Using AudioKit, I am trying to add controller data to a AKMusicTrack which already has some AKMidiNoteData on it, Control data like - like volume, filter resonance, filter cut-off freq, and so on, Im not trying to use the filter effects on this particular instance, but I'm having trouble finding samples of the AKMusicTrack.addController(controller: MIDIByte, value: MIDIByte, position: AKDuration) function. Can anyone point me toward some online samples, or a table with the the control values listed?
For example some manufacturers have pan or envelope attack mapped to midi control 20 or 25 e.t.c
Related
I have a time series of events. The events carry a timestamp and some other metadata. I want to create clusters/buckets of these events based on their temporal proximity in order to discard some redundant events. For example, if I have a cluster of 10 events which are temporally close to each other - let's define temporal proximity as events being no further than x temporal units away from each other -, I can choose one of them as a representative and discard all others. For example, I have the following events
e0-e1-e2-------e3------------------------e5-e6-e7-e8--------------e9-e10
A sample bucketization would be:
(e0,e1,e2)
e3
(e5,e6,e7,e8)
(e9, e10)
I tried to represent the proximity of the events using the dashes above.
I know that there are some elaborate ways to go about this like DBSCAN or k-means but I was hoping that there would be something more straight forward, without any extensive prior knowledge of ML.
I also thought of a sliding window implementation, but the problem becomes quickly very complicated in terms of how to choose the window size and the shifts of the window...
Is there a standard way of doing this?
I want to create similar sound to theremin using touch coordinate on screen. I'm using y axis as frequency, x axis as amplitude.
Due to my small research I believe I can create it using AKOscillator or AKFMOscillator from AudioKit framework (please let me know if any other oscillator works better in this case). I'm open to other frameworks like built-in AudioToolbox (MIDINoteMessage etc.) if I can create similar sound to theremin.
Here it says theremin has two oscillators. One with fixed-frequency on 260kHz and one is dynamic between 257-260kHz. It superimposes their output (it takes difference of them I guess?). And it outputs between frequency between 0-3 kHz.
When I create sounds using AKFMOscillator with baseFrequency between 257-260 kHz, it sounds high-pitched.
When I try with one oscillator range between 0-3kHz it sounds very robotic. How I can simulate timbre of theremin?
How can I make it sound better? Should I mix two oscillators? I tried mixing with AKMixer but when both oscillators use same frequency and amplitude, it makes no difference.
I tried to mapping to nearest note (auto-tune), I tried limiting the frequency between 3-4 octaves. It sounds better but still not good as theremin.
What should use ( AKOscillator or AKFMOscillator, OscillatorBank), with which parameters (rampDuration, baseFrequency, modulationIndex, amplitude) to simulate more thereminish sound?
Update:
I did some more research and played with Synth One presets. Now, I know I need two oscillators mixed (both set to saw-shape wave). Changing ADSR(envelope) values to specific ranges creates richer sound (this gives the instrumental sound type). And a lfo to create the wavy (or spooky) sound effect. Playing notes (specific frequencies) creates good sounds, if you play every frequency in between note frequencies it doesn't sound good.
I'm trying to think of the best way to conduct some sort of analysis between two 3D models of the same object.
The first scan is of the original item and the second scan is after it has been put under some load x.
An example would be trying to find the difference between two types of metal.
I would like to be able to scan the initial metal cylinder, apply a measured load, scan it again, and then finally apply some sort of algorithm to compare the difference.
Is it possible to do this efficiently (maybe using Mablab) over say 50 - 100 items for an object around 5inch^3?
I am assuming I will need to work out some sort of utility function as the total mass should be the same?
Would machine learning be beneficial in this case?
Any suggestions or direction would be amazing.
Thank you :)
EDIT: The scan files are coming through as '.stl'
I want to perform a cross-correlation of two audio files (which are actually NSData objects). I found a vDSP_convD function in accelerate framework. NSData has a property bytes which returns a pointer to an array of voids - that is the parameter of the filter and signal vector.
I struggled with other parameters. What is the length of these vectors or the length of the result vectors?
I guess:
it's the sum of the filter and signal vector.
Could anyone give me an example of using the vDSP_convD function?
Apple reference to the function is here
Thanks
After reading a book - Learning Core Audio, I have made a demo which demonstrates delay between two audio files. I used new iOS 8 API to get samples from the audio files and a good performance optimization.
