficial for mixing sounds and music, even if each channel being mixed is not as high of quality (24/48) as the project.
The music is analogue captured. It may be analogue processed or digitally. It also might be analogue connected to digitally processing units. Finally it's the final mix that is captured/downconverted to 44.1/16 for CD distribution. Entirelly in the digital realm is possible only with Techno/house/etc. music.

Do you mean, let's say, going from 48kHz to 96kHz by simply duplicating every sample? Yes, it's possible, and it's lossless too. :devil:
"Line doubling" is extremely bad for music. Interpolation would be better.

[Mug Funky]

- mixing at a higher samplerate makes no sense. the discrete channels are all sample-aligned, so why would there be an improvement to adding them together in a higher samplerate?

- processing at a higher samplerate often has advantages when distortion is involved - the distortion is typically broadband, meaning it is spread more-or-less evenly throughout the passband. when you upsample, filter, then downsample, the portion of the distortion that is above the output nyquist is filtered out, giving up to half of the noise from distortion you'd otherwise get.

but bit-depth is what you really want if all you're doing is a fold-down from 6 channels to 2 or whatever.

the only gotcha here is if the DAC for whatever reason works better at a given sample-rate. this is usually because of cheap brickwall filters. in this case upsampling with a good digital filter and relying less on a bad analog filter in the DAC makes sense.

[Blue_MiSfit]

^Nailed it

Derek

[vola]

Why not try to build a resample to 192000hz?

WHY NOT BE Practice ?

Is this a technical problem?
Build In FFDSHOW

[MatLz]

"Better" than what ??? :)hum....well....more samples means closer to an analog wave form, right ?
So if we interleave an interpolated sample....
No ?

But I'm pretty sure 192KHz is overkillplacebo.
96KHz should be enough.

[vola]

hum....well....more samples means closer to an analog wave form, right ?
So if we interleave an interpolated sample....
No ?

But I'm pretty sure 192KHz is overkillplacebo.
96KHz should be enough.

With all due respect

Why not try?

Who knows maybe might add?

Is it a large economic investment to try?

Can be on a hit ? OR NOT



I PLAY WITH

Amplifier +7.1+AMD Processor
I hear this series and movies


http://www.fastup.co.il/v.php?file=21197462.png

http://www.fastup.co.il/v.php?file=36635222.png

http://www.fastup.co.il/v.php?file=52606778.png

http://www.fastup.co.il/v.php?file=74139141.png

http://www.fastup.co.il/v.php?file=25586837.png

http://www.fastup.co.il/v.php?file=77572.png

http://www.fastup.co.il/v.php?file=5083901.png

http://www.fastup.co.il/v.php?file=71323725.png


and Sound card 889 or 892 set to 5.1

http://img153.imageshack.us/img153/5660/50758175.png

[Groucho2004]

- mixing at a higher samplerate makes no sense. the discrete channels are all sample-aligned, so why would there be an improvement to adding them together in a higher samplerate?

- processing at a higher samplerate often has advantages when distortion is involved - the distortion is typically broadband, meaning it is spread more-or-less evenly throughout the passband. when you upsample, filter, then downsample, the portion of the distortion that is above the output nyquist is filtered out, giving up to half of the noise from distortion you'd otherwise get.

but bit-depth is what you really want if all you're doing is a fold-down from 6 channels to 2 or whatever.

the only gotcha here is if the DAC for whatever reason works better at a given sample-rate. this is usually because of cheap brickwall filters. in this case upsampling with a good digital filter and relying less on a bad analog filter in the DAC makes sense.

Finally someone with a clue who doesn't recite collected nonsense from various forums.

[Ghitulescu]

No consumer is able to obtain 96/24 sound that is really 96/24, not to mention 192/24.
The consumer gear is limited to some 80dB of dynamics (the now defunct DAT recorders), today even less. Using 24b to store what can be stored in 10b is generally overkill.
The same comes to bandwidth: 96kHz allows up to 30kHz (I know what the theory says :)), which no consumer gear is able to capture.

So why 192/24 and stuff? Because in the studios the signal is several times processed and reprocessed and transferred and so on. Both the bandwidth and the dynamic assures that the signal has enough space of manoeuvre against eg clipping or noise floor. Even in the digital realm, which is not yet THE standard. It's only after the last mixing, when the signal finally becomes a distribution format (CD, DVD, SACD, DVD-A, BD).

Besides, even the reproduction of 192/24 is more a placebo as a reality. 120 dB is the entire spectrum a human ear can feel, from the sound of the "growing grass" to the painful noise of a turbojet. Anyway, 24b should suffice to describe the auditive nature that surrounds us. The placebo consists in each of the following 3 items, separately and in combination:
listening a concert of 144 dB dynamic (the max of 24b) needs the written aproval of the townhall :)
144dB means a voltage difference of 1000000000000000000000000x from the maximal value to the lowest one. Take the SACD which is analogue driven over -10dB wires (500mV): how low must be the lowest voltage :)?
now, can anyone imagine how strong are the intrinsec noise currents compared to the smallest ones (see above)?

So no fully 24b capturing range, no 24b reproduction range. And of course an almost useless "finesse" in capturing and reproducing the 10-16000Hz hearing range.

[pandy]

Could you direct me to a site, article, spec sheet or similar that describe this negative effect? I was convinced about the opposite.


I need to search my datasheet archive - from some time IC manufacturers like Analog Devices or TI limit amount useful data in their datasheets...

but rule is quite simple - cheap DAC mean Delta Sigma - Delta Sigma is limited in bandwidth - internally they use something like oversampling from 32x up to the 256 (maybe higher) ratios - more oversampled data - better processing gain it is clear - so if
feed more data (higher sampling) i reduce oversampling and thus processing gain - restriction is quite obvious - of course new technology still push parameters up - frequency of work is higher thus gain can remain on same figure but as always there is balance between technology and results that can be achieved.


One cannot have a better quality by using a better format when the, in this case, audio definition doesn't allow it? A telephone conversation won't gain anything from being converted to 192/24.
Oh, yes, when the material already lies in 192/24 or 284/48 or whatever, well, now we're talking something, but resampling it from say 48 to 192, apart from dithering noise, is useless.
Most audio HW that work internally with 24b actually take the analog input, not the digital one. If they take a digitla one, say 16b, usually they use 20b, 4+4, to avoid the clipping top and bottom during sound manipulation. Or 24b (6+6). In the end the signal is still 16b, and even if the output is 20 or 24b, the resolution of the sound is still 16b.

Yes, so there is no sense to push lossy compressed and limited in bandwidth signal by upsampler only to artificially increase sampling ratio.

[pandy]

You really think music is made entirely in 16/44.1? I doubt it. It's mixed in a higher samplerate and bit depth and THEN converted to RedBook. It is well known that higher sample rates (24/192) are beneficial for mixing sounds and music, even if each channel being mixed is not as high of quality (24/48) as the project.

but of course but during conversion from such high sampling always low pass filtering and noise shaping is used - after such processing there is no data over stop band of the filter.

[pandy]

No consumer is able to obtain 96/24 sound that is really 96/24, not to mention 192/24.
The consumer gear is limited to some 80dB of dynamics (the now defunct DAT recorders), today even less. Using 24b to store what can be stored in 10b is generally overkill.
The same comes to bandwidth: 96kHz allows up to 30kHz (I know what the theory says :)), which no consumer gear is able to capture.


10bits is only 60dB dynamics, for 80dB 14 bits is required
In real life natural spectrum of many instruments quickly going dow - energy over 25khz can be really low and unhearable.


So why 192/24 and stuff? Because in the studios the signal is several times processed and reprocessed and transferred and so on. Both the bandwidth and the dynamic assures that the signal has enough space of manoeuvre against eg clipping or noise floor. Even in the digital realm, which is not yet THE standard. It's only after the last mixing, when the signal finally becomes a distribution format (CD, DVD, SACD, DVD-A, BD).

