right?
44.1 kHz or 48 kHz?
- KVRAF
- 7641 posts since 2 Sep, 2019
I believe that's the standard for delivery, not for working.
Just like CD audio delivery is 44.1kHz. No professional records at that, though.
Graphic artists don't work with 72dpi .jpg either.
THIS MUSIC HAS BEEN MIXED TO BE PLAYED LOUD SO TURN IT UP
- KVRAF
- 7641 posts since 2 Sep, 2019
It's not the converters you need to worry about so much, it's the anti-aliasing filter. If it's close to Nyquist, as 44.1kHz is, then it needs to be very steep and it causes phase ripple and artifacts (heard as "brittleness") down into the audible range, no matter how much you spent on A/D converters. The ONLY fix for that is to record at double the sample rate or more, so you can move the filter cutoff an octave or more above the audible range, and use a gentler slope.
THIS MUSIC HAS BEEN MIXED TO BE PLAYED LOUD SO TURN IT UP
- KVRAF
- 7018 posts since 19 Apr, 2002 from Utah
At least as far as sampling goes, Jamcat is correct. It is most noticeable when transposing audio. It affects samplers very audibly. That’s also why high transpose interpolation is so desirable when using a sampler. Higher sampling rates push the aliasing up outside of the audible frequency range where it can be more easily filtered out. I won’t say how essential it is when not transposing or pitch shifting audio (I’m not as confident for that part of the theory), but when at least in the case of transposing samples, Jamcat is correct to my knowledge.
Vendor‑Dependent Copy Protection: Customers lose. Pirates win.
(Also: I'm Accused of lying about Linux—it boots, runs my pro audio workflow, stays stable, updates--though yearly dismissed as “niche”. Yet I'm the deluded one.)
(Also: I'm Accused of lying about Linux—it boots, runs my pro audio workflow, stays stable, updates--though yearly dismissed as “niche”. Yet I'm the deluded one.)
- KVRAF
- 16136 posts since 13 Nov, 2012
44.1 kHz is essentially semi-pro, suitable for demos or spoken word.
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- KVRAF
- 2236 posts since 25 Dec, 2005
Says who?PatchAdamz wrote: Mon Sep 11, 2023 6:29 pm 44.1 kHz is essentially semi-pro, suitable for demos or spoken word.
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- KVRian
- 870 posts since 25 Aug, 2019
96khz has twice lower latency than 48khz, maybe the main reason it used in tracking.
Also 96khz sounds better, cleaner.
However, many pros still use 44.1khz so their mixes and recordings sounds slightly shittier.
Also 96khz sounds better, cleaner.
However, many pros still use 44.1khz so their mixes and recordings sounds slightly shittier.
- KVRAF
- 16136 posts since 13 Nov, 2012
The Nyquist frequency is 44.1kHz is 22.05kHz; we can record up to 2.05kHz above the human hearing range. In a 48kHz sample rate, the Nyquist frequency is 24kHz, so we can record up to 4kHz above the human hearing range. We might not be able to hear these frequencies, but by being recorded they can have an impact on audible frequencies and could appear in our recordings as distortions called aliasing.t3toooo wrote: Mon Sep 11, 2023 6:48 pmSays who?PatchAdamz wrote: Mon Sep 11, 2023 6:29 pm 44.1 kHz is essentially semi-pro, suitable for demos or spoken word.
Cons of 44.1kHz
On the cons side, recording with a lower sample rate could increase the risk of aliasing, as the Nyquist frequency in 44.1 kHz is 22.05 kHz, thus only giving 2.05 kHz of headroom for editing and forcing you to resort to an anti-aliasing filter.
Pros of 48kHz
A higher sample rate will make it easy to reduce the risk of aliasing, preventing the frequencies outside the Nyquist frequency limit from bouncing back to lower frequencies. That’s why some producers work with an even higher sampling frequency, such as 96kHz or 192kHz.
An audio sample rate of 48kHz also has a little more headroom for editing than a 44.1 kHz sample rate. This is useful for editing sounds with high frequencies or if you work with audio stretching.
