Merge branch 'master' of https://github.com/sensorium/Mozzi

Author: sensorium

AudioConfigRP2040.h

@@ -5,14 +5,46 @@
 #error This header should be included for RP2040, only
 #endif
 
+
+// AUDIO output modes
+#define PWM_VIA_BARE_CHIP 1  // output using one of the gpio of the board
+#define EXTERNAL_DAC_VIA_I2S 2  // output via external DAC connected to I2S (PT8211 or similar)
+
+//******* BEGIN: These are the defines you may want to change. Best not to touch anything outside this range. ************/
+#define RP2040_AUDIO_OUT_MODE PWM_VIA_BARE_CHIP
+//******* END: These are the defines you may want to change. Best not to touch anything outside this range. ************/
+
+
+#if (RP2040_AUDIO_OUT_MODE == PWM_VIA_BARE_CHIP)  
 #define AUDIO_CHANNEL_1_PIN 0
 #if (AUDIO_CHANNELS > 1)
 // Audio channel pins for stereo or HIFI must be on the same PWM slice (which is the case for the pairs (0,1), (2,3), (4,5), etc.
-#  define AUDIO_CHANNEL_2_PIN 1
+#define AUDIO_CHANNEL_2_PIN 1
 #endif
 
 // The more audio bits you use, the slower the carrier frequency of the PWM signal. 11 bits yields ~ 60kHz on a 133Mhz CPU (which appears to be a reasonable compromise)
 #define AUDIO_BITS 11
+#endif
+
+#if (RP2040_AUDIO_OUT_MODE == EXTERNAL_DAC_VIA_I2S)
+// ****** BEGIN: These are define you may want to change. Best not to touch anything outside this range. ************/
+#define BCLK_PIN 20
+#define WS_PIN (pBCLK+1)  // CANNOT BE CHANGED, HAS TO BE NEXT TO pBCLK
+#define DOUT_PIN 22
+#define LSBJ_FORMAT false   // some DAC, like the PT8211, use a variant of I2S data format called LSBJ
+                           // set this to true to use this kind of DAC or false for plain I2S.
+#define AUDIO_BITS 16   // available values are 8, 16, 24 (LEFT ALIGN in 32 bits type!!) and 32 bits
+// ****** END: These are define you may want to change. Best not to touch anything outside this range. ************/
+
+#define BYPASS_MOZZI_OUTPUT_BUFFER true
+
+// Configuration of the I2S port, especially DMA. Set in stone here as default of the library when this was written.
+// Probably do not change if you are not sure of what you are doing
+#define BUFFERS 8       // number of DMA buffers used
+#define BUFFER_SIZE 256  // total size of the buffer, in samples
+#endif
+
+
 #define AUDIO_BITS_PER_CHANNEL AUDIO_BITS
 
 #define AUDIO_BIAS ((uint16_t) 1<<(AUDIO_BITS-1))

AudioOutput.h

@@ -103,9 +103,7 @@ struct StereoOutput {
   StereoOutput& clip() { _l = CLIP_AUDIO(_l); _r = CLIP_AUDIO(_r); return *this; };
 
   /** @see MonoOutput::fromNBit(), stereo variant */
-  static inline StereoOutput fromNBit(uint8_t bits, int16_t l, int16_t r) { return StereoOutput(SCALE_AUDIO(l, bits), SCALE_AUDIO(r, bits)); }
-  /** @see MonoOutput::fromNBit(), stereo variant, 32 bit overload */
-  static inline StereoOutput fromNBit(uint8_t bits, int32_t l, int32_t r) { return StereoOutput(SCALE_AUDIO(l, bits), SCALE_AUDIO(r, bits)); }
+template<typename T> static inline StereoOutput fromNBit(uint8_t bits, T l, T r) { return StereoOutput(SCALE_AUDIO(l, bits), SCALE_AUDIO(r, bits)); }
   /** @see MonoOutput::from8Bit(), stereo variant */
   static inline StereoOutput from8Bit(int16_t l, int16_t r) { return fromNBit(8, l, r); }
   /** @see MonoOutput::from16Bit(), stereo variant */
@@ -152,9 +150,7 @@ struct MonoOutput {
   /** Construct an audio frame a zero-centered value known to be in the N bit range. Appropriate left- or right-shifting will be performed, based on the number of output
    *  bits available. While this function takes care of the shifting, beware of potential overflow issues, if your intermediary results exceed the 16 bit range. Use proper
    *  casts to int32_t or larger in that case (and the compiler will automatically pick the 32 bit overload in this case) */
-  static inline MonoOutput fromNBit(uint8_t bits, int16_t l) { return MonoOutput(SCALE_AUDIO(l, bits)); }
-  /** 32bit overload. See above. */
-  static inline MonoOutput fromNBit(uint8_t bits, int32_t l) { return MonoOutput(SCALE_AUDIO(l, bits)); }
+  template<typename T> static inline MonoOutput fromNBit(uint8_t bits, T l) { return MonoOutput(SCALE_AUDIO(l, bits)); }
   /** Construct an audio frame from a zero-centered value known to be in the 8 bit range. On AVR, STANDADR or STANDARD_PLUS mode, this is effectively the same as calling the
    * constructor, directly (no scaling gets applied). On platforms/configs using more bits, an appropriate left-shift will be performed. */
   static inline MonoOutput from8Bit(int16_t l) { return fromNBit(8, l); }

