Steve Lascos
6 years ago
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362d0928d2
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9b09021ef0
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/*************************************************************************
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* This demo uses the BALibrary library to provide enhanced control of |
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* the TGA Pro board. |
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*
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* The latest copy of the BA Guitar library can be obtained from |
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* https://github.com/Blackaddr/BALibrary
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*
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* This example demonstrates teh BAAudioEffectsAnalogChorus effect. It can |
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* be controlled using the Blackaddr Audio "Expansion Control Board". |
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*
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* POT1 (left) controls the modulation rate |
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* POT2 (right) controls the modulation depth |
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* POT3 (center) controls the wet/dry mix |
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* SW1 will enable/bypass the audio effect. LED1 will be on when effect is enabled. |
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* SW2 will cycle through the 3 pre-programmed analog filters. LED2 will be on when SW2 is pressed. |
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*
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* Using the Serial Montitor, send 'u' and 'd' characters to increase or decrease |
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* the headphone volume between values of 0 and 9. |
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*/ |
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#define TGA_PRO_REVB // Set which hardware revision of the TGA Pro we're using
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#define TGA_PRO_EXPAND_REV2 // pull in the pin definitions for the Blackaddr Audio Expansion Board.
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#include "BALibrary.h" |
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#include "BAEffects.h" |
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using namespace BAEffects; |
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using namespace BALibrary; |
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AudioInputI2S i2sIn; |
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AudioOutputI2S i2sOut; |
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BAAudioControlWM8731 codec; |
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//#define USE_EXT // uncomment this line to use External MEM0
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#ifdef USE_EXT |
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// If using external SPI memory, we will instantiate a SRAM
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// manager and create an external memory slot to use as the memory
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// for our audio delay
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ExternalSramManager externalSram; |
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ExtMemSlot delaySlot; // Declare an external memory slot.
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// Instantiate the AudioEffectAnalogChorus to use external memory by
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/// passing it the delay slot.
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AudioEffectAnalogChorus analogChorus(&delaySlot); |
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#else |
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AudioEffectAnalogChorus analogChorus; // default chorus delays
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#endif |
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AudioFilterBiquad cabFilter; // We'll want something to cut out the highs and smooth the tone, just like a guitar cab.
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// Simply connect the input to the delay, and the output
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// to both i2s channels
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AudioConnection input(i2sIn,0, analogChorus,0); |
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AudioConnection chorusOut(analogChorus, 0, cabFilter, 0); |
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AudioConnection leftOut(cabFilter,0, i2sOut, 0); |
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AudioConnection rightOut(cabFilter,0, i2sOut, 1); |
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//////////////////////////////////////////
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// SETUP PHYSICAL CONTROLS
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// - POT1 (left) will control the rate
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// - POT2 (right) will control the depth
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// - POT3 (centre) will control the wet/dry mix.
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// - SW1 (left) will be used as a bypass control
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// - LED1 (left) will be illuminated when the effect is ON (not bypass)
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// - SW2 (right) will be used to cycle through the three built in analog filter styles available.
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// - LED2 (right) will illuminate when pressing SW2.
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//////////////////////////////////////////
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// To get the calibration values for your particular board, first run the
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// BAExpansionCalibrate.ino example and
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constexpr int potCalibMin = 1; |
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constexpr int potCalibMax = 1018; |
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constexpr bool potSwapDirection = true; |
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// Create a control object using the number of switches, pots, encoders and outputs on the
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// Blackaddr Audio Expansion Board.
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BAPhysicalControls controls(BA_EXPAND_NUM_SW, BA_EXPAND_NUM_POT, BA_EXPAND_NUM_ENC, BA_EXPAND_NUM_LED); |
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int loopCount = 0; |
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unsigned filterIndex = 0; // variable for storing which analog filter we're currently using.
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constexpr unsigned MAX_HEADPHONE_VOL = 10; |
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unsigned headphoneVolume = 8; // control headphone volume from 0 to 10.
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// BAPhysicalControls returns a handle when you register a new control. We'll uses these handles when working with the controls.
