@erhandemirci This gives you options for fine tuning by ear.... so-far-2.pd
When you save the patch it will save the values of the sliders.
Maybe the pitch/total time relationship can be calculated, but that depends on why the original seems slow.
David.

@erhandemirci Maybe...... added before echo....... so-far.pd
David.

@erhandemirci The [delwrite~] that you copied was in an abstraction....... a patch acting like a subroutine.
The $1 would have taken its value from the first argument of that abstraction.
So here it has no value.
[delwrite~] needs to be [delwrite~ name length-in-milliseconds] so say [delwrite~ mydelay 10000] which will write to memory a buffer of 10 seconds.
[delread~] needs the same name and you can set the point at which it reads with the 3rd argument within.
So [delread~ mydelay 5000] will read from the buffer at 5 seconds.
You can send a mumber that will replace that 3rd argument into the left inlet of [delread~]

A $0 in an object name makes it unique.... it is not a global variable... It takes a value that is assigned as a patch window is created (1st patch opened... 1001... second 1002 etc.), and any sub-patch within the patch has the same value for $0.
Useful for isolating sends and receives etc. from other patches that might be open..... or even from the same patch opened twice.
David.

@erhandemirci But... VERY strangely...... no distortion in Pd.
Try this...... new-play-any.pd
David.

@erhandemirci "play-any-sample-rate.pd" that I posted above will play a 11025Hz file at 16 bits without problems so I would expect 22050Hz to be fine. I don't think bit depth should be a problem.
As @jameslo says a conversion might solve the problem,..... BUT.... playing your file in vlc in Windows it has some distortion towards the end.
Not sure what IEEE codec is.... I will look it up.
David.
In case others know....
But I will try making a 22050/32 file to see what happens...

@erhandemirci The file I posted above will play at natural speed and tone.... but....
... the sample-rate that Pd is running (Pd top menu - Media - Audio Settings...) must be the same as the sample rate of the wav file.
An automatic correction can be applied when using [soundfiler] but not using [readsf~]

This will correct for sample rate......... play-any- sample- rate.pd
Again... some of it you will not need.
It uses [vline~] which is a more accurate version of [line~]
It will play just once (not cylcle as it would using [phasor~]

If you want it to cycle then [phasor~] will be easier to use for that, and the maths will be similar.
David.

@erhandemirci To vary playback speed (rather then pitch shift - constant lemgth) you will need to write the file to an array.

For pitch shifting have a look at G09.pitchshift.pd in your Pd folder... pd/doc/3.audio.examples.
Inside the file (on the right) is an object [pd test-loop] and inside that you will find [soundfiler] writing a wav file to the array.
If you want the file to only play once the use [line~] instead of [phasor~] in that window.
You can play the file at different speed by varying the number sent into the left inlet of [phasor].

Right clicking and selecting help will give you the info you need for usage of an object.

You will find other effects in that folder, such as delay, echo reverb etc.
David.

@erhandemirci Yes, that is not a valid Pd file.
This might help...... if you only want to play the file then [readsf~] is more friendly.
cart-mod.pd

Otherwise, if you intend to modify or manipulate the [array] then you need [soundfiler] to populate the array.

You can delete much of the attached file for your purpose.... stop... loop... etc. if you wish.
David.

EDIT.. I have just re-uploaded the file. The incoming message needs to be tagged as a symbol for $1 in the [open $1( message to process the message correctly.

@DesignDefault Welcome to the Forum...!

Sorry, this will be a lot for a newbie.

You will see in the video @ricky has linked to that the [phasor~] object controls the speed at which the array is being read.
So you will need to increase the number that it receives at its left inlet in order to increase the speed.
It is an audio rate inlet, so you will need [sig~] to convert a control rate 1 to a signal rate 1.
1 will play at normal speed, 2 at double that speed etc.

Unfortunately you will need to "catch" a moment in the [phasor~] loop using [edge~] and then increment that value to [sig~] by whatever increase you want each time that [edge~] outputs a bang.
Catching the exact moment that it loops can be awkward and you might well need to add or subtract a value from the output of [phasor~] before sending it into [edge~]

A more reliable method would be to use a [metro] to increase the [sig~] with the metro rate set by the value from [timer].

Also, before playback (it only needs to be done once for the array after it has been filled) extra data points should be added.
That will stop clicks as [phasor~] wraps back to the beginning of the array.
https://forum.pdpatchrepo.info/topic/14301/add-delete-guard-points-of-an-array-for-4-point-interpolation-of-tabread4-ect
That will add 3 data points to the array, allowing [tabread4~] to correctly interpolate the samples at wrap around.
For that to work properly you should then add 3.... [+ 3] to the total number of samples after the [* 44.1]

Also, if you are running Pd at 48000Hz the [* 44.1] should be changed to [* 48] for the initial playback speed to be correct.
David.

@Pepe_mv98 Does your device now appear in the Pd top menu...... "Media", "Midi Settings..." ?
David.