LowPassFilter.h

@@ -13,146 +13,7 @@
 #ifndef LOWPASS_H_
 #define LOWPASS_H_
 
-#include "IntegerType.h"
-#include "AudioOutput.h"
-
-
-
-/*
-simple resonant filter posted to musicdsp.org by Paul Kellett
-http://www.musicdsp.org/archive.php?classid=3#259
-
-Two versions are available: LowPassFilter and LowPassFilter16.
-LowPassFilter is an optimized version that uses 8bits values to set
-the resonance and the cutoff_freq. It can works on 8bits samples only
-on 8bits platforms.
-LowPassFilter16 consumes more CPU ressources but uses 16bits values
-for resonance and cutoff_freq and can work on samples up to 16bits on
-8bits platforms and up to 32 on 32bits platforms.
-
-// set feedback amount given f and q between 0 and 1
-fb = q + q/(1.0 - f);
-In order to avoid a slow division we use the use a Taylor expansion to approximate 1/(1.0 - f):
-Close to f=0: 1/(1.0-f) approx 1.0+f.
-Hence: fb = q + q * (1.0 + f)
-This approximation is less and less valid with an increasing cutoff, leading to a reduction of the resonance of the filter at high cutoff frequencies.
-
-// for each sample...
-buf0 = buf0 + f * (in - buf0 + fb * (buf0 - buf1));
-buf1 = buf1 + f * (buf0 - buf1);
-out = buf1;
-
-fixed point version of the filter
-"dave's blog of art and programming" http://www.pawfal.org/dave/blog/2011/09/
-*/
-
-// we are using .n fixed point (n bits for the fractional part)
-//#define FX_SHIFT 8
-//#define SHIFTED_1 ((uint8_t)255)
-
-/** A resonant low pass filter for audio signals.
- */
-template<typename su=uint8_t>
-class LowPassFilterNbits
-{
-
-public:
-  /** Constructor.
-   */
-  LowPassFilterNbits() { ; }
-
-
-  /** deprecated.  Use setCutoffFreqAndResonance(su cutoff, su
-  resonance).
-
-  Set the cut off frequency,
-  @param cutoff use the range 0-255 to represent 0-8191 Hz (AUDIO_RATE/2) for LowPassFilter, cutoff use the range 0-65535 to represent 0-AUDIO_RATE/2.
-  Be careful of distortion at the lower end, especially with high resonance.
-  */
-  void setCutoffFreq(su cutoff)
-      {
-    f = cutoff;
-    fb = q + ucfxmul(q, (typename IntegerType<sizeof(su)+sizeof(su)>::unsigned_type) SHIFTED_1 + cutoff);
-  }
-
-  /** deprecated.  Use setCutoffFreqAndResonance(su cutoff, su
-  resonance).
-
-  Set the resonance. If you hear unwanted distortion, back off the resonance.
-  After setting resonance, you need to call setCuttoffFreq() to hear the change!
-  @param resonance in the range 0-255 for LowPassFilter, 0-65535 for LowPassFilter16, with 255/65535 being most resonant
-  @note	Remember to call setCuttoffFreq() after resonance is changed!
-  */
-  void setResonance(su resonance) { q = resonance; }
-
-  /**
-  Set the cut off frequency and resonance.  Replaces setCutoffFreq() and
-  setResonance().  (Because the internal calculations need to be done whenever either parameter changes.)
-  @param cutoff range 0-255 represents 0-8191 Hz (AUDIO_RATE/2) for LowPassFilter, range 0-65535 for LowPassFilter16
