First test for a sample drumplayer.

MicroDexed.ino

@@ -35,6 +35,7 @@
 #include "effect_modulated_delay.h"
 #include "effect_stereo_mono.h"
 #include "effect_mono_stereo.h"
+#include "mixer8.h"
 #ifdef USE_PLATEREVERB
 #include "effect_platervbstereo.h"
 #else
@@ -61,8 +62,8 @@ AudioEffectMonoStereo*     mono2stereo[NUM_DEXED];
 AudioMixer4                microdexed_peak_mixer;
 AudioAnalyzePeak           microdexed_peak;
 #if defined(USE_FX)
-AudioMixer4                reverb_mixer_r;
+AudioMixer8                reverb_mixer_r;
-AudioMixer4                reverb_mixer_l;
+AudioMixer8                reverb_mixer_l;
 #ifdef USE_PLATEREVERB
 AudioEffectPlateReverb     reverb;
 #else
@@ -85,6 +86,14 @@ AudioMixer4                audio_thru_mixer_r;
 AudioMixer4                audio_thru_mixer_l;
 #endif
 
+// Drumset
+#ifdef NUM_DRUMS
+AudioPlaySdRaw*             Drum[NUM_DRUMS];
+AudioEffectMonoStereo*      drum_mono2stereo[NUM_DRUMS];
+AudioMixer8                 drum_mixer_r;
+AudioMixer8                 drum_mixer_l;
+#endif
+
 // Audio chain tail
 #if defined(USE_FX)
 #ifdef USE_PLATEREVERB
@@ -161,6 +170,16 @@ AudioConnection          patchCord23(audio_thru_mixer_r, 0, i2s1, 0);
 AudioConnection          patchCord24(audio_thru_mixer_l, 0, i2s1, 1);
 #endif
 
+#ifdef NUM_DRUMS
+#ifdef USE_FX
+AudioConnection          patchCord25(drum_mixer_r, 0, reverb_mixer_r, 4);
+AudioConnection          patchCord26(drum_mixer_l, 0, reverb_mixer_l, 4);
+#else
+AudioConnection          patchCord25(drum_mixer_r, 0, master_mixer_r, 2);
+AudioConnection          patchCord26(drum_mixer_l, 0, master_mixer_l, 2);
+#endif
+#endif
+
 //
 // Dynamic patching of MicroDexed objects
 //
@@ -172,10 +191,9 @@ AudioConnection * dynamicConnections[NUM_DEXED * 16];
 #else
 AudioConnection * dynamicConnections[NUM_DEXED * 5];
 #endif
-void create_audio_engine_chain(uint8_t instance_id)
+void create_audio_dexed_chain(uint8_t instance_id)
 {
   MicroDexed[instance_id] = new AudioSynthDexed(MAX_NOTES / NUM_DEXED, SAMPLE_RATE);
-  //[instance_id] = new AudioEffectDynamics();
   mono2stereo[instance_id] = new AudioEffectMonoStereo();
 #if defined(USE_FX)
   chorus_modulator[instance_id] = new AudioSynthWaveform();
@@ -216,6 +234,27 @@ void create_audio_engine_chain(uint8_t instance_id)
   dynamicConnections[nDynamic++] = new AudioConnection(*mono2stereo[instance_id], 1, master_mixer_l, instance_id);
 }
 
+//
+// Dynamic patching of Drum objects
+//
+#ifdef NUM_DRUMS
+uint16_t nDrumDynamic = 0;
+AudioConnection* dynamicDrumConnections[NUM_DRUMS * 3];
+void create_audio_drum_chain(uint8_t instance_id)
+{
+  Drum[instance_id] = new AudioPlaySdRaw();
+  drum_mono2stereo[instance_id] = new AudioEffectMonoStereo();
+
+  dynamicDrumConnections[nDrumDynamic++] = new AudioConnection(*Drum[instance_id], 0, *drum_mono2stereo[instance_id], 0);
+  dynamicDrumConnections[nDrumDynamic++] = new AudioConnection(*drum_mono2stereo[instance_id], 0, drum_mixer_r, instance_id);
+  dynamicDrumConnections[nDrumDynamic++] = new AudioConnection(*drum_mono2stereo[instance_id], 1, drum_mixer_l, instance_id);
+
+  drum_mono2stereo[instance_id]->panorama(0.0);
+  drum_mixer_r.gain(instance_id, 1.0);
+  drum_mixer_l.gain(instance_id, 1.0);
+}
+#endif
+
 uint8_t sd_card = 0;
 Sd2Card card;
 SdVolume volume;
@@ -384,7 +423,7 @@ void setup()
     Serial.print(F("Creating MicroDexed instance "));
     Serial.println(instance_id, DEC);
 #endif
-    create_audio_engine_chain(instance_id);
+    create_audio_dexed_chain(instance_id);
   }
 
