/* * AudioEffectCompressor2_F32.h * * Bob Larkin W7PUA 11 December 2020 * * This is a general purpose audio compressor block (C++ class). It works by determining * the average input of the input signal, and based on a pre-determined curve, * changes the gain going through the block. * A good discussion is the Wikipedia page: * https://en.wikipedia.org/wiki/Dynamic_range_compression * This compressor includes up to 5 dB/dB line segments allowing for most of the * features listed. These include * Multi segment compression curves, up to 5 * Limiting * Approximation to "soft knees" * Expansion for suppressing low-level artifacts * Anticipation * Scale offset for use such as hearing-aid audiology * This is derived from the WDRC compressor. Chip Audette (OpenAudio) Feb 2017 * Which was derived From: WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro * As of Feb 2017, CHAPRO license is listed as "Creative Commons?" * * MIT License. Use at your own risk. */ /* Compressor #2. Amplifies input signals by varying amoounts depending * on the signal level. This is controlled by up to 5 line segments specified * by a point at the highest input level for the line along with a compression ratio * (1/slope) for the line segment. This can provide limiting at the highest inputs * by setting compressionRatio[0]=1000.0 (i.e., some big value). Expansion at the * lw levels provides a squelch action, using a very small compression * ratio like 0.01. * * A special case is the [0] segment, that continues at the same slope for inputs up * to any level. For this the kneeDB[0] can be the expected 0.0 or other values * on that line. The output level for kneeDB[0] is the input variable, marginDB. * This allows gain control near clipping but below. marginDB is typically 1 or 2 dB. * Vout dB | | 0.0 + kneeDB[0] marg + kneeDB[1] @******************** | @************ 1:compressionRatio[0] | **** | *** | *** | *** 1:compressionRatio[1] | kneeDB[2] *** | @*** | * | * | * | * | * 1:compressionRatio[2] | * | * === Vout in dB vs. Vin in dB === |* Three segment example * Knees (breakpoints) are shown with '@' *| compressionRatio[] are ratio of: input change (in dB):output change in dB * | * |________|___________________|____________________________|________ vIn dB k1 k2 0.0 * The graph shows the changes in gain on a log or dB scale. A compressionRatio * of 1 represents a constant gain with level. When the compressionRatio is greater * than 1, say 2.0, the voltage gain is decreasing as the input level increases. * * vInDB refers to the time averaged envelope voltage. * The zero reference is the full ADC range output. This is ±1.0 * peak or 0.707 RMS in F32 terminology. * * The curve is for gainOffsetDB = 0.0. This parameter raises and lowers the * input scale for the kneeDB[] parameter. * * Timing: For 44.1 kHz sample rate and 256 samples per update, the update( ) time * runs 240 to 270 icroseconds using Teensy 3.6. */ #ifndef _AUDIO_EFFECT_COMPRESSOR2_F32_H #define _AUDIO_EFFECT_COMPRESSOR2_F32_H #include #include // The following 3 defines are simplified implementations for common uses. // They replace the begin() function that is otherwise required. // See testCompressor2.ino example for how to use these defines. // None of these support offsetting the input scale as done in Tympan. /* limiterBegin(pointerObject, float marDB, float linearInDB) has 2 segments. * It is linear up to an input of linearInDB (typically -15.0f) and * then virtually limits for higher input levels. The output level at * this point is marDB, the margin to prevent clipping, like -2 dB. * This is not a clipper with waveform distortion, but rather decreases * the gain, dB for dB, as the input increases in the limiter region. * pobject is a pointer to the INO AudioEffectCompressor2_F32 object. * This function replaces begin() for the AudioEffectCompressor2_F32 object. */ #define limiterBegin(pobject, marDB, linearInDB) struct compressionCurve _curv={marDB,0.0f,{0.0,linearInDB,-1000.0f,-1000.0f,-1000.0f},{100.0,1.0f,1.0f,1.0f,1.0f}}; pobject->setCompressionCurve(&_curv); pobject->begin(); /* basicCompressorBegin has a 3 segments. It is linear up to an input linearInDB * and then decreases gain according to compressionRatioDB up to an input -10 dB where it * is almost limited, with an increase of output level of 1 dB for a 10 dB increase * in input level. The output level at full input is 1 dB below full output. * This function replaces begin() for the AudioEffectCompressor2_F32 object. */ #define basicCompressorBegin(pobject, linearInDB, compressionRatio) struct compressionCurve _curv={-1.0,0.0f,{0.0,-10.0f,linearInDB,-1000.0f,-1000.0f},{10.0f,compressionRatio,1.0f,1.0f,1.0f}}; pobject->setCompressionCurve(&_curv); pobject->begin(); /* squelchCompressorBegin is similar to basicCompression above, except that there is * an expansion region for low levels. So, the call defines the four regions in * terms of the input levels. squelchInDB sets the lowest input level * before the squelching effect