I found there are multiple ways how to play pre-recorded sound files. For example [Tabread4~], [tabreadsf~] are kind of ready-made players, we just need to add inputs and outputs. [phasor~] on the contrary seems more basic, like low level process, principle (?) that determine playback.

1. can you please explain me how phasor based playback works?
2. if [phasor~] is sound signal and it can play audio file, can I use one sound file (instead of phasor) to play another one?
3. what is [phasor~]? I ask it because of PC everything is just numbers an function, so what this [~] symbol means or how it differ from non-audio date (control?)?

I know these are different questions, but I just want to know possible playback methods and sound file modulations options in PD

@Transcend Of course. It is all just math. You can do what you wish.
This will randomly mess up the read...... mess.pd
Of course, the more you mess it up, the less it will be in any way meaningful to your ears.
David.

@alexandros Yes, but sound file is two dimension array, isn't it? Where X is number of samples(time line) and Y is a numbers of different frequencies, that is spectrum. As I understand phasor~ takes control over X values, but not Y?

@whale-av What exactly happens in this patch?

@Transcend Yes, [phasor~] just scans through the indexes at a desired speed and over a desired range, and [tabread4~] returns the sample values of those indexes at the audio rate.
[phasor~] actually sends a ramp..... 0 to 1, fall instantly to 0 and another ramp...... so a sawtooth 0 to 1 at the frequency that is sent to its left inlet. So to get it to play all the indexes its output needs to be multiplied by the total samples.
The frequency needs to be set that the time for each ramp is the same as the normal time length of the sample....... for playback at normal speed. Changing the frequency will speed up or slow down playback, and inverting the ramp (horizontally) will play backwards.
[mess] uses some math which changes as random numbers arrive........
The [*~ 1] will change the start point and the speed at the same time. They could be separated using a [+~] as well, but I kept it reasonably simple. The speed changes the frequency, but does no stretching (to maintain the time length of what is played) so higher pitch will be shorter, lower pitch longer.
The Grain slider changes the rate at which random numbers arrive, and [random] spews any number between zero and the size of the array..... a value that it gets from [soundfiler].
The [phasor~] gets its frequency from dividing the samplerate (I assumed 44100) by the number of samples, so that if nothing was messed up it would simply play the file at the correct speed from beginning to end repeatedly.
The [spigot] allows the same random numbers to change the frequency of [phasor~] as well.
You can reset that by closing the spigot and clicking the [44100( message. The clicking of the message could be automated by a bang when the spigot is turned off like this........ mess.pd
David.

@whale-av said:

@Transcend Yes, [phasor~] just scans through the indexes at a desired speed and over a desired range, and [tabread4~] returns the sample values of those indexes.

What are these values tabread4~ returns at any index? how pitch can be a single value? Within human hearing range from 20Hz to 20KHz there are a lot o different values. It should return all those values simultaneously at given index, isn't it?

@Transcend Ahh..... that is where the confusion arises. What is being read is individual samples.... at 44100Hz sample rate there are 44100 every second. The pitch is heard because of the rate at which they rise and fall in value.
In Pd a sine wave at 100 Hz would have a sample value of zero (maybe, it depends at what time the wave starts) at index 0, and then rising sample values to a value of 1 at sample number 110 (more or less) falling back to a sample value of 0 at sample 220, continuing to fall to -1 at sample 330 and then rising back to a value of zero at sample number 441....... etc. etc.
When those values are read by [tabread4~] the output would....... eventually, once it gets to your speakers..... push and pull the speaker smoothly 100 times a second, and be heard as 100Hz. Depending on the bit-depth of the dac the actual sample values later in the chain will be much larger, but Pd max/min values are +1/-1 at the output to the [dac~]. Your soundcard and Pd take care of that automatically.

If there is another sound, say at 1KHz., that would modulate that 100Hz wave..... it would look like a ripple on the 100Hz wave........ and so on. A music track will look a real mess when you look at it, with only loud / quiet parts really recognisable. But our ears, or more especially our brains, can make total sense of it.

Samplerate (audio) objects are used to set the indexes because they send a value at the samplerate...... so 44100 times a second...... so every single index is sent to [tabread4~]
Digital audio (FFT) is fiendishly complicated to understand, but this gets you quite a long way into it in only a few minutes........ https://medium.com/@djtech42/explanation-of-sample-rate-in-digital-audio-and-breakdown-of-misconceptions-38f912fb3b1f
Actually it gets you a very long way toward a good understanding in just a few minutes.......
David.

)) wow!

1. As I understand now, for any theoretically possible sound two points or cycle between two points is necessary. And this cycle can be loosely compared with smallest auditory unit (like atoms or anything potentially capable of being experienced). So one index is not enough (that was my FAIL!)
   Your example modified:
   Now In PD (44100 sample rate) if we take sine wave at 22 000 Hz each step from one sample to the next will be equal to one cycle or one possible bit of something audible. But one cycle of sine wave at 20 Hz will go through 4410 indexes(or something like that?)

2. but anyway tabread4~ somehow reads whole complex of that FFT and there should be simultaneous values? Or this asumption is also wrong?

@Transcend Yes. [tabread4~] interpolates using 4 consecutive samples. It uses the one before and the two following the one it is assessing.... and in effect it guesses (by calculation) what the original analogue waveform would have been.
Because samples are taken by an Adc at precise times the individual samples, especially for high frequencies, are necessarily NOT the peak and trough values..... so by doing some math it tries to estimate the original. The more math is done the closer it will get, but it rapidly becomes very expensive in processing. Failure is called aliasing. Usually those frequencies likely to fail are filtered out in the Adc, but of course Pd can make its own. Digital video and even fonts on your computer screen have the same problem...... data points that fall between two possible digital representations of the original.
It is likely that a decent Dac in a quality soundcard will do it better, along with a lot of other clever stuff.
Included in the Pd doc folder in your installation is a sort of tutorial. Interpolation is demonstrated in "B04.tabread4.interpolation.pd"
Work your way through that doc folder (3.audio.examples)..... and it will gently introduce you to FFT..... although you will then probably need to do some external reading as well.
David.

@whale-av Thanks, David!!!