 #ifdef DEBUG
@@ -398,6 +437,18 @@ void setup()
   }
 #endif
 
+#ifdef NUM_DRUMS
+  // create dynamic Drum instances
+  for (uint8_t instance_id = 0; instance_id < NUM_DRUMS; instance_id++)
+  {
+#ifdef DEBUG
+    Serial.print(F("Creating Drum instance "));
+    Serial.println(instance_id, DEC);
+#endif
+    create_audio_drum_chain(instance_id);
+  }
+#endif
+
 #if defined(USE_FX)
   // Init effects
   for (uint8_t instance_id = 0; instance_id < NUM_DEXED; instance_id++)
@@ -494,8 +545,13 @@ void setup()
   master_mixer_r.gain(1, 0.0);
   master_mixer_l.gain(1, 0.0);
 #endif
+#ifdef NUM_DRUMS
+  master_mixer_r.gain(2, 1.0);
+  master_mixer_l.gain(2, 1.0);
+#else
   master_mixer_r.gain(2, 0.0);
   master_mixer_l.gain(2, 0.0);
+#endif
   master_mixer_r.gain(3, 0.0);
   master_mixer_l.gain(3, 0.0);
 
@@ -619,6 +675,7 @@ void loop()
  ******************************************************************************/
 void handleNoteOn(byte inChannel, byte inNumber, byte inVelocity)
 {
+  // Check for MicroDexed
   for (uint8_t instance_id = 0; instance_id < NUM_DEXED; instance_id++)
   {
     if (checkMidiChannel(inChannel, instance_id))
@@ -650,6 +707,50 @@ void handleNoteOn(byte inChannel, byte inNumber, byte inVelocity)
       }
     }
   }
+
+#ifdef NUM_DRUMS
+  // Check for Drum
+  if (inChannel == DRUM_MIDI_CHANNEL)
+  {
+#ifdef DEBUG
+    Serial.print(F("Drum "));
+    Serial.println(inNumber - 48, DEC);
+#endif
+    switch (inNumber - 48)
+    {
+      case 0:
+        Drum[0]->play("drm/bd01.raw");
+        break;
+      case 1:
+        Drum[1]->play("drm/cp02.raw");
+        break;
+      case 2:
+        Drum[2]->play("drm/sd15.raw");
+        break;
+      case 6:
+        Drum[3]->play("drm/hh01.wav");
+        break;
+      case 8:
+        Drum[3]->play("drm/hh02.wav");
+        break;
+      case 10:
+        Drum[3]->play("drm/oh02.wav");
+        break;
+      case 5:
+        Drum[4]->play("drm/lt01.raw");
+        break;
+      case 7:
+        Drum[5]->play("drm/ht01.raw");
+        break;
+      case 13:
+        Drum[6]->play("drm/rd01.raw");
+        break;
+      case 15:
+        Drum[7]->play("drm/rd02.raw");
+        break;
+    }
+  }
+#endif
 }
 
 void handleNoteOff(byte inChannel, byte inNumber, byte inVelocity)
@@ -1930,6 +2031,9 @@ void set_fx_params(void)
   freeverb.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
 #endif
 
+  reverb_mixer_r.gain(4, 1.0);
+  reverb_mixer_l.gain(4, 1.0);
+
   master_mixer_r.gain(3, pseudo_log_curve(mapfloat(configuration.fx.reverb_level, REVERB_LEVEL_MIN, REVERB_LEVEL_MAX, 0.0, 1.0)));
   master_mixer_l.gain(3, pseudo_log_curve(mapfloat(configuration.fx.reverb_level, REVERB_LEVEL_MIN, REVERB_LEVEL_MAX, 0.0, 1.0)));
 

