diff --git a/_audio_effect_8h_source.html b/_audio_effect_8h_source.html index 0b7944bd25..7cf86aa224 100644 --- a/_audio_effect_8h_source.html +++ b/_audio_effect_8h_source.html @@ -73,406 +73,549 @@
2 #include "AudioEffects/AudioParameters.h"
3 #include "AudioEffects/PitchShift.h"
4 #include "AudioTools/AudioLogger.h"
-
5 #include <stdint.h>
-
6 
-
7 namespace audio_tools {
-
8 
-
9 // we use int16_t for our effects
-
10 typedef int16_t effect_t;
-
11 
-
19 class AudioEffect {
-
20 public:
-
21  AudioEffect() = default;
-
22  virtual ~AudioEffect() = default;
-
23 
-
25  virtual effect_t process(effect_t in) = 0;
-
26 
-
28  virtual void setActive(bool value) { active_flag = value; }
-
29 
-
31  virtual bool active() { return active_flag; }
-
32 
-
33  virtual AudioEffect *clone() = 0;
-
34 
-
36  int id() { return id_value; }
-
37 
-
39  void setId(int id) { this->id_value = id; }
-
40 
-
41 protected:
-
42  bool active_flag = true;
-
43  int id_value = -1;
-
44 
-
45  void copyParent(AudioEffect *copy) {
-
46  id_value = copy->id_value;
-
47  active_flag = copy->active_flag;
-
48  }
-
49 
-
51  int16_t clip(int32_t in, int16_t clipLimit = 32767,
-
52  int16_t resultLimit = 32767) {
-
53  int32_t result = in;
-
54  if (result > clipLimit) {
-
55  result = resultLimit;
-
56  }
-
57  if (result < -clipLimit) {
-
58  result = -resultLimit;
-
59  }
-
60  return result;
-
61  }
-
62 };
-
63 
-
72 class Boost : public AudioEffect {
-
73 public:
-
76  Boost(float volume = 1.0) { effect_value = volume; }
-
77 
-
78  Boost(const Boost &copy) = default;
-
79 
-
80  float volume() { return effect_value; }
-
81 
-
82  void setVolume(float volume) { effect_value = volume; }
-
83 
-
84  effect_t process(effect_t input) {
-
85  if (!active())
-
86  return input;
-
87  int32_t result = effect_value * input;
-
88  // clip to int16_t
-
89  return clip(result);
-
90  }
-
91 
-
92  Boost *clone() { return new Boost(*this); }
-
93 
-
94 protected:
-
95  float effect_value;
-
96 };
-
97 
-
105 class Distortion : public AudioEffect {
-
106 public:
-
108  Distortion(int16_t clipThreashold = 4990, int16_t maxInput = 6500) {
-
109  p_clip_threashold = clipThreashold;
-
110  max_input = maxInput;
-
111  }
-
112 
-
113  Distortion(const Distortion &copy) = default;
-
114 
-
115  void setClipThreashold(int16_t th) { p_clip_threashold = th; }
-
116 
-
117  int16_t clipThreashold() { return p_clip_threashold; }
-
118 
-
119  void setMaxInput(int16_t maxInput) { max_input = maxInput; }
-
120 
-
121  int16_t maxInput() { return max_input; }
-
122 
-
123  effect_t process(effect_t input) {
-
124  if (!active())
-
125  return input;
-
126  // the input signal is 16bits (values from -32768 to +32768
-
127  // the value of input is clipped to the distortion_threshold value
-
128  return clip(input, p_clip_threashold, max_input);
-
129  }
-
130 
-
131  Distortion *clone() { return new Distortion(*this); }
-
132 
-
133 protected:
-
134  int16_t p_clip_threashold;
-
135  int16_t max_input;
-
136 };
-
137 
-
145 class Fuzz : public AudioEffect {
-
146 public:
-
148  Fuzz(float fuzzEffectValue = 6.5, uint16_t maxOut = 300) {
-
149  p_effect_value = fuzzEffectValue;
-
150  max_out = maxOut;
-
151  }
-
152 
-
153  Fuzz(const Fuzz &copy) = default;
-
154 
-
155  void setFuzzEffectValue(float v) { p_effect_value = v; }
-
156 
-
157  float fuzzEffectValue() { return p_effect_value; }
-
158 
-
159  void setMaxOut(uint16_t v) { max_out = v; }
-
160 
-
161  uint16_t maxOut() { return max_out; }
-
162 
-
163  effect_t process(effect_t input) {
-
164  if (!active())
-
165  return input;
-
166  float v = p_effect_value;
-
167  int32_t result = clip(v * input);
-
168  return map(result * v, -32768, +32767, -max_out, max_out);
-
169  }
-
170 
-
171  Fuzz *clone() { return new Fuzz(*this); }
-
172 
-
173 protected:
-
174  float p_effect_value;
-
175  uint16_t max_out;
-
176 };
-
177 
-
185 class Tremolo : public AudioEffect {
-
186 public:
-
189  Tremolo(int16_t duration_ms = 2000, uint8_t depthPercent = 50,
-
190  uint32_t sampleRate = 44100) {
-
191  this->duration_ms = duration_ms;
-
192  this->sampleRate = sampleRate;
-
193  this->p_percent = depthPercent;
-
194  int32_t rate_count = sampleRate * duration_ms / 1000;
-
195  rate_count_half = rate_count / 2;
-
196  }
-
197 
-
198  Tremolo(const Tremolo &copy) = default;
-
199 
-
200  void setDuration(int16_t ms) {
-
201  int32_t rate_count = sampleRate * ms / 1000;
-
202  rate_count_half = rate_count / 2;
-
203  }
-
204 
-
205  int16_t duration() { return duration_ms; }
-
206 
-
207  void setDepth(uint8_t percent) { p_percent = percent; }
-
208 
-
209  uint8_t depth() { return p_percent; }
-
210 
-
211  effect_t process(effect_t input) {
-
212  if (!active())
-
213  return input;
-
214 
-
215  // limit value to max 100% and calculate factors
-
216  float tremolo_depth = p_percent > 100 ? 1.0 : 0.01 * p_percent;
-
217  float signal_depth = (100.0 - p_percent) / 100.0;
-
218 
-
219  float tremolo_factor = tremolo_depth / rate_count_half;
-
220  int32_t out = (signal_depth * input) + (tremolo_factor * count * input);
-
221 
-
222  // saw tooth shaped counter
-
223  count += inc;
-
224  if (count >= rate_count_half) {
-
225  inc = -1;
-
226  } else if (count <= 0) {
-
227  inc = +1;
-
228  }
-
229 
-
230  return clip(out);
-
231  }
-
232 
-
233  Tremolo *clone() { return new Tremolo(*this); }
-
234 
-
235 protected:
-
236  int16_t duration_ms;
-
237  uint32_t sampleRate;
-
238  int32_t count = 0;
-
239  int16_t inc = 1;
-
240  int32_t rate_count_half; // number of samples for on raise and fall
-
241  uint8_t p_percent;
-
242 };
-
243 
-
252 class Delay : public AudioEffect {
-
253 public:
-
255  Delay(uint16_t duration_ms = 1000, float depth = 0.5,
-
256  float feedbackAmount = 1.0, uint32_t sampleRate = 44100,
-
257  bool zeroIfBufferEmpty = false) {
-
258  setSampleRate(sampleRate);
-
259  setFeedback(feedbackAmount);
-
260  setDepth(depth);
-
261  setDuration(duration_ms);
-
262  }
-
263 
-
264  Delay(const Delay &copy) {
-
265  setSampleRate(copy.sampleRate);
-
266  setFeedback(copy.feedback);
-
267  setDepth(copy.depth);
-
268  setDuration(copy.duration);
-
269  };
-
270 
-
271  void setDuration(int16_t dur) {
-
272  duration = dur;
-
273  updateBufferSize();
-
274  }
-
275 
-
276  int16_t getDuration() { return duration; }
-
277 
-
278  void setDepth(float value) {
-
279  depth = value;
-
280  if (depth > 1.0)
-
281  depth = 1.0;
-
282  if (depth < 0)
-
283  depth = 0.0;
-
284  }
-
285 
-
286  float getDepth() { return depth; }
-
287 
-
288  void setFeedback(float feed) {
-
289  feedback = feed;
-
290  if (feedback > 1.0)
-
291  feedback = 1.0;
-
292  if (feedback < 0)
-
293  feedback = 0.0;
-
294  }
-
295 
-
296  float getFeedback() { return feedback; }
-
297 
-
298  void setSampleRate(int32_t sample) {
-
299  sampleRate = sample;
-
300  updateBufferSize();
-
301  }
-
302 
-
303  float getSampleRate() { return sampleRate; }
-
304 
-
305  effect_t process(effect_t input) {
-
306  if (!active())
-
307  return input;
-
308 
-
309  // Read last audio sample in each delay line
-
310  int32_t delayed_value = buffer[delay_line_index];
-
311 
-
312  // Mix the above with current audio and write the results back to output
-
313  int32_t out = ((1.0 - depth) * input) + (depth * delayed_value);
-
314 
-
315  // Update each delay line
-
316  buffer[delay_line_index] = clip(feedback * (delayed_value + input));
-
317 
-
318  // Finally, update the delay line index
-
319  if (delay_line_index++ >= delay_len_samples) {
-
320  delay_line_index = 0;
-
321  }
-
322  return clip(out);
-
323  }
-
324 
-
325  Delay *clone() { return new Delay(*this); }
