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#41
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"Ric Oliva" wrote in message ... Ok, so I understand that 44.1k is 44,100 samples per second and 48k is 48,000 samples per second. Obviously 48,000 is better. Correct. The more samples of a waveform you can gather, the easier it is to reproduce it. I'm not exactly sure what bit rate is though? CDs are 16 bit, DVDs are 24. What exactly does that mean though? That's not bit rate, but rather bit depth. MP3 files and other audio file formats are often saved in a format determined by "bit rate", so as to determine the number of bits per actual chunk of time--hence, an HTTP download of an MP3 file can be streamed with highest quality possible if the bit rate is predetermined. Bit depth, however, is the vertical sampling resolution of an audio sample. As you know, 44kHz sample rate is the number of samples per second. On a horizontal waveform drawing, this is the horizontal resolution (think screen resolution on your monitor, 640x480 vs. 1024x768). Bit depth is the number of bits of information per sample, or the vertical resolution. For example, in an 8-bit sample bit depth, a waveform's amplitude in a particular sample can be in any of 256 possible positions (2*8 = 256). Obviously, that's a very small number of possibilities. So a 16-bit sample bit depth the resolution is much higher: 65536. Bit rate is determined by the two resolutions combined. Theoretically, an MP3 file saved with a 64kbps bit rate means that the file must be downloaded from the Internet at a rate of at least 64 kilobits per second in order for it to be streamed through the MP3 player without hiccupping. Another question - if I'm recording a project to audio CD, is it better to just record at 16/44 since that's what the CD will be anyway, and I can save system resources? or should I do 24/48 and then dither it down, essentially changing what I originally heard? In general, with audio it is better to sample at high depth and resolution and then downgrade afterwards than it is to stay true to the final output. Jon |
#42
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"Jay - atldigi" wrote in message
In article , "Arny Krueger" wrote: "Jay - atldigi" wrote in message In article , (White Swan) wrote: Let's take a 1 bit system. Now we have only two volume values: full volume, and full silence. using the 6dB per bit formula, we have values of 0dB and 6 dB. Something to ponder: Why does DSD (SACD) work? Because it's a different data stream and it's not PCM. But how do they make 1 bit work if it can only be off or full 6dB? The point is that digital doesn't actually work that way. As you no doubt know, there are many forms of digital coding that aren't anything like PCM. A common non-PCM digital coding method is PDM or Pulse Duration Modulation, which is frequently used by switchmode power amps. The classic non-PCM coding scheme that most closely resembles DSD is PDM, or Pulse Density Modulation. A DSD data stream is composed of pulses that are basically integrated to produce an analog signal. Pulses have a value of either +1 or -1. Alternate pulses with opposite polarities sum out to zero. If the pulses are predominately +1, then the integrated signal goes positive. The more predominately the pulses are +1, the faster the integrated signal goes positive. If the pulses are predominately -1 then the integrated signal goes minus, and so on. |
#43
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Arny Krueger wrote:
A DSD data stream is composed of pulses that are basically integrated to produce an analog signal. Pulses have a value of either +1 or -1. Alternate pulses with opposite polarities sum out to zero. If the pulses are predominately +1, then the integrated signal goes positive. The more predominately the pulses are +1, the faster the integrated signal goes positive. If the pulses are predominately -1 then the integrated signal goes minus, and so on. Didn't that used to be called 1-bit DPCM? |
#44
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , "Arny Krueger"
writes: 24 bits also adds resolution in any region between -144 dB and full scale. For me, with my limited understanding (or misunderstanding perhaps) of digital theory, the above sentence cuts to the heart of the matter. If I understand what Scott Dorsey and others have said then the change from 16 to 24 bits only adds downward dynamic range and does not increase resolution of signals in the relatively high ranges close to full scale. Maybe I misunderstand but thats what they seem to be saying. On an intuitive level that seems wrong to me and it seems as though resolution even at -10dB should increase (due to less quantization error??) . I infer that thats what Arny is saying in the quote above. Do I have that right? If so, is this, then, the crux of the discussion? I hope you all will bear with my lack of math skills and technical knowledge but I would like to understand as much about this as I can on an intuitive level. Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
#45
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
On Mon, 17 Nov 2003 22:44:52 GMT, Jay - atldigi
wrote: Something to ponder: Why does DSD (SACD) work? Because it's a different data stream and it's not PCM. But how do they make 1 bit work if it can only be off or full 6dB? The point is that digital doesn't actually work that way. But it *is* PCM. Sampling rate and word depth can be traded pretty liberally, I think is your point. Greater word depth reduces the ambiguity of quantization, and ambiguity is "like" noise. I still have two problems with the discussion so far. First, Bob Cain's caveat remains unaddressed: "Within the Nyquist criterion a signal can be produced with any arbitary phase or delay until you consider the quantization of the samples. Then the achievable delays become quantized as well " Second, quantization degrades the theoretical perfection of Nyquist criterion signals to a greater extent for smaller signals, IOW, the conversion is not monotonic. Thanks for your excellent comments, Chris Hornbeck new email address "That is my theory, and what it is too." Anne Elk |