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int bypassHandle, filterHandle, rateHandle, depthHandle, mixHandle, led1Handle, led2Handle; // Handles for the various controls
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void setup() { |
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delay(100); // wait a bit for serial to be available
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Serial.begin(57600); // Start the serial port
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delay(100); |
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// Setup the controls. The return value is the handle to use when checking for control changes, etc.
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// pushbuttons
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bypassHandle = controls.addSwitch(BA_EXPAND_SW1_PIN); // will be used for bypass control
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filterHandle = controls.addSwitch(BA_EXPAND_SW2_PIN); // will be used for stepping through filters
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// pots
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rateHandle = controls.addPot(BA_EXPAND_POT1_PIN, potCalibMin, potCalibMax, potSwapDirection); // control the amount of delay
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depthHandle = controls.addPot(BA_EXPAND_POT2_PIN, potCalibMin, potCalibMax, potSwapDirection);
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mixHandle = controls.addPot(BA_EXPAND_POT3_PIN, potCalibMin, potCalibMax, potSwapDirection);
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// leds
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led1Handle = controls.addOutput(BA_EXPAND_LED1_PIN); |
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led2Handle = controls.addOutput(BA_EXPAND_LED2_PIN); // will illuminate when pressing SW2
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// Disable the audio codec first
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codec.disable(); |
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AudioMemory(128); |
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// Enable and configure the codec
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Serial.println("Enabling codec...\n"); |
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codec.enable(); |
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codec.setHeadphoneVolume(1.0f); // Max headphone volume
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// If using external memory request request memory from the manager
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// for the slot
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#ifdef USE_EXT |
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Serial.println("Using EXTERNAL memory"); |
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// We have to request memory be allocated to our slot.
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externalSram.requestMemory(&delaySlot, 40.0f, MemSelect::MEM0, true); // 40 ms is enough to handle the full range of the chorus delay
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#else |
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Serial.println("Using INTERNAL memory"); |
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#endif |
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// Besure to enable the delay. When disabled, audio is is completely blocked by the effect
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// to minimize resource usage to nearly to nearly zero.
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analogChorus.enable();
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// Set some default values.
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// These can be changed using the controls on the Blackaddr Audio Expansion Board
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analogChorus.bypass(false); |
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analogChorus.rate(0.5f); |
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analogChorus.mix(0.5f); |
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analogChorus.depth(1.0f); |
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//////////////////////////////////
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// AnalogChorus filter selection //
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// These are commented out, in this example we'll use SW2 to cycle through the different filters
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//analogChorus.setFilter(AudioEffectAnalogChorus::Filter::CE2); // The default filter. Naturally bright echo (highs stay, lows fade away)
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//analogChorus.setFilter(AudioEffectAnalogChorus::Filter::WARM); // A warm filter with a smooth frequency rolloff above 2Khz
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//analogChorus.setFilter(AudioEffectAnalogChorus::Filter::DARK); // A very dark filter, with a sharp rolloff above 1Khz
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// Guitar cabinet: Setup 2-stages of LPF, cutoff 4500 Hz, Q-factor 0.7071 (a 'normal' Q-factor)
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cabFilter.setLowpass(0, 4500, .7071); |
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cabFilter.setLowpass(1, 4500, .7071); |
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} |
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void loop() { |
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float potValue; |
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// Check if SW1 has been toggled (pushed)
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if (controls.isSwitchToggled(bypassHandle)) { |
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bool bypass = analogChorus.isBypass(); // get the current state
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bypass = !bypass; // change it
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analogChorus.bypass(bypass); // set the new state
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controls.setOutput(led1Handle, !bypass); // Set the LED when NOT bypassed
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Serial.println(String("BYPASS is ") + bypass); |
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} |
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// Use SW2 to cycle through the filters
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controls.setOutput(led2Handle, controls.getSwitchValue(led2Handle)); |
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if (controls.isSwitchToggled(filterHandle)) { |
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filterIndex = (filterIndex + 1) % 3; // update and potentionall roll the counter 0, 1, 2, 0, 1, 2, ...