-
326 
-
327 protected:
-
328  Vector<effect_t> buffer{0};
-
329  float feedback = 0.0, duration = 0.0, sampleRate = 0.0, depth = 0.0;
-
330  size_t delay_len_samples = 0;
-
331  size_t delay_line_index = 0;
-
332 
-
333  void updateBufferSize() {
-
334  if (sampleRate > 0 && duration > 0) {
-
335  size_t newSampleCount = sampleRate * duration / 1000;
-
336  if (newSampleCount != delay_len_samples) {
-
337  delay_len_samples = newSampleCount;
-
338  buffer.resize(delay_len_samples);
-
339  memset(buffer.data(),0,delay_len_samples*sizeof(effect_t));
-
340  LOGD("sample_count: %u", (unsigned)delay_len_samples);
-
341  }
-
342  }
-
343  }
-
344 };
-
345 
-
361 class ADSRGain : public AudioEffect {
-
362 public:
-
363  ADSRGain(float attack = 0.001, float decay = 0.001, float sustainLevel = 0.5,
-
364  float release = 0.005, float boostFactor = 1.0) {
-
365  this->factor = boostFactor;
-
366  adsr = new ADSR(attack, decay, sustainLevel, release);
-
367  }
-
368 
-
369  ADSRGain(const ADSRGain &ref) {
-
370  adsr = new ADSR(*(ref.adsr));
-
371  factor = ref.factor;
-
372  copyParent((AudioEffect *)&ref);
-
373  };
-
374 
-
375  virtual ~ADSRGain() { delete adsr; }
-
376 
-
377  void setAttackRate(float a) { adsr->setAttackRate(a); }
-
378 
-
379  float attackRate() { return adsr->attackRate(); }
-
380 
-
381  void setDecayRate(float d) { adsr->setDecayRate(d); }
-
382 
-
383  float decayRate() { return adsr->decayRate(); }
-
384 
-
385  void setSustainLevel(float s) { adsr->setSustainLevel(s); }
-
386 
-
387  float sustainLevel() { return adsr->sustainLevel(); }
-
388 
-
389  void setReleaseRate(float r) { adsr->setReleaseRate(r); }
-
390 
-
391  float releaseRate() { return adsr->releaseRate(); }
-
392 
-
393  void keyOn(float tgt = 0) { adsr->keyOn(tgt); }
-
394 
-
395  void keyOff() { adsr->keyOff(); }
-
396 
-
397  effect_t process(effect_t input) {
-
398  effect_t result = factor * adsr->tick() * input;
-
399  return result;
-
400  }
-
401 
-
402  bool isActive() { return adsr->isActive(); }
-
403 
-
404  ADSRGain *clone() { return new ADSRGain(*this); }
-
405 
-
406 protected:
-
407  ADSR *adsr;
-
408  float factor;
-
409 };
-
410 
-
417 class PitchShift : public AudioEffect {
-
418 public:
-
421  PitchShift(float shift_value = 1.0, int buffer_size = 1000) {
-
422  effect_value = shift_value;
-
423  size = buffer_size;
-
424  buffer.resize(buffer_size);
-
425  buffer.setIncrement(shift_value);
-
426  }
-
427 
-
428  PitchShift(const PitchShift &ref) {
-
429  size = ref.size;
-
430  effect_value = ref.effect_value;
-
431  buffer.resize(size);
-
432  buffer.setIncrement(effect_value);
-
433  };
-
434 
-
435  float value() { return effect_value; }
-
436 
-
437  void setValue(float value) {
-
438  effect_value = value;
-
439  buffer.setIncrement(value);
-
440  }
-
441 
-
442  effect_t process(effect_t input) {
-
443  if (!active())
-
444  return input;
-
445  buffer.write(input);
-
446  return buffer.read();
-
447  }
-
448 
-
449  PitchShift *clone() { return new PitchShift(*this); }
-
450 
-
451 protected:
-
452  VariableSpeedRingBuffer<int16_t> buffer;
-
453  float effect_value;
-
454  int size;
-
455 };
-
456 } // namespace audio_tools
-
audio_tools::ADSRGain
ADSR Envelope: Attack, Decay, Sustain and Release. Attack is the time taken for initial run-up oeffec...
Definition: AudioEffect.h:361
-
audio_tools::ADSRGain::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:397
+
5 #include "AudioTools/AudioTypes.h"
+
6 #include <stdint.h>
+
7 
+
8 namespace audio_tools {
+
9 
+
10 // we use int16_t for our effects
+
11 typedef int16_t effect_t;
+
12 
+
20 class AudioEffect {
+
21 public:
+
22  AudioEffect() = default;
+
23  virtual ~AudioEffect() = default;
+
24 
+
26  virtual effect_t process(effect_t in) = 0;
+
27 
+
29  virtual void setActive(bool value) { active_flag = value; }
+
30 
+
32  virtual bool active() { return active_flag; }
+
33 
+
34  virtual AudioEffect *clone() = 0;
+
35 
+
37  int id() { return id_value; }
+
38 
+
40  void setId(int id) { this->id_value = id; }
+
41 
+
42 protected:
+
43  bool active_flag = true;
+
44  int id_value = -1;
+
45 
+
46  void copyParent(AudioEffect *copy) {
+
47  id_value = copy->id_value;
+
48  active_flag = copy->active_flag;
+
49  }
+
50 
+
52  int16_t clip(int32_t in, int16_t clipLimit = 32767,
+
53  int16_t resultLimit = 32767) {
+
54  int32_t result = in;
+
55  if (result > clipLimit) {
+
56  result = resultLimit;
+
57  }
+
58  if (result < -clipLimit) {
+
59  result = -resultLimit;
+
60  }
+
61  return result;
+
62  }
+
63 };
+
64 
+
73 class Boost : public AudioEffect {
+
74 public:
+
77  Boost(float volume = 1.0) { effect_value = volume; }
+
78 
+
79  Boost(const Boost &copy) = default;
+
80 
+
81  float volume() { return effect_value; }
+
82 
+
83  void setVolume(float volume) { effect_value = volume; }
+
84 
+
85  effect_t process(effect_t input) {
+
86  if (!active())
+
87  return input;
+
88  int32_t result = effect_value * input;
+
89  // clip to int16_t
+
90  return clip(result);
+
91  }
+
92 
+
93  Boost *clone() { return new Boost(*this); }
+
94 
+
95 protected:
+
96  float effect_value;
+
97 };
+
98 
+
106 class Distortion : public AudioEffect {
+
107 public:
+
109  Distortion(int16_t clipThreashold = 4990, int16_t maxInput = 6500) {
+
110  p_clip_threashold = clipThreashold;
+
111  max_input = maxInput;
+
112  }
+
113 
+
114  Distortion(const Distortion &copy) = default;
+
115 
+
116  void setClipThreashold(int16_t th) { p_clip_threashold = th; }
+
117 
+
118  int16_t clipThreashold() { return p_clip_threashold; }
+
119 
+
120  void setMaxInput(int16_t maxInput) { max_input = maxInput; }
+
121 
+
122  int16_t maxInput() { return max_input; }
+
123 
+
124  effect_t process(effect_t input) {
+
125  if (!active())
+
126  return input;
+
127  // the input signal is 16bits (values from -32768 to +32768
+
128  // the value of input is clipped to the distortion_threshold value
+
129  return clip(input, p_clip_threashold, max_input);
+
130  }
+
131 
+
132  Distortion *clone() { return new Distortion(*this); }
+
133 
+
134 protected:
+
135  int16_t p_clip_threashold;
+
136  int16_t max_input;
+
137 };
+
138 
+
146 class Fuzz : public AudioEffect {
+
147 public:
+
149  Fuzz(float fuzzEffectValue = 6.5, uint16_t maxOut = 300) {
+
150  p_effect_value = fuzzEffectValue;
+
151  max_out = maxOut;
+
152  }
+
153 
+
154  Fuzz(const Fuzz &copy) = default;
+
155 
+
156  void setFuzzEffectValue(float v) { p_effect_value = v; }
+
157 
+
158  float fuzzEffectValue() { return p_effect_value; }
+
159 
+
160  void setMaxOut(uint16_t v) { max_out = v; }
+
161 
+
162  uint16_t maxOut() { return max_out; }
+
163 
+
164  effect_t process(effect_t input) {
+
165  if (!active())
+
166  return input;
+
167  float v = p_effect_value;
+
168  int32_t result = clip(v * input);
+
169  return map(result * v, -32768, +32767, -max_out, max_out);
+
170  }
+
171 
+
172  Fuzz *clone() { return new Fuzz(*this); }
+
173 
+
174 protected:
+
175  float p_effect_value;
+
176  uint16_t max_out;
+
177 };
+
178 
+
186 class Tremolo : public AudioEffect {
+
187 public:
+
190  Tremolo(int16_t duration_ms = 2000, uint8_t depthPercent = 50,
+
191  uint32_t sampleRate = 44100) {
+
192  this->duration_ms = duration_ms;
+
193  this->sampleRate = sampleRate;
+
194  this->p_percent = depthPercent;
+
195  int32_t rate_count = sampleRate * duration_ms / 1000;
+
196  rate_count_half = rate_count / 2;
+
197  }
+
198 
+
199  Tremolo(const Tremolo &copy) = default;
+
200 
+
201  void setDuration(int16_t ms) {
+
202  int32_t rate_count = sampleRate * ms / 1000;
+
203  rate_count_half = rate_count / 2;
