GitHub Action to find typos and lint Python code

.github/workflows/ci.yml

@@ -0,0 +1,25 @@
+# This Action uses minimal steps to run in ~5 seconds to rapidly:
+# look for typos in the codebase using codespell, and
+# lint Python code using ruff and provide intuitive GitHub Annotations to contributors.
+# https://github.com/codespell-project/codespell#readme
+# https://docs.astral.sh/ruff/
+name: ci
+on:
+  push:
+    # branches: [main, master]
+  pull_request:
+    # branches: [main, master]
+jobs:
+  codespell:
+    runs-on: ubuntu-latest
+    steps:
+    - uses: actions/checkout@v4
+    - uses: codespell-project/actions-codespell@v2
+      with:
+        ignore_words_list: blockin,dout,ethe,fo,reson,rin,shiftin,siz
+        skip: "*.min.js"
+  ruff:
+    runs-on: ubuntu-latest
+    steps:
+    - uses: actions/checkout@v4
+    - uses: astral-sh/ruff-action@v3

Resampler.cpp

@@ -306,7 +306,7 @@ void Resampler::resample(float* input0, float* input1, uint16_t inputLength, uin
         }
     }
     if(outputCount < outputLength){
-        //ouput vector not full -> we ran out of input samples
+        //output vector not full -> we ran out of input samples
         processedLength=inputLength;
     }
     else{

Resampler.h

@@ -161,7 +161,7 @@ class Resampler {
                 }
             }
             if(outputCount < outputLength){
-                //ouput vector not full -> we ran out of input samples
+                //output vector not full -> we ran out of input samples
                 processedLength=inputLength;
             }
             else{

analyze_fft1024.cpp

@@ -97,7 +97,7 @@ void AudioAnalyzeFFT1024::update(void)
 	case 7:
 		blocklist[7] = block;
 		// TODO: perhaps distribute the work over multiple update() ??
-		//       github pull requsts welcome......
+		//       github pull requests welcome......
 		copy_to_fft_buffer(buffer+0x000, blocklist[0]->data);
 		copy_to_fft_buffer(buffer+0x100, blocklist[1]->data);
 		copy_to_fft_buffer(buffer+0x200, blocklist[2]->data);

analyze_tonedetect.cpp

@@ -153,7 +153,7 @@ AudioAnalyzeToneDetect::operator bool()
 
 	//Serial.printf("bool: power=%d, trig=%d\n", power, trigger);
 	return (power >= trigger);
-	// TODO: this should really remember if it's retuned true previously,
+	// TODO: this should really remember if it's returned true previously,
 	// so it can give a single true response each time a tone is seen.
 }
 

control_cs42448.cpp

@@ -127,7 +127,7 @@ bool AudioControlCS42448::inputLevelInteger(int32_t n)
 	return write(CS42448_DAC_Channel_Invert, data, 7);
 }
 
-bool AudioControlCS42448::inputLevelInteger(int chnnel, int32_t n)
+bool AudioControlCS42448::inputLevelInteger(int channel, int32_t n)
 {
 
 	return true;

control_cs42448.h

@@ -66,7 +66,7 @@ class AudioControlCS42448 : public AudioControl
 	bool volumeInteger(uint32_t n);
 	bool volumeInteger(int channel, uint32_t n);
 	bool inputLevelInteger(int32_t n);
-	bool inputLevelInteger(int chnnel, int32_t n);
+	bool inputLevelInteger(int channel, int32_t n);
 	// convert level to volume byte, section 6.9.1, page 50
 	uint32_t volumebyte(float level) {
 		if (level >= 1.0f) return 0;

control_tlv320aic3206.cpp

@@ -244,7 +244,7 @@ bool AudioControlTLV320AIC3206::setMicBias(int n) {
 }
 
 void AudioControlTLV320AIC3206::aic_reset() {
-  if (debugToSerial) Serial.println("INFO: Reseting AIC");
+  if (debugToSerial) Serial.println("INFO: Resetting AIC");
   aic_writePage(0x00, 0x01, 0x01);
   // aic_writeAddress(0x0001, 0x01);
 

