61–70 of 326 posts
Earlier quoted context omitted.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz You are going to have to provide me with a citation to back that up because that goes against everything I've learned and experience in 17 years of working in acoustics.
Humans will hear the impact > 20kHz frequency has on the lower frequencies, not the 30kHz frequency itself. That's been proven a million times.
Earlier quoted context omitted.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz You are going to have to provide me with a citation to back that up because that goes against everything I've learned and experience in 17 years of working in acoustics.
Humans will hear the impact > 20kHz frequency has on the lower frequencies, not the 30kHz frequency itself. That's been proven a million times.
What I would love to have is: independent instrument/vocals tracks along with a default recommended "mix". The default mix would be used for normal playback and independent tracks would be great for custom mix / karaoke etc. Is this too unrealistic to expect? Has something like this been tried before?
There are two reasons I don't think this will happen: 1. People would use the tracks to create custom remixes which they would then distribute. What happens when a remix becomes more popular than the original track? Artists generally have to pay other artists to remix their songs (usually via royalties). 2. Creativity. When an artist creates something they want you to hear it the way it was intended. Allowing you to…
And another important remark: some artists are flattered when someone asks them to make a remix for their song. (Imagine you're an artist and your idol asks you to make a remix of his song.)
There is no point with going over 16 bits, but there is definitely a point with going over 44.1khz, as it allows you to actually reproduce waveforms more accurately than 44.1khz. Try reproducing f.e. a sinewave accurately over 4-5khz with a sample rate of just 44.1khz - it cannot be done, and at this point we haven't even taken into account the issue of varying slew-rate characteristics of the thousands or so differe…
The signal reproduced from your 44.1kHz sampled digital input is not a stair-step like some broken waveform editor might display: On output it goes through a matched reconstruction filter (which may, in fact, be digital and involve an oversampled DAC or it could be analog though those are harder to build without compromise). After the reconstruction filter the output is _EXACT_, assuming the input only contained energy below the nyquist (well, and was sufficiently far away from the reconstruction lowpass).
So even a 5khz sine wave is reproduced perfectly with 44.1kHz sampling.
Earlier quoted context omitted.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz You are going to have to provide me with a citation to back that up because that goes against everything I've learned and experience in 17 years of working in acoustics.
Humans will hear the impact > 20kHz frequency has on the lower frequencies, not the 30kHz frequency itself. That's been proven a million times.
Then you should be able to provide at least one citation.
Earlier quoted context omitted.
Hey cmer, thanks for posting I don't think I understand quite what you're saying and wondered if you could explain more. You and the article both say that humans can't hear above about 20kHz. If there are higher frequencies that create a harmonic at a lower frequency (e.g. a 33kHz harmonic that produces a sound at 16.5kHz) then surely that lower harmonic (16.5kHz in this case) will be recorded by the original recordi…
Let's make things super simple. Let's say you record 4 sine waves at a 192kHz sampling rate: 15kHz, 30kHz, 45kHz and 60kHz. All 4 frequencies will be captured and the 15kHz frequency will sound different to your hear because its harmonics. If you take this recording and master it for a CD (44.1kHz), you'll effectively get up to ~20kHz (since they're a low pass filter starting at around 16-18kHz). This means that only…
Or am I wrong, and the ear is able to detect frequencies above 20kHz?
Earlier quoted context omitted.
The article doesn't suggest only using 48kHz for recording and mixing. I don't think the author would disagree that recording triangles is difficult. He would argue that once you've decided what final audible frequencies you want to present to the listener, the best way to distribute them is at 16-bit 44.1/48kHz. It's a compelling case.
I completely disagree with the article having heard the difference many times myself. You can't record at 192kHz and hope to keep the same quality by distributing the final mix in 44.1kHz. It just doesn't work that way.
Earlier quoted context omitted.
Let's make things super simple. Let's say you record 4 sine waves at a 192kHz sampling rate: 15kHz, 30kHz, 45kHz and 60kHz. All 4 frequencies will be captured and the 15kHz frequency will sound different to your hear because its harmonics. If you take this recording and master it for a CD (44.1kHz), you'll effectively get up to ~20kHz (since they're a low pass filter starting at around 16-18kHz). This means that only…
> Harmonics are never exactly double, triple the fundamental. Those would be mostly inaudible. But you get the idea. Actually, they are: https://en.wikipedia.org/wiki/Harmonics
An extreme example is present on modern pianos, where the high rigidity of the loud, heavy piano strings can cause tuners to stretch the lowest and highest notes as much as a half-semitone so that their harmonics are in tune with the note the next octave down or up. In other words, the first harmonic on the lowest note of a piano can be as much as 1/2 of a note sharp.
And when your oscillator is no longer one-dimensional, most harmonics aren't even close to integer multiples. The harmonics of bells, cymbals and drums are all over the place. That's what gives them their percussive sound. (Edit: some of these modes of vibration aren't harmonics in the linear sense.)
He raises a lot of valid points. However... 192 kHz is clearly overkill for listening. Not so for further editing of the data. Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible. 44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad. A bit more headroom is well needed there. That bit about intermodulation distortion is complete…
Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible No, the difference is not audible at all. At 16 bits of depth on a normal low-level audio signal (~0.3 volts), we're talking about less than 0.000005 volts per amplitude step. This difference gets lost in the THD already at the DAC in your audio output stage. Then it gets lost again in the amplifier. And again in the cable to y…
He raises a lot of valid points. However... 192 kHz is clearly overkill for listening. Not so for further editing of the data. Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible. 44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad. A bit more headroom is well needed there. That bit about intermodulation distortion is complete…
Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible No, the difference is not audible at all. At 16 bits of depth on a normal low-level audio signal (~0.3 volts), we're talking about less than 0.000005 volts per amplitude step. This difference gets lost in the THD already at the DAC in your audio output stage. Then it gets lost again in the amplifier. And again in the cable to y…