mem bug fixes, phaser bypass

src/basic_allpass.h

@@ -16,10 +16,8 @@ template <int N>
 class AudioFilterAllpass
 {
 public:
-	~AudioFilterAllpass()
-	{
-		if (bf) free(bf);
-	}
+	AudioFilterAllpass() { bf = NULL; }
+	~AudioFilterAllpass() { free(bf); }
 	/**
 	 * @brief Allocate the filter buffer in RAM
 	 * 			set the pointer to the allpass coeff
@@ -28,6 +26,7 @@ class AudioFilterAllpass
 	 */
 	bool init(float* coeffPtr)
 	{
+		free(bf);
 		bf = (float *)malloc(N*sizeof(float)); // allocate buffer
 		if (!bf) return false;
 		kPtr = coeffPtr;

src/basic_delay.h

@@ -15,21 +15,19 @@
 
 
 /**
- * @brief Basic delay line with buffer placed in PSRAM
+ * @brief Basic delay line with buffer placed in PSRAM or DMARAM
  * 
  * @tparam N delay length in samples (float)
  */
 //template <int N>
 class AudioBasicDelay
 {
 public:
-	~AudioBasicDelay()
-	{
-		if(bf) free(bf);
-	}
+	AudioBasicDelay() { bf = NULL; }
+	~AudioBasicDelay() { free(bf); }
 	bool init(uint32_t size_samples,  bool psram=false)
 	{
-		if(bf) free(bf);
+		free(bf);
 		use_psram = psram;
 		size = size_samples;
 		if (use_psram) 	bf = (float *)extmem_malloc(size * sizeof(float)); 	// allocate buffer in PSRAM

src/basic_pitch.h

@@ -23,6 +23,8 @@ extern const float music_intevals[];		// semitone intervals -1oct to +2oct
 class AudioBasicPitch
 {
 public:
+	AudioBasicPitch() { bf = NULL; }
+	~AudioBasicPitch() { free(bf); }
 	bool init()
 	{
 		outFilter.init(hp_f, (float *)&hp_gain, lp_f, &lp_gain);
@@ -43,7 +45,6 @@ class AudioBasicPitch
 	}
 	void setTone(float t)
 	{
-		//lp_f = constrain(t, 0.01f, 1.0f);
 		lp_gain = constrain(t, 0.0f, 1.0f);
 	}
 

src/effect_delaystereo_i16.h

@@ -48,7 +48,7 @@ class AudioEffectDelayStereo_i16 : public AudioStream
 	{
 		t = constrain(t, 0.0f, 1.0f);
 		t = t * t;
-		t = map(t, 0.0f, 1.0f, (float32_t)(dly_length-dly_time_min), 0.0f);
+		t = map_sat(t, 0.0f, 1.0f, (float32_t)(dly_length-dly_time_min), 0.0f);
 		__disable_irq();
 		if (force) dly_time = t;
 		dly_time_set = t;
@@ -77,8 +77,8 @@ class AudioEffectDelayStereo_i16 : public AudioStream
 		if (infinite) return;
 		float32_t fb, attn;
         n = constrain(n, 0.0f, 1.0f);
-	    fb = map(n, 0.0f, 1.0f, 0.0f, feedb_max) * hp_feedb_limit;
-        attn = map(n*n*n, 0.0f, 1.0f, 1.0f, 0.4f);
+	    fb = map_sat(n, 0.0f, 1.0f, 0.0f, feedb_max) * hp_feedb_limit;
+        attn = map_sat(n*n*n, 0.0f, 1.0f, 1.0f, 0.4f);
 		inputGain_tmp = attn;
         __disable_irq();
         feedb = fb;
@@ -95,7 +95,7 @@ class AudioEffectDelayStereo_i16 : public AudioStream
 	{
 		n = constrain(n, 0.0f, 1.0f);
 		n = 2.0f * n - (n*n);
-        n = map (n, 0.0f, 1.0f, 10.0f, 0.3f);
+        n = map_sat(n, 0.0f, 1.0f, 10.0f, 0.3f);
         __disable_irq();
         dly_time_step = n;
         __enable_irq();	
@@ -190,7 +190,7 @@ class AudioEffectDelayStereo_i16 : public AudioStream
 	void mod_rate(float32_t r)
 	{
 		r = constrain(r*r*r, 0.0f, 1.0f);
-		r = map(r, 0.0f, 1.0f, 0.0f, lfo_fmax);
+		r = map_sat(r, 0.0f, 1.0f, 0.0f, lfo_fmax);
 		__disable_irq();
 		lfo.setRate(r);
 		__enable_irq();	
@@ -203,7 +203,7 @@ class AudioEffectDelayStereo_i16 : public AudioStream
 	void mod_depth(float32_t d)
 	{
 		d = constrain(d, 0.0f, 1.0f);
-		d = map(d, 0.0f, 1.0f, 0.0f, lfo_ampl_max);
+		d = map_sat(d, 0.0f, 1.0f, 0.0f, lfo_ampl_max);
 		__disable_irq();
 		lfo.setDepth(d);
 		__enable_irq();	

