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In Reply to: Actually, no, i don't consider myself a Guru at all ... posted by Christine Tham on April 27, 2004 at 13:57:39:
"(1) PCM resolution is directly proportional to signal amplitude. if you have a sine wave at maximum amplitude to that the peaks are encoded at maximum values, then yes, that sine wave is recorded at 24 bits resolution. if your sine wave was half that amplitude, you are recording at effectively 23 bits resolution (because the most significant bit is never used therefore it is "wasted"). very low level signals are effectively being recorded using only a few bits of resolution. In most well recorded music, the average peaks of the signal is far below the maximum, so the effective resolution is less than 24 bits. Low level signals are probably being captured at less than 10 bits resolution.DSD on the other hand has an interesting property in that resolution is not linearly proportional to signal amplitude. yes, it is still somewhat dependent on signal strength, but DSD is probably better at capturing low level detail.
to take an extreme example - a zero signal is recorded as a stream of zeros in PCM - effectively with only 1 bit resolution. a zero signal is recorded as a string of alternating 1s and 0s - no bits are "wasted".
"Nonsense, no bits are "waisted". All 24 bits are used to express a sample value.
In digital math zero's and one's are equally important.Another factor is that if you record with a proper gain setting the low level signals are far below the threshold of being able to hear them.
Usually the input gain of the recorder channel is adjusted just below clipping. This is just 'good practise' of the record engineer.
That dsd is somehowe preserving low level detail better is a myth that resulted from clever marketing and pseudo scientific white papers. Hires pcm with dither and noise shaping has far better low level sound quality.
That the lower pcm bits end up burried in the analog stages noise floor is perhaps a beneficial side effect as this masks any errors left over from processing and/or improper dithering after bit reduction from 32 bit processing into 24 bit audio files.
Follow Ups:
as usual, you have missed the pointas an extreme example lets say that the gain is set so that the signal never exceed 12 bits. then effectively the 24-bit PCM recording is indistinguishable from a 12-bit PCM recording. the other 12 bits are never used - therefore "wasted."
*** Usually the input gain of the recorder channel is adjusted just below clipping ***
that is fine if you can predict what the loudest part of the signal will be. in a real life situation, you often can't - that's why you need to leave about 2-4 bits of "headroom" just in case.
if you have ever recorded a live performance and successfully avoided clipping you will know what i'm talking about. i've had too many recordings spoiled by an inadvertent clip caused by too aggressive a gain setting that i stick to my 2-4 bits of headroom philosophy.
You're point misses the mark.
It's simply not an issue.Any half decent record engineer adjust his levels for a maximum input to capture the best possible dynamic range. That was true for tape and is still true for digital recording. Tape saturation was, and still is, used for effect.
Simply ask a drummer, singer or any other participant to make a loud noise and adjust your levels accordingly and add some headroom to compensate for the effect that live performances tend to get louder during the session.
Add to that the knowledge and experience and you know how to compensate for sufficient headroom.
This is just good practice and hold true for any format.
You still refer to 'waisted bits'. Which shows a lack of understanding the basic pcm theory and why it works so well for audio recording.
If you adjust your equipent for lets say a maximum loudness level of 100dB with 6db headroom. You use 23 bits dynamic range but you still encode the sample value with 24 bit precision so no bits are waisted!
Each 6dB headroom 'costs' only 1 bit.Now imagine a soft sound at 40dB during the performance. This sounds dynamic range is captured within 13 bits. (Sample values are still encoded with 24 bit precision!) Any distortion errors due to a lower signal level within the dynamic range window of your recording setup fall far below the hearing threshold.
You're point isn't valid because the end users volume level is usually adjusted in accordance with the highest loudless levels.
Suppose the users listens to this recording at 90dB loudness.
Any artifacts already below hearing threshold will get quiter another 16 dB.
*** Simply ask a drummer, singer or any other participant to make a loud noise and adjust your levels accordingly ***if only it was that simple!
if you've done a reasonable amount of live recording, you will realise it does not work like that in real life.
and never underestimate the dynamic potential of live music. particularly symphony orchestras!
i have been burned many many times.
just what do you do with a symphony that plays at 70-80dB for 95% of the time and then go to 120-130dB right at the end?
if you set the gain to peak at 130dB you will be recording 95% of the symphony at -50dB below peak. if you set the gain any higher the material will clip at the end.