#46
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
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#47
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Chris Hornbeck
writes: Greater word depth reduces the ambiguity inherent in quantization. Thanks Chris, So is this point a matter of contention or is this agreed upon by all? If it is agreed upon then is the argument on the "24 bit sounds no better than 16 bit" side that the effects of the ambiguity are inherently negligable or perhaps that interpolation or something else "repairs" the ambiguity adequately? I hope I have framed thew question well enough to be understood. Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
#48
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
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#49
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
On Sun, 16 Nov 2003 22:57:17 GMT, Jay - atldigi
wrote: The myth is the dynamic equivalent to the argument that 4 samples on a 20kHz sine wave will render it more accurately than 2, and 8 samples even more so. That's not true either. But 4 samples will render it more accurately in *time* than two. Alternatively, a smaller quantization step will also. Quantization could be said to "jitter" the conversion in either time or amplitude. Not such a much, except that the effect's greater for small signals. Thanks, Chris Hornbeck new email address "That is my theory, and what it is too." Anne Elk |
#50
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Carey Carlan wrote in message .205...
"Tommi" wrote in : So, if you're recording, say, someone's vocals at both 16 and 24 bits, and the peaks are at -6dB to 0dB FS, does the 24 bit recording represent more accurately the signal in that region than the 16-bit version? The extra 8 bits give you 48 db more dynamic range between EVERY sample. Between sample value = 0 and sample value = 1 they give you an extra 48 db on the bottom end. On the loud end, 16 bit max value is 32767 (0x7FFF), second value is 32766 (0x7FFE). That equates to 24 bit values 8388352 (0x7FFF00) and 8388096 (0x7FFE00), a difference of 256 values, the equivalent of 48 dB dynamic range. I don't think so. The decibel is used to measure differences between two levels, thus, following the definition U (in decibels) = 20 * log (U_1 / U_0) for Voltages, the relative difference is 0.0006 decibels. So, the 8 bits give you 48 dB more difference between signal and noise, which is a measure for the accuracy, at any level. Which, incidentally, makes 24 bits with 144 dB dynamic range sufficient to accurately reproduce any noise between 0 dB sound pressure level, namely the hearing threshold level, and the sound of a starting jet, which is said to be about 140 dB spl. Assuming of course that the hearing threshold level is also the hearing threshold of detecting differences in amplitude between two tones at other levels. Hendrik |
#51
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Chris Hornbeck
wrote: On Sun, 16 Nov 2003 22:57:17 GMT, Jay - atldigi wrote: The myth is the dynamic equivalent to the argument that 4 samples on a 20kHz sine wave will render it more accurately than 2, and 8 samples even more so. That's not true either. But 4 samples will render it more accurately in *time* than two. Alternatively, a smaller quantization step will also. Quantization could be said to "jitter" the conversion in either time or amplitude. This is a contention of some, but when the system is viewed as a whole including proper dither, time resolution essentially becomes infinite. People Bob Stuart and Tom Holman have suggested that the time issue is significant for imaging if you are dealing with two or more channels. Other digital audio heavy hitters and PhDs point out the seldom understood fact that the right dither has an effect on the time resolution as well as the preventing of truncation distortion we know and love it for. So it is not agreed upon that 4 samples render it more accurately in time. In fact, the science seems to be against it. It's too bad JJ formerly from AT&T isn't around anymore to offer up all the empirical data for us. I wouldn't mind Dick Piecrce or Dave Collins making an appearance either. None of this means that higher sample rates and more bits don't sound better. It's just that the explanations often given are flawed, and the requirements demanded are often excessive. 64kHz 20 bit sampling is probably the minimum necessary. 96/24 offers a margin of safety. More than that may actually cause more problems than it solves. As stated before - upsamling for non linear processes, oversampling, and signal processing with double precision are not the same subject and can be beneficial. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#52
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Chris Hornbeck