-  Be careful of distortion at the lower end, especially with high resonance.
-  @param resonance range 0-255 for LowPassFilter, 0-65535 for LowPassFilter16, 255/65535 is most resonant.
-  */
-  void setCutoffFreqAndResonance(su cutoff, su resonance)
-	{
-    f = cutoff;
-    q = resonance; // hopefully optimised away when compiled, just here for
-                   // backwards compatibility
-    fb = q + ucfxmul(q,(typename IntegerType<sizeof(su)+sizeof(su)>::unsigned_type) SHIFTED_1 + cutoff);
-  }
-
-  /** Calculate the next sample, given an input signal.
-  @param in the signal input. Should not be more than 8bits on 8bits platforms (Arduino) if using LowPassFilter and not LowPassFilter16.
-  @return the signal output.
-  @note Timing: about 11us.
-  */
-  //	10.5 to 12.5 us, mostly 10.5 us (was 14us)
-  inline AudioOutputStorage_t next(AudioOutputStorage_t in)
-	{
-    // setPin13High();
-    buf0 += fxmul(((in - buf0) + fxmul(fb, buf0 - buf1)), f);
-    buf1 += ifxmul(buf0 - buf1, f); // could overflow if input changes fast
-    return buf1;
-  }
-
-private:
-  su q;
-  su f;
-  typename IntegerType<sizeof(su)+sizeof(su)>::unsigned_type fb;
-  AudioOutputStorage_t buf0, buf1;
-  const uint8_t FX_SHIFT = sizeof(su) << 3;
-  const uint8_t FX_SHIFT_M_1 = FX_SHIFT-1;
-  const su SHIFTED_1 = (1<<FX_SHIFT)-1;
-
-  // // multiply two fixed point numbers (returns fixed point)
-  // inline
-  // long fxmul(long a, long b)
-  // {
-  // 	return (a*b)>>FX_SHIFT;
-  // }
-
-  // multiply two fixed point numbers (returns fixed point)
-  inline typename IntegerType<sizeof(su)+sizeof(su)>::unsigned_type ucfxmul(su a, typename IntegerType<sizeof(su)+sizeof(su)>::unsigned_type b)
-	{
-	  return (((typename IntegerType<sizeof(su)+sizeof(su)>::unsigned_type)a * (b >> 1)) >> (FX_SHIFT_M_1));
-  }
-
-  // multiply two fixed point numbers (returns fixed point)
-  inline typename IntegerType<sizeof(AudioOutputStorage_t)+sizeof(su)-1>::signed_type ifxmul(typename IntegerType<sizeof(AudioOutputStorage_t )+sizeof(su)-1>::signed_type a, su b) { return ((a * b) >> FX_SHIFT); } 
-
-  // multiply two fixed point numbers (returns fixed point)
-  inline typename IntegerType<sizeof(AudioOutputStorage_t)+sizeof(AudioOutputStorage_t)>::signed_type fxmul(typename IntegerType<sizeof(AudioOutputStorage_t)+sizeof(AudioOutputStorage_t)>::signed_type a, typename IntegerType<sizeof(AudioOutputStorage_t)+sizeof(su)-1>::signed_type b) { return ((a * b) >> FX_SHIFT); }
-};
-
-typedef LowPassFilterNbits<> LowPassFilter;
-typedef LowPassFilterNbits<uint16_t> LowPassFilter16;
-
-
-/**
-@example 10.Audio_Filters/LowPassFilter/LowPassFilter.ino
-This example demonstrates the LowPassFilter class.
-
-@example 10.Audio_Filters/LowPassFilter/LowPassFilter16.ino
-This example demonstrates the LowPassFilter16 class.
-*/
+#include<ResonantFilter.h>
+#warning This header is deprecated, please use ResonantFilter.h instead.
 
