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!! Sample Rate !!
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jivamukti
IsraTrance Junior Member
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342
Posted : Jun 24, 2007 20:48
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Upavas could you please divide your posts into smaller paragraphs with space between them. My head hurts from reading all written in one big lump.
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When clouds moisten, rain forms. |
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UnderTow
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Posted : Jun 25, 2007 15:42
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On 2007-06-24 10:06, Upavas wrote:
And yes the quality of the converters is important. However most converters in the mid price range do just nicely nowadays,
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Indeed. The quality of cheap converters these days is quite amazing.
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and as to the sample rate as being less important, I don't think so and both arguments, yours and mine have their right.
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My main point is that sample rate does not equal to resolution. It really only equates to bandwidth. At 96Khz (or 88.2Khz or even 60Khz which unfortunately doesn't exist in practise), any of the filter slope issues are more than covered. There really is no point in going to 192Khz. It does not increase resolution.
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Just recently I played a set at Organix and remember the dj after me saying he had trouble keeping up with the overall volume and was wondering how I had the system that loud without distorting everything. Came out he was playing mp3's and I was playing wavs... particularly the bass was what made him trouble. Go figure...
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Not all MP3 encoders are made equal. Some sound better than others. Also some encoders have a 10Hz filter option. If you have that turned on when encoding the MP3, it WILL affect the sound. Especially the bass. These filters are rarely phase linear and as they are not "brickwall filters" they will also affect the level of bass frequencies above 10Hz.
But also, the DJ after you might just have a collection of less well produced tracks... who knows...
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DCPCM or Delta Pulse Code Modulation is generally being used in mp3 compression,
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Aaaah! DPCM. I thought DCPCM might be some different format I had never heard of. (There are so many...)
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oh and who told you that film data is data compressed before it reaches the audience?
No way man. My boss would kill anyone doing such a thing... as a matter of fact he would probably be really pissed off...hahaha...
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He must be really pissed off at the film industry then. I'm talking about DTS, Dolby Digital and such. Nearly everything the audience gets to hear is data compressed.
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Also remember that when you change the sampling rate digitally (not using DA/AD converters the sound will be out of sync with the original... meaning faster or slower depending which way you go...
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This isn't correct. You can easily sample rate convert without affecting length/timing!
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which always is a pain in the @$$ when you have to convert sound with film to tv format... especially in North America where we have 29.97 fps as opposed to the nice even 25fps they have pretty much everywhere else in the world...
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I'm glad I don't have to deal with Drop Frame and stuff like that.
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Most audio for tv film actually is recorded with a sampling rate of 48048 khz, so when it is reduced by .1% which has to be done in North America because the film itself is reduced in Speed by.1% to match the damned 29.97 frames per second...it will be playable on most systems (48k)...It is so much easier to convert when you have even numbers...
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These are actually clocking issues more than anything else. But yeah, I'm not too familiar with all the issues involved with this except in theory. In practise, I work in Europe so I've never had to deal with any issues like that.
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as to analog recording, the recording is always more accurate than digital recording,
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This is absolutely not true. Analogue recording media all have significant levels of distortion. They rarely have flat frequency response and they don't have unlimited bandwidth either.
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the reason why we use digital is because of the static noise that comes with analog recording, the hiss on a tape if you will.
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That is not the only reason but yes, that is one of the issues.
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Since we don't have an option for infinite sampling rates, which you explained well above the recording can never be as accurate as analog, where we get a straight 100% recording.
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I think you have a deep misunderstanding of how digital audio works. I'm guessing you are thinking of a "join the dots" representation of digital audio.
Here's a 20 Khz sine wave as shown in Sound Forge:
http://home.casema.nl/ajohnston/Sampling/20Khz-Join_the_dots.jpg
But this isn't actually what comes out of the converters. This is just the data, not the actual signal!
This is what the same 20Khz sine wave looks like in Audition which reconstructs the wave showing the signal that actually comes out of your converters:
http://home.casema.nl/ajohnston/Sampling/20Khz-Reconstructed.jpg
Like I said in my previous post, you only need two points to fully define a circle (and thus a sine wave). You do not gain any accuracy (resolution) by adding more points (sample frames).
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What are you referring to when you speak of analog format btw. ? You mean the device (Tape, Nagra etc.) ?
