@lacuna [phasor~] only resets at block boundaries..... https://forum.pdpatchrepo.info/topic/3699/phasor-with-sample-accurate-phase-reset
So this might be what you need...... vphasor~.zip
I think this was the previous discussion...... https://forum.pdpatchrepo.info/topic/12865/samphold-noise-phasor-noise-round-off-error-or/3
David.

@ddw_music wrote:

phasor~ for reset, the period is not 100% exact

Now I remember long time ago there has been a thread about very slow <1Hz [osc~] LFO being off.

@y0g1 wrote:

I count only 256 samples, I assume it will be precise then ?

Probably yes.

phasor~is-off.pd

@y0g1 said:

Here "phasor~"'s phase is reset when receiving a non zero signal.
The use case is resetting a sample counter.

Phasor~in puredata doesn't act the same,
Does anyone have a clue how to make a vanilla adaptation ?

I forget who is the pd forum user who taught me this trick, but, here's a neat way to have a signal reset a phasor:

[phasor~]
|  \
|  [samphold~]
|  |
[-~]
|
[wrap~]

Then the reset trigger goes into the right inlet of samphold~ (but IIRC it should be inverted, because samphold samples its input signal when its trigger signal decreases, see its help file).

Always works (think about it a minute, and it becomes clear why).

Btw phasor * sample rate is not safe as a sample counter, due to floating point rounding error. (Maybe ok for one second.) In pd vanilla, [rpole~ 1] can serve as an accumulator but resetting it at signal rate is a bit trickier. I'm not at the computer now but I could track that down later. Or use cyclone's count~.

hjh

@alexandros said:

Am I missing something, or the patch below isn't correct?

[phasor~ 1]
|
[threshold 0.5 10 0.5 10]
|
[o]

This does output a bang whenever [phasor~] resets.

Nice, provided this operates as suggested (haven't tested), this does precisely what I'm looking for.

I literally started with PD last week, so I just haven't encountered the threshold module to consider (I used Max/MSP back in the Nato.0+55+3d days, supercollider in the early 2000s, and patch modulars extensively since the Nord Modular days, but ignored PD for the most part under an impression it lacked refinement).

My aim is quite simple, embarrassingly so: I'm learning how to play back samples loaded into arrays, and wanted to test my bpm calculations after changing the phasor's speed. As a test I wanted to run a metro at the quarter note time of the new BPM, triggering a kick sample, and confirm the hits are falling where I expect them to,... but then I ran into the issue of getting my metro to start precisely in time with the looping sample (which is just freely looping endlessly via a phasor). Since the phasor~ lacked a phase output, I was stumped, and figured getting the array index = 0 = bang would be a simpler approach.

Per the randomF, the multiplicative scaling by 0.1 works, but then limits my range to 0-1. The use of a second random added to these values would fix this easily, but didn't seem as efficient as finding something that might already exist. Good to hear else has a "rand.f" (or whatever is typed above). I have this external already installed, but haven't exhaustively worked through what is included.

Thanks again, one and all.

p.s. Per the playback of arrays, I have learned a method of chopping samples with a dedicated start button using direct indexing of the slice's start in sample size values (fed as arguments into a vline module set to read from tabread4), but haven't learned how to do so when indexing via a phasor~, I suspect I'll get there in the next day or so.

The vline approach feels superior, but I'm not so sure it lends itself to rate control as easily as phasor~. Indexing via phase seems simpler too since it'll always be 0-1.

@_ish said:

1. there used to be a random object that spit floats (randomF).

The ELSE library has [rand.f].

(I wish, when I started with Pd, that someone had pointed me to ELSE... a lot of the things that are missing from vanilla, or "build it yourself," are just there.)

2. Earlier I was working on an array, and really wanted to send a bang every time it looped around to index 0. This feels like it should be really easy, but I couldn't think of anything that would send the active index step as out (the contents of the index step, sure, but not the step number itself).

Are you reading the array in the control or signal domain?

whale-av:

As in that link if using [phasor~] catching it's output as it passes 0 is unreliable because the value of [phasor~] will likely not be 0 as it is captured at a block boundary.

The value of phasor~ isn't reliable for this, but the two-point difference ([rzero~ 1]) is guaranteed to be positive most of the time, and negative only when the phasor jumps down. So this will always work (for a positive phasor~ frequency).

hjh

@lacuna said:

Don't know about phasor's right inlet, but I guess for more precision on the outside of objects we'd have to use 'Pd double precision'.

Yes, I confirmed that my test patch runs without issue under Pd64. Unfortunately, my real patch doesn't

Also remember @ddw_music different approach than [phasor~]:

I think that "different approach" is a different issue--I'm not multipying its output to index into a large table. I'm just using it as an oscillator/LFO. Are you suggesting that there's a way to build an oscillator with [rpole~] that is as frequency-accurate as [phasor~] but without the single vs double precision issue?

On one sample late: It is [snapshot~] who is early, not [phasor~]:
...
I would call this a bug of [snapshot~] !%§$*(

I don't understand your conclusion about the timing of [snapshot~]. Here's how I would analyze your demonstration patch: the top bang is processed in-between audio blocks. Therefore, [tabplay~] and [fexpr~] run in the following audio block. [snapshot~] is different though, because it's a bridge between control rate and audio rate--like bang, it also runs between audio blocks. Therefore, the best it can do (without having to predict the future) is to return the last sample of the previous audio block. So it's not surprising for [fexpr~] to have to reach back into the previous audio block to get the same value as [snapshot~]. If [snapshot~] were to return the first sample of the following audio block, it couldn't pass that value to control rate code until that audio block's processing is complete. Do you see my point?