effect_delay.h

@@ -83,7 +83,7 @@ class AudioEffectDelay : public AudioStream
 	}
 	void disable(uint8_t channel) {
 		if (channel >= 8) return;
-		// diable this channel
+		// disable this channel
 		activemask &= ~(1<<channel);
 		// recompute maxblocks for remaining enabled channels
 		recompute_maxblocks();

effect_granular.cpp

@@ -153,7 +153,7 @@ void AudioEffectGranular::update(void)
 				sample_req = false;
 				allow_len_change = true; // Reduces noise by not allowing the
 						// length to change after the sample has been
-						// recored.  Kind of not too much though
+						// recorded.  Kind of not too much though
 				if (write_head >= glitch_len) {
 					write_head = 0;
 					sample_loaded = true;
@@ -166,7 +166,7 @@ void AudioEffectGranular::update(void)
 
 			if (sample_loaded) {
 				//move it to the middle third of the bank.
-				//3 "seperate" banks are used
+				//3 "separate" banks are used
 				float fade_len = 20.00;
 				int16_t m2 = fade_len;
 

effect_midside.cpp

@@ -63,7 +63,7 @@ void AudioEffectMidSide::update(void)
 			// R[i] = mid[i] - side[i]);
 			// Because the /2 has already been applied in the encoding,
 			// we shouldn't have to add it here.
-			// However... because of the posibility that the user has
+			// However... because of the possibility that the user has
 			// modified mid or side such that
 			// it could overflow, we have to:
 			// a) preventively do a (x/2+y/2)*2 again, causing bit reduction

effect_reverb.h

@@ -3,7 +3,7 @@
  *  Author: Joao Rossi Filho
  *          [email protected]
  *
- *  Soon I'll make a commit with nice comments and discription
+ *  Soon I'll make a commit with nice comments and description
  *
  * Copyright (c) 2016 Joao Rossi FIlho
  *

examples/Analysis/NoteFrequency/NoteFrequency.ino

@@ -82,7 +82,7 @@ void setup() {
      */
     notefreq.begin(.15);
     pinMode(LED_BUILTIN, OUTPUT);
-    // Audio library isr allways gets priority
+    // Audio library isr always gets priority
     playNoteTimer.priority(144);
     playNoteTimer.begin(playNote, 1000);
 }

examples/Dynamic/DynamicPatching/DynamicPatching.ino

@@ -75,7 +75,7 @@ void loop()
   {
     audioUpdates++;
     audioUpdated = true; // audio engine just updated
-    queue.readBuffer(); // not intereseted in contents,
+    queue.readBuffer(); // not interested in contents,
     queue.freeBuffer(); // just that 
   }
 

examples/Effects/Bitcrusher/Bitcrusher.ino

@@ -68,7 +68,7 @@ void setup() {
 
   // Audio connections require memory to work.  For more
   // detailed information, see the MemoryAndCpuUsage example
-  AudioMemory(40); //this is WAY more tha nwe need
+  AudioMemory(40); //this is WAY more than we need
 
   // turn on the output
   audioShield.enable();

examples/HardwareTesting/PassThroughAsyncSpdif/PassThroughAsyncSpdif.ino

@@ -28,7 +28,7 @@ void loop() {
   Serial.print("anti-aliasing attenuation: ");
   Serial.println(spdifIn.getAttenuation());
 
-  Serial.print("resampling goup delay [milli seconds]: ");
+  Serial.print("resampling group delay [milli seconds]: ");
   Serial.println(spdifIn.getHalfFilterLength()/inputFrequency*1e3,2);
 
   Serial.print("half filter length: ");

examples/Queues/PlayQueueDemo/PlayQueueDemo.ino

@@ -5,7 +5,7 @@
  * can be changed to modify the usage of audio blocks, and an
  * ability to execute the loop() function more slowly, with
  * simple waveform generation, or more efficiently / faster,
- * but requring some programmer effort to re-try if sending
+ * but requiring some programmer effort to re-try if sending
  * waveform data fails due to a lack of buffer or queue space.
  *
  * This example code is in the public domain.

examples/Synthesis/Waveforms/Waveforms.ino

@@ -59,7 +59,7 @@ void setup() {
   sgtl5000_1.enable();
   sgtl5000_1.volume(0.8); // caution: very loud - use oscilloscope only!
 