"just what do you do with a symphony that plays at 70-80dB for 95% of the time and then go to 120-130dB right at the end?"Adjust for maximum input if you prefer the purist approach or use (mild) dynamic range compression to prevent trouble.
But again it isn't really a problem because the volume levels in a domestic listening environment are scaled down by the volume setting.
And as a consequence any artifacts with it.With 24 bit artifacts are already very low. With proper dither this is not an issue.
PS 120..130dB levels are hardly reached at the usual mic position.
*** Adjust for maximum input if you prefer the purist approach ***you do realise then you are effectively recording 95% of the symphony at effectively 16 bits resolution (or less), and the full 24 bits are only used for the last 5%?
*** or use (mild) dynamic range compression to prevent trouble. ***
and just exactly how do you do that? if you do it post A/D, don't bother - you've already lost the resolution and "waisted" (sic) the bits. if you do it pre A/D, you'll have to do it in the analog domain.
tell me Frank, just how many times have you actually done a recording, in live conditions? using what equipment?
***you do realise then you are effectively recording 95% of the symphony at effectively 16 bits resolution (or less), and the full 24 bits are only used for the last 5%?***That's not an issue. What is important is that the samples are 24bit accuracy. (actually 21..22bits with the best of the equipment available today)
You fail to see that distortion and artifacts are far below the signals level. If 16 bits are 'used' at a 70 dB signal then distortion is at least -90dB below that relative level.
***and just exactly how do you do that? if you do it post A/D, don't bother - you've already lost the resolution and "waisted" (sic) the bits. if you do it pre A/D, you'll have to do it in the analog domain.***
You need a compressor/limiter in the analog domain.
***tell me Frank, just how many times have you actually done a recording, in live conditions? using what equipment?
***Running out of valid arguments? :)
I have no experience with recording full orchestra's under live conditions (yet). However I don know how it's supposed to be done.
It's important to setup during a rehearsal to get a starting point for the setup. If you are surprised by a simple loudness issue during a performance of a Mahler or Wagner you didn't came prepared very well.
Frank
please don't use an analog dynamic compressor/peak limiter as you suggest.these things do serious damage. you can pretty much forget ever achieving the full potential of a 16 bit recording, let alone a 24 bit recording, if you use them.
*** You fail to see that distortion and artifacts are far below the signals level. ***no, i never said that recording at effectively 16 bits will not yield anything other than 16 bit accuracy.
you seem to be finally acknowledging *my* point, which is that 24-bit PCM recordings do not always have 24 bit resolution - it depends on the signal strength. due to the need to have headroom during the recording process, most 24-bit PCM recordings are effectively utilising less than 24 bits.
*** I have no experience with recording full orchestra's under live conditions (yet). ***
maybe you should actually try it one day. then perhaps you will appreciate what i am talking about.
i'm not saying i am an expert. i have made more botched recordings than acceptable ones. at least i understand how difficult it is to record under live conditions that make me appreciate why it is an art rather than a science, and salute those who do it well.
*** If you are surprised by a simple loudness issue during a performance of a Mahler or Wagner you didn't came prepared very well.
***perhaps, if you've actually experienced recording under live conditions you will realise it is not about not knowing the material, asking the musicians to play their loudest bits first so you can set gain levels etc etc..
no human set of musicians will play the same piece of music at exactly the same loudness levels twice.
i repeat: never underestimate the dynamic potential of live music. those who do not understand this statement have not done live recording before.
just plugging along what shes read in a pamphlet...hope she doesn't do that for a living...or that shes doing the work "pro bono" for the customers' sake...:-)..even then i dont think theyre getting their moneys worth :)
NT
you getting old man, your posts on this thread have contributed nothing but wasted space, step outside and get some fresh air.I can see you wagging your tail and meek bark of yours everytime you see fit to come in defense of your pals that need none. Your pack mentality works elsewhere...not here...move on...
NT
nt
well, wrong again...you should know the sensitivity, dynamic range, and gain applied in the chain...with this you can easily calculate what you should expect as your maximum amplitude...all these things you can find easily in the manufacturer specs (Yep, even for the mic)...the 6 dB rule is for lazy people...
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