From 01cb7ba10556daffaa6d99cd155335014198618b Mon Sep 17 00:00:00 2001
From: Max Huster <maxekinge@googlemail.com>
Date: Sun, 7 Feb 2021 19:42:57 +0100
Subject: [PATCH] add NoiseGate effect

---
 AudioEffectNoiseGate_F32.h | 180 +++++++++++++++++++++++++++++++++++++
 1 file changed, 180 insertions(+)
 create mode 100644 AudioEffectNoiseGate_F32.h

diff --git a/AudioEffectNoiseGate_F32.h b/AudioEffectNoiseGate_F32.h
new file mode 100644
index 0000000..b3e595d
--- /dev/null
+++ b/AudioEffectNoiseGate_F32.h
@@ -0,0 +1,180 @@
+/*
+ * AudioEffectNoiseGate_F32
+ * 
+ * Created: Max Huster, Feb 2021
+ * Purpose: This module mutes the Audio completly, when it's below a given threshold.
+ *          
+ * This processes a single stream fo audio data (ie, it is mono)       
+ *          
+ * MIT License.  use at your own risk.
+*/
+
+#ifndef _AudioEffectNoiseGate_F32_h
+#define _AudioEffectNoiseGate_F32_h
+
+#include <arm_math.h> //ARM DSP extensions.  for speed!
+#include <AudioStream_F32.h>
+
+#ifdef NOISEGATE_EXTENDEDINFO
+const float minLinear = pow10f(-60 / 20.0f);
+#endif
+
+class AudioEffectNoiseGate_F32 : public AudioStream_F32
+{
+  //GUI: inputs:1, outputs:1  //this line used for automatic generation of GUI node
+  //GUI: shortName:NoiseGate
+public:
+  //constructor
+  AudioEffectNoiseGate_F32(void) : AudioStream_F32(1, inputQueueArray_f32){};
+  AudioEffectNoiseGate_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32){};
+
+  //here's the method that does all the work
+  void update(void)
+  {
+
+    //Serial.println("AudioEffectNoiseGate_F32: updating.");  //for debugging.
+    audio_block_f32_t *block;
+    block = AudioStream_F32::receiveWritable_f32();
+    if (!block)
+      return;
+    // create a new audio block for the gain
+    audio_block_f32_t *gainBlock = AudioStream_F32::allocate_f32();
+    // calculate the desired gain
+    calcGain(block, gainBlock);
+    // smooth the "blocky" gain block
+    calcSmoothedGain(gainBlock);
+#ifdef NOISEGATE_EXTENDEDINFO
+
+    float32_t min;
+    float32_t max;
+    uint32_t index;
+    arm_min_f32(gainBlock->data, gainBlock->length, &min, &index);
+    arm_max_f32(gainBlock->data, gainBlock->length, &max, &index);
+
+    _inBetween = min > minLinear && max < (1 - minLinear);
+#endif
+    // multiply it to the input singal
+    arm_mult_f32(gainBlock->data, block->data, block->data, block->length);
+    // release gainBlock
+    AudioStream_F32::release(gainBlock);
+
+    //transmit the block and be done
+    AudioStream_F32::transmit(block);
+    AudioStream_F32::release(block);
+  }
+
+  void setThreshold(float dbfs)
+  {
+    // convert dbFS to linear value to comapre against later
+    linearThreshold = pow10f(dbfs / 20.0f);
+  }
+
+  void setOpeningTime(float timeInSeconds)
+  {
+    openingTimeConst = expf(-1.0f / (timeInSeconds * AUDIO_SAMPLE_RATE));
+  }
+
+  void setClosingTime(float timeInSeconds)
+  {
+    closingTimeConst = expf(-1.0f / (timeInSeconds * AUDIO_SAMPLE_RATE));
+  }
+
+  void setHoldTime(float timeInSeconds)
+  {
+    holdTimeNumSamples = timeInSeconds * AUDIO_SAMPLE_RATE;
+  }
+
+#ifdef NOISEGATE_EXTENDEDINFO
+  bool infoIsOpeningOrClosing()
+  {
+    return _inBetween;
+  }
+
+#endif
+  bool infoIsOpen()
+  {
+    return _isOpenDisplay;
+  }
+
+private:
+  float32_t linearThreshold;
+  float32_t prev_gain_dB = 0;
+  float32_t openingTimeConst, closingTimeConst;
+  float lastGainBlockValue = 0;
+  int32_t counter, holdTimeNumSamples = 0;
+  audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
+  bool falling = false;
+
+  bool _isOpen = false;
+  bool _isOpenDisplay = false;
+
+#ifdef NOISEGATE_EXTENDEDINFO
+  bool _inBetween = false;
+#endif
+  void calcGain(audio_block_f32_t *input, audio_block_f32_t *gainBlock)
+  {
+    _isOpen = false;
+    for (int i = 0; i < input->length; i++)
+    {
+      // take absolute value and compare it to the set threshold
+      bool isAboveThres = abs(input->data[i]) > linearThreshold;
+      _isOpen |= isAboveThres;
+      // if above the threshold set volume to 1 otherwise to 0, we did not account for holdtime
+      gainBlock->data[i] = isAboveThres ? 1 : 0;
+
+      // if we are falling and are above the threshold, the level is not falling
+      if (falling & isAboveThres)
+      {
+        falling = false;
+      }
+      // if we have a falling signal
+      if (falling || lastGainBlockValue > gainBlock->data[i])
+      {
+        // check whether the hold time is not reached
+        if (counter < holdTimeNumSamples)
+        {
+          // signal is (still) falling
+          falling = true;
+          counter++;
+          gainBlock->data[i] = 1.0f;
+        }
+        // otherwise the signal is already muted due to the line: "gainBlock->data[i] = isAboveThres ? 1 : 0;"
+      }
+      // note the last gain value, so we can compare it if the signal is falling in the next sample
+      lastGainBlockValue = gainBlock->data[i];
+    }
+    // note the display value
+    _isOpenDisplay = _isOpen;
+  };
+
+  //this method applies the "opening" and "closing" constants to smooth the
+  //target gain level through time.
+  void calcSmoothedGain(audio_block_f32_t *gain_block)
+  {
+    float32_t gain;
+    float32_t one_minus_opening_const = 1.0f - openingTimeConst;
+    float32_t one_minus_closing_const = 1.0f - closingTimeConst;
+    for (int i = 0; i < gain_block->length; i++)
+    {
+      gain = gain_block->data[i];
+
+      //smooth the gain using the opening or closing constants
+      if (gain < prev_gain_dB)
+      { //are we in the opening phase?
+        gain_block->data[i] = openingTimeConst * prev_gain_dB + one_minus_opening_const * gain;
+      }
+      else
+      { //or, we're in the closing phase
+        gain_block->data[i] = closingTimeConst * prev_gain_dB + one_minus_closing_const * gain;
+      }
+
+      //save value for the next time through this loop
+      prev_gain_dB = gain_block->data[i];
+    }
+
+    //return
+    return; //the output here is gain_block
+  }
+};
+
+#endif