Besides, even the reproduction of 192/24 is more a placebo as a reality. 120 dB is the entire spectrum a human ear can feel, from the sound of the "growing grass" to the painful noise of a turbojet. Anyway, 24b should suffice to describe the auditive nature that surrounds us. The placebo consists in each of the following 3 items, separately and in combination:
listening a concert of 144 dB dynamic (the max of 24b) needs the written aproval of the townhall :)
144dB means a voltage difference of 1000000000000000000000000x from the maximal value to the lowest one. Take the SACD which is analogue driven over -10dB wires (500mV): how low must be the lowest voltage :)?
now, can anyone imagine how strong are the intrinsec noise currents compared to the smallest ones (see above)?

So no fully 24b capturing range, no 24b reproduction range. And of course an almost useless "finesse" in capturing and reproducing the 10-16000Hz hearing range.

Even if usual max audio level is around 2.5VRMS this not change anything, in many cases it is very difficult to create something better than 128dB unless we not go to cryogenics electronics - everything what have temperature higher than 0 K make noise - resistor connected to sensitive AC voltmeter (highly hypothetical) will create some small voltage related to its temperature - it is noise. this is valid for all conductors, joints, soldering point, each contact produce noise... noise is everywhere

https://secure.wikimedia.org/wikipedia/en/wiki/Johnson%E2%80%93Nyquist_noise

48ksps/24bit is more than sufficient in normal life, 96ksps/24bits it way beyond human ear abilities, 192ksps is good for measurements lab ;)

[Ghitulescu]

10bits is only 60dB dynamics, for 80dB 14 bits is requiredI know :), the technique allows 16b but the sources are not :)
Even if usual max audio level is around 2.5VRMS this not change anything, in many cases it is very difficult to create something better than 128dB unless we not go to cryogenics electronics - everything what have temperature higher than 0 K make noise - resistor connected to sensitive AC voltmeter (highly hypothetical) will create some small voltage related to its temperature - it is noise. this is valid for all conductors, joints, soldering point, each contact produce noise... noise is everywhere
Exactly, even with +4dB connections (studio standard) 128dB are hardly achievable (capturing and transporting).48ksps/24bit is more than sufficient in normal life, 96ksps/24bits it way beyond human ear abilities, 192ksps is good for measurements lab ;)
96/24 is probably the best compromise between achievable quality and space requirements.

[pandy]

I know :), the technique allows 16b but the sources are not :)

Exactly, even with +4dB connections (studio standard) 128dB are hardly achievable (capturing and transporting).
96/24 is probably the best compromise between achievable quality and space requirements.

Accordingly to the equation form https://secure.wikimedia.org/wikipedia/en/wiki/Johnson%E2%80%93Nyquist_noise it is impossible to achieve 128dB in 30 - 40kHz bandwidth for nominal 600 ohm studio at the room temperature.

128dB should be possible (barely) with low impedance equipment for 20kHz bandwidth

[Midzuki]

"Line doubling" is extremely bad for music. Interpolation would be better.
:confused: :confused: :confused:

Would you mind explaining why is it so? :)

I've already done that (manual audio upsampling by sample duplication), and my "non-audiophile" auditory cortex didn't detect any "critical injury" :p in the final result. :o

[pandy]

http://www.dsplog.com/2007/03/25/zero-order-hold-and-first-order-hold-based-interpolation/

there are two ways for "Line doubling" and many ways for interpolation...

[Midzuki]

Thanks for the URL.

Honestly, I didn't know that a lossless transformation could ever be worse than a lossy one. :eek:

[CpT]

This...
- mixing at a higher samplerate makes no sense. the discrete channels are all sample-aligned, so why would there be an improvement to adding them together in a higher samplerate?



and Sound card 889 or 892 set to 5.1
http://img153.imageshack.us/img153/5660/50758175.png

After all this you're running onboard sound?

[ramicio]

I can see I'm just banging my head against the wall here trying to make any sense to anyone else about mixing waveforms. High frequencies suffer when mixed together at a low sample rate. If you take a 16/44.1 and make it 24/192 you will have a more coherent waveform and when mixed you will have a more coherent waveform, then you can downsample. No one's nerdy BS talk is going to change my mind on this. Science is not absolute and I see how things go for waveforms with my own eyes. My ears are another thing and I don't care if I can barely tell a difference, I like to have the best quality one can get. I don't see why people need to bring DACs and dB into discussion here. Bits always turns into a damn dB discussion and sample rate turns into a damn hz discussion. So sick of it. I say it only applies to a simple sine waveform. I think about more resolution with either bits or sample rate. You start mixing sounds together at low quality and things will suffer. I get why the OP wants this feature, and I wish it was there too. I wish one could stack filters and use them twice so I could resample, downmix, and resample.

[Ghitulescu]

@ramicio

You're right. However, bandwidth is Hz and bits are dB. Nobody can change this. And, most important, all are numbers.
What you say makes sense in the analogue world, like sinewave and stuff.

Any processing in the digital world (numbers) needs a lot of algorithms, none is perfect.
The very same way one cannot let the PC judge the artistic quality of a painting or a movie, one cannot let the PC judge the music quality. A man can go beyond the numbers and see the logic behind (abstractization) while a PC does only implement pre-created algorithms on data. Same data can be seen as a JPEG or as a WAV or as an AVI. Remember the old CD-players that played noise from Data-CDs? For you it's noise, for them just data that needs to be played. This is the whole point.

[Ghitulescu]

Not very long time ago, some digital mixers worked internally with 20b and output to DATs (for CD factories) only in 16b. How to fit 20b into 16b? Simply, thought the engineers of that time: since the master fader summed all the useful signal in the "top bits", the lower ones are simply floor noise or unused (zeros) data. So they cut the lower 4 bits (in PC language they simply shifted the data 4 positions downwards).
The result was not always acceptable, today most downconverters use noise shaping based dithering algorithms, as they yield the best (for now) results. Which algorithm to use, well that's another issue.

What is importance of this example? Well, the main fader was not graded into bits or Hz, but in dB. A sound engineer uses dB. The whole chain is measured in dB, the specs are in dB. Everything that surrounds an engineer is in dB. Because this is how our senses work, not because someone thought that the job of an A/V technician is so easy that it needs to be complicated with mathematical formulas.

[pandy]

I can see I'm just banging my head against the wall here trying to make any sense to anyone else about mixing waveforms. High frequencies suffer when mixed together at a low sample rate. If you take a 16/44.1 and make it 24/192 you will have a more coherent waveform and when mixed you will have a more coherent waveform, then you can downsample.

this is simple not true...

if You add two samples 44.1/16 together, You will receive 44.1/17 sample - they are completely coherent in time domain and complete coherent from quantization point of view as coherent adding of the 2 bits together can be - there is no magic in adding two 16 bit digits - they give You perfect and fully coherent 17 bit value. What is worst as 192 is not dividable by 44.1 You will receive "incoherent" to source samples after resampling.
High frequencies not suffer at all...


No one's nerdy BS talk is going to change my mind on this. Science is not absolute and I see how things go for waveforms with my own eyes. My ears are another thing and I don't care if I can barely tell a difference, I like to have the best quality one can get. I don't see why people need to bring DACs and dB into discussion here. Bits always turns into a damn dB discussion and sample rate turns into a damn hz discussion. So sick of it. I say it only applies to a simple sine waveform. I think about more resolution with either bits or sample rate. You start mixing sounds together at low quality and things will suffer. I get why the OP wants this feature, and I wish it was there too. I wish one could stack filters and use them twice so I could resample, downmix, and resample.

Do some experiment - 2+3=5 - multiply both by 10 i.e. 20+30=50
divide by 10 and You will receive 5 - nothing nerdy here...
this is simple not even math but arithmetic - definitely You over estimate binary arithmetic if You believe that shifting bits from right to left and add them together can give You better quality.

And believe me - if something apply to the sine waveform it also apply to series of sine (or cosine) waveforms.

If You insist to upsample Your source take this good advice and made this not on 192ksps from 44.1 but on 176.4 or 352.8ksps - after decimation to 44.1 You will receive more "coherent" samples than after up-conversion from 44.1 up to 192ksps.
Why? - search for differences between synchronous and asynchronous sample rate conversion.

btw search also for "golden ears" debate about non-oversampled DAC's in audio ;)

"Non oversampling DACs are by many people (including me) regarded as better sounding than conventional oversampling systems."

http://www.justblair.co.uk/creating-a-diy-non-oversampling-dac-with-pcm1704.html


http://www.sakurasystems.com/articles/Kusunoki.html (http://www.sakurasystems.com/articles/Kusunoki.html)

[vola]

This...




