How does Nyquist's theory apply to digital recording of opera music?

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Nyquist's theory states that to accurately reconstruct a digital recording of sound, one must sample at least twice the highest frequency present in the original signal. In the context of opera music, which comprises various instruments and voices, the sampling rate should be at least twice the highest frequency produced by any of these sources. Most modern recordings of opera are typically done at a sample rate of 96 kHz, allowing for frequencies up to 48 kHz, which exceeds human hearing limits. While sampling at twice the highest frequency is theoretically sufficient for reconstruction, higher sampling rates can improve the quality of the recording by reducing digital noise and enhancing fidelity. Ultimately, for practical purposes, sampling at 96 kHz or higher is recommended to ensure high-quality audio reproduction.
  • #31
Michael C said:
Now is a good time to ask the question: what use is sampling the sound at 192 kHz or 384 kHz?

One of the biggest problems with recreating a digitally recorded audio signal with extreme high fidelity is the low pass or anti-aliasing filter that has to be implemented to ensure that a very minimum amount of energy ever gets above the Nyquist frequency.

The low pass filter in the case of 44 KHz or even 88 KHz and 96 KHz needs to be very, very steep. That introduces some severe phase distortion that varies sharply with frequency. Golden eared audiophiles with high quality analog equipment can definitely hear that, especially with sounds that are percussive such as triangles and cymbals. It's not just the individual sounds from each instrument or voice that matter but the spatial and timing relationships between them and sharp filters do some strange things with those. (I have experience as both a physicist and professional recording and production engineer) The higher sample rates reduce the requirements for the filter and the result should be much reduced phase distortion in the higher audio frequencies.

One of the smarter things that can be done is to upsample the signal to 192 KHz or 384 KHz and then apply the filter though that's not quite as refined as having the original signal there already.
 
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  • #32
Howdy folks.

First post here. I should qualify it by saying that I am a recording engineer, not a scientist.

From a practical standpoint, sampling rate depends as much on delivery format and track count as anything else when it comes to recording audio.

Most adult humans can't hear much above 16 kHz, but we go with the assumption that the average human can hear up to 20 kHz. Can you perceive anything above that? Hard to say, but most evidence points to the fact that higher sampling rates sound better, because the slope of the anti-aliasing filter can be relaxed so that there are no ripples or resonances in the audio band, and the cutoff frequency can be above 20 kHz, rather than in the audio band.

44.1 kHz and 48 kHz were decided upon 30 years ago, since the first commercially available digital audio recorders were adapted video decks, and we seem to be stuck on multiples of these for PCM encoding.

These sampling rates fulfilled two criteria:
1 - they were fast enough to recreate almost the entire bandwidth of human hearing (with rather sharply sloped anti-aliasing filters);
2 - they could function with the frame rates of the video recorders that were being used;

A lot of classical recordings, if they are using PCM, will use a sample rate of 96 kHz. Few (if any) of my colleagues are specifying 192 kHz or above. If track counts are high, chances are it will be recorded at 44.1 kHz for CD release, sometimes 48 kHz.

Nyquist/Shannon works. If you have listened to a CD in the past 30 years and heard all of the instruments, then you can be a witness. Are there some overtones or ultrasonic components missing? Maybe. But it must be said that most microphones and audio processors don't even pass audio signal much beyond 20 kHz. Even the best of the best might have a 50 kHz bandwidth at most. Your speakers sure don't reproduce anything that high.

Which leaves only the quality of the ADC and DAC. High-quality analog components; stable, low-jitter clocks, temperature stability, and such all make far more a difference in the quality of reproduction than the sampling rate.

I'll take a Genex or Mytek converter at 44.1 over a "soundcard" at 384 kHz any day, and if you heard the difference, you would too.
 
  • #33
RobAnderson said:
A lot of classical recordings, if they are using PCM, will use a sample rate of 96 kHz. Few (if any) of my colleagues are specifying 192 kHz or above. If track counts are high, chances are it will be recorded at 44.1 kHz for CD release, sometimes 48 kHz.

A few days ago I participated in a classical concert being recorded for the German radio network. I had a chat with the chief recording engineer, who told me that 48 kHz at a depth of 24 bit is still the standard for radio recordings here. He doesn't see any reason for this standard to change, since the microphones don't have any appreciable output above 24 kHz. He said that some colleagues use 96 kHz for classical recordings, but he would challenge anybody to hear the difference. Higher sampling rates, according to him, only make sense if you are using a lot of digital effects, which isn't the case in a classical recording where you are trying to reproduce the original as faithfully as possible.
 
  • #34
Higher sampling rates, according to him, only make sense if you are using a lot of digital effects, which isn't the case in a classical recording where you are trying to reproduce the original as faithfully as possible.

Yep - 48 kHz is typically used for broadcast or visual media. Bit depth is an important part of this discussion, and probably makes more of a difference than sampling rate - especially in terms of reproducing the dynamic range found in classical music. 24-bit resolution vs 16-bit makes a much bigger difference than 48 vs. 44 kHz sampling rate.

Although his argument is counter to the way many folks on this side of the pond seem to think. For classical recordings, the lower track counts, the desire to maintain the utmost fidelity, and the reluctance to do much in the way of digital processing often leads folks to go with the 88 or 96 k sampling frequencies, while the pop and rock productions often go with the lower sampling rates because of the need for extreme amounts of processing on many tracks.
 
  • #35
Well, you can always run ADC at 384KHz with 8x oversampling, followed by digital anti-aliasing filter and downsampling to 48KHz for transmission/storage. Same thing on the playback side - digitally upsample to 384KHz using proper sinc interpolation before sending it to the DAC. This way analog filters are a lot less critical, noise and jitter are much reduced, basically you get a couple of extra bits of resolution.
 

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