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// cast the index between 0 to 2 to the enum class AudioEffectAnalogChorus::Filter
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analogChorus.setFilter(static_cast<AudioEffectAnalogChorus::Filter>(filterIndex)); // will cycle through 0 to 2
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Serial.println(String("Filter set to ") + filterIndex); |
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} |
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// Use POT1 (left) to control the rate setting
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if (controls.checkPotValue(rateHandle, potValue)) { |
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// Pot has changed
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Serial.println(String("New RATE setting: ") + potValue); |
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analogChorus.rate(potValue); |
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} |
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// Use POT2 (right) to control the depth setting
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if (controls.checkPotValue(depthHandle, potValue)) { |
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// Pot has changed
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Serial.println(String("New DEPTH setting: ") + potValue); |
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analogChorus.depth(potValue); |
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} |
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// Use POT3 (centre) to control the mix setting
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if (controls.checkPotValue(mixHandle, potValue)) { |
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// Pot has changed
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Serial.println(String("New MIX setting: ") + potValue); |
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analogChorus.mix(potValue); |
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} |
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// Use the 'u' and 'd' keys to adjust volume across ten levels.
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if (Serial) { |
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if (Serial.available() > 0) { |
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while (Serial.available()) { |
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char key = Serial.read(); |
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if (key == 'u') {
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headphoneVolume = (headphoneVolume + 1) % MAX_HEADPHONE_VOL; |
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Serial.println(String("Increasing HEADPHONE volume to ") + headphoneVolume); |
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} |
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else if (key == 'd') {
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headphoneVolume = (headphoneVolume - 1) % MAX_HEADPHONE_VOL; |
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Serial.println(String("Decreasing HEADPHONE volume to ") + headphoneVolume); |
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} |
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codec.setHeadphoneVolume(static_cast<float>(headphoneVolume) / static_cast<float>(MAX_HEADPHONE_VOL)); |
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} |
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} |
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} |
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// Use the loopCounter to roughly measure human timescales. Every few seconds, print the CPU usage
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// to the serial port. About 500,000 loops!
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if (loopCount % 524288 == 0) { |
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Serial.print("Processor Usage, Total: "); Serial.print(AudioProcessorUsage()); |
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Serial.print("% "); |
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Serial.print(" AnalogChorus: "); Serial.print(analogChorus.processorUsage()); |
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Serial.println("%"); |
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} |
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loopCount++; |
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} |
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@ -0,0 +1,19 @@ |
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// To give your project a unique name, this code must be
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// placed into a .c file (its own tab). It can not be in
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// a .cpp file or your main sketch (the .ino file).
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#include "usb_names.h" |
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// Edit these lines to create your own name. The length must
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// match the number of characters in your custom name.
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#define MIDI_NAME {'B','l','a','c','k','a','d','d','r',' ','A','u','d','i','o',' ','T','G','A',' ','P','r','o'} |
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#define MIDI_NAME_LEN 23 |
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// Do not change this part. This exact format is required by USB.
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struct usb_string_descriptor_struct usb_string_product_name = { |
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2 + MIDI_NAME_LEN * 2, |
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3, |
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MIDI_NAME |
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}; |
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@ -0,0 +1,196 @@ |
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/**************************************************************************//**
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* @file |
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* @author Steve Lascos |
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* @company Blackaddr Audio |
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* |
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* AudioEffectAnalogChorus is a class for simulating a classic BBD based chorus |
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* like the Boss CE-2. This class works with either internal RAM, or external |
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* SPI RAM. The external RAM uses DMA to minimize load on the |
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* CPU. |
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* |
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* @copyright This program is free software: you can redistribute it and/or modify |
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* it under the terms of the GNU General Public License as published by |
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* the Free Software Foundation, either version 3 of the License, or |
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* (at your option) any later version.* |
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* |
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* This program is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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* GNU General Public License for more details. |
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* |
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* You should have received a copy of the GNU General Public License |
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*****************************************************************************/ |
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#ifndef __BAEFFECTS_BAAUDIOEFFECTANALOGCHORUS_H |
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#define __BAEFFECTS_BAAUDIOEFFECTANALOGCHORUS_H |
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#include <Audio.h> |
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#include "LibBasicFunctions.h" |
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namespace BAEffects { |
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/**************************************************************************//**
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* AudioEffectAnalogChorus models BBD based analog chorus. It provides controls |
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* for rate, depth, mix and output level. All parameters can be |
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* controlled by MIDI. The class supports internal memory, or external SPI |
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* memory by providing an ExtMemSlot. External memory access uses DMA to reduce |
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* process load. |
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*****************************************************************************/ |
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class AudioEffectAnalogChorus : public AudioStream { |
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public: |
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///< List of AudioEffectAnalogChorus MIDI controllable parameters
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enum { |
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BYPASS = 0, ///< controls effect bypass
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RATE, ///< controls the modulate rate of the LFO
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DEPTH, ///< controls the depth of modulation of the LFO
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MIX, ///< controls the the mix of input and chorus signals
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VOLUME, ///< controls the output volume level
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NUM_CONTROLS ///< this can be used as an alias for the number of MIDI controls
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}; |
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enum class Filter { |
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CE2 = 0, |
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WARM, |
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DARK |
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}; |
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/// Construct an analog chorus using internal memory. The chorus will have the
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/// default average delay.