After all this you're running onboard sound?
What a good sound card should be purchased?

I hear only through SPIDF

[nurbs]

@pandy
If you want to understand where ramicios concerns stem from look here (http://www.hydrogenaudio.org/forums/index.php?showtopic=84208&view=findpost&p=726646). Basically his experimentation has lead him to believe that the Nyquist-Shannon sampling theorem is wrong if there is more than one frequency present in an audio file, therefore he oversamples so that at least the downmix isn't affected by that perceived problem.

[ramicio]

If you think you can digitize an analog signal and have it come out exactly the same as it was before it was recorded you are insane. If you think only frequencies that you can audibly hear are the only ones that make a sound sound realistic you are insane. I prefer to have an analog signal as unmolested as possible, so I use high sample rates and a 24 bit depth. Sorry I do what I do, and I feel I should be able to get music is something higher than Redbook. 24/48 would suffice, and no loudness. Do you really think if they would have had say the DVD medium back when CD was being invented that they would have chosen 16/44.1? I think they chose what they chose to be DECENT enough for the storage medium. The other reason being recording off PAL tapes and the sample rate matching BS. I believe if they had more storage they would have chosen better quality. You all argue dynamic range and peak frequency, I argue resolution.

[CpT]

What a good sound card should be purchased?
I hear only through SPIDF

I've used the "Plus" version of this one - no complaints.
http://www.newegg.com/Product/Product.aspx?Item=N82E16829271002

Currently I run m-audio and luv it. But tbh m-audio is a bit overkill for a media center pc.

->Creative sux...

[pandy]

@pandy
If you want to understand where ramicios concerns stem from look here (http://www.hydrogenaudio.org/forums/index.php?showtopic=84208&view=findpost&p=726646). Basically his experimentation has lead him to believe that the Nyquist-Shannon sampling theorem is wrong if there is more than one frequency present in an audio file, therefore he oversamples so that at least the downmix isn't affected by that perceived problem.

hmmmmm i see the source of his faith - maybe this help a bit http://src.infinitewave.ca/

[Groucho2004]

"Non oversampling DACs are by many people (including me) regarded as better sounding than conventional oversampling systems."

Same here. I have a Audio Note DAC1 fed by the SPDIF OUT of a LynxOne card. Sounds great!

[vola]

I've used the "Plus" version of this one - no complaints.
http://www.newegg.com/Product/Product.aspx?Item=N82E16829271002

Currently I run m-audio and luv it. But tbh m-audio is a bit overkill for a media center pc.

->Creative sux...
Thank you
What will give me this card through SPIDF hearing???

How it will manifest itself ???

the card from 2007
not support hd ONLY (AC97)
not support win 7 64 ( Out of a bad experience with ASUS)

Looking for something new

[pandy]

What with older Creative cards with good driver like KX Drivers?

http://kxproject.lugosoft.com/

IMO Creative make quite good audio cards.

[Ghitulescu]

@pandy
If you want to understand where ramicios concerns stem from look here (http://www.hydrogenaudio.org/forums/index.php?showtopic=84208&view=findpost&p=726646). Basically his experimentation has lead him to believe that the Nyquist-Shannon sampling theorem is wrong if there is more than one frequency present in an audio file, therefore he oversamples so that at least the downmix isn't affected by that perceived problem.

Nobody here said ramicio is wrong in his sayings. The problem is that 2 aspects are discussed here: the "feeling" (which is human only) and the numerical representation (which is computational). Music is not dB and Hz and bits, but since this is how the digitised music is all about, we have to restrict ourselves at what we had: bits, Hz and dB.

The only way both aspects can be unified is to use the same parameters from acquisition to reproduction. Any resampling & Co would inherently involve one of the lossy algorithms, so criticised.

In short, upsampling 44k1/48 to 192 or 16b to 24b would only make the sound worse, because of the imperfection of algorithms. Should the source already be present in 192/24 then the manipulation algorithms would work better than on the same source but in 48/16.

[nurbs]

How is he not wrong then? There is stuff that helps with what he does, like doing the downmix at a higher bitdepth, and stuff that doesn't help, like upsampling before downmixing. He does only the latter, because that's what makes him feel better. I get the implications of having a higher quality source, but he doesn't have one in this case.

[Groucho2004]

In short, upsampling 44k1/48 to 192 or 16b to 24b would only make the sound worse, because of the imperfection of algorithms.

Generally, you're probably right. However, there are exceptions like the DACs from Wadia and the DCS upsamplers which employ very sophisticated interpolation algorithms. I have heard a top of the line Wadia DAC which is very impressive.

Still, the only reason I can think of to go this route is to shift the sampling frequency to much higher values (192 KHz or even 384 KHz) so only a simple low order anti-aliasing filter is needed.

My personal choice is however a non-oversampling DAC which sounds very natural to me.
The Audio Note DACs only have a simple passive 3rd order low pass after the DAC and you can certainly see the "steps" when you put a 10 KHz signal on the oscilloscope.
Most electronics engineers would tell you that this concept would lead to horrible sound and one would need an extremely high order low pass to filter out the digital noise. And yet, these DACs sound fantastic which would imply that the human ear appears not to be that sensitive to the digital noise.

[pandy]

"very sophisticated interpolation algorithms" which is always only interpolation - it can be good for some type of signal and wrong for other - not optimal from general point of view... i like Wadia design, i like philosophy behind Wadia but interpolation is only interpolation - filling the gaps between real data samples by "fake" ones - how good are "fake" data - maybe it shall be completely different set of algorithms or some adaptive algorithm, self tuning, different for Miles Davies or Pink Floyd or Beethoven - who knows?

issue with "non-oversampling DAC" is always issue related to the clock recovery - but what in case when we don't recover clock, when we use one master clock for whole system? when all processing is properly synchronized - never ending disputes of "golden ears" - "vinyls are better than CD" - ok, we can assume that this is ok but how to explain that source for vinyl and CD is exactly the same - digital one? on some point such debate turns to faith, believes and religion - there is no place for science on this area.

[Groucho2004]

"very sophisticated interpolation algorithms" which is always only interpolation - it can be good for some type of signal and wrong for other - not optimal from general point of view... i like Wadia design, i like philosophy behind Wadia but interpolation is only interpolation - filling the gaps between real data samples by "fake" ones - how good are "fake" data - maybe it shall be completely different set of algorithms or some adaptive algorithm, self tuning, different for Miles Davies or Pink Floyd or Beethoven - who knows?

issue with "non-oversampling DAC" is always issue related to the clock recovery - but what in case when we don't recover clock, when we use one master clock for whole system? when all processing is properly synchronized - never ending disputes of "golden ears" - "vinyls are better than CD" - ok, we can assume that this is ok but how to explain that source for vinyl and CD is exactly the same - digital one? on some point such debate turns to faith, believes and religion - there is no place for science on this area.

All good points and I agree that filling in missing samples is better left to our brain than to electronics. But that's just my opinion.

I have several albums on CD and also as vinyl and it's not always the vinyl that sounds better.

I believe that nowadays the most damage is done in the mastering, or even worse the "re-mastering" process. The trend is to make music sound louder which mostly means that dynamic compression is applied and the signal is driven into clipping. Not difficult to imagine how that sounds.

[pandy]

Yes, i agree - famous "loudness war" kill high quality sound reproduction - all technological advances was destroyed with mastering targeted to portable mp3 players - mass market doesn't care about quality...

[vola]

What with older Creative cards with good driver like KX Drivers?

http://kxproject.lugosoft.com/

IMO Creative make quite good audio cards.