UI.hpp

@@ -30,6 +30,7 @@
 #include "disp_plus.h"
 #include "effect_modulated_delay.h"
 #include "effect_stereo_mono.h"
+#include "mixer8.h"
 #ifdef USE_PLATEREVERB
 #include "effect_platervbstereo.h"
 #else
@@ -84,8 +85,8 @@ extern AudioEffectMonoStereo*     mono2stereo[NUM_DEXED];
 extern AudioMixer4                microdexed_peak_mixer;
 extern AudioAnalyzePeak           microdexed_peak;
 #if defined(USE_FX)
-extern AudioMixer4                reverb_mixer_r;
+extern AudioMixer8                reverb_mixer_r;
-extern AudioMixer4                reverb_mixer_l;
+extern AudioMixer8                reverb_mixer_l;
 #ifdef USE_PLATEREVERB
 extern AudioEffectPlateReverb     reverb;
 #else

config.h

@@ -108,6 +108,10 @@
 // FX-CHAIN ENABLE/DISABLE
 #define USE_FX 1
 
+// NUMBER OF SAMPLEDRUMS
+#define NUM_DRUMS 8 // Max is currently 8!
+#define DRUM_MIDI_CHANNEL 1
+
 // CHORUS parameters
 #define MOD_DELAY_SAMPLE_BUFFER int32_t(TIME_MS2SAMPLES(20.0)) // 20.0 ms delay buffer. 
 #define MOD_WAVEFORM WAVEFORM_TRIANGLE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE
@@ -145,11 +149,19 @@
 //*************************************************************************************************
 #define SAMPLE_RATE 44100
 #ifdef USE_FX
+#ifdef NUM_DRUMS
+#define AUDIO_MEM (SAMPLE_RATE * NUM_DEXED * DELAY_MAX_TIME / 128000 + 36) + NUM_DRUMS * 3
+#else
 // #define AUDIO_MEM 380 // T_4.1, 2 Instances, 0.5ms delay
 #define AUDIO_MEM (SAMPLE_RATE * NUM_DEXED * DELAY_MAX_TIME / 128000 + 36)
+#endif
+#else
+#ifdef NUM_DRUMS
+#define AUDIO_MEM NUM_DEXED * 15 + NUM_DRUMS * 3
 #else
 #define AUDIO_MEM NUM_DEXED * 15
 #endif
+#endif
 
