From 5c6bd904466efdecacb6924b5efdc07cd0280d97 Mon Sep 17 00:00:00 2001 From: Chip Audette Date: Thu, 29 Dec 2016 18:47:06 -0500 Subject: [PATCH] Add AudioEffectCompressor plus example --- AudioEffectCompressor_F32.h | 165 ++++++++++++++++++ OpenAudio_ArduinoLibrary.h | 1 + .../BasicCompressor_Float.ino | 165 ++++++++++++++++++ keywords.txt | 4 + 4 files changed, 335 insertions(+) create mode 100644 AudioEffectCompressor_F32.h create mode 100644 examples/BasicCompressor_Float/BasicCompressor_Float.ino diff --git a/AudioEffectCompressor_F32.h b/AudioEffectCompressor_F32.h new file mode 100644 index 0000000..607ce21 --- /dev/null +++ b/AudioEffectCompressor_F32.h @@ -0,0 +1,165 @@ +/* + AudioEffectCompressor + + Created: Chip Audette, December 2016 + Purpose; Apply dynamic range compression to the audio stream. + Assumes floating-point data. + + This processes a single stream fo audio data (ie, it is mono) + + MIT License. use at your own risk. +*/ + +#include //ARM DSP extensions. https://www.keil.com/pack/doc/CMSIS/DSP/html/index.html +#include + +class AudioEffectCompressor_F32 : public AudioStream_F32 +{ + public: + //constructor + AudioEffectCompressor_F32(void) : AudioStream_F32(1, inputQueueArray_f32) { + setThresh_dBFS(-20.0f); //default to this threshold + setAttack_sec(0.005f, AUDIO_SAMPLE_RATE); //default to this value + setRelease_sec(0.200f, AUDIO_SAMPLE_RATE); //default to this value + setCompressionRatio(5.0f); //default to this value + setThresh_dBFS(-20.0f); //default to this value + setHPFilterCoeff(); + resetStates(); + }; + + //here's the method that does all the work + void update(void) { + //Serial.println("AudioEffectGain_F32: updating."); //for debugging. + audio_block_f32_t *audio_block; + audio_block = AudioStream_F32::receiveWritable_f32(); + if (!audio_block) return; + + //apply a high-pass filter to get rid of the DC offset + if (use_HP_prefilter) arm_biquad_cascade_df1_f32(&hp_filt_struct, audio_block->data, audio_block->data, audio_block->length); + + //apply the pre-gain...a negative gain value will disable + if (pre_gain > 0.0f) arm_scale_f32(audio_block->data, pre_gain, audio_block->data, audio_block->length); //use ARM DSP for speed! + + //compute the desired gain + audio_block_f32_t *gain_block = AudioStream_F32::allocate_f32(); + calcGain(audio_block, gain_block); //returns through gain_block + + //apply the gain...store it back into audio_block + arm_mult_f32(audio_block->data, gain_block->data, audio_block->data, audio_block->length); + + ///transmit the block and release memory + AudioStream_F32::transmit(audio_block); + AudioStream_F32::release(audio_block); + AudioStream_F32::release(gain_block); + } + + void calcGain(audio_block_f32_t *wav_block, audio_block_f32_t *gain_block) { + + //calculate the signal power...ie, square the signal: wav_pow = wav.^2 + audio_block_f32_t *wav_pow_block = AudioStream_F32::allocate_f32(); + arm_mult_f32(wav_block->data, wav_block->data, wav_pow_block->data, wav_block->length); + + //loop over each sample + float32_t gain_pow; + for (int i = 0; i < wav_pow_block->length; i++) { + + //compute target gain (well, we're actualy calculating gain^2) assuming we want to copress + gain_pow = thresh_pow_FS_wCR / powf(wav_pow_block->data[i], comp_ratio_const); + + //if our signal level is below the threshold, don't compress (set target gain to 0dB, which is 1.0) + if (wav_pow_block->data[i] < thresh_pow_FS) gain_pow = 1.0f; + + //are we in the attack mode or release mode? + float32_t c = attack_const; //at first, assume that we're in the attack phase + if (gain_pow > prev_gain_pow) c = release_const; //here, we decide if we're really in the release phase + + //smooth the gain using the attack or release constants + gain_pow = c*prev_gain_pow + (1.0f-c)*gain_pow; + + //take he sqrt of gain^2 so that we simply get the gain + //arm_sqrt_f32(gain_pow, &(gain_block->data[i])); //should use the DSP acceleration, if the right CMSIS library is used + //gain_block->data[i] = __builtin_sqrtf(gain_pow); //seems to give the same speed as the arm_sqrt_f32 + gain_block->data[i] = sqrtf(gain_pow); //also give the same speed and is more portable + + //save value for the next time through this loop + prev_gain_pow = gain_pow; + } + + //free up the memory and return + release(wav_pow_block); + return; //the output here is gain_block + } + + //methods to set parameters of this module + void resetStates(void) { + prev_gain_pow = 1.0f; + + //initialize the HP filter (it also resets the filter states) + arm_biquad_cascade_df1_init_f32(&hp_filt_struct, hp_nstages, hp_coeff, hp_state); + } + void setPreGain(float g) { pre_gain = g; } + void setPreGain_dB(float gain_dB) { setPreGain(pow(10.0, gain_dB / 20.0)); } + void setCompressionRatio(float cr) { + comp_ratio = max(0.001, cr); //limit to positive values + updateThresholdAndCompRatioConstants(); + } + void setAttack_sec(float a, float fs_Hz) { + attack_sec = a; + attack_const = expf(-1.0f / (attack_sec * fs_Hz)); //expf() is much faster than exp() + } + void setRelease_sec(float r, float fs_Hz) { + release_sec = r; + release_const = expf(-1.0f / (release_sec * fs_Hz)); //expf() is much faster than exp() + } + void setThresh_dBFS(float thresh_dBFS) { setThreshPow(pow(10.0, thresh_dBFS / 10.0)); } + void setThreshPow(float t_pow) { + thresh_pow_FS = t_pow; + updateThresholdAndCompRatioConstants(); + } + void enableHPFilter(boolean flag) { use_HP_prefilter = flag; }; + + //methods to return information about this module + float32_t getPreGain_dB(void) { return 20.0 * log10(pre_gain); } + float32_t getAttack_sec(void) { return attack_sec; } + float32_t getRelease_sec(void) { return release_sec; } + float32_t getThresh_dBFS(void) { return 10.0 * log10(thresh_pow_FS); } + float32_t getCompressionRatio(void) { return comp_ratio; } + float32_t getCurrentGain_dB(void) { return 10.0 * log10(prev_gain_pow); } + + private: + //state-related variables + audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module + float32_t prev_gain_pow = 1.0; //last gain^2 used + + //HP filter state-related variables + arm_biquad_casd_df1_inst_f32 hp_filt_struct; + static const uint8_t hp_nstages = 1; + float32_t hp_coeff[5 * hp_nstages] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later + float32_t hp_state[4 * hp_nstages]; + void setHPFilterCoeff(void) { + //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5 + //Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100 + float32_t b[] = {9.979871156751189e-01, -1.995974231350238e+00, 9.979871156751189e-01}; //from Matlab + float32_t a[] = { 1.000000000000000e+00, -1.995970179642828e+00, 9.959782830576472e-01}; //from Matlab + hp_coeff[0] = b[0]; hp_coeff[1] = b[1]; hp_coeff[2] = b[2]; //here are the matlab "b" coefficients + hp_coeff[3] = -a[1]; hp_coeff[4] = -a[2]; //the DSP needs the "a" terms to have opposite sign vs Matlab + } + + //private parameters related to gain calculation + float32_t attack_const, release_const; //used in calcGain(). set by setAttack_sec() and setRelease_sec(); + float32_t comp_ratio_const, thresh_pow_FS_wCR; //used in calcGain(); set in updateThresholdAndCompRatioConstants() + void updateThresholdAndCompRatioConstants(void) { + comp_ratio_const = 1.0f-(1.0f / comp_ratio); + thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const); //powf() is much faster than pow() + } + + //settings + float32_t attack_sec, release_sec; + float32_t thresh_pow_FS = 1.0f; //threshold for compression, relative to digital full scale + float32_t comp_ratio = 1.0; //compression ratio + float32_t pre_gain = -1.0; //gain to apply before the compression. negative value disables + boolean use_HP_prefilter = false; + +}; + + diff --git a/OpenAudio_ArduinoLibrary.h b/OpenAudio_ArduinoLibrary.h index 43ebd1f..dc2d4e0 100644 --- a/OpenAudio_ArduinoLibrary.h +++ b/OpenAudio_ArduinoLibrary.h @@ -2,5 +2,6 @@ #include #include #include +#include #include diff --git a/examples/BasicCompressor_Float/BasicCompressor_Float.ino b/examples/BasicCompressor_Float/BasicCompressor_Float.ino new file mode 100644 index 0000000..36a5973 --- /dev/null +++ b/examples/BasicCompressor_Float/BasicCompressor_Float.ino @@ -0,0 +1,165 @@ +/* + BasicCompressor + + Created: Chip Audette, Dec 2016 + Purpose: Process audio by applying a single-band compressor + Demonstrates audio processing using floating point data type. + + Uses Teensy Audio Adapter. + Assumes microphones (or whatever) are attached to the