NO older Creative cards

Creative DON'T MAKE audio cards
NOT FOR MOVIES OR Music
Do not know if I working through SPIDF
Why do I need a better card???
No CARD processor information through SPIDF

Do not see why not trying to build Agurittim for 192000HZ
Through SPIDF> PASS THROUGH

Audio card did not analyze the sound Through SPIDF
MY Amplifier gets up 48 BIT 192000HZ

All what I read here is not practical ONLY theories
If the software will BE Supports in 192000
Hardware build By manufacturers .

Just do it
Just do it
Just do it

[Groucho2004]

Creative DON'T MAKE audio cards
NOT FOR MOVIES OR Music
Do not know if I working through SPIDF
Why do I need a better card???
No CARD processor information through SPIDF

Do not see why not trying to build Agurittim for 192000HZ
Through SPIDF> PASS THROUGH

Audio card did not analyze the sound Through SPIDF
MY Amplifier gets up 48 BIT 192000HZ

All what I read here is not practical ONLY theories
If the software will BE Supports in 192000
Hardware build By manufacturers .

Just do it
Just do it
Just do it

Did you run this through some sort of word randomizer before posting?

[vola]

English not MY native language

[pandy]

English not MY native language

Mine to but im trying to improve them and avoiding to make mess on public forums...

[vola]

Mine to but im trying to improve them and avoiding to make mess on public forums...

I do not understand what you want?

[Groucho2004]

Just do it
Just do it
Just do it

This is still about your desire to upsample to 192Kbps in ffdshow, right?

As far as I know, there is no limit in libsamplerate which is used in ffdshow, it just applies a multiplier. I have not looked at the ffdshow source code but I imagine that this would be easy to implement.

Still, lamenting about it in this thread will hardly attract the attention of the ffdshow developers.

[pandy]

NO

Creative DON'T MAKE audio cards
NOT FOR MOVIES OR Music


Creative made a lot of good sound cards, with good drivers they are much better than many similar competitors cards.
But Creative made few, not very good marketing moves and it have bad opinion - but technically they products are above average in terms quality and functionality.



Do not know if I working through SPIDF
Why do I need a better card???
No CARD processor information through SPIDF



So not use card processing - with KX drivers it can be disabled.


Do not see why not trying to build Agurittim for 192000HZ
Through SPIDF> PASS THROUGH

Audio card did not analyze the sound Through SPIDF
MY Amplifier gets up 48 BIT 192000HZ

All what I read here is not practical ONLY theories
If the software will BE Supports in 192000
Hardware build By manufacturers .

Just do it
Just do it
Just do it

At first - S/PDIF may not support 192/24 - it was made for 44.1/16 and work with 48/16 but not many devices support S/PDIF 96/24 and i don't know any device that will support 192/24 (over 9 Mbits), optical S/PDIF (TOS Link) is even more limited and most of optical connection will refuse to work with 96/24 (typical TOS Link transmiter/receiver is approx 4Mbps) - also this cheap plastic connection introduce more problems with quality of the audio (jitter).

For 192/24 probably You must go for HDMI connection - so instead searching for sound card - search for graphics card that have HDMI with embedded Audio on it.

[pandy]

I do not understand what you want?

English is also NOT my native language but im trying to improve my English and im avoiding to make mess on public forums...

Behave instead being like this child http://www.youtube.com/watch?v=HUs_4Aue-fw

[TinTime]

At first - S/PDIF may not support 192/24 - it was made for 44.1/16 and work with 48/16 but not many devices support S/PDIF 96/24 and i don't know any device that will support 192/24 (over 9 Mbits), optical S/PDIF (TOS Link) is even more limited and most of optical connection will refuse to work with 96/24 (typical TOS Link transmiter/receiver is approx 4Mbps) - also this cheap plastic connection introduce more problems with quality of the audio (jitter).

For 192/24 probably You must go for HDMI connection - so instead searching for sound card - search for graphics card that have HDMI with embedded Audio on it.

This Asus (http://uk.asus.com/product.aspx?P_ID=QsEKBPr6ko9pFF2D&templete=2) will do 192/24 over S/PDIF. I'm sure there are other devices and cards as well.

[pandy]

This Asus (http://uk.asus.com/product.aspx?P_ID=QsEKBPr6ko9pFF2D&templete=2) will do 192/24 over S/PDIF. I'm sure there are other devices and cards as well.


So im not so sure - not many S/PDIF receivers are capable to work with 192ksps - issue is not in source but in receiver.
You can transmit data but You must have device that is capable to receive them correctly (conversion from digital to analog is not important at this moment).

[vola]

With all due respect

This is still about your desire to upsample to 192Kbps in ffdshow, right?

YES

As far as I know, there is no limit in libsamplerate which is used in ffdshow,

Please see the picture is limited to 96000HZ

http://img403.imageshack.us/img403/2821/21255289.png

IT'S LIMIT

not many devices support S/PDIF 96/24 and i don't know any device that will support 192/24

All SPIDF support today on 24/96 by manufacturers
Today ASUS + GIGABYTE now support 24 / 192 according to their answers

http://img714.imageshack.us/img714/2664/70734797.png

IF NOT SPIDF
THe last two years all ATI video cards
4000HD +5000 HD +6000HD support through
HDMI in - 24/192


So there is no hardware limit:)

AND If the software will BE Supports 192000
Hardware build By manufacturers going Supports Also 18MBPS
If necessary.

[TinTime]

So im not so sure - not many S/PDIF receivers are capable to work with 192ksps - issue is not in source but in receiver.
You can transmit data but You must have device that is capable to receive them correctly (conversion from digital to analog is not important at this moment).

My fairly elderly Yamaha amp will play it, although all it disables all DSP. I don't know if it truncates to 20bits or not.

[Ghitulescu]

Do not see why not trying to build Agurittim for 192000HZ
Through SPIDF> PASS THROUGH

If you want to say that you would prefer to use your embedded soundcard to send 192kHz/24b over S/P-DIF, well there's a problem as S/P-DIF cannot carry this information, well it can carry 24b but chances are that the receiving end discards the extra 8 bits. 48kHz is the max sampling frequency. More info in IEC 60958.

Should one transport AC-3 signals, well, they may go a bit higher but all the advantage an algorithm in 192kHz would have is lost due to compression. More in IEC 61937.

[Groucho2004]

Please see the picture is limited to 96000HZ

http://img403.imageshack.us/img403/2821/21255289.png

IT'S LIMIT


You've taken my statement out of context and you obviously don't have a clue what I'm talking about.

I give up.

[pandy]

But Yamaha is very good manufacturer - they made good audio from ear and scope point of view - after fast search seems that Only ONE company make S/PDIF receivers capable to 192ksps - Cirrus (former Crystal) - IC's from TI and AD probably not supporting 192ksps on S/PDIF

[vola]

if you want to say that you would prefer to use your embedded soundcard to send 192khz/24b ober s/p-dif, well there's a problem as s/p-dif cannot carry this information
so hdmi 1.3a or 1.4 in hd 6000 ati

[pandy]

With all due respect



YES



Please see the picture is limited to 96000HZ

http://img403.imageshack.us/img403/2821/21255289.png

IT'S LIMIT


NO, put 192000 manually and FFDSHOW will mark output as 192000 but there is something wrong and when i tried to play ac3 file it was played twice slower (maybe issue with drivers on my notebook)


All SPIDF support today on 24/96 by manufacturers
Today ASUS + GIGABYTE now support 24 / 192 according to their answers

http://img714.imageshack.us/img714/2664/70734797.png

IF NOT SPIDF
THe last two years all ATI video cards
4000HD +5000 HD +6000HD support through
HDMI in - 24/192


So there is no hardware limit:)

AND If the software will BE Supports 192000
Hardware build By manufacturers going Supports Also 18MBPS
If necessary.

LOL - i dont believe that someone use Realtek chips as a good sound source example...