Audio engineers and sound designers prefer higher sample rates because they have advantages for films, television, and digital video. Currently, 48kHz is the preferred sample rate for digital video.
You will find that professional studios use 48kHz or higher sampling rates for more then the science, for the ears...
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- KVRAF
- 2236 posts since 25 Dec, 2005
PatchAdamz wrote: Mon Sep 11, 2023 8:38 pmThe Nyquist frequency is 44.1kHz is 22.05kHz; we can record up to 2.05kHz above the human hearing range. In a 48kHz sample rate, the Nyquist frequency is 24kHz, so we can record up to 4kHz above the human hearing range. We might not be able to hear these frequencies, but by being recorded they can have an impact on audible frequencies and could appear in our recordings as distortions called aliasing.t3toooo wrote: Mon Sep 11, 2023 6:48 pmSays who?PatchAdamz wrote: Mon Sep 11, 2023 6:29 pm 44.1 kHz is essentially semi-pro, suitable for demos or spoken word.
Cons of 44.1kHz
On the cons side, recording with a lower sample rate could increase the risk of aliasing, as the Nyquist frequency in 44.1 kHz is 22.05 kHz, thus only giving 2.05 kHz of headroom for editing and forcing you to resort to an anti-aliasing filter.
Pros of 48kHz
A higher sample rate will make it easy to reduce the risk of aliasing, preventing the frequencies outside the Nyquist frequency limit from bouncing back to lower frequencies. That’s why some producers work with an even higher sampling frequency, such as 96kHz or 192kHz.
An audio sample rate of 48kHz also has a little more headroom for editing than a 44.1 kHz sample rate. This is useful for editing sounds with high frequencies or if you work with audio stretching.
Audio engineers and sound designers prefer higher sample rates because they have advantages for films, television, and digital video. Currently, 48kHz is the preferred sample rate for digital video.
You will find that professional studios use 48kHz or higher sampling rates for more then the science, for the ears...
That's your opinion.
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- KVRAF
- 8475 posts since 12 Feb, 2006 from Helsinki, Finland
44.1kHz vs. 48kHz makes absolutely no meaningful difference with regards to aliasing.PatchAdamz wrote: Mon Sep 11, 2023 8:38 pm On the cons side, recording with a lower sample rate could increase the risk of aliasing, as the Nyquist frequency in 44.1 kHz is 22.05 kHz, thus only giving 2.05 kHz of headroom for editing and forcing you to resort to an anti-aliasing filter.
Since both 44.1kHz and 48kHz are specced for 20kHz useful bandwidth, what 48kHz does is allow for about half as steep anti-aliasing/anti-imaging/resampling filters. This is the reason 48kHz as a standard sampling rate exists: when manufacturing the cheapest piece of junk hardware you can get away with half the filter quality and potentially save a few cents in the process.
Now... more gentle filters (and potentially half the latency for linear-phase filters) is not a bad thing for audio processing... but the idea that such a small difference in sampling rate makes any other meaningful difference is just wishful thinking.
- KVRAF
- 7641 posts since 2 Sep, 2019
At 20kHz, 2.05 kHz is not even a whole tone. So that needs to be a super steep filter. Even with the 4kHz headroom that 48kHz sample rate affords, that's still not even ⅓ an octave for the filter.PatchAdamz wrote: Mon Sep 11, 2023 8:38 pm The Nyquist frequency is 44.1kHz is 22.05kHz; we can record up to 2.05kHz above the human hearing range. In a 48kHz sample rate, the Nyquist frequency is 24kHz, so we can record up to 4kHz above the human hearing range. We might not be able to hear these frequencies, but by being recorded they can have an impact on audible frequencies and could appear in our recordings as distortions called aliasing.
Cons of 44.1kHz
On the cons side, recording with a lower sample rate could increase the risk of aliasing, as the Nyquist frequency in 44.1 kHz is 22.05 kHz, thus only giving 2.05 kHz of headroom for editing and forcing you to resort to an anti-aliasing filter.
THIS MUSIC HAS BEEN MIXED TO BE PLAYED LOUD SO TURN IT UP
- KVRAF
- 16136 posts since 13 Nov, 2012
Anti-aliasing filters are the issue here.
