Current surround formats on receivers/preamps

Obviously this should have been done a while ago...

Encoding Formats:

Dolby Digital 5.1 - L/C/R + SL/SR + LFE (384 or 448k/s)

Dolby Digital 5.1 EX - L/C/R + SL/SR + LFE + matrix Rear Centre

dts 5.1 - L/C/R + SL/SR + LFE (768 or 1,536k/s)

dts '6.1'* ES-Matrix - L/C/R + SL/SR + LFE + matrix Rear Centre

dts 6.1 ES-Discrete - L/C/R + SL/SR + LFE + discrete Rear Centre

*obviously they call it "6.1" but it is only 5.1 in reality.

Decoding Format:

THX EX Surround 7.1 - L/C/R + SL/SR + LFE + two matrix rear centres; depending on the product, the matrix rears can be summed mono or derived stereo. This is applied to Dolby 5.1 EX titles. Later Dolby 5.1 EX was introduced as THX 7.1 was too expensive for the majority of systems.

Certifications:

THX Ultra2 - update for 7.1 systems, rated for 3,000+ cu.ft rooms

THX Ultra - 5.1 systems, rated for 3,000+ cu.ft rooms (same as the old THX cert.)

THX Select - 5.1 or 7.1 systems, rated for 2,000+ cu.ft rooms

Other THX formats aren't worth anything.

Current virtualising formats for two channel:

Circle Surround 5.1 ("CS 5.1") - derived centre + SL/SR + LFE

Dolby Pro Logic II - derived centre + SL/SR + LFE

dts Neo 6 - derived centre + SL/SR + LFE

Logic 7 - derived centre + SL/SR + SBL/SBR + LFE can be applied to either stereo or 5.1/6.1 formats depending on sophistication of processor. Limited to Lexicon and Harmon Kardon equipment.

 

Hehe, Good one Tony.

 

Quite obviously I have a lot to do at work

I might also add that you will often see:

Linear PCM 48kHz audio at 16-bit (1,536k/s) or at 24-bit (2,304k/s).

Dolby Digital 2.0 Stereo or Surround at anything from 96 to 448k/s.

There is also an emerging dts 24/96 format at 768k/s with 5.1 or 6.1 channels. I think they call it "dts 96/24".

Details on DVD-Audio and SACD will come some other day and hopefully by someone other than me.

T.

 

wow, good job Tony

Now me to contribute a bit on DVD audio like u requested.

2 types : SuperAudio CD (SACD) and DVD-Audio (DVD-A)

SACD - signal frequecy response up to 100kHz. Normal Cds only 20 KHz.

DVD-A - signal frequency response of up to 192kHz.

DVD-A uses the same coding method of Cds by extending the word length from 16 to 24 bit. Thus the improved performance.

SACD are found in most new Philips and Sony Dvd players. ( Cos they co-develope it)

 

The Dvd format does not actually have a set sample rate. It is determined by [m X 44.1] or [n X 48] where m and n = 1, 2, 3 and so on.

 

Me thought it has a maximum of 192kHz.

 

I think you'll find that it is the sampling rate which is a maximum of 192kHz, giving a frequency response up to 96kHz.

And it was my understanding that DVD-A uses MLP encoding rather than raw PCM like a CD.

AB

 

I see i see

ahhh....since u all know more about dvd audio why dont post a thread about it?

 

I never said I knew more, I simply said that it was my understanding that it uses MLP (and I'm happy to be corrected on that). I spose in retrospect, because MLP is a lossless compression scheme (hence the name) it could be classes as a psuedo-RAW format because every sample is preserved exactly without loss of information. But in my mind a true RAW format stores each sample without compression of any sort. But that's subjective.

I'm sorry I offended you but as this is an informational sticky, I thought it best to double check our facts. If I end up being wrong and you are right then I'm happy to delete my posts to clear up the clutter.

AB

 

As requested:

DVD-A supported formats:



As can be seen, we are both right in that the format supports BOTH raw PCM and MLP. Are we all friends now

Also, remember the nequist theorem which states that you require a sample frequency at least twice the highest frequency your are sampling. ie a sample rate of 192kHz means the highest frequency you can reproduce without introducing folding and aliasing is 96kHz.