wrote: On Mon, 17 Nov 2003 22:44:52 GMT, Jay - atldigi wrote: Something to ponder: Why does DSD (SACD) work? Because it's a different data stream and it's not PCM. But how do they make 1 bit work if it can only be off or full 6dB? The point is that digital doesn't actually work that way. But it *is* PCM. Sampling rate and word depth can be traded pretty liberally, I think is your point. Greater word depth reduces the ambiguity of quantization, and ambiguity is "like" noise. DSD can essentially be thought of as PCM that is 1 bit with a very high sample rate, severely noise shaped, and greatly reduced filter worries (I'd say no filters, but many manufacturers suggest using some filtering to prevent the high level ultrasonics due to the noise shaping from damaging equipment). The noise shaping is one reason the high sample rate helps so much; you can shape all that noise far away from your desired passband (20-20k) and get the equivalent of around a 120 dB signal to noise ratio within that limited bandwidth. However, the noise in the ultrasonic range is ridiculous. It's like recording right off the oversampling ADC without taking a trip through quantization. It's 1 bit. That's all there is. But it seems to work, doesn't it? It doesn't just output 6dB square waves. It plays music. This couldn't be possible under the criteria that some are trying to impose. But it works. It works due to the trade off Chris alludes to. But there's no free lunch. All that noise has to go somewhere, but after it does, you can hear music in the 20-20k range, even though there's only 1 bit. And it's not because of the "rising or falling" illustration used with 1 bit recording or you'd never be able to start a song in the middle. I know, it's hard to wrap your head around, but viewing a digital audio system as a whole instead of in pieces, and taking into account proper design and implimentation, digital audio works, even though some of the concepts seem pretty wierd in practice. Some things when illustrated on paper are good for learning and visualization, but you eventually have to move away from the illustrations and into the wierd world where there are no stairsteps and you only gain more dynamic range with more bits. Add filtering, use crappy filters, forget to dither, or use the wrong dither, and it can all fall apart, and all of these isolated evils that people worry about can actually come into being. Do everything right and this stuff isn't a problem within the limits of the system, which admittedly do exist. The filter issues (ripple, group delay, ringing, noise or poor performance from the analog stages) that certainly affect the audible band in the CD standard, poor filters which may not entirely prevent aliases or images (that would be **** poor design with no excuse these days) or amplify the normal filter issues, the noise floor which also can be heard in the CD standard and prevent the lowest level details from being captured (assuming the source didn't have a bunch more noise to begin with), and poor practice in preparing the masters, whether it be dither problems or poor processing due to bad algorithms or insufficient resolution (we won't even mention clocking and jitter), and you have plenty of land mines to screw you up. 24/96 solves or lessens some of those problems, gives us some margin of safety, and offers possibilities for currently unused, simpler filter techniques that could really help. I'm not saying 24/96 isn't better than 16/44.1. I'm just saying that many reasons we see given aren't always correct, and the extent of the problems are sometimes overstated. What do the details mean to the guy just recording some music? Not much usually. But from the standpoint of tecnical learning, the fine distinctions are worth making. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#53
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Jay - atldigi wrote:
White Sawn's satement seems to indicate that the extra bits are within the same dynamic range, thereby giving you greater detail within that range. You can't into the trap of viewing digital audio like it's digital imagery. Unfortunately, 24 bits leaves the top 96db range of 16 bit alone, but lowers the noise floor and allows the recording of audio events that are even smaller, at a lower level, i.e. below -96dB. Arny Krueger wrote: 24 bits puts 16 extra levels between each pair of levels that exist with 16 bits. Thus, the resolution is increased at any level, not just the smallest one. In article , (Scott Dorsey) wrote: Not really. It gives you more dynamic range, which is often wasted anyway. 96 dB is an awful lot. Scott's not arguing what you're arguing there, Jay. He's just curmudgeoning about the fact nobody's going to use the available dynamic range anyway. Which is a different discussion altogether. Arny's right on this one. While it's true that the additional bits tack your extended resolution onto "the bottom" of the dynamic range, it clearly increases the resolution at all levels. You can have a -100dB component to a -1dB signal, and you still want to hear it. The simplest way to think about this is to imagine a digital audio recording that contained two simultaneous sounds: One a -1dBFS and the other at -111dBFS. It should be clear that in a 16-bit recording, the -111dBFS sound will be buried in the noise floor and will not be heard, while in a 24-bit recording it will be above the theoretical noise floor. Now, suppose I told you that the -1dBFS signal was my guitar; and the -111dBFS signal was some subtle overtone of that guitar sound. Maybe it's some fret buzz, maybe it's some room reflection. Now it should be obvious that whether or not you hear the -111dBFS signal will affect the level of detail you hear in the -1dBFS signal. Once again I present my favorite digital audio analogy, cash denominations: Having some pennies in your pocket allows you to pay a more precise amount even if you're spending thousands of dollars. ulysses |
#54
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Thanks for this explanation, Arny. Idunno if it helped Tommi but it