 #endif /* LOWPASS_H_ */

Mozzi

@@ -37,6 +37,7 @@ void setupMozziADC(int8_t speed) {
 
 
 
+
 //// BEGIN AUDIO OUTPUT code ///////
 #include <driver/i2s.h>   // for I2S-based output modes
 #include <driver/timer.h> // for EXTERNAL_AUDIO_OUTPUT
@@ -103,17 +104,19 @@ void CACHED_FUNCTION_ATTR timer0_audio_output_isr(void *) {
 static void startAudio() {
 #if (BYPASS_MOZZI_OUTPUT_BUFFER != true)  // for external audio output, set up a timer running a audio rate
   static intr_handle_t s_timer_handle;
+  const int div = 2;
   timer_config_t config = {
-    .alarm_en = true,
-    .counter_en = false,
-    .intr_type = TIMER_INTR_LEVEL,
+    .alarm_en = (timer_alarm_t)true,
+    .counter_en = (timer_start_t)false,
+    .intr_type = (timer_intr_mode_t) TIMER_INTR_LEVEL,
     .counter_dir = TIMER_COUNT_UP,
-    .auto_reload = true,
-    .divider = 1 // For max available precision: The APB_CLK clock signal is running at 80 MHz, i.e. 1/80 uS per tick
+    .auto_reload = (timer_autoreload_t) true,
+    .divider = div // For max available precision: The APB_CLK clock signal is running at 80 MHz, i.e. 2/80 uS per tick
+                 // Min acceptable value is 2
   };
   timer_init(TIMER_GROUP_0, TIMER_0, &config);
   timer_set_counter_value(TIMER_GROUP_0, TIMER_0, 0);
-  timer_set_alarm_value(TIMER_GROUP_0, TIMER_0, 80000000UL / AUDIO_RATE);
+  timer_set_alarm_value(TIMER_GROUP_0, TIMER_0, 80000000UL / AUDIO_RATE / div);
   timer_enable_intr(TIMER_GROUP_0, TIMER_0);
   timer_isr_register(TIMER_GROUP_0, TIMER_0, &timer0_audio_output_isr, nullptr, 0, &s_timer_handle);
   timer_start(TIMER_GROUP_0, TIMER_0);

MozziGuts_impl_ESP32.hpp

@@ -83,6 +83,7 @@ void setupMozziADC(int8_t speed) {
   );
 
   uint dma_chan = dma_claim_unused_channel(true);
+  rp2040_adc_dma_chan = dma_chan;
   static dma_channel_config cfg = dma_channel_get_default_config(dma_chan);
 
   // Reading from constant address, writing to incrementing byte addresses
@@ -102,12 +103,14 @@ void setupMozziADC(int8_t speed) {
 
   // we want notification, when a sample has arrived
   dma_channel_set_irq0_enabled(dma_chan, true);
-  irq_set_exclusive_handler(DMA_IRQ_0, rp2040_adc_queue_handler);
+  irq_add_shared_handler(DMA_IRQ_0, rp2040_adc_queue_handler, PICO_SHARED_IRQ_HANDLER_DEFAULT_ORDER_PRIORITY);
   irq_set_enabled(DMA_IRQ_0, true);
+  dma_channel_start(dma_chan);
 }
 
 void rp2040_adc_queue_handler() {
-  dma_hw->ints0 = 1u << rp2040_adc_dma_chan;  // clear interrupt flag
+  if (!dma_channel_get_irq0_status(rp2040_adc_dma_chan)) return; // shared handler may get called on unrelated events
+  dma_channel_acknowledge_irq0(rp2040_adc_dma_chan); // clear interrupt flag  
   //adc_run(false);  // adc not running continuous
   //adc_fifo_drain(); // no need to drain fifo, the dma transfer did that
   dma_channel_set_trans_count(rp2040_adc_dma_chan, 1, true);  // set up for another read
@@ -119,16 +122,17 @@ void rp2040_adc_queue_handler() {
 ////// BEGIN audio output code //////
 #define LOOP_YIELD tight_loop_contents();  // apparently needed, among other things, to service the alarm pool
 