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I mean the recording medium. So tape, vinyl, optical, whatever.
UnderTow |
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UnderTow
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1448
Posted : Jun 25, 2007 15:46
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On 2007-06-24 07:00, shamantrixx wrote:
Do you have any pleasure in life beside calling someone a no brain? I guess not!!!
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You are such a fool. If you would have been paying attention you would have seen that that was just a joke in response to Upavas comment on hearing. The smiley was a pretty obvious hint...
UnderTow |
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Upavas
Upavas
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Posted : Jun 25, 2007 21:46
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You should try Nagra, with the right mics or sound source there will be no distortion whatsoever, provided you make sure there is no proximity effect when recording mics and that the levels aren't too hot. Distortion always ocurs when yor recording levels are to high or through a proximity effect that occurs on your mic when the soundsource is too close to the diaphragm of the mike...
and not all analog recording devices have significant levels of distortion. Most do, yes, but not all of them...
Yes, the dynamic range in the analog realm is not as good as a 24 bit digital recording. However, technically the analog recording will always be more precise than any digital recording. (one reason why my cat does not like cd's...
On another note lets say here that I prefer analog to digital distortion, especially when it comes to loud noises, gunshots etc., simply because it is recorded seamlesssly analog and not digitally sampled!
As to you thinking I have a deep misunderstanding of how digital audio works, I have actually learned all that stuff years ago @ school, so my understanding of the digital audio realm is quite ok.
Yes , for the human ear it might be precise at 44.1, technically you should know as well as I do it never is 100% precise! Higher frequencies will still get aliased as previously mentioned (Niquist Theorem) even if not audible by the human ear they will still be present and as sound pressure level affect the human being though not his or her hearing.
Hence the lowpassfilter...
As to the myth that nearly all audio data in the movie industry is compressed, you obviously do not know what you're talking about. Under no circumstances would any sound supervisor or even sound editor ever do such a thing, unless they work in Game audio, that is a totally different story.
Hell, I told a supervising soundeditor of this remark who works in Hollywood, currently involved in a production with Owen Wilson, and he said after looking at me that how could I have asked such a question and then laughing out loud as well that this is absolute bolloks.
I recently finished a post production in 5.1 surround with DTS and nothing of the audio was ever compressed. It would take too much out of the movie. As a matter of fact it would sound too hollow!
And yes, not all mp3 converters are made equal, if this would be a onetimer with the dj who played after me I would probably tend to agree with you Undertow, since this happened to me again and again on many events however I am pretty sure that it is just the case that mp3's just do not have the same sound quality as wavs.
And if you think about it is is perfectly logical and plausible...The bigger soundsystems seem to reflect this a lot more acurately...
I know you only need two points to fully define a circle, however, you also need to consider that if you depict the wav as a sinecurve instead of a circle, which is more often done and by no means more or less accurate, if your sinewave is too long for the distance of your 2 points the recorded sample will be aliased and therefore not precisely recorded.
As to join the dots, I have no idea what you mean by that.
Besides, we are getting into the age old arguments between digital and analog domain and this could probably go on until the end of time.
Geez, I am not even talking about resolution and you still are bugging me with it.
Oh and Shamantrixx, Ithink this thread is good to read for you as it probably will increase your understanding of audio in general.
As a matter of fact I rather enjoy this open discussion...
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Spindrift
Spindrift
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33
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1560
Posted : Jun 25, 2007 23:19
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On 2007-06-25 21:46, Upavas wrote:
As to the myth that nearly all audio data in the movie industry is compressed, you obviously do not know what you're talking about. Under no circumstances would any sound supervisor or even sound editor ever do such a thing, unless they work in Game audio, that is a totally different story.
Hell, I told a supervising soundeditor of this remark who works in Hollywood, currently involved in a production with Owen Wilson, and he said after looking at me that how could I have asked such a question and then laughing out loud as well that this is absolute bolloks.
I recently finished a post production in 5.1 surround with DTS and nothing of the audio was ever compressed. It would take too much out of the movie. As a matter of fact it would sound too hollow!
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Maybe you and your 'supervising soundeditor' should read up on what the DTS format is.
What you are saying is equal to claiming that you made an mp3 that was not compressed.