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AudioEffectAnalogChorus(); |
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/// Construct an analog chorus using external SPI via an ExtMemSlot. The chorus will have
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/// the default average delay.
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/// @param slot A pointer to the ExtMemSlot to use for the delay.
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AudioEffectAnalogChorus(BALibrary::ExtMemSlot *slot); // requires sufficiently sized pre-allocated memory
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virtual ~AudioEffectAnalogChorus(); ///< Destructor
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// *** PARAMETERS ***
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/// Set the chorus average delay in milliseconds
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/// The value should be between 0.0f and 1.0f
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void setDelayConfig(float averageDelayMs, float delayRangeMs); |
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/// Set the chorus average delay in number of audio samples
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/// The value should be between 0.0f and 1.0f
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void setDelayConfig(size_t averageDelayNumSamples, size_t delayRangeNumSamples); |
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/// Bypass the effect.
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/// @param byp when true, bypass wil disable the effect, when false, effect is enabled.
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/// Note that audio still passes through when bypass is enabled.
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void bypass(bool byp) { m_bypass = byp; } |
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/// Get if the effect is bypassed
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/// @returns true if bypassed, false if not bypassed
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bool isBypass() { return m_bypass; } |
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/// Toggle the bypass effect
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void toggleBypass() { m_bypass = !m_bypass; } |
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/// Set the LFO frequency where 0.0f is MIN and 1.0f is MAX
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void rate(float rate); |
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/// Set the depth of LFO modulation.
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/// @param lfoDepth must be a float between 0.0f and 1.0f
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void depth(float lfoDepth) { m_lfoDepth = lfoDepth; } |
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/// Set the amount of blending between dry and wet (echo) at the output.
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/// @param mix When 0.0, output is 100% dry, when 1.0, output is 100% wet. When
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/// 0.5, output is 50% Dry, 50% Wet.
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void mix(float mix) { m_mix = mix; } |
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/// Set the output volume. This affect both the wet and dry signals.
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/// @details The default is 1.0.
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/// @param vol Sets the output volume between -1.0 and +1.0
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void volume(float vol) {m_volume = vol; } |
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// ** ENABLE / DISABLE **
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/// Enables audio processing. Note: when not enabled, CPU load is nearly zero.
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void enable() { m_enable = true; } |
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/// Disables audio process. When disabled, CPU load is nearly zero.
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void disable() { m_enable = false; } |
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// ** MIDI **
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/// Sets whether MIDI OMNI channel is processig on or off. When on,
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/// all midi channels are used for matching CCs.
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/// @param isOmni when true, all channels are processed, when false, channel
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/// must match configured value.
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void setMidiOmni(bool isOmni) { m_isOmni = isOmni; } |
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/// Configure an effect parameter to be controlled by a MIDI CC
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/// number on a particular channel.
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/// @param parameter one of the parameter names in the class enum
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/// @param midiCC the CC number from 0 to 127
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/// @param midiChannel the effect will only response to the CC on this channel
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/// when OMNI mode is off.
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void mapMidiControl(int parameter, int midiCC, int midiChannel = 0); |
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/// process a MIDI Continous-Controller (CC) message
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/// @param channel the MIDI channel from 0 to 15)
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/// @param midiCC the CC number from 0 to 127
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/// @param value the CC value from 0 to 127
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void processMidi(int channel, int midiCC, int value); |
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// ** FILTER COEFFICIENTS **
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/// Set the filter coefficients to one of the presets. See AudioEffectAnalogChorus::Filter
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/// for options.