src/effect_phaserStereo_i16.cpp

@@ -80,7 +80,14 @@ void AudioEffectPhaserStereo_i16::update()
 
  	if (!bypass_process(&blockL, &blockR, BYPASS_MODE_PASS, bp))
 		return;
-
+	if (bp)
+	{
+		transmit(blockL, 0);
+		transmit(blockR, 1);
+		release(blockL);
+		release(blockR);
+		return;
+	}
     blockMod = receiveReadOnly(2);      // bipolar/int16_t control input
     if (!blockMod)  internalLFO = true;         // no modulation input provided -> use internal LFO
 
@@ -117,9 +124,6 @@ void AudioEffectPhaserStereo_i16::update()
         modSigL = modSigL * _lfo_scaler + _lfo_bias;
         modSigR = modSigR * _lfo_scaler + _lfo_bias;
 
-
-
-
         drySigL = ((float32_t)blockL->data[i] / 32768.0f) * (1.0f - abs(fdb)*0.25f);  // attenuate the input if using feedback
         inSigL = drySigL + last_sampleL * fdb;
         drySigR = ((float32_t)blockR->data[i] / 32768.0f)  * (1.0f - abs(fdb)*0.25f);
@@ -207,7 +211,7 @@ bool AudioEffectPhaserStereo_i16::bypass_process(audio_block_t** p_blockL, audio
 				*p_blockL = allocate();								// try to allocate a new block
 				if( !*p_blockL)
 				{
-					if (*p_blockR) release(*p_blockR);						// if the Rch is present, release/discard it
+					if (*p_blockR) release(*p_blockR);										// if the Rch is present, release/discard it
 					result = false;
 					break;																	// operation failed due to no audio memory left
 				}
@@ -218,7 +222,7 @@ bool AudioEffectPhaserStereo_i16::bypass_process(audio_block_t** p_blockL, audio
 				*p_blockR = allocate();
 				if( !*p_blockR) 															// no memory for a new block, but we might have L channel zeroed
 				{
-					if (*p_blockL) 	release(*p_blockL);					// blockL is available and contains audio. Discard it
+					if (*p_blockL) 	release(*p_blockL);										// blockL is available and contains audio. Discard it
 					result = false;
 					break;																	// and sigbnal failed operation
 				}

src/effect_phaserStereo_i16.h

@@ -147,11 +147,22 @@ class AudioEffectPhaserStereo_i16 : public AudioStream
     /**
      * @brief Controls the feedback parameter
      * 
-     * @param fdb ffedback value in range 0.0f to 1.0f
+     * @param fdb ffedback value in range -1.0f to 1.0f
      */
     void feedback(float32_t fdb)
     {
-        feedb = constrain(fdb, -1.0f, 1.0f);
+		float t = constrain(fdb, -1.0f, 1.0f);
+		t = fabs(t);
+		t = 2*t - (t*t);
+
+		if (fdb < 0.0f)
+		{
+			t *= -1.0;			
+		}
+		t = t * feedb_max;
+		__disable_irq()
+        feedb = t;
+		__enable_irq();
     }
     /**
      * @brief Dry / Wet mixer ratio. Classic Phaser sound uses 0.5f for 50% dry and 50%Wet
@@ -161,7 +172,9 @@ class AudioEffectPhaserStereo_i16 : public AudioStream
      */
     void mix(float32_t ratio)
     {
+		__disable_irq()
         mix_ratio = constrain(ratio, 0.0f, 1.0f);
+		__enable_irq();
     }
     /**
      * @brief Sets the number of stages used in the phaser
@@ -173,7 +186,9 @@ class AudioEffectPhaserStereo_i16 : public AudioStream
     {
         if (st && st == ((st >> 1) << 1) && st <= PHASER_STEREO_STAGES) // only 2, 4, 6, 8, 12 allowed
         {
+			__disable_irq()
             stg = st;
+			__enable_irq();
         }
     }
     /**
@@ -183,6 +198,7 @@ class AudioEffectPhaserStereo_i16 : public AudioStream
      */
     void bypass_set(bool state) {bp = state;}
     bool bypass_tgl(void) {bp ^= 1; return bp;}
+	bool bypass_get() { return bp;}
 