helped me. Every time I finally grasp another "big concept" in digital audio I'm always amazed at how incredibly clever it is. Those old French mathmen must have been giddy as hell when they figured this stuff out. ulysses In article , Arny Krueger wrote: "Tommi" wrote in message I may well be suffering the myth, but my understanding is that it matters whether you sample a sine wave 2 or 8 times. Tests have been made where subjects had to determine which sound came first from their headphones. The same signal was fed to both L and R channels, only the other one was delayed by 5-15 _micro_seconds. Some of the people were able to "localize" the sound source even when it was delayed only by 5 microseconds. This implies that a sampling rate of 192kHz(which results in 5.2 microsecond's sample intervals), for example, is not only pushing the nyquist rate to the ultrasonic range, but also presents better channel separation on multichannel systems. For the purpose of discussion, I'll stipulate that your facts are correct to this point. I really don't know that, but it would help me make an important point if we don't argue over that part of your comments. So, it doesn't necessarily matter if you sample a sine wave 2 or 8 times on a mono system, but on a multichannel system higher sample rates result in better localization. The myth here is that signals in a digital system can have interchannel timing differences that are only integer numbers of sample periods. IOW this myth as applied to 44,100 Hz sampling is that interchannel timing differences can only be multiples of 22.675736961451247165532879818594 microseconds. I agree that this seems to be intuitively clear. But it is also quite wrong. The myth comes from the idea that two signals in different channels that are displaced in time can only be expressed as the same set of sample values, but time-shifted. This is not the case. Two signals in different channels that are displaced in time can be expressed as different sample values. For example, if two slowly-increasing (ramp) signals are displaced in time, one signal might have a set of sample values that starts out 0, 10, 20, 30... This is a ramp that starts at t = 0. The time-delayed version of this signal in another channel could have a set of values that is 0 at t = 0, but is 5, 15, 25... for successive samples. If you looked at these two signals over time, you'd say that the second signal is time-shifted by an amount of time equal to half a sample period. And, that is how it would sound. The correct time resolution of sampled signals is the sample period divided by the number of distinct amplitude levels. In the case of 16/44 this would be 5.1418904674492623958124444033093e-10 seconds or 514.18904674492623958124444033093 picoseconds. This is a tiny, tiny number. In reality, it is lost in the noise. |
#55
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Garthrr wrote:
So is this point a matter of contention or is this agreed upon by all? If it is agreed upon then is the argument on the "24 bit sounds no better than 16 bit" side that the effects of the ambiguity are inherently negligable or perhaps that interpolation or something else "repairs" the ambiguity adequately? I hope I have framed thew question well enough to be understood. The fact that it isn't agreed upon by all doesn't mean it's really a matter of contention. This is a math question, and it has a correct answer. The disagreement only comes from those who don't know the correct answer. That sounds snotty, but I'm saying it aside from any declaration of who's right and who's wrong. I'm saying that after hashing it out, it's not going to be a matter of opinion. Furthermore, I don't think by any means that Jay is claiming 24 bits sounds not better than 16 bits. If you read his discussion on his website, he very clearly considers more than 16 bits necessary for transparent audio. He's simply saying that the benefit comes only in the lowest audio levels. In a way he's right and in a way he's wrong. It sounds like he's saying that increased bit depth can't add any resolution to loud sources, that it only adds the ability to reproduce quiet sounds. But I think he knows that a loud source can have quiet elements in it. Music is complex and it can be thought of as a bunch of simultaneous sounds at multiple amplitudes and frequencies. If Jay is suggesting that the benefit of "8 more bits" only exists when there are no signals above -96dBFS present, then he is wrong. If he is saying that the "increased resolution" on a full-scale signal is nothing more than the added ability to resolve the quietest overtones, then he's right and is actually in full agreement with Arny. This is one of those areas where describing audio in words gets kind of tricky. ulysses |
#56
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin
Ulysses Morse wrote: saying that increased bit depth can't add any resolution to loud sources, that it only adds the ability to reproduce quiet sounds. But I think he knows that a loud source can have quiet elements in it. Music is complex and it can be thought of as a bunch of simultaneous sounds at multiple amplitudes and frequencies. I could swear I've already said that in this thread... maybe that was another thread. that the benefit of "8 more bits" only exists when there are no signals above -96dBFS present Of course not... "increased resolution" on a full-scale signal is nothing more than the added ability to resolve the quietest overtones, then he's right and is actually in full agreement with Arny. At least somebody understands me, but I thought I had already said this somewhere in the thread. It's those quieter components that you are getting from the extra bits. The louder components aren't represented any better. In the end, it can be a more precise and better sounding recording (provided the source is of a quality to benefit), but it's because of the little things you can now record, not that the big ones are better. So late... falling asleep... good night... -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#57
