Merge remote-tracking branch 'refs/remotes/PaulStoffregen/master'

effect_chorus.cpp

@@ -78,7 +78,7 @@ void AudioEffectChorus::update(void)
   // do passthru
   // It stores the unmodified data in the delay line so that
   // it isn't as likely to click
-  if(num_chorus < 1) {
+  if(num_chorus <= 1) {
     // Just passthrough
     block = receiveWritable(0);
     if(block) {
@@ -93,6 +93,7 @@ void AudioEffectChorus::update(void)
       transmit(block,0);
       release(block);
     }
+    return;
   }
 
   //          L E F T  C H A N N E L
@@ -100,6 +101,7 @@ void AudioEffectChorus::update(void)
   block = receiveWritable(0);
   if(block) {
     bp = block->data;
+    uint32_t tmp = delay_length/(num_chorus - 1) - 1;
     for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
       l_circ_idx++;
       if(l_circ_idx >= delay_length) {
@@ -110,7 +112,7 @@ void AudioEffectChorus::update(void)
       c_idx = l_circ_idx;
       for(int k = 0; k < num_chorus; k++) {
         sum += l_delayline[c_idx];
-        if(num_chorus > 1)c_idx -= delay_length/(num_chorus - 1) - 1;
+        if(num_chorus > 1)c_idx -= tmp;
         if(c_idx < 0) {
           c_idx += delay_length;
         }

effect_delay_ext.cpp

@@ -157,6 +157,15 @@ static int16_t testmem[8000]; // testing only
 
 #define SPISETTING SPISettings(20000000, MSBFIRST, SPI_MODE0)
 
+// While 20 MHz (Teensy actually uses 16 MHz in most cases) and even 24 MHz
+// have worked well in testing at room temperature with 3.3V power, to fully
+// meet all the worst case timing specs, the SPI clock low time would need
+// to be 40ns (12.5 MHz clock) for the single chip case and 51ns (9.8 MHz
+// clock) for the 6-chip memoryboard with 74LCX126 buffers.
+//
+// Timing analysis and info is here:
+// https://forum.pjrc.com/threads/29276-Limits-of-delay-effect-in-audio-library?p=97506&viewfull=1#post97506
+
 void AudioEffectDelayExternal::read(uint32_t offset, uint32_t count, int16_t *data)
 {
 	uint32_t addr = memory_begin + offset;

effect_flange.cpp

@@ -77,7 +77,7 @@ if(0) {
   // initial index
   l_delay_rate_index = 0;
   l_circ_idx = 0;
-  delay_rate_incr = delay_rate/44100.*2147483648.; 
+  delay_rate_incr = delay_rate / (2147483648.0 * AUDIO_SAMPLE_RATE_EXACT);
 //Serial.println(delay_rate_incr,HEX);
 
   delay_offset_idx = delay_offset;
@@ -100,7 +100,7 @@ boolean AudioEffectFlange::voices(int delay_offset,int d_depth,float delay_rate)
   
   delay_depth = d_depth;
 
-  delay_rate_incr = delay_rate/44100.*2147483648.;
+  delay_rate_incr = delay_rate / (2147483648.0 * AUDIO_SAMPLE_RATE_EXACT);
   
   delay_offset_idx = delay_offset;
   // Allow the passthru code to go through

gui/index.html

@@ -1053,6 +1053,13 @@ span.mainfunction {color: #993300; font-weight: bolder}
 		Disabling the audio library interrupt for too long may cause audible
 		dropouts or glitches.
 	</p>
+	<p>An experimental SD library optimization exists, which can remove these
+		SD library restrictions.  It also allows reliable playback of more
+		files at the same time.  To enable this special code, find and edit
+		the SD_t3.h file within your Arduino folder.  See the comments within
+		that file for details.
+	</p>
+
 </script>
 <script type="text/x-red" data-template-name="AudioPlaySdWav">
 	<div class="form-row">