-  // Confirgure both to use "myWaveform" for WAVEFORM_ARBITRARY
+  // Configure both to use "myWaveform" for WAVEFORM_ARBITRARY
   waveform1.arbitraryWaveform(myWaveform, 172.0);
   waveform2.arbitraryWaveform(myWaveform, 172.0);
 
@@ -114,7 +114,7 @@ void loop() {
         break;
       case WAVEFORM_TRIANGLE_VARIABLE:
         current_waveform = WAVEFORM_ARBITRARY;
-        Serial.println("Arbitary Waveform");
+        Serial.println("Arbitrary Waveform");
         break;
       case WAVEFORM_ARBITRARY:
         current_waveform = WAVEFORM_PULSE;

examples/Synthesis/WaveformsModulated/WaveformsModulated.ino

@@ -61,7 +61,7 @@ void setup() {
   sgtl5000_1.enable();
   sgtl5000_1.volume(0.8); // caution: very loud - use oscilloscope only!
 
-  // Confirgure both to use "myWaveform" for WAVEFORM_ARBITRARY
+  // Configure both to use "myWaveform" for WAVEFORM_ARBITRARY
   waveformMod1.arbitraryWaveform(myWaveform, 172.0);
 
   // Configure for middle C note without modulation
@@ -114,7 +114,7 @@ void loop() {
         break;
       case WAVEFORM_TRIANGLE_VARIABLE:
         current_waveform = WAVEFORM_ARBITRARY;
-        Serial.println("Arbitary Waveform");
+        Serial.println("Arbitrary Waveform");
         break;
       case WAVEFORM_ARBITRARY:
         current_waveform = WAVEFORM_PULSE;

examples/Tutorial/Part_2_08_Oscillators/Part_2_08_Oscillators.ino

@@ -132,7 +132,7 @@ void loop() {
   }
 
   // after 4 seconds of inactivity, go back to
-  // steady listening intead of the envelope
+  // steady listening instead of the envelope
   if (mixer2_envelope == true && timeout > 4000) {
     mixer2.gain(0, 0.15);
     mixer2.gain(1, 0.0);

extras/miditones/miditones.c

@@ -70,7 +70,7 @@
 *  tone generators without any volume or tone controls.
 *
 *  Volume ("velocity") and instrument specifications in the MIDI files are discarded.
-*  All the tracks are prcoessed and merged into a single time-ordered stream of
+*  All the tracks are processed and merged into a single time-ordered stream of
 *  "note on", "note off", and "delay" commands.
 *
 *  This was written for the "Playtune" Arduino library, which plays polyphonic music
@@ -101,7 +101,7 @@
 *     miditones chopin
 *
 *  It will create a file in the same directory called "chopin.c" which contains
-*  the C-language statement to intiialize an array called "score" with the bytestream.
+*  the C-language statement to initialize an array called "score" with the bytestream.
 *
 *
 *  The general form for command line execution is this:
@@ -134,7 +134,7 @@
 *       The default is 6 tone generators, and the maximum is 16.
 *       The program will report how many notes had to be discarded because there
 *       weren't enough tone generators.  Note that for the Arduino Playtunes
-*       library, it's ok to have the bytestream use more tone genreators than
+*       library, it's ok to have the bytestream use more tone generators than
 *       exist on your processor because any extra notes will be ignored, although
 *       it does make the file bigger than necessary . Of course, too many ignored
 *       notes will make the music sound really strange!
@@ -811,7 +811,7 @@ int main(int argc,char *argv[]) {
     }
 