[vola]

NO, put 192000 manually and FFDSHOW will mark output as 192000 but there is something wrong and when i tried to play ac3 file it was played twice slower (maybe issue with drivers on my notebook)

my notebook
A full model sound card?
What format do you want to play it? 192 000?
Which player?
What speakers?
What Amplifier ?

i dont believe that someone use Realtek chips as a good sound source example

I HAVE Asus Xonar D2X 7.1
AND PRODIGY LT 7.1

AND I LOVE SPIDF HD
ON Motherboard

Sound Cards sound AS Polished (I DON'T LIKE AC97 ON PCI )

[pandy]

A full model sound card?
What format do you want to play it? 192 000?
Which player?
What speakers?
What Amplifier ?


hp6930p
ad1984a?
directx*
B&W802**
TACT T2**


I HAVE Asus Xonar D2X 7.1
AND PRODIGY LT 7.1

AND I LOVE SPIDF HD
ON Motherboard

Sound Cards sound AS Polished (I DON'T LIKE AC97 ON PCI )

You know that for S/PIDF such audio card act like UART...


*PC and windows software players are not good environment for anything in real time.

**joke - I use headphones - Sennheiser HD215 with my notebook

[vola]

hp6930p
ad1984a?
directx*
B&W802**
TACT T2**

you have
Beautiful parts
I can not know their ability to play your file
Because I do not know the FILE AND YOUR SPIDF
Day-Old in production
Therefore possibility the playr can't play 192 000 Through SPIDF

But the point if you do not play SPIDF So HDMI will play

The question who will put 192 000 algorithm IN FFDSHOW
" resample " ???

Who Can I Turn???

It could be a hit or flop
But then we know

[Ghitulescu]

Please use google translate - do us a favour, pliz pliz pliz

[ramicio]

When did I say that I don't downsample after I mix the audio? In my head I think of upsampling say from 48khz to 192khz almost like what a DAC does. So if there so much error and loss with upsampling, then the same will happen within a DAC when making the signal analog.

And what is so wrong with Sound Blaster cards? Yea their drivers suck anymore, but their hardware is amazing. I tried onboard sound ONCE and will never go back to any C-media or any generic garbage again. I have had zero problems with my PCI-express X-fi card.

People seem to think S/PDIF is an actual interface, which it is not at all. Yes S/PDIF has standards for hardware, but it is not hardware itself, just a way to code the audio. TOSLINK now supports really high bandwidth (125 mbps), so we only all need to wait for sound card makers and receiver makers to allow raw PCM to be transferred optically. That will never happen because of the music and movie industries. You could use TOSLINK for a network if you wanted to. Coax, TOSLINK, and HDMI are only interconnects, they can transfer any data you want to.

[TinTime]

TOSLINK now supports really high bandwidth (125 mbps), so we only all need to wait for sound card makers and receiver makers to allow raw PCM to be transferred optically.

And coax doesn't support high bitrates?

[ramicio]

I never said it physically can't, but TOSLINK already has it implemented, we just need to wait for audio hardware makers to take advantage of it. HDMI is still the best choice but it is too new and so restricted by big corporations.

[nurbs]

When did I say that I don't downsample after I mix the audio? In my head I think of upsampling say from 48khz to 192khz almost like what a DAC does. So if there so much error and loss with upsampling, then the same will happen within a DAC when making the signal analog.
So what you do is upsample, downmix, downsample. There won't be any problems with that, at least in the 48 kHz to 192 kHz case, could be a little more tricky with a 44.1 kHz source, but probably no real trouble anyway. Still two of those three steps are useless. There is no reason to upsample to begin with and the downsampling is only there because you want the original sampling rate back.
There are reasons why DACs upsample. It allows the use of a lower resolution DAC, like for instance the 1bit DACs that are standard (I think) in portable players, to convert higher resolution input for example. It also helps with quantization noise. The point is they do it because it helps with the actual digital to analog conversion. That has nothing to do with mixing a couple of channels together.
So if you want to do a good quality downmix leave the sampling rate alone and use a higher bitdepth.

[TinTime]

I never said it physically can't, but TOSLINK already has it implemented, we just need to wait for audio hardware makers to take advantage of it. HDMI is still the best choice but it is too new and so restricted by big corporations.

High bitrates over coax are also already implemented.

[ramicio]

Yeah that's exactly what I do for movie tracks, which are 5.1 24/48 lossless, and I just downsample to 24/96 then just because I don't care about space. I'm done beating my head into a wall here. Everyone tries to clutch onto something they like and invest so much into rather than accepting change when there is something much better available.

[ramicio]

High bitrates over coax are also already implemented.

In what aspect? Industry? Definitely not for home use. I know of all the crazy coax BNC crap out there, and industrial stuff is not much better than home stuff, just different so they can charge more for it. Anyone can have any technology that isn't classified by a military, if they have the money for it.

[nurbs]

OK so your workflow is this:
5.1 24/48 -> 5.1 24/192 -> 2.0 24/192 -> 2.0 24/96

A better workflow would be:
5.1 24/48 -> 2.0 24/48

No difference in quality, but the latter would be half the size.

If the audiophile magic in your workflow makes you happier that's fine, but it doesn't make the result any better, it just makes it bigger.

[ramicio]

As I have said before, no.

5.1 24/48 -> wavs

I mix each channel independently where they belong into a stereo 24/192, then go to 24/96. As I said before I don't care about space, and I use FLAC as the track that actually gets muxed into the movie itself.

[TinTime]

In what aspect? Industry? Definitely not for home use. I know of all the crazy coax BNC crap out there, and industrial stuff is not much better than home stuff, just different so they can charge more for it. Anyone can have any technology that isn't classified by a military, if they have the money for it.

I'm not sure what your point is because 125Mbps TOSLINK isn't supported for home use either.

[nurbs]

Yeah, because separating the source into mono wavs makes all the difference. :rolleyes:

It may well be necessary for your workflow, but it doesn't make what you do different from my description.

I did a short test. Your workflow resulted in a 40% bigger file. That might not be representative because I only tested one movie, but it goes to show that even with lossless compression there is still a significant overhead.
You might not care about filesize, but most people wouldn't waste bits on something that doesn't offer any benefit at all.

[Ghitulescu]

And what is so wrong with Sound Blaster cards? Yea their drivers suck anymore, but their hardware is amazing. I tried onboard sound ONCE and will never go back to any C-media or any generic garbage again. I have had zero problems with my PCI-express X-fi card.

The wrong is, apart from its catastrophic drivers, that it requires 3 IRQs (erh, not shared ones!). Other drawbacks can be derived from the drivers and the software it comes with.

For the same price one can get a decent semipro card (like M-Audio or Echo, they are many) will all the advantages such a card might bring. This depends however also on your priorities, because a semipro card doesn't care too much about 3D gaming acceleration and stuff.

Onboard soundcards suck, so better a SB than the onboard one.

[Ghitulescu]

Anyone can have any technology that isn't classified by a military, if they have the money for it.

For some gear they have to invest part of this money into a one-man-company, as they are not sold to individuals. :)

ANd if we're discussing about "the money", the military technology is also available.

[Ghitulescu]

You might not care about filesize, but most people wouldn't waste bits on something that doesn't offer any benefit at all.

Those people won't feel then any difference between 44.1 and 48 or between 16 and 24b, not to mention MP3 vs. WAV/FLAC. Because if they do, then they would care about.

[nurbs]

Those situations aren't the same at all. There might not be a perceptible difference between MP3 and WAV, but there is an actual difference so depending on what you want to do keeping the lossless track can make perfect sense.
Here the source is 24 bit 48 kHz, the operation is downmixing from 5.1 to stereo and the output is 24 bit 96 kHz. Upsampling doesn't provide any benefit in that case. It makes about as much sense as taking a 128 kbps MP3 and reencoding it at 192 kbps to improve the quality, except that the only drawback here is higher filesize since the resampling won't reduce the quality, unlike the reencoding would.

[vola]

Please use google translate - do us a favour, pliz pliz pliz

If you think I write wrong writing to tell me exactly what?
I have no idea what you want from me?



pliz pliz pliz

[Ghitulescu]

you have
Beautiful parts
I can not know their ability to play your file
Because I do not know the FILE AND YOUR SPIDF
Day-Old in production
Therefore possibility the playr can't play 192 000 Through SPIDF

But the point if you do not play SPIDF So HDMI will play

The question who will put 192 000 algorithm IN FFDSHOW
" resample " ???

Who Can I Turn???

It could be a hit or flop
But then we know

I don't understand what is written here, well, yes I know the words.