+
204  }
+
205 
+
206  int16_t duration() { return duration_ms; }
+
207 
+
208  void setDepth(uint8_t percent) { p_percent = percent; }
+
209 
+
210  uint8_t depth() { return p_percent; }
+
211 
+
212  effect_t process(effect_t input) {
+
213  if (!active())
+
214  return input;
+
215 
+
216  // limit value to max 100% and calculate factors
+
217  float tremolo_depth = p_percent > 100 ? 1.0 : 0.01 * p_percent;
+
218  float signal_depth = (100.0 - p_percent) / 100.0;
+
219 
+
220  float tremolo_factor = tremolo_depth / rate_count_half;
+
221  int32_t out = (signal_depth * input) + (tremolo_factor * count * input);
+
222 
+
223  // saw tooth shaped counter
+
224  count += inc;
+
225  if (count >= rate_count_half) {
+
226  inc = -1;
+
227  } else if (count <= 0) {
+
228  inc = +1;
+
229  }
+
230 
+
231  return clip(out);
+
232  }
+
233 
+
234  Tremolo *clone() { return new Tremolo(*this); }
+
235 
+
236 protected:
+
237  int16_t duration_ms;
+
238  uint32_t sampleRate;
+
239  int32_t count = 0;
+
240  int16_t inc = 1;
+
241  int32_t rate_count_half; // number of samples for on raise and fall
+
242  uint8_t p_percent;
+
243 };
+
244 
+
253 class Delay : public AudioEffect {
+
254 public:
+
256  Delay(uint16_t duration_ms = 1000, float depth = 0.5,
+
257  float feedbackAmount = 1.0, uint32_t sampleRate = 44100,
+
258  bool zeroIfBufferEmpty = false) {
+
259  setSampleRate(sampleRate);
+
260  setFeedback(feedbackAmount);
+
261  setDepth(depth);
+
262  setDuration(duration_ms);
+
263  }
+
264 
+
265  Delay(const Delay &copy) {
+
266  setSampleRate(copy.sampleRate);
+
267  setFeedback(copy.feedback);
+
268  setDepth(copy.depth);
+
269  setDuration(copy.duration);
+
270  };
+
271 
+
272  void setDuration(int16_t dur) {
+
273  duration = dur;
+
274  updateBufferSize();
+
275  }
+
276 
+
277  int16_t getDuration() { return duration; }
+
278 
+
279  void setDepth(float value) {
+
280  depth = value;
+
281  if (depth > 1.0)
+
282  depth = 1.0;
+
283  if (depth < 0)
+
284  depth = 0.0;
+
285  }
+
286 
+
287  float getDepth() { return depth; }
+
288 
+
289  void setFeedback(float feed) {
+
290  feedback = feed;
+
291  if (feedback > 1.0)
+
292  feedback = 1.0;
+
293  if (feedback < 0)
+
294  feedback = 0.0;
+
295  }
+
296 
+
297  float getFeedback() { return feedback; }
+
298 
+
299  void setSampleRate(int32_t sample) {
+
300  sampleRate = sample;
+
301  updateBufferSize();
+
302  }
+
303 
+
304  float getSampleRate() { return sampleRate; }
+
305 
+
306  effect_t process(effect_t input) {
+
307  if (!active())
+
308  return input;
+
309 
+
310  // Read last audio sample in each delay line
+
311  int32_t delayed_value = buffer[delay_line_index];
+
312 
+
313  // Mix the above with current audio and write the results back to output
+
314  int32_t out = ((1.0 - depth) * input) + (depth * delayed_value);
+
315 
+
316  // Update each delay line
+
317  buffer[delay_line_index] = clip(feedback * (delayed_value + input));
+
318 
+
319  // Finally, update the delay line index
+
320  if (delay_line_index++ >= delay_len_samples) {
+
321  delay_line_index = 0;
+
322  }
+
323  return clip(out);
+
324  }
+
325 
+
326  Delay *clone() { return new Delay(*this); }
+
327 
+
328 protected:
+
329  Vector<effect_t> buffer{0};
+
330  float feedback = 0.0, duration = 0.0, sampleRate = 0.0, depth = 0.0;
+
331  size_t delay_len_samples = 0;
+
332  size_t delay_line_index = 0;
+
333 
+
334  void updateBufferSize() {
+
335  if (sampleRate > 0 && duration > 0) {
+
336  size_t newSampleCount = sampleRate * duration / 1000;
+
337  if (newSampleCount != delay_len_samples) {
+
338  delay_len_samples = newSampleCount;
+
339  buffer.resize(delay_len_samples);
+
340  memset(buffer.data(),0,delay_len_samples*sizeof(effect_t));
+
341  LOGD("sample_count: %u", (unsigned)delay_len_samples);
+
342  }
+
343  }
+
344  }
+
345 };
+
346 
+
362 class ADSRGain : public AudioEffect {
+
363 public:
+
364  ADSRGain(float attack = 0.001, float decay = 0.001, float sustainLevel = 0.5,
+
365  float release = 0.005, float boostFactor = 1.0) {
+
366  this->factor = boostFactor;
+
367  adsr = new ADSR(attack, decay, sustainLevel, release);
+
368  }
+
369 
+
370  ADSRGain(const ADSRGain &ref) {
+
371  adsr = new ADSR(*(ref.adsr));
+
372  factor = ref.factor;
+
373  copyParent((AudioEffect *)&ref);
+
374  };
+
375 
+
376  virtual ~ADSRGain() { delete adsr; }
+
377 
+
378  void setAttackRate(float a) { adsr->setAttackRate(a); }
+
379 
+
380  float attackRate() { return adsr->attackRate(); }
+
381 
+
382  void setDecayRate(float d) { adsr->setDecayRate(d); }
+
383 
+
384  float decayRate() { return adsr->decayRate(); }
+
385 
+
386  void setSustainLevel(float s) { adsr->setSustainLevel(s); }
+
387 
+
388  float sustainLevel() { return adsr->sustainLevel(); }
+
389 
+
390  void setReleaseRate(float r) { adsr->setReleaseRate(r); }
+
391 
+
392  float releaseRate() { return adsr->releaseRate(); }
+
393 
+
394  void keyOn(float tgt = 0) { adsr->keyOn(tgt); }
+
395 
+
396  void keyOff() { adsr->keyOff(); }
+
397 
+
398  effect_t process(effect_t input) {
+
399  effect_t result = factor * adsr->tick() * input;
+
400  return result;
+
401  }
+
402 
+
403  bool isActive() { return adsr->isActive(); }
+
404 
+
405  ADSRGain *clone() { return new ADSRGain(*this); }
+
406 
+
407 protected:
+
408  ADSR *adsr;
+
409  float factor;
+
410 };
+
411 
+
418 class PitchShift : public AudioEffect {
+
419 public:
+
422  PitchShift(float shift_value = 1.0, int buffer_size = 1000) {
+
423  effect_value = shift_value;
+
424  size = buffer_size;
+
425  buffer.resize(buffer_size);
+
426  buffer.setIncrement(shift_value);
+
427  }
+
428 
+
429  PitchShift(const PitchShift &ref) {
+
430  size = ref.size;
+
431  effect_value = ref.effect_value;
+
432  buffer.resize(size);
+
433  buffer.setIncrement(effect_value);
+
434  };
+
435 
+
436  float value() { return effect_value; }
+
437 
+
438  void setValue(float value) {
+
439  effect_value = value;
+
440  buffer.setIncrement(value);
+
441  }
+
442 
+
443  effect_t process(effect_t input) {
+
444  if (!active())
+
445  return input;
+
446  buffer.write(input);
+
447  return buffer.read();
+
448  }
+
449 
+
450  PitchShift *clone() { return new PitchShift(*this); }
+
451 
+
452 protected:
+
453  VariableSpeedRingBuffer<int16_t> buffer;
+
454  float effect_value;
+
455  int size;
+
456 };
+
457 
+
458 
+
466 class Compressor : public AudioEffect {
+
467 
+
468  Compressor(const Compressor &copy) = default;
+
469 
+
470  Compressor(uint32_t sampleRate = 44100, int32_t attackMs=30, int32_t releaseMs=20, int32_t holdMs=10, uint8_t thresholdPercent=10, float compressionRatio=0.5){
+
471  //assuming 1 sample = 1/96kHz = ~10us
+
472  //Attack -> 30 ms -> 3000
+
473  //Release -> 20 ms -> 2000
+
474  //Hold -> 10ms -> 1000
+
475  sample_rate = sample_rate * attackMs / 1000;
+
476  attack_count = sample_rate * attackMs / 1000;
+
477  release_count = sample_rate * releaseMs / 1000;
+
478  hold_count = sample_rate * holdMs / 1000;
+
479 
+
480  //threshold -20dB below limit -> 0.1 * 2^31
+
481  threshold = 0.01f * thresholdPercent * NumberConverter::maxValueT<effect_t>();
+
482  //compression ratio: 6:1 -> -6dB = 0.5
+
483  gainreduce = compressionRatio;
+
484  //initial gain = 1.0 -> no compression
+
485  gain = 1.0f;
+
486  recalculate();
+
487  }
+
488 
+
489  Compressor *clone() { return new Compressor(*this); }
+
490 
+
491  void setAttack(int32_t attackMs){
+
492  attack_count = sample_rate * attackMs / 1000;
+
493  recalculate();
+
494  }
+
495 
+
496  void setRelease(int32_t releaseMs){
+
497  release_count = sample_rate * releaseMs / 1000;
+
498  recalculate();
+