An anti-aliasing filter is effectively a low-pass filter that is placed on the input of an analog-digital converter (ADC). In theory, any type of active low-pass filter with unity gain can be used as an anti-aliasing filter. Any anti-aliasing filter design intends to remove high frequency content from the signal you want to sample with the goal of preventing aliasing. In this way, the goal in anti-aliasing filter design is to set the filter’s -3 dB cutoff frequency equal to half the sampling frequency.
When sampling an analog signal with a specific center frequency, aliasing will occur if the sampling frequency is less than double the center frequency. In other words, if the signal’s center frequency is more than half the sampling frequency, then aliasing will occur, and the ADC will detect a low alias frequency signal in error, rather than detecting the desired center frequency of the input high frequency signal. In the case when aliasing occurs, the ADC will output a series of digital pulses that quantifies a signal with frequency equal to the difference between the sampling rate and the center frequency. This particular sampling frequency at which the system can prevent aliasing is called the Nyquist frequency.An anti-aliasing filter is effectively a low-pass filter that is placed on the input of an analog-digital converter (ADC). In theory, any type of active low-pass filter with unity gain can be used as an anti-aliasing filter. Any anti-aliasing filter design intends to remove high frequency content from the signal you want to sample with the goal of preventing aliasing. In this way, the goal in anti-aliasing filter design is to set the filter’s -3 dB cutoff frequency equal to half the sampling frequency.
When sampling an analog signal with a specific center frequency, aliasing will occur if the sampling frequency is less than double the center frequency. In other words, if the signal’s center frequency is more than half the sampling frequency, then aliasing will occur, and the ADC will detect a low alias frequency signal in error, rather than detecting the desired center frequency of the input high frequency signal. In the case when aliasing occurs, the ADC will output a series of digital pulses that quantifies a signal with frequency equal to the difference between the sampling rate and the center frequency. This particular sampling frequency at which the system can prevent aliasing is called the Nyquist frequency.
The higher the sample rate, the less need for anti-aliasing filters and thus, the main reason for higher sampling rates.
An anti-aliasing filter is effectively a low-pass filter that is placed on the input of an analog-digital converter (ADC). In theory, any type of active low-pass filter with unity gain can be used as an anti-aliasing filter. Any anti-aliasing filter design intends to remove high frequency content from the signal you want to sample with the goal of preventing aliasing. In this way, the goal in anti-aliasing filter design is to set the filter’s -3 dB cutoff frequency equal to half the sampling frequency.
When sampling an analog signal with a specific center frequency, aliasing will occur if the sampling frequency is less than double the center frequency. In other words, if the signal’s center frequency is more than half the sampling frequency, then aliasing will occur, and the ADC will detect a low alias frequency signal in error, rather than detecting the desired center frequency of the input high frequency signal. In the case when aliasing occurs, the ADC will output a series of digital pulses that quantifies a signal with frequency equal to the difference between the sampling rate and the center frequency. This particular sampling frequency at which the system can prevent aliasing is called the Nyquist frequency.An anti-aliasing filter is effectively a low-pass filter that is placed on the input of an analog-digital converter (ADC). In theory, any type of active low-pass filter with unity gain can be used as an anti-aliasing filter. Any anti-aliasing filter design intends to remove high frequency content from the signal you want to sample with the goal of preventing aliasing. In this way, the goal in anti-aliasing filter design is to set the filter’s -3 dB cutoff frequency equal to half the sampling frequency.
When sampling an analog signal with a specific center frequency, aliasing will occur if the sampling frequency is less than double the center frequency. In other words, if the signal’s center frequency is more than half the sampling frequency, then aliasing will occur, and the ADC will detect a low alias frequency signal in error, rather than detecting the desired center frequency of the input high frequency signal. In the case when aliasing occurs, the ADC will output a series of digital pulses that quantifies a signal with frequency equal to the difference between the sampling rate and the center frequency. This particular sampling frequency at which the system can prevent aliasing is called the Nyquist frequency.
The higher the sample rate, the less need for anti-aliasing filters and thus, the main reason for higher sampling rates.