     /* Continue processing all tracks, in an order based on the simulated time.
-    This is not unlike multiway merging used for tape sorting algoritms in the 50's! */
+    This is not unlike multiway merging used for tape sorting algorithms in the 50's! */
 
     tracknum = 0;
     if (!parseonly) do { /* while there are still track notes to process */

gui/index.html

@@ -935,7 +935,7 @@ <h3>Functions</h3>
 	<p class=desc>Returns the sample rate of incoming data, if the PLL has locked, or returns 0 if the audio sample rate is unknown.
 	</p>
 	<p class=func><span class=keyword>getTargetLantency</span>();</p>
-	<p class=desc>Returns the target latency in seconds. The latency is the time from the moment a sample is received by the Teensy SPDIF hardware receiver until it is transmitted by the asrc intput. The audio samples arrive at the asrc input in chunks of 32 samples per channel. The target latency consists of these 32 samples + some buffer that is needed to compensate for timing variations.
+	<p class=desc>Returns the target latency in seconds. The latency is the time from the moment a sample is received by the Teensy SPDIF hardware receiver until it is transmitted by the asrc input. The audio samples arrive at the asrc input in chunks of 32 samples per channel. The target latency consists of these 32 samples + some buffer that is needed to compensate for timing variations.
 	</p>
 	<p class=func><span class=keyword>getAttenuation</span>();</p>
 	<p class=desc>
@@ -955,7 +955,7 @@ <h3>Examples</h3>
 	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughAsyncSpdif
 	</p>
 	<h3>Notes</h3>
-	<p>AsyncAudioInputSPDIF3 is not able to clock the audio pipline (never has the 'update_responsibility'). At least 1 other input or output must be used to cause the entire Audio library to update.
+	<p>AsyncAudioInputSPDIF3 is not able to clock the audio pipeline (never has the 'update_responsibility'). At least 1 other input or output must be used to cause the entire Audio library to update.
 	</p>
 
 	<p>AsyncAudioInputSPDIF3 can optionally take parameters to alter its resampling behavior.
@@ -1055,7 +1055,7 @@ <h3>Notes</h3>
 	<p>Algorithm for automatic DC bias tracking</p>
 	<p>Noise coupling from digital circuitry inside the chip is always a problem when using
 		ADC inputs pin.  It's never as quiet as the audio shield and any good quality
-		audio ADC chip.  Stong low impedance drive to the analog input pin is critical
+		audio ADC chip.  Strong low impedance drive to the analog input pin is critical
 		to minimizing noise coupling.  If an opamp is used, connect a low value resistor
 		(eg, 100 to 1000 ohms)
 		between the opamp output and ADC input pin, and a 1nF capacitor from the ADC pin
@@ -1824,7 +1824,7 @@ <h3>Examples</h3>
 	<h3>Credits</h3>
 	<p><a href="https://github.com/FrankBoesing" target="_blank">Frank Boesing</a>
 		developed the AudioOutputSPDIF code.  The original
-		<a href="https://forum.pjrc.com/threads/28639-S-pdif" target="_blank">forum disussion</a>
+		<a href="https://forum.pjrc.com/threads/28639-S-pdif" target="_blank">forum discussion</a>
 		included valuable input and code from "kpc".
 	<h3>Notes</h3>
 	<p>S/PDIF output uses the I2S hardware.  This object can not be used
@@ -1876,7 +1876,7 @@ <h3>Examples</h3>
 	<h3>Credits</h3>
 	<p><a href="https://github.com/FrankBoesing" target="_blank">Frank Boesing</a>
 		developed the AudioOutputSPDIF code.  The original
-		<a href="https://forum.pjrc.com/threads/28639-S-pdif" target="_blank">forum disussion</a>
+		<a href="https://forum.pjrc.com/threads/28639-S-pdif" target="_blank">forum discussion</a>
 		included valuable input and code from "kpc".
 	<h3>Notes</h3>
 	<p>S/PDIF output uses the I2S2 hardware.  This object can not be used
@@ -1979,7 +1979,7 @@ <h3>Examples</h3>
 	<h3>Credits</h3>
 	<p>Frank Boesing and Benjamin developed this PT8211 object.  Details can be
 		found on this
-		<a href="https://forum.pjrc.com/threads/29284-Dual-channel-16bit-dac-PT8211/page3" target="_blank">forum disussion</a>.
+		<a href="https://forum.pjrc.com/threads/29284-Dual-channel-16bit-dac-PT8211/page3" target="_blank">forum discussion</a>.
 	<h3>Notes</h3>
 	<p>
 	</p>
@@ -2025,7 +2025,7 @@ <h3>Examples</h3>
 	<h3>Credits</h3>
 	<p>Frank Boesing and Benjamin developed this PT8211 object.  Details can be
 		found on this
-		<a href="https://forum.pjrc.com/threads/29284-Dual-channel-16bit-dac-PT8211/page3" target="_blank">forum disussion</a>.
+		<a href="https://forum.pjrc.com/threads/29284-Dual-channel-16bit-dac-PT8211/page3" target="_blank">forum discussion</a>.
 	<h3>Notes</h3>
 	<p>
 	</p>
@@ -4229,7 +4229,7 @@ <h3>Hardware</h3>
     </table>
 	</p>
 	<p>When <span class=literal>AUDIO_MEMORY_CY15B104</span> is used, a single
-		CY15B104 FRAM chip is used, with these pins:
+		CY15B104 F-RAM chip is used, with these pins:
     <table class=doc align=center cellpadding=3>
         <tr class=top><th>Pin</th><th>Signal</th></tr>
         <tr class=odd><td align=center>6</td><td>CS</td></tr>
@@ -4641,7 +4641,7 @@ <h3>Known Issues</h3>
 		the length to 4.
 	</p>
 	<p>The impulse response must be given in reverse order.  Many filters have
-		symetrical impluse response, making this a non-issue.  If your filter has
+		symmetrical impluse response, making this a non-issue.  If your filter has
 		a non-symetrical response, make sure the data is in reverse time order.
 	</p>
 	<h3>Notes</h3>
@@ -5278,7 +5278,7 @@ <h3>Signal Levels</h3>
 31: 1.16 Volts p-p
 </pre>
 	<p class=func><span class=keyword>lineOutLevel</span>(left, right);</p>
-	<p class=desc>Adjust the line level outout voltage range, with separate
+	<p class=desc>Adjust the line level output voltage range, with separate
 		settings for left and right.  The same settings (13 to 31) are available.
 	</p>
 