LINE IN (stereo) + + MIT License. use at your own risk. +*/ + +//These are the includes from the Teensy Audio Library +#include //Teensy Audio Librarya +#include +#include +#include +#include + +#include //for AudioConvert_I16toF32, AudioConvert_F32toI16, and AudioEffectGain_F32 + +//create audio library objects for handling the audio +AudioControlSGTL5000_Extended sgtl5000; //controller for the Teensy Audio Board +AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo. +AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo +AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h +AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h +AudioEffectCompressor_F32 comp1, comp2; + +//Make all of the audio connections +AudioConnection patchCord1(i2s_in, 0, int2Float1, 0); //connect the Left input to the Left Int->Float converter +AudioConnection patchCord2(i2s_in, 1, int2Float2, 0); //connect the Right input to the Right Int->Float converter +AudioConnection_F32 patchCord10(int2Float1, 0, comp1, 0); //Left. makes Float connections between objects +AudioConnection_F32 patchCord11(int2Float2, 0, comp2, 0); //Right. makes Float connections between objects +AudioConnection_F32 patchCord12(comp1, 0, float2Int1, 0); //Left. makes Float connections between objects +AudioConnection_F32 patchCord13(comp2, 0, float2Int2, 0); //Right. makes Float connections between objects +AudioConnection patchCord20(float2Int1, 0, i2s_out, 0); //connect the Left float processor to the Left output +AudioConnection patchCord21(float2Int2, 0, i2s_out, 1); //connect the Right float processor to the Right output + +// which input on the audio shield will be used? +const int myInput = AUDIO_INPUT_LINEIN; +//const int myInput = AUDIO_INPUT_MIC; + +//I have a potentiometer on the Teensy Audio Board +#define POT_PIN A1 //potentiometer is tied to this pin + +//define a function to setup the Teensy Audio Board how I like it +void setupMyAudioBoard(void) { + sgtl5000.enable(); //start the audio board + sgtl5000.inputSelect(myInput); //choose line-in or mic-in + sgtl5000.volume(0.8); //volume can be 0.0 to 1.0. 0.5 seems to be the usual default. + sgtl5000.lineInLevel(10,10); //level can be 0 to 15. 5 is the Teensy Audio Library's default + sgtl5000.adcHighPassFilterDisable(); //reduces noise. https://forum.pjrc.com/threads/27215-24-bit-audio-boards?p=78831&viewfull=1#post78831 + sgtl5000.micBiasEnable(3.0); //enable the mic bias voltage...only in AudioControlSGTL5000_Extended +} + +//define a function to configure the left and right compressors +void setupMyCompressors(boolean use_HP_filter, float knee_dBFS, float comp_ratio, float attack_sec, float release_sec) { + comp1.enableHPFilter(use_HP_filter); comp2.enableHPFilter(use_HP_filter); + comp1.setThresh_dBFS(knee_dBFS); comp2.setThresh_dBFS(knee_dBFS); + comp1.setCompressionRatio(comp_ratio); comp2.setCompressionRatio(comp_ratio); + + float fs_Hz = AUDIO_SAMPLE_RATE; + comp1.setAttack_sec(attack_sec, fs_Hz); comp2.setAttack_sec(attack_sec, fs_Hz); + comp1.setRelease_sec(release_sec, fs_Hz); comp2.setRelease_sec(release_sec, fs_Hz); +} + +// define the overall setup() function, the function that is called once when the device is booting +void setup() { + Serial.begin(115200); //open the USB serial link to enable debugging messages + delay(500); //give the computer's USB serial system a moment to catch up. + Serial.println("Teensy Hearing Aid: BasicCompressor_Float..."); //identify myself over the USB serial + + // Audio connections require memory, and the record queue + // uses this memory to buffer incoming audio. + AudioMemory(12); //allocate Int16 audio data blocks + AudioMemory_F32(10); //allocate Float32 audio data blocks + + // Enable the audio shield, select input, and enable output + setupMyAudioBoard(); + + //choose the compressor parameters...note that preGain is set by the potentiometer in the main loop() + boolean use_HP_filter = true; //enable the software HP filter to get rid of DC? + float knee_dBFS, comp_ratio, attack_sec, release_sec; + if (false) { + Serial.println("Configuring Compressor for fast response for use as a limitter."); + knee_dBFS = -15.0; comp_ratio = 8.0; attack_sec = 0.005; release_sec = 0.200; + } else { + Serial.println("Configuring Compressor for slow response more like an automatic volume control."); + knee_dBFS = -50.0; comp_ratio = 5.0; attack_sec = 1.0; release_sec = 2.0; + } + + //configure the left and right compressors with the desired settings + setupMyCompressors(use_HP_filter, knee_dBFS, comp_ratio, attack_sec, release_sec); + + // setup any other other features + pinMode(POT_PIN, INPUT); //set the potentiometer's input pin as an INPUT + +} //end setup() + + +// define the loop() function, the function that is repeated over and over for the life of the device +unsigned long updatePeriod_millis = 100; //how many milliseconds between updating gain reading? +unsigned long lastUpdate_millis = 0; +unsigned long curTime_millis = 0; +int prev_gain_dB = 0; +unsigned long lastMemUpdate_millis=0; +void loop() { + //choose to sleep ("wait for interrupt") instead of spinning our wheels doing nothing but consuming power + asm(" WFI"); //ARM-specific. Will wake on next interrupt. The audio library issues tons of interrupts, so we wake up often. + + //has enough time passed to try updating the GUI? + curTime_millis = millis(); //what time is it right now + if (curTime_millis < lastUpdate_millis) lastUpdate_millis = 0; //handle wrap-around of the clock + if ((curTime_millis - lastUpdate_millis) > updatePeriod_millis) { //is it time to update the user interface? + + //read potentiometer + float32_t val = float(analogRead(POT_PIN)) / 1024.0f; //0.0 to 1.0 + val = 0.1*(float)((int)(10.0*val + 0.5)); //quantize so that it doesn't chatter + + //compute desired digital gain + const float min_gain_dB = -20.0, max_gain_dB = 40.0; //set desired gain range + float gain_dB = min_gain_dB + (max_gain_dB - min_gain_dB)*val; //computed desired gain value in dB + + //if the gain is different than before, set the new gain value + if (abs(gain_dB - prev_gain_dB) > 1.0) { //is it different than before + comp1.setPreGain_dB(gain_dB); //set the gain of the Left-channel gain processor + comp2.setPreGain_dB(gain_dB); //set the gain of the Right-channel gain processor + Serial.print("Setting Digital Pre-Gain dB = "); Serial.println(gain_dB); //print text to Serial port for debugging + prev_gain_dB = gain_dB; //we will use this value the next time around + } + + lastUpdate_millis = curTime_millis; //we will use this value the next time around. + } // end if + + + //print status information to the Serial port + if ((curTime_millis - lastMemUpdate_millis) > 2000) { // print a summary of the current & maximum usage + //printCompressorState(&Serial); + printCPUandMemoryUsage(&Serial); + lastMemUpdate_millis = curTime_millis; //we will use this value the next time around. + } + +} //end loop(); + +void printCompressorState(Stream *s) { + s->print("Current Compressor: Pre-Gain (dB) = "); + s->print(comp1.getPreGain_dB()); + s->print(", Dynamic Gain L/R (dB) = "); + s->print(comp1.getCurrentGain_dB()); + s->print(", "); + s->print(comp2.getCurrentGain_dB()); + s->println(); +}; + +void printCPUandMemoryUsage(Stream *s) { + s->print("Usage/Max: "); + s->print("comp1 CPU = "); s->print(comp1.processorUsage()); s->print("/"); s->print(comp1.processorUsageMax());s->print(", "); + s->print("all CPU = " ); s->print(AudioProcessorUsage()); s->print("/"); s->print(AudioProcessorUsageMax());s->print(", "); + s->print("Int16 Mem = ");s->print(AudioMemoryUsage()); s->print("/"); s->print(AudioMemoryUsageMax());s->print(", "); + s->print("Float Mem = ");s->print(AudioMemoryUsage_F32());s->print("/"); s->print(AudioMemoryUsageMax_F32()); s->print(", "); + s->println(); +}; + diff --git a/keywords.txt b/keywords.txt index 508385a..283d58c 100644 --- a/keywords.txt +++ b/keywords.txt @@ -2,14 +2,18 @@ OpenAudio_ArduinoLibrary KEYWORD1 #data type / class / static function names +float32_t KEYWORD1 audio_block_f32_t KEYWORD1 AudioStream_F32 KEYWORD1 AudioConnection_F32 KEYWORD1 AudioConvert_I16toF32 KEYWORD1 AudioConvert_F32toI16 KEYWORD1 AudioEffectGain_F32 KEYWORD1 +AudioEffectCompressor_F32 KEYWORD1 AudioMemory_F32 KEYWORD1 AudioMemoryUsage_F32 KEYWORD1 AudioMemoryUsageMax_F32 KEYWORD1 AudioMemoryUsageMaxReset_F32 KEYWORD1 +AudioControlSGTL5000_Extended KEYWORD1 +micBiasEnable KEYWORD2 \ No newline at end of file