 #ifdef TEENSY_AUDIO_BOARD
 /*

mixer8.cpp

@@ -0,0 +1,132 @@
+/* Audio Library for Teensy 3.X
+ * Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
+ *
+ * Development of this audio library was funded by PJRC.COM, LLC by sales of
+ * Teensy and Audio Adaptor boards.  Please support PJRC's efforts to develop
+ * open source software by purchasing Teensy or other PJRC products.
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice, development funding notice, and this permission
+ * notice shall be included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <Arduino.h>
+#include "mixer8.h"
+#include "utility/dspinst.h"
+
+#if defined(__ARM_ARCH_7EM__)
+#define MULTI_UNITYGAIN 65536
+
+static void applyGain(int16_t *data, int32_t mult)
+{
+	uint32_t *p = (uint32_t *)data;
+	const uint32_t *end = (uint32_t *)(data + AUDIO_BLOCK_SAMPLES);
+
+	do {
+		uint32_t tmp32 = *p; // read 2 samples from *data
+		int32_t val1 = signed_multiply_32x16b(mult, tmp32);
+		int32_t val2 = signed_multiply_32x16t(mult, tmp32);
+		val1 = signed_saturate_rshift(val1, 16, 0);
+		val2 = signed_saturate_rshift(val2, 16, 0);
+		*p++ = pack_16b_16b(val2, val1);
+	} while (p < end);
+}
+
+static void applyGainThenAdd(int16_t *data, const int16_t *in, int32_t mult)
+{
+	uint32_t *dst = (uint32_t *)data;
+	const uint32_t *src = (uint32_t *)in;
+	const uint32_t *end = (uint32_t *)(data + AUDIO_BLOCK_SAMPLES);
+
+	if (mult == MULTI_UNITYGAIN) {
+		do {
+			uint32_t tmp32 = *dst;
+			*dst++ = signed_add_16_and_16(tmp32, *src++);
+			tmp32 = *dst;
+			*dst++ = signed_add_16_and_16(tmp32, *src++);
+		} while (dst < end);
+	} else {
+		do {
+			uint32_t tmp32 = *src++; // read 2 samples from *data
+			int32_t val1 = signed_multiply_32x16b(mult, tmp32);
+			int32_t val2 = signed_multiply_32x16t(mult, tmp32);
+			val1 = signed_saturate_rshift(val1, 16, 0);
+			val2 = signed_saturate_rshift(val2, 16, 0);
+			tmp32 = pack_16b_16b(val2, val1);
+			uint32_t tmp32b = *dst;
+			*dst++ = signed_add_16_and_16(tmp32, tmp32b);
+		} while (dst < end);
+	}
+}
+
+#elif defined(KINETISL)
+#define MULTI_UNITYGAIN 256
+
+static void applyGain(int16_t *data, int32_t mult)
+{
+	const int16_t *end = data + AUDIO_BLOCK_SAMPLES;
+
+	do {
+		int32_t val = *data * mult;
+		*data++ = signed_saturate_rshift(val, 16, 0);
+	} while (data < end);
+}
+
+static void applyGainThenAdd(int16_t *dst, const int16_t *src, int32_t mult)
+{
+	const int16_t *end = dst + AUDIO_BLOCK_SAMPLES;
+
+	if (mult == MULTI_UNITYGAIN) {
+		do {
+			int32_t val = *dst + *src++;
+			*dst++ = signed_saturate_rshift(val, 16, 0);
+		} while (dst < end);
+	} else {
+		do {
+			int32_t val = *dst + ((*src++ * mult) >> 8); // overflow possible??
+			*dst++ = signed_saturate_rshift(val, 16, 0);
+		} while (dst < end);
+	}
+}
+
+#endif
+
+void AudioMixer8::update(void)
+{
+	audio_block_t *in, *out=NULL;
+	unsigned int channel;
+
+	for (channel=0; channel < 8; channel++) {
+		if (!out) {
+			out = receiveWritable(channel);
+			if (out) {
+				int32_t mult = multiplier[channel];
+				if (mult != MULTI_UNITYGAIN) applyGain(out->data, mult);
+			}
+		} else {
+			in = receiveReadOnly(channel);
+			if (in) {
+				applyGainThenAdd(out->data, in->data, multiplier[channel]);
+				release(in);
+			}
+		}
+	}
+	if (out) {
+		transmit(out);
+		release(out);
+	}
+}

mixer8.h

@@ -0,0 +1,69 @@
+/* Audio Library for Teensy 3.X
+ * Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
+ *
+ * Development of this audio library was funded by PJRC.COM, LLC by sales of
+ * Teensy and Audio Adaptor boards.  Please support PJRC's efforts to develop
+ * open source software by purchasing Teensy or other PJRC products.
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice, development funding notice, and this permission
+ * notice shall be included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#ifndef mixer8_h_
+#define mixer8_h_
+
+#include "Arduino.h"
+#include "AudioStream.h"
+
+class AudioMixer8 : public AudioStream
+{
+#if defined(__ARM_ARCH_7EM__)
+public:
+	AudioMixer8(void) : AudioStream(8, inputQueueArray) {
+		for (int i=0; i<8; i++) multiplier[i] = 65536;
+	}
+	virtual void update(void);
+	void gain(unsigned int channel, float gain) {
+		if (channel >= 8) return;
+		if (gain > 32767.0f) gain = 32767.0f;
+		else if (gain < -32767.0f) gain = -32767.0f;
+		multiplier[channel] = gain * 65536.0f; // TODO: proper roundoff?
+	}
+private:
+	int32_t multiplier[8];
+	audio_block_t *inputQueueArray[8];
+
+#elif defined(KINETISL)
+public:
+	AudioMixer8(void) : AudioStream(8, inputQueueArray) {
+		for (int i=0; i<8; i++) multiplier[i] = 256;
+	}
+	virtual void update(void);
+	void gain(unsigned int channel, float gain) {
+		if (channel >= 8) return;
+		if (gain > 127.0f) gain = 127.0f;
+		else if (gain < -127.0f) gain = -127.0f;
+		multiplier[channel] = gain * 256.0f; // TODO: proper roundoff?
+	}
+private:
+	int16_t multiplier[8];
+	audio_block_t *inputQueueArray[8];
+#endif
+};
+
+#endif