AB

 

I think u are mistaken......i am not offended.

Cos the more u guys post, the more i know.

Thanx.

 

i don't like nequist, he shouldn't be used in audio so much. remembering if you use a sampling frequency exsactly twice the sampled frequency, the best you can end up with is a triangle or square wave output, and the worst you can end up with is silience (you can manage to always sample the wave at its zero point) if you sample a frequency VERY close to it you'll get a triangle wave that pulses its intencity.

so you realy need to keep the sampled frequency decently below half the sampling frequency.



actualy as i wrote that i realised that the amount you need to be below the half way point is probably something like 300Hz, as a pulsing of sound at 300Hz isn't noticable.


but then you will also notice weird sounds if you sample exsactly 1/3rd or 1/4 the sampling frequency as well. i guess though as you get down to things like 1/5th or 1/6th any such artifacts shouldn't be noticable.

oh and if you think you can't hear such artifacts your wrong, its just that they are rare

 

talk about being out of my depth...

can anyone explain this lot please.

 

Where are you getting that from? Aiasing artifacts do show up close to half the sampling frequency, but at 192kHz sampling, you'd never hear it. Granted 44.1kHz was a compromise when CDs came out, most of the artifacts came from the attempts to filter out the aliasing. Those artifacts were in the range of hearing. The ones from filtering around 96kHz, if even needed, won't be heard. You might see them on a scope, but not hear them with your ears.

 

good to see you agree with me.

 

lol guys slow down . some stuff to add to the encoding formats
SDDS which is Sony Dynamic Digital Sound
its a 7.1 system and it uses Atrac coding
its freq responce is 20Hz to 20kHz +-1db
sampl freq is 44.1khz
and it has a 90dB min
distortion 0.07%
20db head room
and a compresion ratio of 5.1

oh and the samp rate of DTS is 48khz at 20 bit res

and THX is not actully a surround sound format but a standerd, and what i mean buy that is anything product that has met the qualification will have the THX approval which means will have the damn logo on the device and is good enough to play "lucasfilm" movies on it.

now with the DVD Freq responce. it come down to this. yes a DVD can reproduce 192khz at 24 bit. but it dont mean u cant hear it . the better the freq responce and bit rate is the close the audio will be to the orignal analog signal (remeber our Ear is analog).
the human ear can hear from 20hz to 20khz but it the extra infinty hz that make the sound more alive

i know all of this from doing audio engineering at SAE. maybe u guys should do it cause u seem to know what ur talking about!!!!

 

I dont want to hijack this thread, but the myth that humans can hear 20hz - 20khz is old and has been proven wrong several times since the 70's. I've seen hearing tests from ppl who can hear has low as 10hz to as high as 40khz. There is also Debrovsky's condition (cant remember the correct term) which can cause people to be hyper sensitive to certain things and can manifest in extreme aural sensitivity, making people "hear"or "feel" sounds far above 20khz.

 

i know this. the 20 to 20k is a average that the human ear can hear. as i was saying the extra freqs in the sound wave that is out of the 20 to 20k range makes the sound more alive which means the human ear might "hear" and "Feel" the freqs out of the 20 to 20k spectrum. lol did u understand that

 

It must have been "the human ear can hear from 20hz to 20khz" that made me comment about it. You should really make comments like that clearer, for a start is has very little to do with the ear itself, or the cochlea, it is the brain that ignores frequencies above a below what is "useful". The cochlea itself can pick up a vast range, most of which the brain ignores. Also the 20-20 range is not an average hearing range.

 

i think you've got the whole idea of nyquist stuffed if i can remember my dsp stuff

sampling frequency has to be > 2 fmax where fmax is the maximum frequency present in the signal. At this sampling frequency you can correctly construct the signal being sampled. at lower frequencies stuff like aliasing and folding come into it. if you oversample i.e. sample at greater than 2 fmax the output waveform is much better.

you can't disregard nyquist as it is one of the corner stones of dsp theory.