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/// @details See AudioEffectAnalogChorusFIlters.h for more details.
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/// @param filter the preset filter. E.g. AudioEffectAnalogChorus::Filter::WARM
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void setFilter(Filter filter); |
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/// Override the default coefficients with your own. The number of filters stages affects how
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/// much CPU is consumed.
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/// @details The effect uses the CMSIS-DSP library for biquads which requires coefficents.
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/// be in q31 format, which means they are 32-bit signed integers representing -1.0 to slightly
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/// less than +1.0. The coeffShift parameter effectively multiplies the coefficients by 2^shift. <br>
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/// Example: If you really want +1.5, must instead use +0.75 * 2^1, thus 0.75 in q31 format is
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/// (0.75 * 2^31) = 1610612736 and coeffShift = 1.
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/// @param numStages the actual number of filter stages you want to use. Must be <= MAX_NUM_FILTER_STAGES.
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/// @param coeffs pointer to an integer array of coefficients in q31 format.
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/// @param coeffShift Coefficient scaling factor = 2^coeffShift.
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void setFilterCoeffs(int numStages, const int32_t *coeffs, int coeffShift); |
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virtual void update(void); ///< update automatically called by the Teesny Audio Library
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private: |
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static constexpr float m_DEFAULT_DELAY_MS = 20.0f; ///< default average delay of chorus in milliseconds
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static constexpr float m_DELAY_RANGE = 15.0f; ///< default range of delay variation in milliseconds
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static constexpr float m_LFO_MIN_RATE = 2.0f; ///< slowest possible LFO rate in milliseconds
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static constexpr float m_LFO_RANGE = 8.0f; ///< fastest possible LFO rate in milliseconds
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audio_block_t *m_inputQueueArray[1]; |
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bool m_isOmni = false; |
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bool m_bypass = true; |
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bool m_enable = false; |
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bool m_externalMemory = false; |
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BALibrary::AudioDelay *m_memory = nullptr; |
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BALibrary::LowFrequencyOscillatorVector<float> m_lfo; |
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size_t m_maxDelaySamples = 0; |
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audio_block_t *m_previousBlock = nullptr; |
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audio_block_t *m_blockToRelease = nullptr; |
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BALibrary::IirBiQuadFilterHQ *m_iir = nullptr; |
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// Controls
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int m_midiConfig[NUM_CONTROLS][2]; // stores the midi parameter mapping
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size_t m_delaySamples = 0; |
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float m_lfoDepth = 0.0f; |
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float m_mix = 0.0f; |
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float m_volume = 1.0f; |
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void m_preProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet); |
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void m_postProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet); |
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// Coefficients
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void m_constructFilter(void); |
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}; |
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} |
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#endif /* __BAEFFECTS_BAAUDIOEFFECTAnalogChorus_H */ |
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@ -0,0 +1,83 @@ |
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/**************************************************************************//**
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* @file |
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* @author Steve Lascos |
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* @company Blackaddr Audio |
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* |
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* This file constains precomputed co-efficients for the AudioEffectAnalogChorus |
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* class. |
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* |
||||
* @copyright This program is free software: you can redistribute it and/or modify |
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* it under the terms of the GNU General Public License as published by |
||||
* the Free Software Foundation, either version 3 of the License, or |
||||
* (at your option) any later version.* |
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* |
||||
* This program is distributed in the hope that it will be useful, |
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of |
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
||||
* GNU General Public License for more details. |
||||
* |
||||
* You should have received a copy of the GNU General Public License |
||||
* along with this program. If not, see <http://www.gnu.org/licenses/>.
|
||||
*****************************************************************************/ |
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#include <cstdint> |
||||
|
||||
namespace BAEffects { |
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|
||||
// The number of stages in the analog-response Biquad filter
|
||||
constexpr unsigned MAX_NUM_FILTER_STAGES = 4; |
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constexpr unsigned NUM_COEFFS_PER_STAGE = 5; |
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|
||||
// Matlab/Octave can be helpful to design a filter. Once you have the IIR filter (bz,az) coefficients
|
||||
// in the z-domain, they can be converted to second-order-sections. AudioEffectAnalogChorus is designed
|
||||
// to accept up to a maximum of an 8th order filter, broken into four, 2nd order stages.