It was an advice, I know that many people around here, including me, are not native English speakers, but at least they try to make them understood. It wasn't disrespectful.

Translate.google.com can translate from Hebrew into English, how well, sorry, I can't tell, I don't speak Hebrew. But translations from/to FR, IT, DE, ES (some of the languages I know) are surprisingly good.

[ramicio]

People make CDs from mp3, then a friend rips that CD to mp3, and it goes on and on and I have heard how songs sound after a while. I might just try out your theory that just mixing them all together @48khz doesn't make a difference. And I don't like your comment about separating the WAVs making a difference. What the hell does that even mean? I do it so I can mix them in another program. I don't like the downmix scheme that eac3to applies so I do it manually. As I said before a GOOD SOUNDING and properly mastered CD isn't mixed in 16/44.1. You keep believing that. Maybe some cheap indie crap or any ignorant pop star CD.

And what programs do people need for a sound card? It's a device that plays sound. I don't recall of any sound card that ever came with a proprietary program that played music and became popular. For someone listening to music EAX and other DSP crap stay off. Drivers work fine. I view semipro or pro cards as what people need to get the best in recording, not so much playback. Who cares about IRQ anymore? Things are managed no so you don't have to mess with things. I remember trying to get my Sound Blaster 16 trying to work because of IRQ stuff, and even as recently as Live!. The Live! era was definitely the worst in Creative history, they are getting better.

[Ghitulescu]

Who cares about IRQ anymore? Things are managed no so you don't have to mess with things. I remember trying to get my Sound Blaster 16 trying to work because of IRQ stuff, and even as recently as Live!. The Live! era was definitely the worst in Creative history, they are getting better.

I do.

Besides, a card that needs 2-3 IRQs in an exclusive mode, while others can share 3 cards an IRQ, makes me think about how that card was designed in other places. Was it still "Who cares?" approach used or not? Remember that it was the Creative Live! the card that destroyed the fame of the emu10k.

[pandy]

As I said before a GOOD SOUNDING and properly mastered CD isn't mixed in 16/44.1. You keep believing that. Maybe some cheap indie crap or any ignorant pop star CD.

it is not mixed but it is rendered to 44.1/16 bit - in many cases at the maximum efficient way - it is like turning picture with 256 optimal color pallete and FS dither back to the 24 bit... how to recover 24 bit from 8?

[pandy]

Remember that it was the Creative Live! the card that destroyed the fame of the emu10k.

But EMU10K was designed with 48ksps on mind and for completely different purposes than PC audio cards. What is really funny - this card is still used by many musicians but with KX or EMU drivers - not Creative ones.

[Ghitulescu]

...Exactly :)

[vola]

I don't understand what is written here, well, yes I know the words.

It was an advice, I know that many people around here, including me, are not native English speakers, but at least they try to make them understood. It wasn't disrespectful.

Translate.google.com can translate from Hebrew into English, how well, sorry, I can't tell, I don't speak Hebrew. But translations from/to FR, IT, DE, ES (some of the languages I know) are surprisingly good.

I'm sorry I did not understand what you want from me I checked the inscription
Everything correctly, what do you want?

How can I understand what you write if you does not explain what the problem?
Do you see spelling Problems ?
and Where ?
Do words not understood to you?
and Where ?

[vola]

If you want to say that you would prefer to use your embedded soundcard to send 192kHz/24b over S/P-DIF, well there's a problem as S/P-DIF cannot carry this information, well it can carry 24b but chances are that the receiving end discards the extra 8 bits. 48kHz is the max sampling frequency. More info in IEC 60958.

Should one transport AC-3 signals, well, they may go a bit higher but all the advantage an algorithm in 192kHz would have is lost due to compression. More in IEC 61937.

http://en.wikipedia.org/wiki/S/PDIF
soundcard to send 192kHz/24b over S/P-DIF, well there's a problem as S/P-DIF cannot carry this information

As I wrote back and saying ALC 892 (realtek)
They say they found a way to deliver 24/192
I attach again the Document:
http://img714.imageshack.us/img714/2664/70734797.png
AND
Optical fiber again as in the picture:
There are regular SPIDF and SPIDF Optical
Again only to GIGABYTE ALC 892
http://img10.imageshack.us/img10/8862/88088124.png

Again regardless of the document or SPIDF
I wrote if you do not User SPIDF use HDMI :

Originally Posted by vola View Post
All SPIDF support today on 24/96 by manufacturers
Today ASUS + GIGABYTE now support 24 / 192 according to their answers

http://img714.imageshack.us/img714/2664/70734797.png

IF NOT SPIDF
THe last two years all ATI video cards
4000HD +5000 HD +6000HD support through
HDMI in - 24/192


So there is no hardware limit

AND If the software will BE Supports 192000
Hardware build By manufacturers going Supports Also 18MBPS
If necessary.


1. Ghitulescu- Do you understand the words literally ?
2.Ghitulescu- Do you not agree WITH something
3. I do not understand what all the debate?
There is now an appropriate hardware

AND If there is limited The manufacturers will find a way
Produce an appropriate hardware .

Now we have to make Alagoritm of 192000HZ
Again in "resample" !

[Ghitulescu]

Where does it say that your S/P-DIF transport more than 48kHz/24b in LPCM mode? First this is the highest format the S/P-DIF is allowed by standard to carry*. Secondly, any extensions to this must however either be standardized or are proprietary techniques (like ADAT).
You may wanna say it's DTS or DD, well, they are transported as user data format, and the sender and the receiver must agree on the format (channel status byte 1). You can send 96kHz or even 192kHz but in this case one must use lower bitrate. Like for MP3, if you want 20 songs instead of 15, then you have to pick a lower bitrate. Which makes ridiculous the whole idea of upsampling 48 to 192 for quality reasons if one then use aggressive compression upon it.

For your information: wikipedia is not authoritative, it's mostly written by regular people. I have an account there and if I want to change the article about S/P-DIF to write there that S/P-DIF can carry 24 channels of 384kHz and 48b in LPCM, I can do it. And until this error will be corrected by other people, some readers may believe me, because "it's wiki".

*For LPCM there are only 2 (eventual 4) channels that have a max. sampling of 48kHz. While the standard defines other sampling rates, up to 192 and low as 22.05, these are used in junction with compressed data (like DTS or DD). An indication of multichannel was also provided, in junction with the 4chn mode (which is not used any longer, I think) or it may be used with DTS or DD or any other "user data format".

As I said before, should the Gigabyte of yours deviate from the standard, then it's proprietary, and proprietary solutions means often incompatibilities with other manufacturers.

[Groucho2004]

You can send 96kHz or even 192kHz but in this case one must use lower bitrate.

This doesn't make sense. The bitrate is the product of sampling frequency and sample width (16, 24 bit, or whatever).

*For LPCM there are only 2 (eventual 4) channels that have a max. sampling of 48kHz.
Eventual? What does that mean? It starts with 2 channels and then turns into 4????

[Ghitulescu]

This doesn't make sense. The bitrate is the product of sampling frequency and sample width (16, 24 bit, or whatever).


Eventual? What does that mean? It starts with 2 channels and then turns into 4????

There's a bandwidth limitation, for the first point. To transport more one needs compression, which transform a higher bitrate into a lower one. Legal compression algorithms are MPEG-2, DTS, DD and the new AAC and another new one I forgot its name.

There were trials on 4 channels long time ago, this modus corresponds 1:1 to one of the 32kHz modi of a pro DAT (as the consumer ones cannot record/play more than 2 channels; it's called LP on these). AFAIK, the 4 channels was used on some SAT transmissions, in some TV stations, and for this reason was implemented in DATs and therefore in S/P-DIF.

If you want more details just read the standard.

Eventual NOT EQUAL eventually.

[Groucho2004]

Eventual NOT EQUAL eventually.

Seriously? You are giving me a lecture in English?