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin
Ulysses Morse wrote: Arny Krueger wrote: 24 bits puts 16 extra levels between each pair of levels that exist with 16 bits. Thus, the resolution is increased at any level, not just the smallest one. In article , (Scott Dorsey) wrote: Not really. It gives you more dynamic range, which is often wasted anyway. 96 dB is an awful lot. Scott's not arguing what you're arguing there, Jay. He's just curmudgeoning about the fact nobody's going to use the available dynamic range anyway. Which is a different discussion altogether. I think he's saying both, but I'd have to let him speak for himself. Arny's right on this one. Unless I misunderstand him (certainly possible), I don't think so. While it's true that the additional bits tack your extended resolution onto "the bottom" of the dynamic range, it clearly increases the resolution at all levels. You can have a -100dB component to a -1dB signal, and you still want to hear it. But that's exactly my point: only the -100 component is what you've gained. The -1 component is not rendered any better than it was before. The simplest way to think about this is to imagine a digital audio recording that contained two simultaneous sounds: One a -1dBFS and the other at -111dBFS. It should be clear that in a 16-bit recording, the -111dBFS sound will be buried in the noise floor and will not be heard, while in a 24-bit recording it will be above the theoretical noise floor. Right. That's what I said, isn't it? The noise floor goes down and you can record smaller events; they don't necessarily have to be a fundamental that is very low - it can be low level overtones making a violin sound more real, or a little incidental sound, or the sound of the hall and the natural reverb tail, but it's still the low level stuff that you are gaining at the higher bit depth. It doesn't mean that the whole recording has to stay below -96 dB. Have I not made this clear? I guess not; or it was buried too deeply in the 16 bit noise floor... Once again I present my favorite digital audio analogy, cash denominations: Having some pennies in your pocket allows you to pay a more precise amount even if you're spending thousands of dollars. But the value of the dollars don't change when you add a few pennies. The total value changes, but each dollar is still worth a dollar. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
#58
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Jay - atldigi wrote:
At least somebody understands me, but I thought I had already said this somewhere in the thread. It's those quieter components that you are getting from the extra bits. The louder components aren't represented any better. In the end, it can be a more precise and better sounding recording (provided the source is of a quality to benefit), but it's because of the little things you can now record, not that the big ones are better. See, you are in full agreement with Arny. It just depends on whether you're thinking of the music as a collection of sounds or one big sound. As a collection of sounds, your extra bits are only revealing the quiet ones; the loud components were already represented by 16 bits. But when you step back and listen to the whole thing, what that MEANS is greater detail in the music, even where it's loud. ulysses |
#59
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin Ulysses
Morse writes: Furthermore, I don't think by any means that Jay is claiming 24 bits sounds not better than 16 bits. If you read his discussion on his website, he very clearly considers more than 16 bits necessary for transparent audio. He's simply saying that the benefit comes only in the lowest audio levels. Actually I was under the impresion that Jay was on the other side of the fence--that he was saying that 24 bit was really no better than 16 bit for any sort of real world audio. Perhaps I misunderstood his stance. I thought it was Arny who was contending that there is resolution to be gained by 24 bit, resolution which exists even in the not-so-low level signal. Disclaimer: Please forgive me if I accidently put words in anyone's mouth while trying to paraphrase something. If I do its just my ignorance of the subject matter and not any desire to spin. In a way he's right and in a way he's wrong. It sounds like he's saying that increased bit depth can't add any resolution to loud sources, that it only adds the ability to reproduce quiet sounds. But I think he knows that a loud source can have quiet elements in it. Music is complex and it can be thought of as a bunch of simultaneous sounds at multiple amplitudes and frequencies. If Jay is suggesting that the benefit of "8 more bits" only exists when there are no signals above -96dBFS present, then he is wrong. If he is saying that the "increased resolution" on a full-scale signal is nothing more than the added ability to resolve the quietest overtones, then he's right and is actually in full agreement with Arny. This is one of those areas where describing audio in words gets kind of tricky. ulysses Thanks Ulysses. There is still a nagging question I have about all this but I want to try to think of a way to phrase it properly. Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
#60
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin Ulysses