|
||||
//
|
||||
// Second order sections can be created with:
|
||||
// [sos] = tf2sos(bz,az);
|
||||
// The results coefficents must be converted the Q31 format required by the ARM CMSIS-DSP library. This means
|
||||
// all coefficients must lie between -1.0 and +0.9999. If your (bz,az) coefficients exceed this, you must divide
|
||||
// them down by a power of 2. For example, if your largest magnitude coefficient is -3.5, you must divide by
|
||||
// 2^shift where 4=2^2 and thus shift = 2. You must then mutliply by 2^31 to get a 32-bit signed integer value
|
||||
// that represents the required Q31 coefficient.
|
||||
|
||||
// BOSS DM-3 Filters
|
||||
// b(z) = 1.0e-03 * (0.0032 0.0257 0.0900 0.1800 0.2250 0.1800 0.0900 0.0257 0.0032)
|
||||
// a(z) = 1.0000 -5.7677 14.6935 -21.3811 19.1491 -10.5202 3.2584 -0.4244 -0.0067
|
||||
constexpr unsigned CE2_NUM_STAGES = 4; |
||||
constexpr unsigned CE2_COEFF_SHIFT = 2; |
||||
constexpr int32_t CE2[5*MAX_NUM_FILTER_STAGES] = { |
||||
536870912, 988616936, 455608573, 834606945, -482959709, |
||||
536870912, 1031466345, 498793368, 965834205, -467402235, |
||||
536870912, 1105821939, 573646688, 928470657, -448083489, |
||||
2339, 5093, 2776, 302068995, 4412722 |
||||
}; |
||||
|
||||
|
||||
// Blackaddr WARM Filter
|
||||
// Butterworth, 8th order, cutoff = 2000 Hz
|
||||
// Matlab/Octave command: [bz, az] = butter(8, 2000/44100/2);
|
||||
// b(z) = 1.0e-05 * (0.0086 0.0689 0.2411 0.4821 0.6027 0.4821 0.2411 0.0689 0.0086_
|
||||
// a(z) = 1.0000 -6.5399 18.8246 -31.1340 32.3473 -21.6114 9.0643 -2.1815 0.2306
|
||||
constexpr unsigned WARM_NUM_STAGES = 4; |
||||
constexpr unsigned WARM_COEFF_SHIFT = 2; |
||||
constexpr int32_t WARM[5*MAX_NUM_FILTER_STAGES] = { |
||||
536870912,1060309346,523602393,976869875,-481046241, |
||||
536870912,1073413910,536711084,891250612,-391829326, |
||||
536870912,1087173998,550475248,835222426,-333446881, |
||||
46,92,46,807741349,-304811072 |
||||
}; |
||||
|
||||
// Blackaddr DARK Filter
|
||||
// Chebychev Type II, 8th order, stopband = 60db, cutoff = 1000 Hz
|
||||
// Matlab command: [bz, az] = cheby2(8, 60, 1000/44100/2);
|
||||
// b(z) = 0.0009 -0.0066 0.0219 -0.0423 0.0522 -0.0423 0.0219 -0.0066 0.0009
|
||||
// a(z) = 1.0000 -7.4618 24.3762 -45.5356 53.1991 -39.8032 18.6245 -4.9829 0.5836
|
||||
constexpr unsigned DARK_NUM_STAGES = 4; |
||||
constexpr unsigned DARK_COEFF_SHIFT = 1; |
||||
constexpr int32_t DARK[5*MAX_NUM_FILTER_STAGES] = { |
||||
1073741824,-2124867808,1073741824,2107780229,-1043948409, |
||||
1073741824,-2116080466,1073741824,2042553796,-979786242, |
||||
1073741824,-2077777790,1073741824,1964779896,-904264933, |
||||
957356,-1462833,957356,1896884898,-838694612 |
||||
}; |
||||
|
||||
}; |
||||
Loading…
Reference in new issue