[Ghitulescu]

I pass over what my English teacher taught me :)

[vola]

Where does it say that your S/P-DIF transport more than 48kHz/24b in LPCM mode?
I've included here twice document ALC + GIGABYTE:
What they SPIDF Through ALC 892 can play if you have a technical question to ALC + GIGABYTE
I'd love to give them your question (With an explanation and evidence)

Now the main issue
If all due respect
"resample to 192000hz"
If you do not believe the manufacturer ( ALC + GIGABYTE )
Use HDMI
What is the problem ? Use HDMI ?
I understand like you THE problem of standard
( Like you I looked at WEB and NOT found material on SPIDF +24 / 192 )

I GOING TO GIVE Prize FOR "resample to 192000hz"
Player + CODE How to watch movies Originally IN 24/25/29 FPS
Turn them to 60/75 FPS! AND MORE !!!
( like frame scale Studio version )

The film will show you an amazing quality LIKE poster
Sharpness, contrast Maximum, every hair, dust you can see ALL

picture :
http://img440.imageshack.us/img440/4826/666hi.png
http://img89.imageshack.us/img89/1676/667h.png
http://img525.imageshack.us/img525/4504/668.png

As you can see
Achieved an average frame rate
Picture 666 47.7 FPS
Picture 667 60.62 FPS
Picture 668 65.59 FPS

[Inspector.Gadget]

Congratulations: in those screenshots you failed to hide the fact that you were watching pirated content. Worse yet, one of the titles is "Sex and the City 2". You ought to be embarassed.

[ramicio]

I was looking at DACs yesterday and other USB to S/PDIF high end stuff, and most of them supported transporting 24/96 over coax, some even did 24/192.

[vola]

Congratulations: in those screenshots you failed to hide the fact that you were watching pirated content. Worse yet, one of the titles is "Sex and the City 2". You ought to be embarassed.

No need to hide it
Bad movie they should make up for time

THIS IS TRAILER

[leeperry]

Why not give the possibility to make UPSCALE TO 192000HZ with audio ? Through ffdshow audio
how about typing it in, duh: http://www.pixelz.fr/0/d/4/7d64ab8d9006322780e91f0f53c68.png

It does work. This thread is full of fail :o

Your Realtek prolly has such a large jitter that your 24bit will never get through to the other side anyway: http://www.agoraquest.com/viewtopic.php?topic=34057&forum=51
The difference is not loss of Data, but the corruption of the timing relationship between the Data and the 4 timing clocks in the S/PDIF Bitstream. This timing discrepancy is called jitter and for 16 bit data there can be no more than 100 picoseconds (1 picosecond =1/1,000,000,000,000th of a second) of jitter for full resolution whereas 24 bit Data can have no more than 0.5 picoseconds of jitter to maintain full resolution.

Don't forget that S/PDIF is bound to fail in the first place, hence the need for a low jitter and a clean signal: http://www.gearslutz.com/board/so-much-gear-so-little-time/172143-spdif-vs-word-clock-question.html
S/PDIF is a horrendously poorly designed interface. This is because it combines the clock and audio coding onto the same signal. The receiver is supposed to recover the clock from this signal as well as extract the audio data. This turns out to be a non-trivial task, and one that almost always leaves the recovered clock contaminated with signal correlated jitter artefacts.

And 24bit only matters for post-processing, well...except if you regularly listen >96dB: http://www.head-fi.org/forum/thread/415361/24bit-vs-16bit-the-myth-exploded

Gregorio is a ProTools instructor ^

I upsample when feeding Reclock to decrease aliasing and improve my VST post-processing, but for bit-perfect audio all it does is coloring the sound/increasing the THD/THD+N IME(easy to check w/ WaveSpectra).

You can see that some pro-audio plugins also oversample on purpose to increase the processing accuracy: http://www.gearslutz.com/board/5101666-post296.html
The oversampling is necessary to avoid aliasing with the saturation, but also helps to get a more accurate response in the upper end. Just as you guys said, we use a steep IIR LP filter as oversampling filter.


Many ppl feel that NOS is a terrible idea: http://www.head-fi.org/forum/thread/438220/nos-dac-marketing-bs

look at those measurements on a NOS DAC, I giggle to no end: http://www.head-fi.org/forum/thread/378459/valab-dac-first-impressions/1185#post_5483734

http://www.head-fi.org/image/id/24007/width/525/height/525/flags/LL

And some other ppl feel that upsampling is uncalled for when you have a properly oversampling DAC: http://www.audioholics.com/education/audio-formats-technology/upsampling-vs-oversampling-for-digital-audio/upsampling-vs-oversampling-for-digital-audio-page-2
The effects of upsampling are greatly debated. While it is true that upsampling does help us in attenuating the amount of jitter caused by sampling errors and an inaccurate clock, whether this jitter is audible or not is a point of contention. There is no doubt that wide bit words and super-high sampling rates that are touted by the latest products are largely marketing. Oversampling has been around for a very long time and has been used extensively in audio products to not only improve sound quality through 'better' filtering but to make these same products much cheaper. Upsampling, on the other hand, is relatively newer and debated greatly. The effects of upsampling are no doubt overstated. By carefully designing the sampler, ADC, digital processing path, and oversampling DAC, the upsampling and asynchronous rate transfer can, in my opinion, be avoided.

and last, but not least, bit-perfect can easily end up clipping due to intersample peaks: http://www.gearslutz.com/board/tips-techniques/334385-intersample-peaks.html

if you care about SQ over PCM S/PDIF, you might wanna look into:
-getting a low jitter transport to chew the receiver job at recovering the clock
-using coax instead of toslink...its jitter is horrid(like +700ps) when coax off a good transport is <100ps
-attenuate the volume to avoid intersample clipping
-using a properly oversampling DAC and avoid upsampling
-using a bit-perfect audio renderer such as KS/ASIO/WASAPI
-play lossless audio files

[Inspector.Gadget]

No need to hide it
Bad movie they should make up for time

THIS IS TRAILER

Trailers are named like "scene" releases now and have the same runtime as the full movie? :rolleyes: Read the rules, the last thing this forum needs is DMCA complaints.

[vola]

leeperry

My eyes are happy with
Tears

When I saw the picture

http://img94.imageshack.us/img94/1818/82677878.png

[pandy]

leeperry

My eyes are happy with
Tears

When I saw the picture

http://img94.imageshack.us/img94/1818/82677878.png

Then all that thread can end after http://forum.doom9.org/showthread.php?p=1451199#post1451199

[vola]

Then all that thread can end after http://forum.doom9.org/showthread.php?p=1451199#post1451199

I'm sorry to write just 192 000 not working
It sounds like 96 000 HZ

No Alagoritm reach

What's the point to register just 192000HZ
If you do not have Alagoritm to turning it to 192 000 HZ?

[leeperry]

192 000 [..] sounds like 96 000 HZ
O RLY? :scared:

[nurbs]

Upsampling doesn't improve quality. Since the 192 kHz output sounds like the 96 kHz output to you we can conclude that everything works as expected. If you could hear a difference that would be an indication that something went wrong during the upsampling or of problems with your hardware.

[vola]

Upsampling doesn't improve quality. Since the 192 kHz output sounds like the 96 kHz output to you we can conclude that everything works as expected. If you could hear a difference that would be an indication that something went wrong during the upsampling or of problems with your hardware.

do you have Receiver to check through the HDMI port 1.3a or 1.4?
24/192 Through ATI video card

If there is no difference between 96 000 HZ to 192000hz in hdmi
so 192 000 is not working at all

problem in the Algorithm or elsewhere .

[nurbs]

If there is no difference between 96 000 HZ to 192000hz in hdmi
so 192 000 is not working at all
Why? Why should there be a difference?

Does your receiver show the input as 96 kHz when you set 192 kHz in ffdshow? That would be a problem. If your problem is that both sound the same then you don't have a problem.

[pandy]

I'm sorry to write just 192 000 not working
It sounds like 96 000 HZ


I verified ffdshow output in graphedit - definitely output pin from ffdshow is flagged as 192ksps - then if You can't tell the difference seems that You need to have ear upgrade.


No Alagoritm reach


Algorithm works OK


What's the point to register just 192000HZ
If you do not have Alagoritm to turning it to 192 000 HZ?

But everything is fine - only yours ears are flaky and they need to be upgraded - maybe some cochlear implant?