Morse writes: Once again I present my favorite digital audio analogy, cash denominations: Having some pennies in your pocket allows you to pay a more precise amount even if you're spending thousands of dollars. Following this analogy -- and I just know I'm gonna be wrong here but this is just how it seems to me -- if we say, for example, that 16 bit audio is like having a pocket full of 10 dimes then isnt 24 bit audio a pocket full of 100 pennies? Finer divisions of the same whole--the ability to describe finer voltage differences? I understand that the dynamic range increases with higher bit depth and I guess in this money analogy we could think of that as having a dollar fifty or something instead of the original dollar but it still seems like you get finer resolution even in the first dollar. Is this question not analogous to the number of pixels in a digital photograph? The more pixels, the higher resolution the picture (all else being equal). That being analogous to bit depth in audio then the rate of frames per second in a moving picture would be analogous to sample rate. Is that a reasonable comparison? Sooner or later I'll phrase this question in enough different ways as to clearly communicate what I want to ask! Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
#61
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"S O'Neill" wrote in message
Arny Krueger wrote: A DSD data stream is composed of pulses that are basically integrated to produce an analog signal. Pulses have a value of either +1 or -1. Alternate pulses with opposite polarities sum out to zero. If the pulses are predominately +1, then the integrated signal goes positive. The more predominately the pulses are +1, the faster the integrated signal goes positive. If the pulses are predominately -1 then the integrated signal goes minus, and so on. Didn't that used to be called 1-bit DPCM? Here are block diagrams of a DPCM coder and decoder http://ce.sharif.edu/~m_amiri/Projects/MWIPC/dpcm1.htm On page 7 of http://www.hit.bme.hu/people/papay/edu/Acrobat/DSD.pdf there is a block diagram of a DSD decoder. Don't look the same to me. |
#62
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Garthrr wrote:
Following this analogy -- and I just know I'm gonna be wrong here but this is just how it seems to me -- if we say, for example, that 16 bit audio is like having a pocket full of 10 dimes then isnt 24 bit audio a pocket full of 100 pennies? Finer divisions of the same whole--the ability to describe finer voltage differences? Yes. For the sake of discussion, let's say you have a thousand dollars in either dimes or pennies. Now, say you're going to make a single purchase of something that costs $1.87. At 16 bits, you're forced to tell the clerk to "keep the change" because all you have are dimes. No big deal, you're out $0.03. You'll never miss it. Most of us wouldn't bother stooping to pick up three pennies. But suppose you're going around town buying a whole bunch of different things, and every time you do, you have to say, "keep the change." Eventually, it starts to add up and you wish you had some pennies. Suppose you record live to 2-track at 16 bits and you just make a single "transaction" where you maybe run an EQ, a gain boost, and a little peak limiting all in one pass. You're using 24-bit DSP but you have to stuff the result back into a 16-bit package. Not a real big deal, your "clerk" rings you up and says that'll be $45.58. You only have to say "keep the change" once. But now what if you've got a bunch of different processes to run, incrementally, that you evaluate before you move on to the next process? Maybe you're multi-tracking and you're processing each track differently. There's all kinds of "keep the change" adding up. In fact, it's not only "adding up" but it's also "multiplying up." The error in your first process will get multiplied in your next step. I guess that would be something like if you bought 1000 of something that should cost $0.13 apiece but since you're paying in dimes you're paying $0.20 apiece. Suddenly you're out $70. Your accountant is gonna be ****ed. This is why more processing means you should start with more bits. But you know, decimal places on a calculator is probably a better analogy. Should I start again? ulysses I understand that the dynamic range increases with higher bit depth and I guess in this money analogy we could think of that as having a dollar fifty or something instead of the original dollar but it still seems like you get finer resolution even in the first dollar. Is this question not analogous to the number of pixels in a digital photograph? The more pixels, the higher resolution the picture (all else being equal). That being analogous to bit depth in audio then the rate of frames per second in a moving picture would be analogous to sample rate. Is that a reasonable comparison? Sooner or later I'll phrase this question in enough different ways as to clearly communicate what I want to ask! Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Garthrr wrote:
Actually I was under the impresion that Jay was on the other side of the fence--that he was saying that 24 bit was really no better than 16 bit for any sort of real world audio. Perhaps I misunderstood his stance. I thought it was Arny who was contending that there is resolution to be gained by 24 bit, resolution which exists even in the not-so-low level signal. You've understood Arny correctly but have gotten the wrong idea about what Jay is trying to say. On his website he advocates going to at least 20 bits for a release format, or better yet 24 bits. The perceived disagreement (which turned out not really to be a disagreement at all, if you ask me) revolved around the *reason* for needing more bits. Jay seems to think 24 bits is better than 16 bits because of the extra low-level resolution. I actually disagree about the need for more bits in the delivery medium. I think 16 is enough to deliver the full fidelity of any real-world finished production, even though 24 bits are needed during tracking, mixdown, and mastering. ulysses |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Ok, considering the post below, then the question is "Where is the disagreement