499  }
+
500 
+
501  void setHold(int32_t holdMs){
+
502  hold_count = sample_rate * holdMs / 1000;
+
503  recalculate();
+
504  }
+
505 
+
506  void setThresholdPercent(uint8_t thresholdPercent){
+
507  threshold = 0.01f * thresholdPercent * NumberConverter::maxValueT<effect_t>();
+
508  }
+
509 
+
510  void setCompressionRatio(float compressionRatio){
+
511  if (compressionRatio<1.0){
+
512  gainreduce = compressionRatio;
+
513  }
+
514  recalculate();
+
515  }
+
516 
+
517  effect_t process(effect_t inSample) {
+
518  float inSampleF = (float)inSample;
+
519 
+
520  if (fabs(inSampleF) > threshold) {
+
521  if (gain >= gainreduce) {
+
522  if (State==S_NoOperation) {
+
523  State=S_Attack;
+
524  timeout = attack_count;
+
525  }
+
526  else if (State==S_Release) {
+
527  State=S_Attack;
+
528  timeout = attack_count;
+
529  }
+
530  }
+
531  if (State==S_GainReduction) timeout = hold_count;
+
532 
+
533  }
+
534 
+
535  if (fabs(inSampleF) < threshold && gain <= 1.0f) {
+
536  if ( timeout==0 && State==S_GainReduction) {
+
537  State=S_Release;
+
538  timeout = release_count;
+
539  }
+
540  }
+
541 
+
542  switch (State) {
+
543  case S_Attack:
+
544  if ( timeout>0 && gain > gainreduce) {
+
545  gain -= gain_step_attack;
+
546  timeout--;
+
547  }
+
548  else {
+
549  State=S_GainReduction;
+
550  timeout = hold_count;
+
551  }
+
552  break;
+
553 
+
554 
+
555  case S_GainReduction:
+
556  if ( timeout>0) timeout--;
+
557  else {
+
558  State=S_Release;
+
559  timeout = release_count;
+
560  }
+
561  break;
+
562 
+
563 
+
564  case S_Release:
+
565  if ( timeout>0 && gain<1.0f) {
+
566  timeout--;
+
567  gain += gain_step_release;
+
568  }
+
569  else {
+
570  State=S_NoOperation;
+
571  }
+
572  break;
+
573 
+
574  case S_NoOperation:
+
575  if (gain < 1.0f) gain = 1.0F;
+
576  break;
+
577 
+
578  default:
+
579 
+
580  break;
+
581 
+
582  }
+
583 
+
584  float outSampleF = inSample*gain;
+
585 
+
586  return (int) outSampleF;
+
587  }
+
588 
+
589 protected:
+
590  enum CompStates {S_NoOperation, S_Attack, S_GainReduction, S_Release };
+
591  enum CompStates State = S_NoOperation;
+
592 
+
593  int32_t attack_count, release_count, hold_count, timeout;
+
594  float gainreduce, gain_step_attack, gain_step_release, gain, threshold;
+
595  uint32_t sample_rate;
+
596 
+
597  void recalculate() {
+
598  gain_step_attack = (1.0f - gainreduce) / attack_count;
+
599  gain_step_release = (1.0f - gainreduce) / release_count;
+
600  }
+
601 
+
602 };
+
603 
+
604 
+
605 } // namespace audio_tools
+
audio_tools::ADSRGain
ADSR Envelope: Attack, Decay, Sustain and Release. Attack is the time taken for initial run-up oeffec...
Definition: AudioEffect.h:362
+
audio_tools::ADSRGain::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:398
audio_tools::ADSR
Generates ADSR values between 0.0 and 1.0.
Definition: AudioParameters.h:51
-
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:19
-
audio_tools::AudioEffect::active
virtual bool active()
determines if the effect is active
Definition: AudioEffect.h:31
-
audio_tools::AudioEffect::setActive
virtual void setActive(bool value)
sets the effect active/inactive
Definition: AudioEffect.h:28
-
audio_tools::AudioEffect::setId
void setId(int id)
Allows to identify an effect.
Definition: AudioEffect.h:39
-
audio_tools::AudioEffect::id
int id()
Allows to identify an effect.
Definition: AudioEffect.h:36
+
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:20
+
audio_tools::AudioEffect::active
virtual bool active()
determines if the effect is active
Definition: AudioEffect.h:32
+
audio_tools::AudioEffect::setActive
virtual void setActive(bool value)
sets the effect active/inactive
Definition: AudioEffect.h:29
+
audio_tools::AudioEffect::setId
void setId(int id)
Allows to identify an effect.
Definition: AudioEffect.h:40
+
audio_tools::AudioEffect::id
int id()
Allows to identify an effect.
Definition: AudioEffect.h:37
audio_tools::AudioEffect::process
virtual effect_t process(effect_t in)=0
calculates the effect output from the input
-
audio_tools::AudioEffect::clip
int16_t clip(int32_t in, int16_t clipLimit=32767, int16_t resultLimit=32767)
generic clipping method
Definition: AudioEffect.h:51
-
audio_tools::Boost
Boost AudioEffect.
Definition: AudioEffect.h:72
-
audio_tools::Boost::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:84
-
audio_tools::Boost::Boost
Boost(float volume=1.0)
Definition: AudioEffect.h:76
-
audio_tools::Delay
Delay/Echo AudioEffect. See https://wiki.analog.com/resources/tools-software/sharc-audio-module/barem...
Definition: AudioEffect.h:252
-
audio_tools::Delay::Delay
Delay(uint16_t duration_ms=1000, float depth=0.5, float feedbackAmount=1.0, uint32_t sampleRate=44100, bool zeroIfBufferEmpty=false)
e.g. depth=0.5, ms=1000, sampleRate=44100
Definition: AudioEffect.h:255
-
audio_tools::Delay::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:305
-
audio_tools::Distortion
Distortion AudioEffect.
Definition: AudioEffect.h:105
-
audio_tools::Distortion::Distortion
Distortion(int16_t clipThreashold=4990, int16_t maxInput=6500)
Distortion Constructor: e.g. use clipThreashold 4990 and maxInput=6500.
Definition: AudioEffect.h:108
-
audio_tools::Distortion::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:123
-
audio_tools::Fuzz
Fuzz AudioEffect.
Definition: AudioEffect.h:145
-
audio_tools::Fuzz::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:163
-
audio_tools::Fuzz::Fuzz
Fuzz(float fuzzEffectValue=6.5, uint16_t maxOut=300)
Fuzz Constructor: use e.g. effectValue=6.5; maxOut = 300.
Definition: AudioEffect.h:148
-
audio_tools::PitchShift
Shifts the pitch by the indicated step size: e.g. 2 doubles the pitch.
Definition: AudioEffect.h:417
-
audio_tools::PitchShift::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:442
-
audio_tools::PitchShift::PitchShift
PitchShift(float shift_value=1.0, int buffer_size=1000)
Definition: AudioEffect.h:421
-
audio_tools::Tremolo
Tremolo AudioEffect.
Definition: AudioEffect.h:185
-
audio_tools::Tremolo::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:211
-
audio_tools::Tremolo::Tremolo
Tremolo(int16_t duration_ms=2000, uint8_t depthPercent=50, uint32_t sampleRate=44100)
Definition: AudioEffect.h:189
+
audio_tools::AudioEffect::clip
int16_t clip(int32_t in, int16_t clipLimit=32767, int16_t resultLimit=32767)
generic clipping method
Definition: AudioEffect.h:52
+
audio_tools::Boost
Boost AudioEffect.
Definition: AudioEffect.h:73
+
audio_tools::Boost::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:85
+
audio_tools::Boost::Boost
Boost(float volume=1.0)
Definition: AudioEffect.h:77
+
audio_tools::Compressor
Compressor inspired by https://github.com/YetAnotherElectronicsChannel/STM32_DSP_COMPRESSOR/blob/mast...
Definition: AudioEffect.h:466
+
audio_tools::Delay
Delay/Echo AudioEffect. See https://wiki.analog.com/resources/tools-software/sharc-audio-module/barem...
Definition: AudioEffect.h:253
+
audio_tools::Delay::Delay
Delay(uint16_t duration_ms=1000, float depth=0.5, float feedbackAmount=1.0, uint32_t sampleRate=44100, bool zeroIfBufferEmpty=false)
e.g. depth=0.5, ms=1000, sampleRate=44100
Definition: AudioEffect.h:256
+
audio_tools::Delay::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:306
+
audio_tools::Distortion
Distortion AudioEffect.
Definition: AudioEffect.h:106
+
audio_tools::Distortion::Distortion
Distortion(int16_t clipThreashold=4990, int16_t maxInput=6500)
Distortion Constructor: e.g. use clipThreashold 4990 and maxInput=6500.
Definition: AudioEffect.h:109
+
audio_tools::Distortion::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:124
+
audio_tools::Fuzz
Fuzz AudioEffect.