DTS is a format that is using lossy compression:
http://en.wikipedia.org/wiki/Digital_Theatre_System
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There are significant technical differences between commercial/theatrical and home variants: the former being a traditional ADPCM compression system and the latter a sophisticated hybrid perceptual and signal-redundancy compressor based on ADPCM called APTX-100. |
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Not that I work in film, but what I understand the issue is with DTS that the audio needs to fit on one CD that will be synced and play with the movie in a theatre with DTS.
Also dolby digital uses compression. The reason there is that the space they use to fit audio information on the actual film (between the sprocket holes) is very limited.
http://en.wikipedia.org/wiki/Image:35mm_film_audio_macro.jpg
Luckily your impression that it would sound hollow is only placebo, as clearly presented by the fact that you never heard that the audio is compressed when you was under the belief that it was not.
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UnderTow
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Posted : Jun 26, 2007 00:51
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On 2007-06-25 21:46, Upavas wrote:
You should try Nagra
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Which one? Alot of the Nagra devices are digital.
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Distortion always ocurs when yor recording levels are to high or through a proximity effect that occurs on your mic when the soundsource is too close to the diaphragm of the mike...
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No, distortion always occures period. This goes both for analogue and digital recording. The question is how much and what type of distortion. Digital will give you less distortion. Much less.
With digital recording, the distortion is below what we can percieve. With analogue tape it is actually always perceivable but, when kept within reasonable limits, it is a type of distortion that we find pleasant: Harmonic distortion.
You are probably thinking of audible amounts of distorion when things oversaturate or overload. That is not what I am talking about.
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and not all analog recording devices have significant levels of distortion. Most do, yes, but not all of them...
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I am using the term eaxctly how it is meant: The distortion signifies something aka we can hear it.
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However, technically the analog recording will always be more precise than any digital recording. (one reason why my cat does not like cd's...
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You are not backing this up with any data or explanation. But thats ok because it isn't correct. As for the cat comment, that sounds like antropomorphism to me... You have no idea why your cat does or doesn't like CDs. Maybe it just doesn't like your brand of CD player.
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On another note lets say here that I prefer analog to digital distortion, especially when it comes to loud noises, gunshots etc.,
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Agreed. Analogue is more forgiving.
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simply because it is recorded seamlesssly analog and not digitally sampled!
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Bingo! That is exactly what I mean with the "join the dots" understanding of digital audio. Digital audio, once it leaves the converters, IS seamless! There are no stair steps, no sample points that get joined in straight lines etc. Only seamless fluid continuous voltages. Just like analogue. (Well of course, once the signal leaves the converters it is analogue).
Actually any straight lines or sharp corners would be equal to high frequencies beyond half nyquist. Those are out of band signals (frequencies) and should not be part of digital sampling. That is why digital distortion sounds so bad. They are illegal signals.
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As to you thinking I have a deep misunderstanding of how digital audio works, I have actually learned all that stuff years ago @ school, so my understanding of the digital audio realm is quite ok.
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I'm sorry to say but your above comment about seamlessness in analogue reveals your lack of understanding. Go and have a good look at the pictures I referenced in my previous post.
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Yes , for the human ear it might be precise at 44.1, technically you should know as well as I do it never is 100% precise!
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No indeed it is never 100% precise. The inaccuracies (ignoring bad filter design and clocking issues etc for a second) reveals itself in noise. But, as you pointed out yourself, the noise floor in digital is much lower than in analogue!
And as you again point out yourself, it is precise enough for the human ear. That is all that is needed.
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Higher frequencies will still get aliased as previously mentioned (Niquist Theorem) even if not audible by the human ear they will still be present and as sound pressure level affect the human being though not his or her hearing.
Hence the lowpassfilter...
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This paragraph doesn't hold water for the simple reason that Nyquist's Theory includes the filters! When properly implemented, Nyquist's theory and Shanon's theorem (He proved the maths making it a theorem rather than a theory), gives you a non aliased signal.
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As to the myth that nearly all audio data in the movie industry is compressed, you obviously do not know what you're talking about. Under no circumstances would any sound supervisor or even sound editor ever do such a thing, unless they work in Game audio, that is a totally different story.
Hell, I told a supervising soundeditor of this remark who works in Hollywood, currently involved in a production with Owen Wilson, and he said after looking at me that how could I have asked such a question and then laughing out loud as well that this is absolute bolloks.