between the two camps-- the one camp who says 16 bit is as good as 24 bit for anything but very, very low level audio and the other camp that says 24 bit is better even at higher recording levels? Where is the point at which the two camps begin to disagree? Garth~ In article , Justin Ulysses Morse writes: Yes. For the sake of discussion, let's say you have a thousand dollars in either dimes or pennies. Now, say you're going to make a single purchase of something that costs $1.87. At 16 bits, you're forced to tell the clerk to "keep the change" because all you have are dimes. No big deal, you're out $0.03. You'll never miss it. Most of us wouldn't bother stooping to pick up three pennies. But suppose you're going around town buying a whole bunch of different things, and every time you do, you have to say, "keep the change." Eventually, it starts to add up and you wish you had some pennies. Suppose you record live to 2-track at 16 bits and you just make a single "transaction" where you maybe run an EQ, a gain boost, and a little peak limiting all in one pass. You're using 24-bit DSP but you have to stuff the result back into a 16-bit package. Not a real big deal, your "clerk" rings you up and says that'll be $45.58. You only have to say "keep the change" once. But now what if you've got a bunch of different processes to run, incrementally, that you evaluate before you move on to the next process? Maybe you're multi-tracking and you're processing each track differently. There's all kinds of "keep the change" adding up. In fact, it's not only "adding up" but it's also "multiplying up." The error in your first process will get multiplied in your next step. I guess that would be something like if you bought 1000 of something that should cost $0.13 apiece but since you're paying in dimes you're paying $0.20 apiece. Suddenly you're out $70. Your accountant is gonna be ****ed. This is why more processing means you should start with more bits. But you know, decimal places on a calculator is probably a better analogy. Should I start again? ulysses "I think the fact that music can come up a wire is a miracle." Ed Cherney |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin Ulysses
Morse writes: I actually disagree about the need for more bits in the delivery medium. I think 16 is enough to deliver the full fidelity of any real-world finished production, even though 24 bits are needed during tracking, mixdown, and mastering. Yeah this is a good point IMO. The reason is that in the real world where things are sometimes done in a hurry and levels are not always set to optimum, having that overkill is a good thing. Not to mention the fact that tracks get EQed and compressed all to hell which of course brings up the noise floor and exposes low level signals more than they would be otherwise. I can think of a session I did a week ago where the drummer used brushes on one song very quietly and I didnt feel like resetting the levels of all ten tracks so I just left them, knowing that I would be fine with my 24 bit system. In the delivery medium you can be pretty sure (especially these days with the "level wars") that the full potential of the medium is going to be exploited. Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
Justin Ulysses Morse wrote:
| |While it's true that the additional bits tack your extended resolution |onto "the bottom" of the dynamic range, it clearly increases the |resolution at all levels. You can have a -100dB component to a -1dB |signal, and you still want to hear it. Is the ear even capable of hearing the -100 component against the much louder -1? I thought masking pervented this. Phil |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"Justin Ulysses Morse" wrote in message m... Thanks for this explanation, Arny. Idunno if it helped Tommi but it helped me. Every time I finally grasp another "big concept" in digital audio I'm always amazed at how incredibly clever it is. Those old French mathmen must have been giddy as hell when they figured this stuff out. ulysses Yes, it was indeed an informative reply from Arny! |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
wrote in message ... Justin Ulysses Morse wrote: | |While it's true that the additional bits tack your extended resolution |onto "the bottom" of the dynamic range, it clearly increases the |resolution at all levels. You can have a -100dB component to a -1dB |signal, and you still want to hear it. Is the ear even capable of hearing the -100 component against the much louder -1? I thought masking pervented this. Phil Masking, it is frequency-dependent. However, this leads to thinking about the fact that the human ear actually compresses dynamics at higher sound pressures. My understanding is that we have roughly 80dB's worth of dynamic range at a time, which we then move according to the sound pressure levels of the sound sources. For example, if you'd be listening something at 110dB SPL for 5 minutes, after that you couldn't hear the same sound with 2dB SPL for a while. It works the other way round too: If you're listening something at 5dB spl for a while, and then suddenly the same sound source produces a 120dB spl sound, your ear would compress it lower(by stretching the eardrum, moving the hammer away from it etc) in order to protect your hearing mechanism. This, however isn't true with very short peaks because your protection mechanism takes some time to wake up. |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin
