/*
AudioEffectMine
Created : Chip Audette , December 2016
Purpose ; Here is the skeleton of a audio processing algorithm that will
( hopefully ) make it easier for people to start making their own
algorithm .
This processes a single stream fo audio data ( ie , it is mono )
MIT License . use at your own risk .
*/
# include <arm_math.h> //ARM DSP extensions. https://www.keil.com/pack/doc/CMSIS/DSP/html/index.html
# include <AudioStream_F32.h>
class AudioEffectMine_F32 : public AudioStream_F32
{
public :
//constructor
AudioEffectMine_F32 ( void ) : AudioStream_F32 ( 1 , inputQueueArray_f32 ) {
//do any setup activities here
} ;
//here's the method that is called automatically by the Teensy Audio Library
void update ( void ) {
//Serial.println("AudioEffectMine_F32: doing update()"); //for debugging.
audio_block_f32_t * audio_block ;
audio_block = AudioStream_F32 : : receiveWritable_f32 ( ) ;
if ( ! audio_block ) return ;
//do your work
applyMyAlgorithm ( audio_block ) ;
///transmit the block and release memory
AudioStream_F32 : : transmit ( audio_block ) ;
AudioStream_F32 : : release ( audio_block ) ;
}
// !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!11
// Here is where you can add your algorithm.
// This function gets called block-wise...which is usually hard-coded to every 128 samples
void applyMyAlgorithm ( audio_block_f32_t * audio_block ) {
//Add whatever you'd like here. I'm going to show a couple of examples below.
//you can delete all of these examples. They're just to illustrate.
//More examples of fast (DSP accelerated) operations are at: https://github.com/chipaudette/OpenAudio/blob/master/Docs/Programming%20Algorithms/Using%20DSP%20Exentions.md
// //apply a fixed gain...the whole block is processed with this one command
//float32_t gain = 2.0; %here's 6 dB of gain
//arm_scale_f32(audio_block->data, gain, audio_block->data, audio_block->length); //uses ARM DSP for speed!
// //compute the power of each sample...the whole block is processed with this one command
// audio_block_f32_t *audio_pow_block = AudioStream_F32::allocate_f32();
// arm_mult_f32(audio_block->data, audio_block->data, audio_pow_block->data, audio_block->length);
// //loop over each sample and do something on a point-by-point basis (when it cannot be done as a block)
//for (int i=0; i < audio_block->length; i++) {
// //as a boring example, let's add the user_parameter value to every sample. yes, this
// //addition operation could have been done with a DSP-accelerated function, but I'm
// //simply trying to illustrate how to loop on your audio samples.
// audio_block->data[i] = audio_block->data[i] + user_parameter;
//}
// //clean up...if you allocated any memory in the lines above
//Audiostream_F32::release(audio_pow_block);
// return your processed audio by putting it back into audio_block->data
} //end of applyMyAlgorithms
// !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
// Call this method in your main program if you want to set the value of your user parameter.
// The user parameter can be used in your algorithm above. The user_parameter variable was
// created in the "private:" section of this class, which appears a little later in this file.
// Feel free to create more user parameters (and to use better names for your variables)
// for use in this class.
float32_t setUserParameter ( float val ) {
return user_parameter = val ;
}
private :
//state-related variables
audio_block_f32_t * inputQueueArray_f32 [ 1 ] ; //memory pointer for the input to this module
//this value can be set from the outside (such as from the potentiometer) to control
//a parameter within your algorithm
float32_t user_parameter = 0.0 ;
} ; //end class definition for AudioEffectMine_F32