starts. */ #define squelchCompressorBegin(pobject, squelchInDB, linearInDB, compressionInDB, compressionRatio) struct compressionCurve _curv={-1.0,0.0f,{0.0,compressionInDB,linearInDB,squelchInDB,-1000.0f},{10.0,compressionRatio,1.0f,0.1f,1.0f}}; pobject->setCompressionCurve(&_curv); pobject->begin(); // Basic definition of compression curve: struct compressionCurve { float marginDB; float offsetDB; float kneeDB[5]; float compressionRatio[5]; }; // --------------------------------------------------------------------------- class AudioEffectCompressor2_F32 : public AudioStream_F32 { //GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node //GUI: shortName: Compressor2 public: AudioEffectCompressor2_F32(void): AudioStream_F32(1, inputQueueArray) { setAttackReleaseSec(0.005f, 0.100f); } AudioEffectCompressor2_F32(const AudioSettings_F32 &settings): AudioStream_F32(1, inputQueueArray) { //setSampleRate_Hz(settings.sample_rate_Hz); setAttackReleaseSec(0.005f, 0.100f); } virtual void update(void); void begin(void); void setCompressionCurve(struct compressionCurve*); // A delay of 256 samples is 256/44100 = 0.0058 sec = 5.8 mSec void setDelayBufferSize(int16_t _delaySize) { // Any power of 2, i.e., 256, 128, 64, etc. delaySize = _delaySize; delayBufferMask = _delaySize - 1; in_index = 0; } void printOn(bool _printIO) { printIO = _printIO; } // Diagnostics ONLY. Not for general INO float getCurrentInputDB(void) { return sampleInputDB; } float getCurrentGainDB(void) { return sampleGainDB; } float getvInMaxDB(void) { float vRet = vInMaxDB; vInMaxDB = -1000.0f; // Reset for next max measure return vRet; } //convert time constants from seconds to unitless parameters, from CHAPRO, agc_prepare.c void setAttackReleaseSec(const float atk_sec, const float rel_sec) { // convert ANSI attack & release times to filter time constants float ansi_atk = atk_sec * sample_rate_Hz / 2.425f; float ansi_rel = rel_sec * sample_rate_Hz / 1.782f; alpha = (float) (ansi_atk / (1.0f + ansi_atk)); oneMinusAlpha = 1.0f - alpha; beta = (float) (ansi_rel / (1.0f + ansi_rel)); } private: audio_block_f32_t *inputQueueArray[1]; float delayData[256]; // The circular delay line for the signal uint16_t in_index = 0; // Pointer to next block update entry // And a mask to make the circular buffer limit to a power of 2 uint16_t delayBufferMask = 0X00FF; uint16_t delaySize = 256; float sample_rate_Hz = 44100; float attackSec = 0.005f; // Q: Can this be reduced with the delay line added to the signal path?? float releaseSec = 0.100f; // This alpha, beta for 5 ms attack, 100ms release, about 0.07 dB max ripple at 1000 Hz float alpha = 0.98912216f; float oneMinusAlpha = 0.01087784f; float beta = 0.9995961f; /* Definition of the compression curve. * Input and output are normally referenced to the full scale range (-1.0 to 1.0 in float). * The full scale point is called 0 dB. * There are 5 slopes, specified by the top input level for each line segment (kneeDB[]) * and the Compression Ratio (1/slope) of the segment (compressionRatio[]). * These are numbered from the highest to the lowest allowing * the number of segments to be adjusted, i.e., there is always a segment 0 * but there may not be a segment 3 and 4, for instance. This becomes very flexible, * as there can be, say, compression for top levels and expansion for very low levels. * * A special case is segment 0. On a plot of output dB vs input db, an input * level of kneeDB[0] produces an output of marginDB. A typical marginDB might * be -1.0 or -2.0. This margin allows controlling the gain without clipping in the DAC. * * This structure can have multiple curve structures like cCurve that can be used by * cmpr1.setCompressionCurve(&cCurve); * cmpr1.begin(); * * Unused segments can have knees like kneeDB[4] at, say -1000.0f. * Values below -500 will slightly speed up the update() function. if the bottom * two segments are not used, kneeDB[3] would also be set to -1000.0f amd so forth. * * Finally, there is a variable offsetDB that allows for a shift in the input scale. * It simply shifts the definition input levels, in dB. * This allows converting from DSP scales that have 0dB at full DSP scale * to auditory scales such as are used in the Tympan library. An offsetDB=119.0 * would allow all inputs to be in "SPL" units with a maximum input value of 119. */ struct compressionCurve curve0 = { -2.0f, 0.0f, // margin, offset {0.0f, -10.0f, -20.0f, -30.0f, -1000.0f}, // kneeDB[] { 100.0f, 2.0f, 1.5f, 1.0f, 1.0f} }; // compressionRatio // VoutDB at each knee, needed to find gain; determined at begin(): float outKneeDB[5]={-2.0f, -3.0f, -8.0f -978.0f, -978.0f}; // slopes are 1/compressionRatio determined at begin() to save update{} time float slope[5] = {0.01f, 0.5f, 0.666667f, 1.0f, 1.0f}; // Save time in update() by ignoring unused (low level) segments: uint16_t firstIndex = 3; float vPeakSave = 0.0f; float vInMaxDB = -1000.0f; // Only for reporting bool printIO = false; // Diagnostics Only float sampleInputDB, sampleGainDB; }; #endif