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See Spindrift's comment... Nearly all sound delivered to cinemas is data compressed. This usually happens after the post studios have handed over the project. I am talking about the various delivery formats used. There are many and it depends on what any particular cinema has and what the different film studios and distribution companies use.
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I recently finished a post production in 5.1 surround with DTS and nothing of the audio was ever compressed. It would take too much out of the movie. As a matter of fact it would sound too hollow!
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There are various types of DTS encoding, many of them lossy.
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And yes, not all mp3 converters are made equal, if this would be a onetimer with the dj who played after me I would probably tend to agree with you Undertow, since this happened to me again and again on many events however I am pretty sure that it is just the case that mp3's just do not have the same sound quality as wavs.
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They don't but when well encoded, you can't tell on a big PA. Also, you mention this happening on many occasions which means that you knew the DJ was using MP3s. I suspect the placebo effect at work here.
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And if you think about it is is perfectly logical and plausible...The bigger soundsystems seem to reflect this a lot more acurately...
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Actually it is the other way round. PAs arn't nearly as accurate as good studio monitors, and there is more often than not processing (like limiting) in the signal chain. Also people get pretty much desensitised to subtle changes very quickly due to the SPL levels of the sound.
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I know you only need two points to fully define a circle, however, you also need to consider that if you depict the wav as a sinecurve instead of a circle, which is more often done and by no means more or less accurate, if your sinewave is too long for the distance of your 2 points the recorded sample will be aliased and therefore not precisely recorded.
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Actually the problem is with wave cycles that are too short. Too short means too high in frequency. That is why the signal needs to be band limited. That is the whole basis of Nyquist's sampling theory.
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As to join the dots, I have no idea what you mean by that.
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Aka not seamless. See above.
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Besides, we are getting into the age old arguments between digital and analog domain and this could probably go on until the end of time.
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That depends wether people are willing to accept the science or prefer to take it as a religion. So far we are talking about the facts so thats cool. It's only when people at some point refuse to accept the science that the discussion becomes pointless. (No pun intended ).
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Geez, I am not even talking about resolution and you still are bugging me with it.
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I'm just trying to clear up misunderstandings about audio in general.
UnderTow |
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Upavas
Upavas
Started Topics :
150
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3315
Posted : Jun 26, 2007 03:47
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"Which one? Alot of the Nagra devices are digital."
try any analog Nagra 4 track, not the digital ones...
As to why my cat will not like cd's, she can perceive much higher frequencies than us and even with the best filters there will still be some aliasing, that is unless you spend big money...
Analog recording devices have some harmonic distortion, yes, however it is not perceived by humans directly, so since you so abundantly made clear with the human ear and digital recording, it does not matter.
What matters is that Analog records directly, from a soundsource into a converter which converts it into voltage and then to your speaker which converts it back without taking single samples like digital recording, it simply records the whole thing, not in incrementations of 44.1oo per second, just the whole thing at once... NO ALIASING!!!
Since we do not have unlimited amounts of 1's and 0's yet we cannot EVER record anything digitally a 100% which is why lp filters are being used.
So in that sense any tibetan singing bowl which has a very large spectrum of overtones that goes a lot higher than 20k will never sound the same digitally recorded, simply because there is a lowpass filter which cuts of anything above 20k, that is if it is a good one... you have a much better chance recording that whole sound more true analog.
As even that will not be 100% the same (noisefloor and distortion, it is still much closer than digital recording. And yes, these frequencies may not be perceived by the human ear, it still has an impact on our bodies and our feeling just like the 7hz frequency does...
As to the placebo effect, no, I don't always know what anybody uses who plays before or after me, after all I do not care, I generally use wavs, so I know mine will be ok... so you are suspecting wrongly in that instance...
As you made clear once the signal is converted it IS analog. So no aliasing. Unless you have not used the filters of course and your digital version had aliasing... however nothing above 20k will be there, so where is the accuracy? You may not be able to hear the difference, however it still has an impact on you since the ears are not all of our senses...
Natuarally the wave file gets aliased if to short for the sampling rate, sorry about that but it was late late last night...
my brand of cd player is called Pioneer 100, any more questions on that one?
I said the big PA's seem to , not do. When we hear something more loud, so we hear more things that may not have had enough loudness with the studio monitors. A studio monitor will of course reflect spl more accurately, however in a studio at least I don't crank it up like on a party. Now don't you start complaining about my monitors, I know they are good...