[vola]

Why? Why should there be a difference?

with all due respect

Why is there a difference between 48 000 TO 96000HZ
WHY ?

Why not be a difference between 96 000 HZ TO 192 000
WHY ?

Does your receiver show the input as 96 kHz when you set 192 kHz in ffdshow?

And this is the problem that FFDSHOW not Converter Alagoritm of 192 000? to receiver ?

If your problem is that both sound the same then you don't have a problem.

Or we all have a problem, we do not understand there is a problem !

I wait to my VGA card+
HDMI then I'll Thorough checks
But I have a problem because I do not know if HD4850
Support - 192000hz

The manufacturer registrar :
"" Subject to digital rights management limitations; maximum supported audio stream bandwidth is 6.144 Mbps

http://www.amd.com/us/products/desktop/graphics/ati-radeon-hd-4000/hd-4850/Pages/ati-radeon-hd-4850-specifications.aspx

So what card support - Dolby TrueHD and DTS Master Audio formats
If the DTS-HD MA supports variable bit rates up to 24.5 Mbit / s on a Blu-ray Disc and up to 18.0 Mbit / s
http://en.wikipedia.org/wiki/DTS-HD_Master_Audio

so How thay write that the manufacturer AMD - HDMI
Support -
Dolby TrueHD and DTS Master Audio formats
How will 18 MB Work in 6 MB ?

What format transmitter FFDSHOW 192/24 ?

[vola]

I verified ffdshow output in graphedit - definitely output pin from ffdshow is flagged as 192ksps - then if You can't tell the difference seems that You need to have ear upgrade.

But everything is fine - only yours ears are flaky and they need to be upgraded - maybe some cochlear implant?

pandy I will give you the answer of nurbs
Does your receiver show the input as 96 kHz when you set 192 kHz in ffdshow? That would be a problem. If your problem is that both sound the same then you don't have a problem.

[nurbs]

Do all of us a favor and get somebody who speaks English better than you to help you with this stuff.

Also get your story straight. In post #62 you say there is no hardware limit and now suddenly you say your ATI card limits the output after complaining about ffdshow for several pages. The limitation you posted is consistent with uncompressed 8 channel 16 bit 48 kHz audio. It would be interesting to know if you can use higher bitdepths and samplerates if you have less channels. It is possible that this idiotic limitation is only for PCM and the card can still bitstream compressed audio.

You still haven't told us what your receiver displays. Maybe that sentence you quoted is supposed to be an answer, but I don't get it.

Assuming the card only outputs 48 kHz regardless of channels or sampling rate since you state you can hear a difference between 48 and 96 kHz, but 192 kHz sounds the same (as 96) it's entirely possible that your card uses a bad downsampling algorithm and that's what causes the difference.
If that's true congratulations. You have spent the last six pages desperately trying to upsample 48 kHz audio even though there aren't any benefits on playback and all you have achieved is worse audio quality because of your hardware limitations.

[vola]

Again you perverse the reality
I mentioned at the end :

AND If the software will BE Supports 192000
Hardware build By manufacturers going Supports Also 18MBPS
If necessary.
"nurbs" do you Read English?
get somebody who speaks English better than you to help you with this stuff ?
Also get your story straight "nurbs"

and in
http://en.wikipedia.org/wiki/DTS-HD_Master_Audio
Specifically listed:
DTS-HD MA supports variable bit rates up to 24.5 Mbit/s on a Blu-ray Disc and up to 18.0 Mbit/s

If a device / manufacturer said
DTS-HD MA supports variable bit rates up to 24.5 Mbit / s on a Blu-ray Disc and up to 18.0 Mbit / s
Where do I need to know that GIGABYTE
Lying?

If they sent me this article :
http://en.wikipedia.org/wiki/HDMI#Version_1.3
What do I need to understand?

GIGABYTE Write me No problem easily supported by DTS-HD MA / Dolby TrueHD
Where do I need to know that GIGABYTE mean up to 6 MB ???
Read their site if registered up to 6MB ???
Where do I need to know they Lie ?


OK I've included more and give you in writing
Document GIGABYET + ALC
Support -192/24
All SPIDF support today on 24/96 by manufacturers
Today ASUS + GIGABYTE now support 24 / 192 according to their answers

http://img714.imageshack.us/img714/2664/70734797.png

[nurbs]

If you want to pretend that what you post is anywhere close to proper English and that if I don't understand it I'm the one who is the problem I'm not going to bother with this anymore. I hope you realize that by starting that last post like you did you are only making a fool out of yourself.

Let me give you one last piece of advice: If you want to get help with your problem then at least try to answer questions people ask.
I haven't been able to figure out yet how you determine that the upsampling doesn't work since according to pandy ffdshow outputs the right sampling rate and you haven't told us what your receiver displays.

[pandy]

im trying to be sarcastic in my previous answer but i will say this one more - if there is no hear-able difference between 96 and 192ksps it doesn't mean that upsampling not work but it can mean that for 96ksps You are in the point where Your system (equipment, ears, brain, listening environment) limit 192ksps (search for discussions about capabilities of human abilities to hear), why assumption that after living in noisy environment, listening high level of various sound you have still more than perfect hearing system?
Connect 44.1kHz wave sound with some two close frequencies (IM test) to the ffdshow, choose 192000, to the output of ffdshow connect wav dest filter and write file - then open file in decent audio editor and verify what is produced by ffdshow.

[vola]

People do not know how to dance
They claim the floor curve
in post 88 I wrote:
If you think I write wrong writing to tell me exactly what?
I have no idea what you want from me?

http://forum.doom9.org/showthread.php?p=1452161#post1452161

I noticed that some people a bit rude
They are not sure of thay Answer they given
so they Suggest it's because my English!

Of course they do not show the problem with my English
Or where the error
But even if they are wrong ,they make it because of me
It's rude, poor !

...last post like you did you are only making a fool out of yourself.
no
I keep my dignity
From people like you " rude "
I have no problem you Trying to prove where I am wrong in English

But instead of proving me that I right Because
You continue to" bark" (your English, your English)
It's not serious

Tell me where, tell me what exactly
tell me how this disrupts your answers ?

If you want to get help with your problem then at least try to answer questions people ask
You are wrong again
I answer all
Except rude people

You want a good attitude
Give good attitude

[vola]

I HAVE succes to resample to 192000hz
:):):):):)

It sounds sharper from 96 000 HZ
A higher level of acuity

There is a clear difference from 96 000 HZ
192000 sharper
This WOW

http://img704.imageshack.us/img704/7947/17986971.png

I changed some things on FFDSOHW....
and i cross / pass to "digital output optical "
http://img688.imageshack.us/img688/7740/27694609.png

There are some difficult passages hearing ( With different sound types )
That compression 192 000
Too big for SPIDF
Directions must be done

I will try soon to see if there are Difference with HDMI

the Volume with my 48 000HZ = 55%
now
the Volume with my 192000HZ = 35%
the sound sharp, clear
:):):):):)

One problem sometimes you're resample to 192000hz do not get caught/ not encoded to 192000
Have to try several times

A second problem to do resample to 192000hz Or 96000HZ
Should target every time again to 192000hz Or 96000HZ
After the start of the song or movie
if you leave the resample in 192000hz Or 96000HZ
At the beginning of the next song or movie sound card will not work with ffdshow !

[Groucho2004]

It sounds sharper from 96 000 HZ
A higher level of acuity

Prove it! :D:D

[vola]

Prove it! :D:D
I added more pictures
have a good life to all
:thanks::thanks::thanks::thanks::thanks:

[Groucho2004]

That compression 192 000
Too big for SPIDF
Directions must be done

And don't forget that during the night it's colder than outside.
;)

[leeperry]

There is a clear difference from 96 000 HZ
192000 sharper
This WOW
yay, you did it \o/

[MatLz]

I wonder why he didn't yet asked for 384KHz :confused:
:D

[pandy]

I wonder why he didn't yet asked for 384KHz :confused:
:D

It is obvious - next step will be 768 ksps not flaky 384.

[leeperry]

I wonder why he didn't yet ask for 384KHz
over S/PDIF? good luck :D