Definition: AudioEffect.h:146
+
audio_tools::Fuzz::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:164
+
audio_tools::Fuzz::Fuzz
Fuzz(float fuzzEffectValue=6.5, uint16_t maxOut=300)
Fuzz Constructor: use e.g. effectValue=6.5; maxOut = 300.
Definition: AudioEffect.h:149
+
audio_tools::PitchShift
Shifts the pitch by the indicated step size: e.g. 2 doubles the pitch.
Definition: AudioEffect.h:418
+
audio_tools::PitchShift::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:443
+
audio_tools::PitchShift::PitchShift
PitchShift(float shift_value=1.0, int buffer_size=1000)
Definition: AudioEffect.h:422
+
audio_tools::Tremolo
Tremolo AudioEffect.
Definition: AudioEffect.h:186
+
audio_tools::Tremolo::process
effect_t process(effect_t input)
calculates the effect output from the input
Definition: AudioEffect.h:212
+
audio_tools::Tremolo::Tremolo
Tremolo(int16_t duration_ms=2000, uint8_t depthPercent=50, uint32_t sampleRate=44100)
Definition: AudioEffect.h:190
audio_tools::VariableSpeedRingBuffer::read
T read()
reads a single value
Definition: PitchShift.h:238
audio_tools::VariableSpeedRingBuffer::write
bool write(T sample)
write add an entry to the buffer
Definition: PitchShift.h:255
audio_tools
Generic Implementation of sound input and output for desktop environments using portaudio.
Definition: AnalogAudio.h:10
diff --git a/_audio_effects_8h_source.html b/_audio_effects_8h_source.html index 0f70258873..b119ea93b0 100644 --- a/_audio_effects_8h_source.html +++ b/_audio_effects_8h_source.html @@ -498,7 +498,7 @@
audio_tools::AudioEffectCommon::addEffect
void addEffect(AudioEffect &effect)
Adds an effect object (by reference)
Definition: AudioEffects.h:27
audio_tools::AudioEffectCommon::findEffect
AudioEffect * findEffect(int id)
Finds an effect by id.
Definition: AudioEffects.h:50
audio_tools::AudioEffectCommon::clear
void clear()
deletes all defined effects
Definition: AudioEffects.h:39
-
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:19
+
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:20
audio_tools::AudioEffectStream
EffectsStream supporting variable bits_per_sample. This class is only available when __cplusplus >= 2...
Definition: AudioEffects.h:395
audio_tools::AudioEffectStreamT
EffectsStreamT: the template class describes an input or output stream to which one or multiple effec...
Definition: AudioEffects.h:208
audio_tools::AudioEffectStreamT::size
size_t size()
Provides the actual number of defined effects.
Definition: AudioEffects.h:363
diff --git a/_audio_effects_suite_8h_source.html b/_audio_effects_suite_8h_source.html index c9ce67b86c..a5aa2a7c5d 100644 --- a/_audio_effects_suite_8h_source.html +++ b/_audio_effects_suite_8h_source.html @@ -1191,7 +1191,7 @@
1586 };
1587 
1588 } // namespace effectsuite_tools
-
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:19
+
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:20
audio_tools::DelayEffectBase
A Base class for delay based digital effects. Provides the basic methods that are shared amongst Flan...
Definition: AudioEffectsSuite.h:396
audio_tools::DelayEffectBase::setDelayBuffReadIndex
void setDelayBuffReadIndex(effectsuite_t index)
Definition: AudioEffectsSuite.h:559
audio_tools::DelayEffectBase::~DelayEffectBase
~DelayEffectBase()
Definition: AudioEffectsSuite.h:412
diff --git a/_audio_f_e_c_8h_source.html b/_audio_f_e_c_8h_source.html index e86f152e0e..9bb88232b7 100644 --- a/_audio_f_e_c_8h_source.html +++ b/_audio_f_e_c_8h_source.html @@ -72,6 +72,7 @@
1 #pragma once
2 // Forward Error Corrections
3 #include "AudioFEC/FECReedSolomon.h"
+
4 #include "AudioFEC/FECHamming.h"
diff --git a/_audio_s_t_k_8h_source.html b/_audio_s_t_k_8h_source.html index e669cfc563..37a84f7a0b 100644 --- a/_audio_s_t_k_8h_source.html +++ b/_audio_s_t_k_8h_source.html @@ -300,7 +300,7 @@
307 
308 }
309 
-
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:19
+
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:20
audio_tools::GeneratedSoundStream
Source for reading generated tones. Please note.
Definition: AudioStreams.h:598
audio_tools::NumberConverter::maxValue
static int64_t maxValue(int value_bits_per_sample)
provides the biggest number for the indicated number of bits
Definition: AudioTypes.h:248
audio_tools::STKChorus
Chorus Effect.
Definition: AudioSTK.h:139
diff --git a/_f_e_c_hamming_8h_source.html b/_f_e_c_hamming_8h_source.html new file mode 100644 index 0000000000..b6d5644965 --- /dev/null +++ b/_f_e_c_hamming_8h_source.html @@ -0,0 +1,401 @@ + + + + + + + +arduino-audio-tools: src/AudioTools/AudioFEC/FECHamming.h Source File + + + + + + + + + +
+
+ + + + + + +
+
arduino-audio-tools +
+
+
+ + + + + + + + +
+
+ + +
+ +
+ + +
+
+
+
FECHamming.h
+
+
+
1 #pragma once
+
2 #include "AudioConfig.h"
+
3 #include "AudioTools/AudioStreams.h"
+
4 
+
5 #include <stdio.h>
+
6 #include <stdlib.h>
+
7 #include <stdbool.h>
+
8 #include <math.h>
+
9 #include <string.h>
+
10 
+
11 namespace audio_tools {
+
12 
+
28 template <int bytecount, class block_t>
+
29 class Hamming : public Stream {
+
30  public:
+
31  Hamming(Stream &stream){
+
32  p_stream = &stream;
+
33  p_print = &stream;
+
34  }
+
35 
+
36  Hamming(Print &print){
+
37  p_print = &print;
+
38  }
+
39 
+
40  int availableForWrite() override {
+
41  return bytecount;
+
42  }
+
43 
+
44  size_t write(const uint8_t* data, size_t len)override{
+
45  if (p_print==nullptr) return 0;
+
46  for (int j=0;j<len;j++){
+
47  raw.write(data[j]);
+
48  if(raw.availableForWrite()==0){
+
49  encode(raw.data(), raw.size());
+
50  raw.reset();
+
51  }
+
52  }
+
53  return len;
+
54  }
+
55 
+
56  int available() override{
+
57  return bytecount;
+
58  }
+
59 
+
60  size_t readBytes(uint8_t* data, size_t len)override{
+
61  if (p_stream==nullptr) return 0;
+
62  if (encoded.isEmpty()){
+
63  // read encoded data from surce
+
64  p_stream->readBytes(encoded.data(), encodedSize());
+
65  encoded.setAvailable(encodedSize());
+
66  // decode data
+
67  if(decode((block_t*)encoded.data(), bytecount)){
+
68  raw.setAvailable(bytecount);
+
69  }
+
70  }
+
71  return raw.readArray(data, len);
+
72  }
+
73 
+
75  size_t write(uint8_t c) override {
+
76  return write(&c, 1);
+
77  }
+
78 
+
80  int read() override {
+
81  uint8_t ch;
+
82  size_t len = readBytes(&ch, 1);
+
83  return len == 1 ? ch : -1;
+
84  }
+
86  int peek() override {return -1;}
+
87 
+
88 
+
89  protected:
+
90  SingleBuffer<uint8_t> raw{bytecount};
+
91  SingleBuffer<uint8_t> encoded{encodedSize()};
+
92  Stream* p_stream = nullptr;
+
93  Print* p_print = nullptr;
+
94 
+
95  int getBlocks(){
+
96  // Amount of bits in a block_t //
+
97  int bits = sizeof(block_t) * 8;
+
98 
+
99  // Amount of bits per block_t used to carry the message //
+
100  int messageBits = bits - log2(bits) - 1;
+
101 
+
102  // Amount of block_ts needed to encode message //
+
103  int block_ts = ceil((float) bytecount / messageBits);
+
104 
+
105  return block_ts;
+
106  }
+
107 
+
108  int encodedSize(){
+
109  return getBlocks()+1*sizeof(block_t);
+
110  }
+
111 
+
112  int encode(uint8_t *input, int len) {
+
113 
+
114  // Amount of bits in a block_t //
+
115  int bits = sizeof(block_t) * 8;
+
116 
+
117  // Amount of bits per block_t used to carry the message //
+
118  int messageBits = bits - log2(bits) - 1;
+
119 
+
120  // Amount of block_ts needed to encode message //
