Chip Audette
8 years ago
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/*
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AudioEffectMine |
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Created: Chip Audette, December 2016 |
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Purpose; Here is the skeleton of a audio processing algorithm that will |
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(hopefully) make it easier for people to start making their own
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algorithm. |
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This processes a single stream fo audio data (ie, it is mono) |
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MIT License. use at your own risk. |
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*/ |
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#include <arm_math.h> //ARM DSP extensions. https://www.keil.com/pack/doc/CMSIS/DSP/html/index.html |
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#include <AudioStream_F32.h> |
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class AudioEffectMine_F32 : public AudioStream_F32 |
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{ |
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public: |
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//constructor
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AudioEffectMine_F32(void) : AudioStream_F32(1, inputQueueArray_f32) { |
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//do any setup activities here
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}; |
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//here's the method that is called automatically by the Teensy Audio Library
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void update(void) { |
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//Serial.println("AudioEffectMine_F32: doing update()"); //for debugging.
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audio_block_f32_t *audio_block; |
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audio_block = AudioStream_F32::receiveWritable_f32(); |
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if (!audio_block) return; |
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//do your work
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applyMyAlgorithm(audio_block); |
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///transmit the block and release memory
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AudioStream_F32::transmit(audio_block); |
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AudioStream_F32::release(audio_block); |
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} |
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// !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!11
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// Here is where you can add your algorithm.
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// This function gets called block-wise...which is usually hard-coded to every 128 samples
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void applyMyAlgorithm(audio_block_f32_t *audio_block) { |
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//Add whatever you'd like here. I'm going to show a couple of examples below.
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//you can delete all of these examples. They're just to illustrate.
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//More examples of fast (DSP accelerated) operations are at: https://github.com/chipaudette/OpenAudio/blob/master/Docs/Programming%20Algorithms/Using%20DSP%20Exentions.md
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// //apply a fixed gain...the whole block is processed with this one command
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//float32_t gain = 2.0; %here's 6 dB of gain
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//arm_scale_f32(audio_block->data, gain, audio_block->data, audio_block->length); //uses ARM DSP for speed!
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// //compute the power of each sample...the whole block is processed with this one command
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// audio_block_f32_t *audio_pow_block = AudioStream_F32::allocate_f32();
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// arm_mult_f32(audio_block->data, audio_block->data, audio_pow_block->data, audio_block->length);
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// //loop over each sample and do something on a point-by-point basis (when it cannot be done as a block)
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//for (int i=0; i < audio_block->length; i++) {
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// //as a boring example, let's add the user_parameter value to every sample. yes, this
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// //addition operation could have been done with a DSP-accelerated function, but I'm
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// //simply trying to illustrate how to loop on your audio samples.
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// audio_block->data[i] = audio_block->data[i] + user_parameter;
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//}
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// //clean up...if you allocated any memory in the lines above
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//Audiostream_F32::release(audio_pow_block);
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} //end of applyMyAlgorithms
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// !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
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// Call this method in your main program if you want to set the value of your user parameter.
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// The user parameter can be used in your algorithm above. The user_parameter variable was
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// created in the "private:" section of this class, which appears a little later in this file.
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// Feel free to create more user parameters (and to use better names for your variables)
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// for use in this class.
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float32_t setUserParameter(float val) { |
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return user_parameter = val; |
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} |
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private: |
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//state-related variables
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audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
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//this value can be set from the outside (such as from the potentiometer) to control
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//a parameter within your algorithm
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float32_t user_parameter = 0.0;
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}; //end class definition for AudioEffectMine_F32
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/*
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BasicGain |
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Created: Chip Audette, Dec 2016 |
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Purpose: Be a blank canvas for adding your own floating-point audio processing. |
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Uses Teensy Audio Adapter. |
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Assumes microphones (or whatever) are attached to the LINE IN (stereo) |
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Listens potentiometer mounted to Audio Board to provde a control signal. |
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MIT License. use at your own risk. |
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*/ |
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//These are the includes from the Teensy Audio Library
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#include <Audio.h> //Teensy Audio Library |
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#include <Wire.h> |
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#include <SPI.h> |
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#include <SD.h> |
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#include <SerialFlash.h> |
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#include <OpenAudio_ArduinoLibrary.h> //for AudioConvert_I16toF32, AudioConvert_F32toI16, and AudioEffectGain_F32 |
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#include "AudioEffectMine_F32.h"; |
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//create audio library objects for handling the audio
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AudioControlSGTL5000 sgtl5000_1; //controller for the Teensy Audio Board
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AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo.
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AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
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AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h
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AudioEffectMine_F32 effect1, effect2; //This is your own algorithms
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AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h
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//Make all of the audio connections
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AudioConnection patchCord1(i2s_in, 0, int2Float1, 0); //connect the Left input to the Left Int->Float converter
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AudioConnection patchCord2(i2s_in, 1, int2Float2, 0); //connect the Right input to the Right Int->Float converter
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AudioConnection_F32 patchCord10(int2Float1, 0, effect1, 0); //Left. makes Float connections between objects
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AudioConnection_F32 patchCord11(int2Float2, 0, effect2, 0); //Right. makes Float connections between objects
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AudioConnection_F32 patchCord12(effect1, 0, float2Int1, 0); //Left. makes Float connections between objects
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AudioConnection_F32 patchCord13(effect2, 0, float2Int2, 0); //Right. makes Float connections between objects
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AudioConnection patchCord20(float2Int1, 0, i2s_out, 0); //connect the Left float processor to the Left output
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AudioConnection patchCord21(float2Int2, 0, i2s_out, 1); //connect the Right float processor to the Right output
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// which input on the audio shield will be used?