YOU can't tell between mp3 format and wav format, however I can and many other people I know seem to be able to as well. Even if 320kbps... it is till not the same sound, sorry.
  Upavas - Here And Now (Sangoma Rec.) new EP out Oct.29th, get it here:
http://timecode.bandcamp.com
http://upavas.com
http://soundcloud.com/upavas-1/ |
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Upavas
Upavas
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150
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Posted : Jun 26, 2007 04:13
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Spindrift, not all DTS uses lossy compression!
So why don't you read some more about DTS ?
Adaptive Differential Pulse code modulation (ADPCM)is basically used in the game audio industry, not so much in movies. It is generally Proprietary, meaning there are many different forms that are not compatible on other systems, purely marketing strategy... And yes, that one is lossy, although it anticipates what the next sample does and appears therefore less lossy than mp3 copmpression, however is not!. However ADPCM is not very widely used in Film. At least not here in North America.
  Upavas - Here And Now (Sangoma Rec.) new EP out Oct.29th, get it here:
http://timecode.bandcamp.com
http://upavas.com
http://soundcloud.com/upavas-1/ |
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UnderTow
Started Topics :
9
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1448
Posted : Jun 26, 2007 15:00
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On 2007-06-26 03:47, Upavas wrote:
"Which one? Alot of the Nagra devices are digital."
try any analog Nagra 4 track, not the digital ones...
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Well I don't have one available but I am familiar with the sound of tape. (1/4" or 1/2").
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As to why my cat will not like cd's, she can perceive much higher frequencies than us and even with the best filters there will still be some aliasing, that is unless you spend big money...
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That is just guessing on your part. Anyway, there are no higher frequencies on the CD. Any frequencies that were aliased during sampling or sample rate conversion, can get aliased back all over the frequency spectrum Of course they are more likely to be likely to be close to 1/2 Nyquist...
Anyway, who says that cats don't like aliasing, or more specificaly, don't like any potential aliasing from band limiting? There are all sorts of unnatural sounds in audio. Alot of synths alias so most music has some amount of aliasing in it.
Maybe your cat doesn't like the typical crushed sound most CDs have these days... Actually I will contradict what I wrote above: When digital clipping is used to increase the level (or accidentaly) there are bursts of higher frequencies depending on the DACs. (It isn't for nothing that this is called an illegal signal).
So don't clip your mixes if you want to make cat friendly music.
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Analog recording devices have some harmonic distortion, yes, however it is not perceived by humans directly, so since you so abundantly made clear with the human ear and digital recording, it does not matter.
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Actually it is perceived directly. That is the whole point of using analogue in the 21st century! Euphonic distortion is what it is all about.
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What matters is that Analog records directly, from a soundsource into a converter which converts it into voltage and then to your speaker which converts it back without taking single samples like digital recording, it simply records the whole thing, not in incrementations of 44.1oo per second, just the whole thing at once... NO ALIASING!!!
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There you go again with those steps. That is not how digital works in practise. I seriously recommend you read Nika Aldrich's book mentioned in a previous post.
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Since we do not have unlimited amounts of 1's and 0's yet we cannot EVER record anything digitally a 100% which is why lp filters are being used.
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Yes, to band limit the signal. Did you try the experiment I described previously? Try that. It should demonstrate that those higher frequencies are not audible. That is what counts.
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So in that sense any tibetan singing bowl which has a very large spectrum of overtones that goes a lot higher than 20k will never sound the same digitally recorded, simply because there is a lowpass filter which cuts of anything above 20k, that is if it is a good one... you have a much better chance recording that whole sound more true analog.
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Again, try the experiment I described. If you can't hear those frequencies, they are not called sound. They are ultra sound. We don't need to record ultra sounds for music.
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And yes, these frequencies may not be perceived by the human ear, it still has an impact on our bodies and our feeling just like the 7hz frequency does...
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There is one study that claims having detected physiological effects from ultra sounds. No one has managed to reproduce that study. It is highly contested amongst the pros and the methodology of the study has been questioned.
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As to the placebo effect, no, I don't always know what anybody uses who plays before or after me, after all I do not care, I generally use wavs, so I know mine will be ok... so you are suspecting wrongly in that instance...