 private:
     uint8_t stg;                                    // number of stages
@@ -192,6 +208,7 @@ class AudioEffectPhaserStereo_i16 : public AudioStream
 	float32_t allpass_y[2][PHASER_STEREO_STAGES];      // allpass outputs
 	float32_t mix_ratio;                            // 0 = dry. 1.0 = wet
     float32_t feedb;                                // feedback 
+	static constexpr float32_t feedb_max = 0.95f;
     float32_t last_sampleL;
     float32_t last_sampleR;
     uint32_t lfo_phase_acc;                         // interfnal lfo 

src/effect_platereverb_i16.cpp

@@ -113,14 +113,13 @@ bool AudioEffectPlateReverb_i16::begin()
 
 	flags.bypass = 1;
     flags.freeze = 0;
-	initialised = true;
 	return true;
 }
 
 void AudioEffectPlateReverb_i16::update()
 {
 #if defined(__IMXRT1062__)	
-	if (!initialised) return;
+	if (!initialized) return;
     audio_block_t *blockL, *blockR;
 	float sampleL, sampleR;
 	int16_t i;

src/effect_platereverb_i16.h

@@ -43,7 +43,7 @@
 class AudioEffectPlateReverb_i16 :  public AudioStream
 {
 public:
-    AudioEffectPlateReverb_i16() : AudioStream(2, inputQueueArray) { begin();}
+    AudioEffectPlateReverb_i16() : AudioStream(2, inputQueueArray) { initialized = begin();}
 	~AudioEffectPlateReverb_i16(){};
     virtual void update();
 
@@ -395,7 +395,7 @@ class AudioEffectPlateReverb_i16 :  public AudioStream
 		pitchRatio = s;
 		__enable_irq();
 	}
-	friend bool bypass_process(AudioStream* stream, audio_block_t** p_blockL, audio_block_t** p_blockR, bypass_mode_t mode, bool state);
+	bool isInitialized() { return initialized;}
 private:
     struct flags_t
     {
@@ -507,7 +507,7 @@ class AudioEffectPlateReverb_i16 :  public AudioStream
     const float freeze_lodamp_k = 0.0f;
     const float freeze_hidamp_k = 1.0f;
 
-	bool initialised = false;
+	bool initialized = false;
 	uint16_t block_size = AUDIO_BLOCK_SAMPLES;
 
 	bool bypass_process(audio_block_t** p_blockL, audio_block_t** p_blockR, bypass_mode_t mode, bool state);

src/effect_reverbsc_i16.cpp

@@ -52,13 +52,13 @@ AudioEffectReverbSC_i16::AudioEffectReverbSC_i16(bool use_psram) : AudioStream(2
 		if (external_psram_size == 0) 
 		{
 			flags.mem_fail = 1;
-			initialised = true;
+			initialized = true;
 			return;
 		}
 		aux_ = (float32_t *) extmem_malloc(aux_size_bytes);
 		#else
 			flags.mem_fail = 1;
-			initialised = true;
+			initialized = true;
 			return;
 		#endif
 	}
@@ -82,7 +82,7 @@ AudioEffectReverbSC_i16::AudioEffectReverbSC_i16(bool use_psram) : AudioStream(2
 	}
 	mix(0.5f);
 
-	initialised = true;
+	initialized = true;
 }
 
 static int DelayLineMaxSamples(float32_t sr, float32_t i_pitch_mod, int n)
@@ -168,7 +168,7 @@ void AudioEffectReverbSC_i16::update()
 	int buffer_size; /* Local copy */
 	float32_t damp_fact = damp_fact_;
 
-	if (!initialised) return;
+	if (!initialized) return;
 	// special case if memory allocation failed, pass the input signal directly to the output
 	if (flags.mem_fail) bp_mode = BYPASS_MODE_PASS;
 

src/effect_reverbsc_i16.h

@@ -153,7 +153,7 @@ class AudioEffectReverbSC_i16 : public AudioStream
 	float32_t i_pitch_mod_;
     float32_t sample_rate_;
     float32_t damp_fact_, damp_fact_tmp;
-    bool initialised = false;
+    bool initialized = false;
     ReverbScDl_t delay_lines_[8];
     float32_t *aux_; // main delay line storage buffer, placed either in RAM2 or PSRAM
 	const uint32_t aux_size_bytes = REVERBSC_I16_DLYBUF_SIZE*sizeof(float32_t);