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kilgore88
New member
What exactly does this mean? And what does it affect? Someone please send some knowledge my way. Thanks!
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M.Brane
New member
The higher the bitrate the more dynamic range you have to work with i.e. you have a lower noise floor. Obviously this is more important for music that has a wide dynamic range than for your average rock/metal stuff.
The higher the samplerate, the more frequency response you get. While CDs at 44.1 are good to 22.5khz (well above the hearing of the average person) it's a good idea to track, and mix at a higher rate to avoid artifacts/anomolies that can occur near the nyquist freq. especially during processing.
The higher the samplerate, the more frequency response you get. While CDs at 44.1 are good to 22.5khz (well above the hearing of the average person) it's a good idea to track, and mix at a higher rate to avoid artifacts/anomolies that can occur near the nyquist freq. especially during processing.
BigRay
New member
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frankieballsss
New member
whats the dynamic range difference from 16 bit to 24 bit, in headroom? Like how many decibels more of headroom would you get from 24 bit? Am I thinking correctly here?
BigRay
New member
frankieballsss said:
The FAQ above covers this: www.24bitfaq.org
he overriding concept here is called dynamic range, and is measured in dB. The dynamic range of a recording is the difference between its loudest point and its quietest point.
To elaborate further, each bit gives us the ability to represent about 6dB of dynamic range. A passage that is 6dB louder than another passage is said to be twice as loud as the other passage. In the 4-bit example, we theoretically have 24dB of dynamic range that can be used. But what if recording doesn’t take advantage of all that dynamic range? What if the recording never peaks beyond 6dB of its maximum possible limit? In this case, the recording would only take advantage of 3 of what we call the least significant (or left-most) bits, meaning 18dB of dynamic range. 16-bit recordings are capable of a theoretical maximum limit of 96dB of dynamic range. This means that a single wave could have up to 65536 discrete values that can be used to represent it. But if the same wave recorded at 16-bit peaks at 48dB below its maximum possible limit, then there would only be 256 discrete values that can be used to represent it, taking advantage of only 8 of the least significant bits. The 8 most significant bits would contain no information whatsoever, and would remain unused. In the case of 24-bit recording, you’d have a maximum of 16,777,216 values to choose from, and in the case of a wave peaking at 48dB below its maximum possible limit, the wave would still have 65536 possible discrete amplitude values that could be used to represent it.
Now, have you ever heard any of the early 8-bit computer recordings that floated around in the early days of home computers? Didn’t they sound just awful? I mean, you were impressed because you had a snippet of music that you could recognize playing from your computer, but you wouldn’t want to listen to it for more than a minute or two. I personally remember playing back an 8-bit digitized 5 second snippet of Van Halen’s rendition of the Kinks’ “You Really Got Me” over and over again on my Atari 800 until I couldn’t take it anymore. The thrill soon had me building an 8-bit digitizing device with a microphone input jack and a connector for the joystick port. Ah, those were the days… but I digress.
Perhaps many are more familiar with 8-bit audio from real-time internet sources like RealAudio. It’s good enough for speech recognition, but leaves all too much to be desired for music.
Now here’s the kicker in the16-bit realm. While the volume level of a recorded low-E note struck on an acoustic guitar might take advantage all 16 available bits (for instance, where the peak on the DAT deck reaches 0dB), the squeak of the fingers on the string, the scratch of the pick hitting the string, and the 5 or 10 audible harmonic overtones of that note may never reach a point beyond 48dB shy of the 96dB maximum. Yes, all of these additional by-products of that low-E string that make the guitar sound alive and compelling receive all of the fidelity of that scratchy, distorted, computerized sound of that 8-bit sample from long ago. And as the basic low-E note fades out, it too gets the same butcher treatment from the ever decreasing number of discrete amplitude values. Yikes!
Now record with a 24-bit word length, and put the CD quality back into those string squeaks, pick scratches, and overtones. With 24 bits, you can hear the clarity of the cymbals decaying as they keep ringing smoothly down to complete silence. The little low-level smack of the bass pedal head hitting the bass drum skin that sounded barely like a small click before (if audible at all) now sounds like a smack, complete with its own smoothly reverberating decay. Even the low-level acoustical reflection from the wall behind the band now contributes to the experience with added detail and a sense of ambience, not simply low-level distortion. Finally, because of this improvement, no more does the recordist have to risk overloading and clipping the recording in effort to achieve maximum fidelity. Levels can be set conservatively with the assurance that a high degree of fidelity is maintained.
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Robert D
New member
By the way, when you're done having fun with the math.......most of us have come to the conclusion that 24 bits at 44.1k is the format that makes the most sense for a bound for CD project. SRC (sample rate conversion) introduces a degree of quality loss that at the very least negates any quality gain from recording at 48Khz, and higher sample rates like 96K are so resource intensive that it's usually not practical to use across 24 tracks or more. But if you have the gear to support it, it's worth doing some A/B tests of your own.
Cheers, RD
Cheers, RD
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frankieballsss
New member
Thanks for the info.
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kilgore88
New member
So, is 24-bit going to sound alot better than 16-bit? Is it worth buying a new interface that supports 24-bit/96Khz?
Blue Bear Sound
New member
The very short answer is yes....kilgore88 said:
The long answer is that it depends on your engineering skills and the rest of your gear. If you can't already get good sound out of a modest 16-bit-based rig, then it's quite likely that going to 24-bit is not going to help you at all.
Most people underuse the equipment they have, mistakenly thinking that it's the gear that gives them the sound. More often, it's what's happening BEFORE the gear that's the issue.