Also, I'm not saying [phasor~] is early, I'm just saying its right inlet isn't a mechanism for restoring its state. It's just for what the documentation says it's for: to reset its phase to a given value.

Suppose you want to capture the state of a [phasor~] in order to return to it later, i.e. you want it to output the exact same sequence of samples as it did from the moment you captured its state. As a first attempt, you might [snapshot~] the [phasor~]'s output and then send that value to the [phasor~]'s right inlet sometime later to restore it. Of course you also want to capture the state of its frequency inlet (and record it if it's changing). But there are two subtleties to consider. Firstly, the value you write to the right inlet is the next value that the [phasor~] will output, but when you [snapshot~]ed it, the next value it output was a little greater, greater by the amount of the last frequency divided by the current sample rate. So really, this greater value (call it "snap++") is the one you should be writing to the right inlet.

The second subtlety has to do with the limits of Pd's single precision floats. Internally, [phasor~] is accumulating into a double precision float, so in a way, the right inlet overwrites the state of the phasor with something that can only approximate its state. The only solution I could find is to immediately write snap++ to the right inlet of the running [phasor~], so that the current and all future runs output the exact same sequence of samples. This might not be acceptable in your application if you are reliant on [phasor~]'s double precision accumulation because you'd be disrupting it in exchange for repeatability.

Here's a test patch that demonstrates the issue:
phasor snapshot restore test.pd

@nicnut said:

There is a number box labeled "transpose." The first step is to put a value in there, 1 being the original pitch, .5 is an octave down, 2 is an octave up, etc.

Here, you'd need to calculate the phasor frequency corresponding to normal playback speed, which is 1 / dur, where dur is the soundfile duration in seconds. Soundfile duration = soundfile size in samples / soundfile sample rate, so the phasor~ frequency can be expressed as soundfile sample rate / soundfile size in samples. Then you multiply this by a transpose factor, ok.

But I don't see you accounting for the soundfile's size or sample rate, so you have no guarantee of actually hitting normal playback speed.

After that you can change the frequency of the phasor in the other number box labled "change phasor frequency without changing pitch."

If you change the phasor frequency here, you will also change the transposition.

This way, I can make the phasor frequency, say .5 and the transposition 2,

Making the eventual playback speed 1 (assuming it was 1 to begin with), undoing the transposition.

which I don't think I can do using line~ with tabread4~.

The bad news is, you can't do this with phasor either.

hjh

@nicnut said:

Yes line~ would be better, but one thing I am doing is also playing the file for longer periods of time, by lowering the phaser~ frequency, and doing some math the transpose the pitch of the playback that I would like to keep. With line~ I don't know how to separate the pitch and the playback speed.

If I understand you right, separating pitch and playback speed would require pitch shifting...?

With phasor~, you can play the audio faster or slower, and the pitch will be higher or lower accordingly.

With line~, you can play the audio faster or slower, with the same result.

To take a concrete example -- if you want to play 0-10 seconds in the audio file, you'd use a phasor~ frequency of 0.1 and map the phasor's 0-1 range onto 0 .. (samplerate*10) samples. If instead, the phasor~ frequency were 0.05, then it would play 10 seconds of audio within 20 seconds = half speed, one octave down.

With line~, if you send it messages "0, $1 10000" where $1 = samplerate*10, then it would play the same 10 seconds' worth of audio, in 10 seconds.

To change the rate, you'd only need to change the amount of time: 10000 / 0.5 (half speed) = 20000. 10 seconds of audio, over 20 seconds -- exactly the same as the phasor~ 0.05 result.

frequency = 1 / time, time = 1 / frequency. Whatever you multiply phasor~ frequency by, just divide line~ time by the same amount.

(line~ doesn't support changing rate in the middle though. There's another possible trick for that but perhaps not simpler than phasor~.)

hjh

@earlyjp There might well be a better way than the two options that I can think of straight away.

You could play the audio file from disk, and send a bang into [delay] objects as you start playback, with the delays set to trigger the messages at the correct time.
I am not sure how long a [delay] you can create though because of the precision limit of a number size that can be saved in Pd. You might need to use [timer] banged by a [metro]

Or as you say you could load the file into RAM using [soundfiler] and play the track in the array using [phasor~] and trigger the messages as the output of [phasor~] reaches certain values.
Using [phasor~] the sample will loop though.
To play just once you could use [line~] or [vline~]
You would need to catch the values from [phasor~] [line~] or [vline~] using [snapshot~].
[<=] followed by [once] would be a good idea for catching, as catching the exact value would likely be unreliable...... once.pd
The same will be true if you use [timer] as the [metro] will not make [timer] output every value.

Again, number precision for catching the value could be a problem with long audio files.

Help for the second method....... https://forum.pdpatchrepo.info/topic/9151/for-phasor-and-vline-based-samplers-how-can-we-be-sure-all-samples-are-being-played

The [soundfiler] method will use much more cpu if that is a problem.
If there is a playback object that outputs it's index then use that, but I have not seen one.
David.