@@ -5292,7 +5292,7 @@ <h3>Signal Conditioning</h3>
 	<p class=func><span class=keyword>adcHighPassFilterFreeze</span>();</p>
 	<p class=desc>By default, the analog input (either line-level inputs or mic)
 		is high-pass filtered, to remove any DC component.  This function
-		freezes the filter, so the current DC component is still substracted, but
+		freezes the filter, so the current DC component is still subtracted, but
 		the filter stops tracking any DC or low frequency changes.
 	</p>
 	<p class=func><span class=keyword>adcHighPassFilterDisable</span>();</p>
@@ -5375,12 +5375,12 @@ <h3>Audio Processor</h3>
 	</p>
 	<p class=desc>If <em>hardLimit</em> is 0, a 'soft
 	knee' compressor is used to progressively compress louder values which are near to or above the
-	threashold (the louder they are, the greater the compression). If it is 1, a hard compressor
-	is used (all values above the threashold are the same loudness). The <em>threashold</em> is specified
+	threshold (the louder they are, the greater the compression). If it is 1, a hard compressor
+	is used (all values above the threshold are the same loudness). The <em>threshold</em> is specified
 	as a float in the range 0dBFS to -96dBFS, where -18dBFS is a typical value.
 	<em>attack</em> is a float controlling the rate of decrease in gain when the signal is over
-	threashold, in dB/s. <em>decay</em> controls how fast gain is restored once the level
-	drops below threashold, again in dB/s. It is typically set to a longer value than attack.
+	threshold, in dB/s. <em>decay</em> controls how fast gain is restored once the level
+	drops below threshold, again in dB/s. It is typically set to a longer value than attack.
 	</p>
 	<p class=func><span class=keyword>autoVolumeEnable</span>();</p>
 	<p class=desc>Enables auto volume control, using the previously specified settings.