Github Project.
a call would look like this:
vDSP_conv ( signal *, signalStride, filter *, filterStride, result*, resultStride, resultLenght, filterLength );
where we have:
signal*: a pointer to the first element of your signal array
signalStride: the lets call it "step size" throug your signal array. 1 is every element, 2 is every second ...
same for filter and result array
length for result and filter array
How long do the arrays have to be?:
As stated in the docs you linked our signal array has to be lenResult + lenFilter - 1 which it is where it gets a little messy. You can find a demonstration of this by Apple here or a shorter answer by SO user Rasman here.
You have to do the zero padding of the signal array by yourself so the vector functions can apply the sliding window without preparation.
Note: You might consider using the Fast-Fourier-Transformation for this, because when you work with audio files i assume, that you have quite some data and there is a significant performance increase from a certain point onwards when using:
FFT -> complex multiplication in frequency domain (which results in a correlation in time domain) -> reverse FFT
here you can find a useful piece of code for this!
I have a DirectShow Transform filter written in Delphi 6 using the DSPACK component library. It is a simple audio mixer that creates a new input pin whenever a new connection is attempted. I say simple because once its media format is set, all connections to the its input pins or singular output pin are forced to conform to that media format. I build the filter chain manually, making all pin connections explicitly myself. I do not use any of the "intelligent rendering" calls, unless there is some way to trigger that unwanted behavior (in my case) accidentally.
NOTE: The Capture Filter is a standard DirectShow filter external to my application. My push source audio filter and simple audio mixer filters are being used as private, unregistered filters and are internal to my application.
I am having a weird problem that only occurs when I try to make multiple input connections to my mixer, which does indeed accept them. Currently, I am attempting to connect both a Capture Filter and my custom Push Source audio filter to my mixer filter. Whenever I try to do that the second upstream filter connection fails. Regardless of whether I connect the Capture Filter first or Push Source audio filter first, the second upstream filter connection always fails.
The first test I ran was to try connecting just the Capture Filter to the mixer. That worked fine.
The second test I ran was to try connecting just the Push Source audio filter to the mixer. That worked fine.
But as soon as try to do both I get a "no combination of intermediate filters could be found" error. I did several hours of deep digging into the media negotiation calls hitting my filter from the graph builder and then I found the problem. For some reason, the filter graph is dragging in the ancient "Indeo (R) Audio Software" codec into the chain.
I discovered this because despite the fact that codec did have a media format that matched my filter in almost every regard (major type, sub type, format type, wave format parameters), it had an extra 2 bytes at the end of it's pbFormat data member and that was enough to fail the equals test since that test does a comparison between the source and target pbFormat areas by comparing the cbFormat value of each media type. The Indeo codec has a cbFormat value of 20 while my filter has a cbFormat value of 18, which is the size of a _tWAVEFORMATEX data structure. In a way it's a good thing the Indeo pbFormat has that weird size because the first 18 bytes of its 20 byte area were exactly equal to the pbFormat area of my mixer filter's supported media type. Without that anomaly I never would have known that ancient codec was being drug in. I'm surprised it's being drug in at all since it has known exploits and vulnerabilities. What is most confusing is that this is happening on my mixer filter's output pin, not one of the input pins, and I have not made a single downstream connection yet when building up my pin connections.
Can anyone tell me why DirectShow is trying to drag in that codec despite the fact that the media formats for the both incoming filters, the Capture Filter and the Push Source filter, are identical and don't need any intermediate filters at all since they match my mixer filter's input pins supported format exactly? How can I fix this problem?
Also, I noticed that even in the single filter attachment tests above that succeeded, my mixer output pin was still getting queried for media formats. Why is that when as I said, at this point in building up my pin connections I have not connected anything to the output pin of my mixer filter?
--------------------------- UPDATE: 1 ----------------------------
I have learned that you can avoid the "intelligent connection" behavior entirely by using IFilterGraph.ConnectDirect() instead of IGraphBuilder.Connect(). I switched over to DirectConnect() and turns out that the input pin on my mixer filter is coming back as "already connected". That may be what is causing the graph builder to drag in the Indeo codec filter. Now that I have this new diagnostic information I will correct the problem and update this post with my results.
--------------------------- RESOLUTION ----------------------------
The root problem of all of this was my re-use of the input pin I obtained from the first destination/downstream filter I connected to my simple audio mixer filter, at the top of my application code. In other words my filter was working correctly, but I was not getting a fresh input pin with each upstream filter I tried to connect to it. Once I started doing that the connection process worked fine. I don't know why the code behind the IGraphBuilder.Connect() interface tried to bring in the Indeo codec filter, perhaps something to do with trying to connect to an already connected input pin, but it did. For my needs, I prefer the tight control that IFilterGraph.ConnectDirect() provides since it eliminates any interference from the intelligent connection code in IGraphBuilder, but I could see when video filters get involved it could become useful.