4 ms·

There’s one thing that bothers me about this. Sure, PCM sampling is a lossless representation of the low frequency portions of a continuous signal. But it is no

by amluto 10mo ago

There’s one thing that bothers me about this. Sure, PCM sampling is a lossless representation of the low frequency portions of a continuous signal. But it is not a latency-free representation. To recover a continuous signal covering the low frequencies (up to 20kHz) from PCM pulses at a sampling frequency f_s (f_s >= 40kHz), you turn each pulse into the appropriate kernel (sinc works and is ideal in a sense, but you probably want to low-pass filter the result as well), and that gives you the decoded signal. But it’s not causal! To recover the signal at time t, you need some pulses from times beyond t. If you’re using the sinc kernel, you need quite a lot of lookahead, because sinc decays very slowly and you don’t want to cut it off until it’s decayed enough. So if you want to take a continuous (analog) signal, digitize it, then convert back to analog, you are fundamentally adding latency. And if you want to do DSP operations on a digital signal, you also generally add some latency. And the higher the sampling rate, the lower the latency you can achieve, because you can use more compact approximations of sinc that are still good enough below 20kHz. None of this matters, at least in principle, for audio streaming over the Internet or for a stored library — there is a ton of latency, and up to a few ms extra is irrelevant as long as it’s managed correctly when at synchronizing different devices. But for live sound, or for a potentially long chain of DSP effects, I can easily imagine this making a difference, especially at 44.1ksps. I don’t work in audio or DSP, and I haven’t extensively experimented. And I haven’t run the numbers. But I suspect that a couple passes of DSP effects or digitization at 44.1ksps may become audible to ordinary humans in terms of added latency if there are multiple different speakers with different effects or if A/V sync is carelessly involved.

Sesse__ 10mo ago
This is all true, but it is also true for most _other_ filters and effects, too; you always get some added delay. You generally don't have a lot of conversions in your chain, and they are more on the order of 16 samples and such, so the extra delay from chunking/buffering (you never really process sample-by-sample from the sound card, the overhead would be immense) tends to be more significant.
wildzzz 10mo ago
Wouldn't each sample be just an amplitude(say, 16bit), not a since function? You can't recover frequency data without a significant number of pulses but that's what the low pass filter is for. Digital audio is cool but PCM is just a collection of analog samples. There's no reason why it couldnt be an energy signal.
- amluto 10mo ago
  This is the sampling theorem. You start with a continuous band-limited signal (e.g. sound pressure [0], low-pass filtered such that there is essentially no content above 20kHz [1]). You then sample it by measuring and recording the pressure, f_s times per second (e.g. 48 kHz). The result is called PCM (Pulse Code Modulation). Now you could play it back wrong by emitting a sharp pulse f_s times per second with the indicated level. This will have a lot of frequency content above 20kHz and, in fact, above f_s/2. It will sounds all kinds of nasty. In fact, it’s what you get by multiplying the time-domain signal by a pulse train, which is equivalent to convolving the frequency-domain signal with some sort of comb, and the result is not pretty. Or you do what the sampling theorem says and emit a sinc-shaped pulse for each sample, and you get exactly the original signal. Except that sinc pulses are infinitely long in both directions. [0] Energy is proportional to pressure squared. You’re sampling pressure, not energy. [1] This is necessary to prevent aliasing. If you feed this algorithm a signal at f_s/2 + 5kHz, it would come back out at f_s - 5kHz, which may be audible.
  - brewmarche 10mo ago
    > Now you could play it back wrong by emitting a sharp pulse f_s times per second with the indicated level. This will have a lot of frequency content above 20kHz and, in fact, above f_s/2. It will sounds all kinds of nasty. Wouldn’t the additional frequencies be inaudible with the original frequencies still present? Why would that sound nasty?
    - amluto 10mo ago
      Because the rest of the system is not necessarily designed to tolerate high frequency content gracefully. Any nonlinearities can easily cause that high frequency junk to turn back into audible junk. This is like the issues xiphmont talks about with trying to reproduce sound above 20kHz, but worse, as this would be (trying to) play back high energy signals that weren’t even present in the original recording.

by amluto 10mo ago

Sesse__ 10mo ago

This is all true, but it is also true for most _other_ filters and effects, too; you always get some added delay. You generally don't have a lot of conversions in your chain, and they are more on the order of 16 samples and such, so the extra delay from chunking/buffering (you never really process sample-by-sample from the sound card, the overhead would be immense) tends to be more significant.

wildzzz 10mo ago

Wouldn't each sample be just an amplitude(say, 16bit), not a since function? You can't recover frequency data without a significant number of pulses but that's what the low pass filter is for. Digital audio is cool but PCM is just a collection of analog samples. There's no reason why it couldnt be an energy signal.

amluto 10mo ago
This is the sampling theorem. You start with a continuous band-limited signal (e.g. sound pressure [0], low-pass filtered such that there is essentially no content above 20kHz [1]). You then sample it by measuring and recording the pressure, f_s times per second (e.g. 48 kHz). The result is called PCM (Pulse Code Modulation). Now you could play it back wrong by emitting a sharp pulse f_s times per second with the indicated level. This will have a lot of frequency content above 20kHz and, in fact, above f_s/2. It will sounds all kinds of nasty. In fact, it’s what you get by multiplying the time-domain signal by a pulse train, which is equivalent to convolving the frequency-domain signal with some sort of comb, and the result is not pretty. Or you do what the sampling theorem says and emit a sinc-shaped pulse for each sample, and you get exactly the original signal. Except that sinc pulses are infinitely long in both directions. [0] Energy is proportional to pressure squared. You’re sampling pressure, not energy. [1] This is necessary to prevent aliasing. If you feed this algorithm a signal at f_s/2 + 5kHz, it would come back out at f_s - 5kHz, which may be audible.
- brewmarche 10mo ago
  > Now you could play it back wrong by emitting a sharp pulse f_s times per second with the indicated level. This will have a lot of frequency content above 20kHz and, in fact, above f_s/2. It will sounds all kinds of nasty. Wouldn’t the additional frequencies be inaudible with the original frequencies still present? Why would that sound nasty?
  - amluto 10mo ago
    Because the rest of the system is not necessarily designed to tolerate high frequency content gracefully. Any nonlinearities can easily cause that high frequency junk to turn back into audible junk. This is like the issues xiphmont talks about with trying to reproduce sound above 20kHz, but worse, as this would be (trying to) play back high energy signals that weren’t even present in the original recording.