+
121  int block_ts = ceil((float) len / messageBits);
+
122 
+
123  // Array of encoded block_ts //
+
124  block_t encoded[block_ts+1];
+
125 
+
126  // Loop through each block_t //
+
127  for (int i = 0; i < block_ts+1; i++) {
+
128 
+
129  // Final encoded block_t variable //
+
130  block_t thisBlock = 0;
+
131 
+
132  // Amount of "skipped" bits (used for parity) //
+
133  int skipped = 0;
+
134 
+
135  // Count of how many bits are "on" //
+
136  int onCount = 0;
+
137 
+
138  // Array of "on" bits //
+
139  int onList[bits];
+
140 
+
141  // Loop through each message bit in this block_t to populate final block_t //
+
142  for (int j = 0; j < bits; j++) {
+
143 
+
144  // Skip bit if reserved for parity bit //
+
145  if ((j & (j - 1)) == 0) { // Check if j is a power of two or 0
+
146  skipped++;
+
147  continue;
+
148  }
+
149 
+
150  bool thisBit;
+
151 
+
152  if (i != block_ts) {
+
153 
+
154  // Current overall bit number //
+
155  int currentBit = i*messageBits + (j-skipped);
+
156 
+
157  // Current uint8_tacter //
+
158  int currentChar = currentBit/(sizeof(uint8_t)*8); // int division
+
159 
+
160  // Value of current bit //
+
161  thisBit = currentBit < len*sizeof(uint8_t)*8 ? getCharBit(input[currentChar], currentBit-currentChar*8) : 0;
+
162  }
+
163 
+
164  else {
+
165  thisBit = getBit(len/8, j-skipped+(sizeof(block_t)*8-messageBits));
+
166  }
+
167 
+
168  // If bit is "on", add to onList and onCount //
+
169  if (thisBit) {
+
170  onList[onCount] = j;
+
171  onCount++;
+
172  }
+
173 
+
174  // Populate final message block_t //
+
175  thisBlock = modifyBit(thisBlock, j, thisBit);
+
176  }
+
177 
+
178  // Calculate values of parity bits //
+
179  block_t parityBits = multipleXor(onList, onCount);
+
180 
+
181  // Loop through skipped bits (parity bits) //
+
182  for (int k = 1; k < skipped; k++) { // skip bit 0
+
183 
+
184  // If bit is "on", add to onCount
+
185  if (getBit(parityBits, sizeof(block_t)*8-skipped+k)) {
+
186  onCount++;
+
187  }
+
188 
+
189  // Add parity bit to final block_t //
+
190  thisBlock = modifyBit(thisBlock, (int) pow(2, skipped-k-1) , getBit(parityBits, sizeof(block_t)*8-skipped+k));
+
191  }
+
192 
+
193  // Add overall parity bit (total parity of onCount) //
+
194  thisBlock = modifyBit(thisBlock, 0, onCount & 1);
+
195 
+
196  // Add block_t to encoded block_ts //
+
197  encoded[i] = thisBlock;
+
198  }
+
199 
+
200  // Write encoded message to file /
+
201  return p_print->write((const uint8_t*)encoded, sizeof(block_t)*block_ts+1);
+
202  }
+
203 
+
204  bool decode(block_t input[], int len) {
+
205  uint8_t* output = (uint8_t*)raw.data();
+
206 
+
207  // Amount of bits in a block_t //
+
208  int bits = sizeof(block_t) * 8;
+
209 
+
210  for (int b = 0; b < (len/sizeof(block_t)); b++) {
+
211 
+
212  // Count of how many bits are "on" //
+
213  int onCount = 0;
+
214 
+
215  // Array of "on" bits //
+
216  int onList[bits];
+
217 
+
218  // Populate onCount and onList //
+
219  for (int i = 1; i < bits; i++) {
+
220  getBit(input[b], i);
+
221  if (getBit(input[b], i)) {
+
222  onList[onCount] = i;
+
223  onCount++;
+
224  }
+
225  }
+
226 
+
227  // Check for single errors //
+
228  int errorLoc = multipleXor(onList, onCount);
+
229 
+
230  if (errorLoc) {
+
231 
+
232  // Check for multiple errors //
+
233  if (!(onCount & 1 ^ getBit(input[b], 0))) { // last bit of onCount (total parity) XOR first bit of block_t (parity bit)
+
234  printf("\nMore than one error detected. Aborting.\n");
+
235  return false;
+
236  }
+
237 
+
238  // Flip error bit //
+
239  else {
+
240  input[b] = toggleBit(input[b], (bits-1) - errorLoc);
+
241  }
+
242  }
+
243  }
+
244 
+
245  // Amount of bits per block_t used to carry the message //
+
246  int messageBits = bits - log2(bits) - 1;
+
247 
+
248  int currentBit, currentChar;
+
249 
+
250  int uint8_ts = 0;
+
251 
+
252  for (int i = 0; i < len/sizeof(block_t); i++) {
+
253 
+
254  // Initialise variable to store amount of parity bits passed //
+
255  int skipped = 0;
+
256 
+
257  // Loop through each message bit in this block_t to populate final block_t //
+
258  for (int j = 0; j < bits; j++) {
+
259 
+
260  // Skip bit if reserved for parity bit //
+
261  if ((j & (j - 1)) == 0) { // Check if j is a power of two or 0
+
262  skipped++;
+
263  continue;
+
264  }
+
265 
+
266  // Current overall bit number //
+
267  currentBit = i*messageBits + (j-skipped);
+
268 
+
269  // Current uint8_tacter //
+
270  currentChar = currentBit/(sizeof(uint8_t)*8); // int division
+
271 
+
272  // Value of current bit //
+
273  bool thisBit = getBit(input[i], j);
+
274 
+
275  if (i != len/sizeof(block_t)-1) {
+
276 
+
277  // Populate final decoded uint8_tacter //
+
278  output[currentChar] = modifyCharBit(output[currentChar], currentBit-currentChar*sizeof(uint8_t)*8, thisBit);
+
279  }
+
280 
+
281  else {
+
282  uint8_ts = modifyBit(uint8_ts, j-skipped+(sizeof(block_t)*8-messageBits), thisBit);
+
283  }
+
284 
+
285  }
+
286  }
+
287 
+
288 
+
289  return uint8_ts;
+
290  }
+
291 
+
292  int multipleXor(int *indicies, int len) {
+
293  int val = indicies[0];
+
294  for (int i = 1; i < len; i++) {
+
295  val = val ^ indicies[i];
+
296  }
+
297  return val;
+
298  }
+
299 
+
300  block_t modifyBit(block_t n, int p, bool b) {
+
301  return ((n & ~(1 << (sizeof(block_t)*8-1-p))) | (b << (sizeof(block_t)*8-1-p)));
+
302  }
+
303 
+
304  uint8_t modifyCharBit(uint8_t n, int p, bool b) {
+
305  return ((n & ~(1 << (sizeof(uint8_t)*8-1-p))) | (b << (sizeof(uint8_t)*8-1-p)));
+
306  }
+
307 
+
308  bool getBit(block_t b, int i) {
+
309  return (b << i) & (int) pow(2, (sizeof(block_t)*8-1));
+
310  }
+
311 
+
312  bool getCharBit(uint8_t b, int i) {
+
313  return (b << i) & (int) pow(2, (sizeof(uint8_t)*8-1));
+
314  }
+
315 
+
316  block_t toggleBit(block_t b, int i) {
+
317  return b ^ (1 << i);
+
318  }
+
319 
+
320  int inList(uint8_t **list, uint8_t *testString, int listLen) {
+
321  for (int i = 0; i < listLen; i++) {
+
322  if (strcmp(list[i], testString) == 0) {
+
323  return i;
+
324  }
+
325  }
+
326  return 0;
+
327  }
+
328 };
+
329 
+
330 } // ns
+
audio_tools::BaseBuffer::readArray
virtual int readArray(T data[], int len)
reads multiple values
Definition: Buffers.h:40
+
audio_tools::Hamming::write
size_t write(uint8_t c) override
To be avoided.
Definition: FECHamming.h:75
+
audio_tools::Hamming::peek
int peek() override
To be avoided.
Definition: FECHamming.h:86
+
audio_tools::Hamming::read
int read() override
To be avoided.
Definition: FECHamming.h:80
+
audio_tools::SingleBuffer< uint8_t >
+
audio_tools::SingleBuffer::data
T * data()
Provides address of actual data.
Definition: Buffers.h:239
+
audio_tools::SingleBuffer::setAvailable
size_t setAvailable(size_t available_size)
Definition: Buffers.h:251
+
audio_tools::SingleBuffer::write
bool write(T sample) override
write add an entry to the buffer
Definition: Buffers.h:181
+
audio_tools::SingleBuffer::availableForWrite
int availableForWrite() override
provides the number of entries that are available to write
Definition: Buffers.h:211
+
audio_tools::SingleBuffer::reset
void reset() override
clears the buffer
Definition: Buffers.h:241
+
audio_tools
Generic Implementation of sound input and output for desktop environments using portaudio.