Toki987
Rock Steady
M.Brane said:
Alan, what is "the nyquist freq" ?
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M.Brane
New member
Ken:
In a nutshell it's the frequency cutoff point of you're top end The theory states that in order to prevent aliasing at high frequencies you have sample at twice the rate of your desired frequency hence to get 22.5khz you have to sample at 44.1khz.
In theory it's pretty simple, but in practice it's a little more complicated due to the nature of analog low-pass filters. This is why higher quality converters cost more.
In a nutshell it's the frequency cutoff point of you're top end The theory states that in order to prevent aliasing at high frequencies you have sample at twice the rate of your desired frequency hence to get 22.5khz you have to sample at 44.1khz.
In theory it's pretty simple, but in practice it's a little more complicated due to the nature of analog low-pass filters. This is why higher quality converters cost more.
Toki987
Rock Steady
M.Brane said:Ken:
In a nutshell it's the frequency cutoff point of you're top end The theory states that in order to prevent aliasing at high frequencies you have sample at twice the rate of your desired frequency hence to get 22.5khz you have to sample at 44.1khz.
In theory it's pretty simple, but in practice it's a little more complicated due to the nature of analog low-pass filters. This is why higher quality converters cost more.
Thanks man. It makes good sense. I didnt know it had a name for the phenomena though. Let me spin it back in my words to see if I`m on it. If the sampling rate is not fast as the frequencies themselves approach the number recordings per sec in each sample, it`s possible, that if the sampling hits are not in perfect proper phase or time, with the frequency pulses themselves some or or all could be missed with the dip in the pulse gettin recorded instead of the pulse itself. Doubling the shots, or greater garrantees that each pulse in a frequency gets recorded fully, including the rise and fall of each pulse in the frequency. ?
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cultureofgreed
New member
Robert D said:
Just about any Modern CPU and Harddisk have more then enough power to do a normal session of 24 to 48 tracks at 24bit 96K. In fact, I hit a track count of 80 in Cubase at 24bit 96K on my old AMD XP 2800+ 1 gig of ram with a 7200RPM IDE hard disk before my I got any errors.
With the new 16meg cache 7200RPM SATA drives and the new 64bit AMD I imagine a track count of 100+ in Cubase at 24 bit 96K would not be unreasonable.
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M.Brane
New member
Toki987 said:
I read that a few times, and I'm still a bit fuzzy on it.
In a perfect world you could have a perfectly band-limited signal. Of course this is not the case in practice because even the steepest of low-pass filters has a slope, and leaves some signal above the cutoff frequecy. Steep filters also produce phase shift, and a peak just before the cutoff that can cause harmonic ringing.
So since the actual A/D conversion takes place after the filter the quality of the filter makes all the difference. Otherwise you're sampling a highly distorted signal.
Farview
Well-known member
That would really depend on how many plugins you are using. If you have a quality EQ on everything and compressors on half with a few verbs going, the computer will choke. The hard drive may be up to the task of reading the files, but the processing power will be cut in half going to 96k.cultureofgreed said:
Just about any Modern CPU and Harddisk have more then enough power to do a normal session of 24 to 48 tracks at 24bit 96K. In fact, I hit a track count of 80 in Cubase at 24bit 96K on my old AMD XP 2800+ 1 gig of ram with a 7200RPM IDE hard disk before my I got any errors.
With the new 16meg cache 7200RPM SATA drives and the new 64bit AMD I imagine a track count of 100+ in Cubase at 24 bit 96K would not be unreasonable.
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cultureofgreed
New member
Farview said:
Plugins don't lower track count in a session, CPU power lowers the amount of plugins you can have available at any time. The CPU has little or nothing to do with how fast a larger file (as in 96k) is read off a drive as compared to smaller files (44.1k). Hard drive I/O speed is the primary determinant of the actual number of tracks you can playback or record at once.
Of course, you could be talking about the 96k processing inside the plugins themselves, which could increase CPU load. In practice, however, I think the benefits outweigh the cost. In my experience this does not cut my CPU power for plug ins in half at 96k compared to 44.1k. YMMV
If your not recording at 96k strictly because your afraid of the CPU load IMHO your missing out.
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Farview
Well-known member
To me, the tracks without the plugins are useless. Who cares how many tracks you can have if you can't do anything with them?
I've recorded at 96k and I don't find the sound quality of the finished product worth the limitations of available processing power and doubling my harddrive space. Maybe once i upgrade my computer, it won't be a big deal. I really don't hear more than a very subtle difference between 44.1 and 96k and it usually gets washed out in mastering.
I've recorded at 96k and I don't find the sound quality of the finished product worth the limitations of available processing power and doubling my harddrive space. Maybe once i upgrade my computer, it won't be a big deal. I really don't hear more than a very subtle difference between 44.1 and 96k and it usually gets washed out in mastering.
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Mo-Kay
Dragon Soul Productions
I actually agree with BlueBear!
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cultureofgreed
New member
Farview said:
Some people also claim that the sound quality is degraded when you truncate the sample rate down to 44.1, or that 88.2 is better because it is exactly double of 44.1 and is easier math. Those are the biggest gripe I have heard about 96K.
I find that I cannot tell the difference between a single track at 44.1 and a single track at 96K. I can tell the difference between 24 tracks at 96K and 24 tracks at 44.1 though. It all gets butchered in the end when it ends up on a CD anyways.
But for sure, 24bit is the way to go.
Blue Bear Sound
New member
No friggin' way!!!Mo-Kay said:I actually agree with BlueBear!