synth_dc.h

@@ -32,20 +32,17 @@
 // compute (a - b) / c
 // handling 32 bit interger overflow at every step
 // without resorting to slow 64 bit math
-#if 0
-// TODO: write this in assembly....
-static inline int32_t substract_32_then_divide(int32_t a, int32_t b, int32_t c) __attribute__((always_inline, unused));
-static inline int32_t substract_32_then_divide(int32_t a, int32_t b, int32_t c)
+#if defined(KINETISK)
+static inline int32_t substract_int32_then_divide_int32(int32_t a, int32_t b, int32_t c) __attribute__((always_inline, unused));
+static inline int32_t substract_int32_then_divide_int32(int32_t a, int32_t b, int32_t c)
 {
-	int32_t diff = substract_32_saturate(a, b);
-	// if only C language provided a way to test Q status bit....
-	if (diff != 0x7FFFFFFF && diff != 0x80000000) {
-		return diff / c;
-	} else {
-		diff = substract_32_saturate(a/2, b/2);
-		if (c == 1 || c == -1) c *= 2; // <-- horrible, incorrect hack
-		return (diff / c) * 2;
-	}
+ 	int r;
+ 	r = substract_32_saturate(a,b);
+ 	if ( !get_q_psr() ) return (r/c);
+ 	clr_q_psr();
+ 	if ( c==0 ) r=0;
+ 	if (__builtin_abs(c)<=1) return r;
+ 	return (a/c)-(b/c);
 }
 #else
 // compute (a - b) / c  ... handling 32 bit interger overflow without slow 64 bit math

synth_tonesweep.cpp

@@ -34,7 +34,7 @@
 
 boolean AudioSynthToneSweep::play(float t_amp,int t_lo,int t_hi,float t_time)
 {
-  double tone_tmp;
+  float tone_tmp;
   
 if(0) {
   Serial.print("AudioSynthToneSweep.begin(tone_amp = ");
@@ -52,8 +52,8 @@ if(0) {
   if(t_amp > 1)return false;
   if(t_lo < 1)return false;
   if(t_hi < 1)return false;
-  if(t_hi >= 44100/2)return false;
-  if(t_lo >= 44100/2)return false;
+  if(t_hi >= (int) AUDIO_SAMPLE_RATE_EXACT / 2)return false;
+  if(t_lo >= (int) AUDIO_SAMPLE_RATE_EXACT / 2)return false;
   if(t_time < 0)return false;
   tone_lo = t_lo;
   tone_hi = t_hi;
@@ -69,7 +69,7 @@ if(0) {
     tone_sign = -1;
     tone_tmp = tone_lo - tone_hi;
   }
-  tone_tmp = tone_tmp/t_time/44100.;
+  tone_tmp = tone_tmp / t_time / AUDIO_SAMPLE_RATE_EXACT;
   tone_incr = (tone_tmp * 0x100000000LL);
   sweep_busy = 1;
   return(true);
@@ -95,22 +95,23 @@ void AudioSynthToneSweep::update(void)
   block = allocate();
   if(block) {
     bp = block->data;
+    uint32_t tmp  = tone_freq >> 32; 
+    uint64_t tone_tmp = (0x400000000000LL * (int)(tmp&0x7fffffff)) / (int) AUDIO_SAMPLE_RATE_EXACT;
     // Generate the sweep
     for(i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
       *bp++ = (short)(( (short)(arm_sin_q31((uint32_t)((tone_phase >> 15)&0x7fffffff))>>16) *tone_amp) >> 16);
-      uint64_t tone_tmp = (0x400000000000LL * (int)((tone_freq >> 32)&0x7fffffff))/44100;
 
       tone_phase +=  tone_tmp;
       if(tone_phase & 0x800000000000LL)tone_phase &= 0x7fffffffffffLL;
 
       if(tone_sign > 0) {
-        if((tone_freq >> 32) > tone_hi) {
+        if(tmp > tone_hi) {
           sweep_busy = 0;
           break;
         }
         tone_freq += tone_incr;
       } else {
-        if((tone_freq >> 32) < tone_hi) {
+        if(tmp < tone_hi) {
           sweep_busy = 0;
 
           break;
@@ -121,8 +122,7 @@ void AudioSynthToneSweep::update(void)
     while(i < AUDIO_BLOCK_SAMPLES) {
       *bp++ = 0;
       i++;
-    }
-    //b_count++;
+    }    
     // send the samples to the left channel
     transmit(block,0);
     release(block);

utility/dspinst.h

@@ -338,7 +338,7 @@ static inline uint32_t get_q_psr(void) __attribute__((always_inline, unused));
 static inline uint32_t get_q_psr(void)
 {
   uint32_t out;
-  asm volatile("mrs %0, APSR" : "=r" (out));
+  asm ("mrs %0, APSR" : "=r" (out));
   return (out & 0x8000000)>>27;
 }
 
@@ -347,9 +347,8 @@ static inline void clr_q_psr(void) __attribute__((always_inline, unused));
 static inline void clr_q_psr(void)
 {
   uint32_t t;
-  asm volatile("mrs %0,APSR " : "=r" (t));
-  asm volatile("bfc %0, #27, #1" : "=r" (t));
-  asm volatile("msr APSR_nzcvq,%0" : "=r" (t)); 
+  asm ("mov %[t],#0\n"
+       "msr APSR_nzcvq,%0\n" : [t] "=&r" (t)::"cc"); 
 }
 
 #endif