Definition: AnalogAudio.h:10
+
+ +
+Generated by doxygen 1.9.1 +
+ + diff --git a/_f_e_c_reed_solomon_8h_source.html b/_f_e_c_reed_solomon_8h_source.html index d4f89a90d7..5c2e6737c5 100644 --- a/_f_e_c_reed_solomon_8h_source.html +++ b/_f_e_c_reed_solomon_8h_source.html @@ -77,63 +77,78 @@
6 
7 namespace audio_tools {
8 
-
15 template <int msglen, int ecclen>
-
16 class FECReedSolomon : public Stream {
-
17  public:
-
18 
-
19  FECReedSolomon(Stream &stream){
-
20  p_stream = &stream;
-
21  p_print = &stream;
-
22  }
-
23 
-
24  FECReedSolomon(Print &print){
-
25  p_print = &print;
-
26  }
-
27 
-
28  int availableForWrite() {
-
29  return msglen;
-
30  }
-
31 
-
32  size_t write(const uint8_t* data, size_t len){
-
33  if (p_print==nullptr) return 0;
-
34  for (int j=0;j<len;j++){
-
35  raw.write(data[j]);
-
36  if(raw.availableForWrite()==0){
-
37  rs.Encode(raw.data(), encoded.data());
-
38  p_print->write(encoded.data(), msglen+ecclen);
-
39  raw.reset();
-
40  }
-
41  }
-
42  return len;
-
43  }
-
44 
-
45  int available() {
-
46  return msglen;
-
47  }
-
48 
-
49  size_t readBytes(uint8_t* data, size_t len){
-
50  if (p_stream==nullptr) return 0;
-
51  if (encoded.isEmpty()){
-
52  int read_len = msglen + ecclen;
-
53  p_stream->readBytes(encoded.data(), read_len);
-
54  encoded.setAvailable(read_len);
-
55  }
-
56  return encoded.readArray(data, len);
-
57  }
-
58 
-
59  protected:
-
60  SingleBuffer<uint8_t> raw{msglen};
-
61  SingleBuffer<uint8_t> encoded{msglen+ecclen};
-
62  RS::ReedSolomon<msglen, ecclen> rs;
-
63  Stream* p_stream = nullptr;
-
64  Print* p_print = nullptr;
-
65 
-
66 };
-
67 
-
68 }
- +
18 template <int bytecount, int additional_bytes>
+
19 class FECReedSolomon : public Stream {
+
20  public:
+
21 
+
22  FECReedSolomon(Stream &stream){
+
23  p_stream = &stream;
+
24  p_print = &stream;
+
25  }
+
26 
+
27  FECReedSolomon(Print &print){
+
28  p_print = &print;
+
29  }
+
30 
+
31  int availableForWrite() override {
+
32  return bytecount;
+
33  }
+
34 
+
35  size_t write(const uint8_t* data, size_t len) override {
+
36  if (p_print==nullptr) return 0;
+
37  for (int j=0;j<len;j++){
+
38  raw.write(data[j]);
+
39  if(raw.availableForWrite()==0){
+
40  rs.Encode(raw.data(), encoded.data());
+
41  p_print->write(encoded.data(), bytecount+additional_bytes);
+
42  raw.reset();
+
43  }
+
44  }
+
45  return len;
+
46  }
+
47 
+
48  int available()override {
+
49  return bytecount;
+
50  }
+
51 
+
52  size_t readBytes(uint8_t* data, size_t len) override{
+
53  if (p_stream==nullptr) return 0;
+
54  if (encoded.isEmpty()){
+
55  int read_len = bytecount + additional_bytes;
+
56  p_stream->readBytes(encoded.data(), read_len);
+
57  encoded.setAvailable(read_len);
+
58  }
+
59  return encoded.readArray(data, len);
+
60  }
+
61 
+
63  size_t write(uint8_t c) override {
+
64  return write(&c, 1);
+
65  }
+
66 
+
68  int read() override {
+
69  uint8_t ch;
+
70  size_t len = readBytes(&ch, 1);
+
71  return len == 1 ? ch : -1;
+
72  }
+
74  int peek() override {return -1;}
+
75 
+
76 
+
77  protected:
+
78  SingleBuffer<uint8_t> raw{bytecount};
+
79  SingleBuffer<uint8_t> encoded{bytecount+additional_bytes};
+
80  RS::ReedSolomon<bytecount, additional_bytes> rs;
+
81  Stream* p_stream = nullptr;
+
82  Print* p_print = nullptr;
+
83 
+
84 };
+
85 
+
86 }
+
audio_tools::BaseBuffer::readArray
virtual int readArray(T data[], int len)
reads multiple values
Definition: Buffers.h:40
-
audio_tools::FECReedSolomon
Forward error correction using Reed-Solomon: write is encoding and readBytes does the decoding.
Definition: FECReedSolomon.h:16
+
audio_tools::FECReedSolomon
Forward error correction using Reed-Solomon: write is encoding and readBytes does the decoding.
Definition: FECReedSolomon.h:19
+
audio_tools::FECReedSolomon::write
size_t write(uint8_t c) override
To be avoided.
Definition: FECReedSolomon.h:63
+
audio_tools::FECReedSolomon::peek
int peek() override
To be avoided.
Definition: FECReedSolomon.h:74
+
audio_tools::FECReedSolomon::read
int read() override
To be avoided.
Definition: FECReedSolomon.h:68
audio_tools::Print
Definition: NoArduino.h:58
audio_tools::SingleBuffer< uint8_t >
audio_tools::SingleBuffer::data
T * data()
Provides address of actual data.
Definition: Buffers.h:239
diff --git a/_f_f_t_windows_8h.html b/_f_f_t_windows_8h.html index 0bc7565845..0fd9a346cf 100644 --- a/_f_f_t_windows_8h.html +++ b/_f_f_t_windows_8h.html @@ -96,7 +96,7 @@ class  FlatTop  FlatTop FFT Window function. More...
  -class  Hamming +class  Hamming< bytecount, block_t >  Hamming FFT Window function. More...
  class  Hann diff --git a/_synthesizer_8h_source.html b/_synthesizer_8h_source.html index f1a7a62492..ed53f86670 100644 --- a/_synthesizer_8h_source.html +++ b/_synthesizer_8h_source.html @@ -370,7 +370,7 @@
338 
339 } // namespace
340 
-
audio_tools::ADSRGain
ADSR Envelope: Attack, Decay, Sustain and Release. Attack is the time taken for initial run-up oeffec...
Definition: AudioEffect.h:361
+
audio_tools::ADSRGain
ADSR Envelope: Attack, Decay, Sustain and Release. Attack is the time taken for initial run-up oeffec...
Definition: AudioEffect.h:362
audio_tools::AbstractSynthesizerChannel
Defines the sound generation for one channel. A channel is used to process an indivudual key so that ...
Definition: Synthesizer.h:22
audio_tools::AbstractSynthesizerChannel::keyOff
virtual void keyOff()=0
Provides the key off event to ADSR to stop the sound.
audio_tools::AbstractSynthesizerChannel::begin
virtual void begin(AudioInfo config)
Start the sound generation.
@@ -383,8 +383,8 @@
audio_tools::AudioEffectCommon::size
size_t size()
Provides the actual number of defined effects.
Definition: AudioEffects.h:45
audio_tools::AudioEffectCommon::addEffect
void addEffect(AudioEffect &effect)
Adds an effect object (by reference)
Definition: AudioEffects.h:27
audio_tools::AudioEffectCommon::findEffect
AudioEffect * findEffect(int id)
Finds an effect by id.
Definition: AudioEffects.h:50
-
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:19
-
audio_tools::AudioEffect::setId
void setId(int id)
Allows to identify an effect.
Definition: AudioEffect.h:39
+
audio_tools::AudioEffect
Abstract Base class for Sound Effects.
Definition: AudioEffect.h:20
+
audio_tools::AudioEffect::setId
void setId(int id)
Allows to identify an effect.
Definition: AudioEffect.h:40
audio_tools::DefaultSynthesizerChannel
Default implementation for a Channel. You can provide the Sound Generator as parameter to the effects...
Definition: Synthesizer.h:47
audio_tools::DefaultSynthesizerChannel::DefaultSynthesizerChannel
DefaultSynthesizerChannel()=default
Default constructor.
audio_tools::DefaultSynthesizerChannel::keyOn
virtual void keyOn(int note, float tgt) override
start to play a note - note expects the frequency of the note!