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If you don't know, how can you tell it was MP3s then? That a bit of a contradiction there. Anyway, MP3 DJs might be more likely to use suboptimaly produced music (just a guess) and as I have said before, not all encoders are equal. Especially if there are hipass filters affecting the bass...
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As you made clear once the signal is converted it IS analog. So no aliasing. Unless you have not used the filters of course and your digital version had aliasing... however nothing above 20k will be there, so where is the accuracy?
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There is accuracy within the limited frequency band that we call sound. Thats what counts. But reguarldess, no format has infinite bandwidth. So where do we set the limit? 20Khz? 30Khz? 100Khz? 10Mhz? 25Ghz? It has to be set somewhere. Many many studies have shown that arround 20Khz is a good place.
To be on the safe side, a bit more might have been a good idea but 192Khz sampling rates are just plain bollocks. That sets the Nyquist frequency at 96Khz which is ridiculous.
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You may not be able to hear the difference, however it still has an impact on you since the ears are not all of our senses...
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No hearing isn't our only sense but the other senses don't perceive sound. Except innordinate amounts of bass...
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my brand of cd player is called Pioneer 100, any more questions on that one?
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The comment about the cat not liking a particular brand of CD players was actually a joke. Of course any particular set of converters might cause sounds that certain animals don't like. Like those Echo soundcards that produce a very high pitched tone that we can't hear. (No idea if anyone has ever tested how animals react to this).
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I said the big PA's seem to , not do. When we hear something more loud, so we hear more things that may not have had enough loudness with the studio monitors. A studio monitor will of course reflect spl more accurately, however in a studio at least I don't crank it up like on a party. Now don't you start complaining about my monitors, I know they are good...
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Frankly, I hear more detail in studio monitors than PA speakers.
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YOU can't tell between mp3 format and wav format, however I can and many other people I know seem to be able to as well. Even if 320kbps... it is till not the same sound, sorry.
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Have you ever performed a double blind test under controlled conditions? These tests have been done quite a few times. The results are always statisticaly no better than random.
UnderTow |
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Upavas
Upavas
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Posted : Jun 28, 2007 03:34
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1. If you know tape then you know that not all tapes sound the same. There are profound differences in noise ratio, distortion etc. depending on which media you use.
2. I don't clip my sound ever, for that there are always pot's and faders !
3. Audibility is not what you were referring to previously, as you were referring to precision, I merely pointed out that yes, it is precise up to 20k, and even with very good filters there still will be higher frequs creeping in. And yes, I do know that.
4. I know because many dj's that generally use mp3's come up to me and ask if I was using wavs, and then explain they had trouble keeping up with the same SPL level. You simply keep assuming here, lets keep to the facts.
5. I know all I need to about sampling. No books are needed, thank you.
6. As to the thought that only ears perceive sound, you might be surprised to learn your whole body does, start listening with it and you might find out otherwise. This view is not necessarily true for western minds, for eastern minds it seems to be a fact. Alone the idea that plants that do not have ears, are attracted to some music, repulsed by other music and neutral to lets say western classic has proven that. A plant that is repulsed from music will grow away from a speaker... this is proven science. As plants perceive sounds without ears, so do other animals, by vibration, see snakes with their tongues. So on that point I think you should read a little more. And yes the frequencies have something to do with that! And not just the low frequencies...
7. As to the suggestion of a blind test no, however the fact that many dj's that spin mp3's come to me AFTER and say, "man you playing wavs, really hard to me to keep up with the overall loudness for me" speaks for itself. And even I have heard the difference, as have many friends of mine.
8. CD's will always have minimal amounts of Aliasing in them as no filter is perfect. This cannot be perceived by human ears, but nevertheless it is not precise as you keep telling.
9. I do not say that cats in general don't like aliasing, again you did not listen, I said that my cat seems to not like it. THERE IS A DIFFERENCE!
Summary, yes as far as human hearing goes, you are right, sampling works fine up to 20k on a 44.1 samplerate. It is precise for human ears, but NEVER PRECISE, no matter what, filter and all.
The human ears may be all that matters to you, other people who have learned to not just listen with their ears might beg to differ. Both have their points!
  Upavas - Here And Now (Sangoma Rec.) new EP out Oct.29th, get it here:
http://timecode.bandcamp.com
http://upavas.com
http://soundcloud.com/upavas-1/ |
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