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const int myInput = AUDIO_INPUT_LINEIN; |
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//const int myInput = AUDIO_INPUT_MIC;
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//I have a potentiometer on the Teensy Audio Board
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#define POT_PIN A1 //potentiometer is tied to this pin
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// define the setup() function, the function that is called once when the device is booting
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void setup() { |
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Serial.begin(115200); //open the USB serial link to enable debugging messages
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delay(500); //give the computer's USB serial system a moment to catch up.
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Serial.println("OpenAudio_ArduinoLibrary: MyAudioEffect_Float..."); |
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// Audio connections require memory
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AudioMemory(10); //allocate Int16 audio data blocks
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AudioMemory_F32(10); //allocate Float32 audio data blocks
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// Enable the audio shield, select input, and enable output
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sgtl5000_1.enable(); //start the audio board
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sgtl5000_1.inputSelect(myInput); //choose line-in or mic-in
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sgtl5000_1.volume(0.8); //volume can be 0.0 to 1.0. 0.5 seems to be the usual default.
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sgtl5000_1.lineInLevel(10, 10); //level can be 0 to 15. 5 is the Teensy Audio Library's default
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sgtl5000_1.adcHighPassFilterDisable(); //reduces noise. https://forum.pjrc.com/threads/27215-24-bit-audio-boards?p=78831&viewfull=1#post78831
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// setup any other other features
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pinMode(POT_PIN, INPUT); //set the potentiometer's input pin as an INPUT
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} //end setup()
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// define the loop() function, the function that is repeated over and over for the life of the device
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void loop() { |
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//choose to sleep ("wait for interrupt") instead of spinning our wheels doing nothing but consuming power
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asm(" WFI"); //ARM-specific. Will wake on next interrupt. The audio library issues tons of interrupts, so we wake up often.
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//service the potentiometer...if enough time has passed
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servicePotentiometer(millis()); |
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//update the memory and CPU usage...if enough time has passed
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printMemoryAndCPU(millis()); |
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} //end loop()
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//servicePotentiometer: listens to the blue potentiometer and sends the new pot value
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// to the audio processing algorithm as a control parameter
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void servicePotentiometer(unsigned long curTime_millis) { |
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static unsigned long updatePeriod_millis = 100; //how many milliseconds between updating the potentiometer reading?
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static unsigned long lastUpdate_millis = 0; |
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static float prev_val = 0; |
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//has enough time passed to update everything?
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if (curTime_millis < lastUpdate_millis) lastUpdate_millis = 0; //handle wrap-around of the clock
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if ((curTime_millis - lastUpdate_millis) > updatePeriod_millis) { //is it time to update the user interface?
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//read potentiometer
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float val = float(analogRead(POT_PIN)) / 1024.0; //0.0 to 1.0
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val = 0.1 * (float)((int)(10.0 * val + 0.5)); //quantize so that it doesn't chatter
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//add code here to change the potentiometer value to something useful (like gain_dB?)
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// ..... add code here if you'd like ......
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//send the potentiometer value to your algorithm as a control parameter
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if (abs(val - prev_val) > 0.05) { //is it different than befor?
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Serial.print("Sending new value to my algorithms: "); Serial.println(val); |
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effect1.setUserParameter(val); effect2.setUserParameter(val); |
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} |
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prev_val = val; //use the value the next time around
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} // end if
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} //end servicePotentiometer();
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void printMemoryAndCPU(unsigned long curTime_millis) { |
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static unsigned long updatePeriod_millis = 2000; //how many milliseconds between updating gain reading?
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static unsigned long lastUpdate_millis = 0; |
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//has enough time passed to update everything?
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if (curTime_millis < lastUpdate_millis) lastUpdate_millis = 0; //handle wrap-around of the clock
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if ((curTime_millis - lastUpdate_millis) > updatePeriod_millis) { //is it time to update the user interface?
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Serial.print("CPU: Usage, Max: "); |
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Serial.print(AudioProcessorUsage()); |
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Serial.print(", "); |
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Serial.print(AudioProcessorUsageMax()); |
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Serial.print(" "); |
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Serial.print("Int16 Memory: "); |
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Serial.print(AudioMemoryUsage()); |
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Serial.print(", "); |
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Serial.print(AudioMemoryUsageMax()); |
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Serial.print(" "); |
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Serial.print("Float Memory: "); |
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Serial.print(AudioMemoryUsage_F32()); |
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Serial.print(", "); |
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Serial.print(AudioMemoryUsageMax_F32()); |
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Serial.println(); |
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lastUpdate_millis = curTime_millis; //we will use this value the next time around.
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} |
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} |
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