-#include <hardware/pwm.h>
 
+#if (RP2040_AUDIO_OUT_MODE == PWM_VIA_BARE_CHIP) || (EXTERNAL_AUDIO_OUTPUT == true)
+#include <hardware/pwm.h>
 #if (EXTERNAL_AUDIO_OUTPUT != true) // otherwise, the last stage - audioOutput() - will be provided by the user
 inline void audioOutput(const AudioOutput f) {
   pwm_set_gpio_level(AUDIO_CHANNEL_1_PIN, f.l()+AUDIO_BIAS);
 #if (AUDIO_CHANNELS > 1)
   pwm_set_gpio_level(AUDIO_CHANNEL_2_PIN, f.r()+AUDIO_BIAS);
 #endif
 }
-#endif
+#endif  // #if (EXTERNAL_AUDIO_OUTPUT != true)
 
 #include <pico/time.h>
 /** Implementation notes:
@@ -150,8 +154,56 @@ void audioOutputCallback(uint) {
     // NOTE: hardware_alarm_set_target returns true, if the target was already missed. In that case, keep pushing samples, until we have caught up.
   } while (hardware_alarm_set_target(audio_update_alarm_num, next_audio_update));
 }
+ 
+#elif (RP2040_AUDIO_OUT_MODE == EXTERNAL_DAC_VIA_I2S)
+#include <I2S.h>
+I2S i2s(OUTPUT);
+
+
+inline bool canBufferAudioOutput() {
+  return (i2s.availableForWrite());
+}
+
+inline void audioOutput(const AudioOutput f) {
+
+#if (AUDIO_BITS == 8)
+#if (AUDIO_CHANNELS > 1)
+  i2s.write8(f.l(), f.r());
+#else
+  i2s.write8(f.l(), 0);
+#endif
+  
+#elif (AUDIO_BITS == 16)
+#if (AUDIO_CHANNELS > 1)
+  i2s.write16(f.l(), f.r());
+#else
+  i2s.write16(f.l(), 0);
+#endif
+  
+#elif (AUDIO_BITS == 24)
+#if (AUDIO_CHANNELS > 1)
+  i2s.write24(f.l(), f.r());
+#else
+  i2s.write24(f.l(), 0);
+#endif
+  
+#elif (AUDIO_BITS == 32)
+#if (AUDIO_CHANNELS > 1)
+  i2s.write32(f.l(), f.r());
+#else
+  i2s.write32(f.l(), 0);
+#endif
+#else
+  #error The number of AUDIO_BITS set in AudioConfigRP2040.h is incorrect
+#endif  
+
+  
+}
+#endif 
+
 
 static void startAudio() {
+#if (RP2040_AUDIO_OUT_MODE == PWM_VIA_BARE_CHIP) || (EXTERNAL_AUDIO_OUTPUT == true) // EXTERNAL AUDIO needs the timers set here
 #if (EXTERNAL_AUDIO_OUTPUT != true)
   // calling analogWrite for the first time will try to init the pwm frequency and range on all pins. We don't want that happening after we've set up our own,
   // so we start off with a dummy call to analogWrite:
@@ -185,9 +237,30 @@ static void startAudio() {
     next_audio_update = make_timeout_time_us(micros_per_update);
     // See audioOutputCallback(), above. In _theory_ some interrupt stuff might delay us, here, causing us to miss the first beat (and everything that follows)
   } while (hardware_alarm_set_target(audio_update_alarm_num, next_audio_update));
+
+#elif (RP2040_AUDIO_OUT_MODE == EXTERNAL_DAC_VIA_I2S)
+  i2s.setBCLK(BCLK_PIN);
+  i2s.setDATA(DOUT_PIN);
+  i2s.setBitsPerSample(AUDIO_BITS);
+  
+#if (AUDIO_BITS > 16)
+  i2s.setBuffers(BUFFERS, (size_t) (BUFFER_SIZE/BUFFERS), 0);
+#else
+  i2s.setBuffers(BUFFERS, (size_t) (BUFFER_SIZE/BUFFERS/2), 0);
+#endif
+#if (LSBJ_FORMAT == true)
+  i2s.setLSBJFormat();
+#endif
+  i2s.begin(AUDIO_RATE);
+#endif
 }
 
 void stopMozzi() {
+#if (RP2040_AUDIO_OUT_MODE == PWM_VIA_BARE_CHIP) || (EXTERNAL_AUDIO_OUTPUT == true)
   hardware_alarm_set_callback(audio_update_alarm_num, NULL);
+#elif (RP2040_AUDIO_OUT_MODE == EXTERNAL_DAC_VIA_I2S)
+  i2s.end();
+#endif
+  
 }
 ////// END audio output code //////