Ulysses Morse wrote: Jay - atldigi wrote: At least somebody understands me, but I thought I had already said this somewhere in the thread. It's those quieter components that you are getting from the extra bits. The louder components aren't represented any better. In the end, it can be a more precise and better sounding recording (provided the source is of a quality to benefit), but it's because of the little things you can now record, not that the big ones are better. See, you are in full agreement with Arny. It just depends on whether you're thinking of the music as a collection of sounds or one big sound. As a collection of sounds, your extra bits are only revealing the quiet ones; the loud components were already represented by 16 bits. But when you step back and listen to the whole thing, what that MEANS is greater detail in the music, even where it's loud. ulysses Perhaps all the extra discussion gets in the way of the simple truths. Here's the most simple way I think my point can be stated: 16 bits is perfectly capable of reproducing 96 dB of dynamic range. With dither, the system is linear. You can get better than that. It's linear within 96 dB (a little less if you count the dither's added noise floor, a little more if you count what the ear can hear within the noise floor due to averaging of noise in our brain). You can't get better than "what you put in is what you get out". However, that's not all there is to audio, and we can hear about 120 dB of dynamic range, so 16 bits can be a limitation and 20 or 24 can certainly sound better. It doesn't have to be in an area where there's nothing above -96 either. however, it in no way makes 16 bits' limited 96db range any less accurate. That's the point people seem to miss. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Justin
Ulysses Morse wrote: Suppose you record live to 2-track at 16 bits and you just make a single "transaction" where you maybe run an EQ, a gain boost, and a little peak limiting all in one pass. You're using 24-bit DSP but you have to stuff the result back into a 16-bit package. Not a real big deal, your "clerk" rings you up and says that'll be $45.58. You only have to say "keep the change" once. But now what if you've got a bunch of different processes to run, incrementally, that you evaluate before you move on to the next process? Maybe you're multi-tracking and you're processing each track differently. There's all kinds of "keep the change" adding up. You're really making the case for higher intermediate wordlengths. If you have a 16 bit file, preferably you'll process even higher than 24 bits. Let's take 48 bit for purposes of discussion. Between processes, however, if you keep "stuffing" it back to 16 as you say, then you indeed are going to have trouble, especially cumlatively. In the best case scenario, a DAW would hand 48 bits from one process to the next and you'd never come back down until delivery. Unfortunately, most DAWs, even ones that process at 48, hand 24 bit words between processors (some do allow 32 float to be saved as an intermediate). Also, external processors, even those that work at greater than 24 bits of precision, can only receive and transmit 24 bit words as AES and SPDIF etc. only support up to 24 bit words. So, you dither from 48 to 24 before handing it off, but don't drop below 24 until delivery, and that clerk won't be keeping your change. This is not to say that nothing above 16 bit capture or delivery is ever beneficial - just that this is more the argument for processing with longer wordlengths. Simply staring with a 16 bit file to process doesn't mean you have to go back to 16 after every process you apply. -- Jay Frigoletto Mastersuite Los Angeles promastering.com |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
"Tommi" wrote in message
"Justin Ulysses Morse" wrote in message m... Thanks for this explanation, Arny. Idunno if it helped Tommi but it helped me. Every time I finally grasp another "big concept" in digital audio I'm always amazed at how incredibly clever it is. Those old French mathmen must have been giddy as hell when they figured this stuff out. ulysses Yes, it was indeed an informative reply from Arny! blush |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
wrote in message
Justin Ulysses Morse wrote: While it's true that the additional bits tack your extended resolution onto "the bottom" of the dynamic range, it clearly increases the resolution at all levels. You can have a -100dB component to a -1dB signal, and you still want to hear it. Is the ear even capable of hearing the -100 component against the much louder -1? I thought masking prevented this. The threshold of reliable perception of spurious signals and noise is on the order of from -60 to -70 dB when the music has reasonably sustained peaks at 0 dB. This is one reason why it's fair to say that the practical benefits of going past 16 bits are non-existent at high levels. |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Jay -
atldigi writes: Absolutely not. I know there's a lot of posting going on, and I've written a lot in this thread, but I know I've stated several times that the above is not what I'm saying. Instead of adding even more confustion, please try to go back and read my posts again. Sorry Jay. I was hoping to avoid that. Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
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16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain
In article , Jay -
atldigi writes: I think you're making the mistake of assuming that the "low level stuff" means that if the average recording level is above 16 bits' -96dB limit then the extra bits are somehow not helpful. No, I understand why you would think that but I am aware of what you mean in that there are low level components to audio with a high average level and I understand that they would benefit from the added dynamic range. That scenario, however is not the crux of my question. Garth~ "I think the fact that music can come up a wire is a miracle." Ed Cherney |
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