Definition: Synthesizer.h:96
diff --git a/annotated.html b/annotated.html index c77e6b82c4..d59568a84d 100644 --- a/annotated.html +++ b/annotated.html @@ -200,379 +200,380 @@  CCodec2EncoderEncoder for Codec2 - Depends on https://github.com/pschatzmann/arduino-libcodec2  CCodecNOPDummy no implmentation Codec. This is used so that we can initialize some pointers to decoders and encoders to make sure that they do not point to null  CCommonHeader - CConfigEquilizer3BandsConfiguration for 3 Band Equilizer: Set channels,bits_per_sample,sample_rate. Set and update gain_low, gain_medium and gain_high to value between 0 and 1.0 - CContainerMP4Minimum flexible parser for MPEG4 data (which is based on the Quicktime format). Small atoms will be make available via a callback method. The big (audio) content is written to the Print object which was specified in the constructor. Depends on https://github.com/pschatzmann/arduino-libhelix! - CContainerTargetContainerTarget: forwards requests to both the output and the encoder/decoder and sets up the output chain for Containers. We also manage the proper sequence of the output classes - CContainerTargetPrint - CConverter1ChannelConverter for 1 Channel which applies the indicated Filter - CConverterAutoCenterMakes sure that the avg of the signal is set to 0 - CConverterAutoCenterTMakes sure that the avg of the signal is set to 0 - CConverterFillLeftAndRightMake sure that both channels contain any data - CConverterNChannelsConverter for n Channels which applies the indicated Filter - CConverterScalerMultiplies the values with the indicated factor adds the offset and clips at maxValue. To mute use a factor of 0.0! - CConverterStreamBoth the data of the read or write operations will be converted with the help of the indicated converter - CConverterSwitchLeftAndRightSwitches the left and right channel - CConverterToInternalDACFormatSpecial case for internal DAC output, the incomming PCM buffer needs to be converted from signed 16bit to unsigned - CCopyDecoderDummy Decoder which just copies the provided data to the output - CCopyEncoderDummy Encoder which just copies the provided data to the output - CCsvOutputStream Wrapper which can be used to print the values as readable ASCII to the screen to be analyzed in the Serial Plotter The frames are separated by a new line. The channels in one frame are separated by a , - CDebouncerHelper class to debounce user input from a push button - CDecimateProvides reduced the sampling rates - CDecimateTProvides reduced sampling rates - CDecoderBase64DecoderBase64 - Converts a Base64 encoded Stream into the original data stream. Decoding only gives a valid result if we start at a limit of 4 bytes. We therefore use by default a newline to determine a valid start boundary - CDecoderBasicDecoderBasic - supports mime type audio/basic Requires https://github.com/pschatzmann/arduino-libg7xx The content of the "audio/basic" subtype is single channel audio encoded using 8bit ISDN mu-law [PCM] at a sample rate of 8000 Hz - CDecoderFloatDecoderFloat - Converts Stream of floats into 2 byte integers - CDecoderHelixMP3 and AAC Decoder using libhelix: https://github.com/pschatzmann/arduino-libhelix. We dynamically create a MP3 or AAC decoder dependent on the provided audio format - CDecoderL16DecoderL16 - Converts an 16 Bit Stream into 16Bits network byte order - CDecoderL8DecoderL8 - Converts an 8 Bit Stream into 16Bits Most microcontrollers can not output 8 bit data directly. 8 bit data however is very memory efficient and helps if you need to store audio on constrained resources. This decoder translates 8bit data into 16bit data. By default the encoded data is represented as uint8_t, so the values are from 0 to 255 - CDefaultSynthesizerChannelDefault implementation for a Channel. You can provide the Sound Generator as parameter to the effects: e.g. DefaultSynthesizerChannel<AudioEffects<SineWaveGenerator<int16_t>>> *channel = new DefaultSynthesizerChannel<AudioEffects<SineWaveGenerator<int16_t>>>(); - CDelayDelay/Echo AudioEffect. See https://wiki.analog.com/resources/tools-software/sharc-audio-module/baremetal/delay-effect-tutorial Howver the dry value and wet value were replace by the depth parameter - CDelayEffectBaseA Base class for delay based digital effects. Provides the basic methods that are shared amongst Flanger, Delay, Chorus and Phaser - CDistortionDistortion AudioEffect - CDriverPWMBaseBase Class for all PWM drivers - ►CDynamicMemoryStreamMemoryStream which is written and read using the internal RAM. For each write the data is allocated on the heap - CDataNode - CEffectSuiteBaseBase Class for Effects - CEncodedAudioOutputA more natural Print class to process encoded data (aac, wav, mp3...). Just define the output and the decoder and write the encoded data - CEncodedAudioStreamA more natural Stream class to process encoded data (aac, wav, mp3...) which also supports the decoding by calling readBytes() - CEncoderBase64EncoderBase64s - Encodes the input data into a Base64 string. By default each audio frame is followed by a new line, so that we can easily resynchronize the reading of a data stream. The generation of the new line can be configured with the setNewLine() method - CEncoderBasicEncoderBasic - supports mime type audio/basic. The content of the "audio/basic" subtype is single channel audio encoded using 8bit ISDN mu-law [PCM] at a sample rate of 8000 Hz. Requires https://github.com/pschatzmann/arduino-libg7xx - CEncoderFloatEncoderFloats - Encodes 16 bit PCM data stream to floats data - CEncoderL16EncoderL16s - Condenses 16 bit PCM data stream to 8 bits data. Most microcontrollers can not process 8 bit audio data directly. 8 bit data however is very memory efficient and helps if you need to store audio on constrained resources. This encoder translates 16bit data into 8bit data - CEncoderL8EncoderL8s - Condenses 16 bit PCM data stream to 8 bits data. Most microcontrollers can not process 8 bit audio data directly. 8 bit data however is very memory efficient and helps if you need to store audio on constrained resources. This encoder translates 16bit data into 8bit data. By default the encoded data is represented as uint8_t, so the values are from 0 to 255 - CEnvelopeFilterEnvelopeFilter - ►CEquilizer3Bands3 Band Equilizer inspired from https://www.musicdsp.org/en/latest/Filters/236-3-band-equaliser.html - CEQSTATE - CESP3288AudioOutputStream Adapter for ESP8288-Audio AudioOutput - CESPNowStreamESPNow as Arduino Stream - CESPNowStreamConfigConfiguration for ESP-NOW protocol - CExponentialVolumeControlSimple exponentional volume control using the formula pow(2.0, input) - 1.0; - CFadeFade In and Fade out in order to prevent popping sound when the audio is started or stopped. The fade in/out is performed over the length of the buffer - CFadeConverterConverter which does a fade out or fade in - CFadeStreamStream which can be used to manage fade in and fade out. Before you read or write data you need to call setAudioInfo() to provide the bits_per_sample and channels - CFastSineGeneratorSine wave which is based on a fast approximation function - CFaustStreamIntegration into Faust DSP see https://faust.grame.fr/ To generate code from faust, select src and cpp - CFECReedSolomonForward error correction using Reed-Solomon: write is encoding and readBytes does the decoding - CFFTDisplay - CFFTDriverAbstract Class which defines the basic FFT functionality - CFFTDriverCmsisFFTDriver for Cmsis-FFT see https://arm-software.github.io/CMSIS_5/DSP - CFFTDriverESP32FFTDriver for ESP32-FFT https://github.com/pschatzmann/esp32-fft
+ CCompressorCompressor inspired by https://github.com/YetAnotherElectronicsChannel/STM32_DSP_COMPRESSOR/blob/master/code/Src/main.c + CConfigEquilizer3BandsConfiguration for 3 Band Equilizer: Set channels,bits_per_sample,sample_rate. Set and update gain_low, gain_medium and gain_high to value between 0 and 1.0 + CContainerMP4Minimum flexible parser for MPEG4 data (which is based on the Quicktime format). Small atoms will be make available via a callback method. The big (audio) content is written to the Print object which was specified in the constructor. Depends on https://github.com/pschatzmann/arduino-libhelix! + CContainerTargetContainerTarget: forwards requests to both the output and the encoder/decoder and sets up the output chain for Containers. We also manage the proper sequence of the output classes + CContainerTargetPrint + CConverter1ChannelConverter for 1 Channel which applies the indicated Filter + CConverterAutoCenterMakes sure that the avg of the signal is set to 0 + CConverterAutoCenterTMakes sure that the avg of the signal is set to 0 + CConverterFillLeftAndRightMake sure that both channels contain any data + CConverterNChannelsConverter for n Channels which applies the indicated Filter + CConverterScalerMultiplies the values with the indicated factor adds the offset and clips at maxValue. To mute use a factor of 0.0! + CConverterStreamBoth the data of the read or write operations will be converted with the help of the indicated converter + CConverterSwitchLeftAndRightSwitches the left and right channel + CConverterToInternalDACFormatSpecial case for internal DAC output, the incomming PCM buffer needs to be converted from signed 16bit to unsigned + CCopyDecoderDummy Decoder which just copies the provided data to the output + CCopyEncoderDummy Encoder which just copies the provided data to the output + CCsvOutputStream Wrapper which can be used to print the values as readable ASCII to the screen to be analyzed in the Serial Plotter The frames are separated by a new line. The channels in one frame are separated by a , + CDebouncerHelper class to debounce user input from a push button + CDecimateProvides reduced the sampling rates + CDecimateTProvides reduced sampling rates + CDecoderBase64DecoderBase64 - Converts a Base64 encoded Stream into the original data stream. Decoding only gives a valid result if we start at a limit of 4 bytes. We therefore use by default a newline to determine a valid start boundary + CDecoderBasicDecoderBasic - supports mime type audio/basic Requires https://github.com/pschatzmann/arduino-libg7xx The content of the "audio/basic" subtype is single channel audio encoded using 8bit ISDN mu-law [PCM] at a sample rate of 8000 Hz + CDecoderFloatDecoderFloat - Converts Stream of floats into 2 byte integers + CDecoderHelixMP3 and AAC Decoder using libhelix: https://github.com/pschatzmann/arduino-libhelix. We dynamically create a MP3 or AAC decoder dependent on the provided audio format + CDecoderL16DecoderL16 - Converts an 16 Bit Stream into 16Bits network byte order + CDecoderL8DecoderL8 - Converts an 8 Bit Stream into 16Bits Most microcontrollers can not output 8 bit data directly. 8 bit data however is very memory efficient and helps if you need to store audio on constrained resources. This decoder translates 8bit data into 16bit data. By default the encoded data is represented as uint8_t, so the values are from 0 to 255 + CDefaultSynthesizerChannelDefault implementation for a Channel. You can provide the Sound Generator as parameter to the effects: e.g. DefaultSynthesizerChannel<AudioEffects<SineWaveGenerator<int16_t>>> *channel = new DefaultSynthesizerChannel<AudioEffects<SineWaveGenerator<int16_t>>>(); + CDelayDelay/Echo AudioEffect. See https://wiki.analog.com/resources/tools-software/sharc-audio-module/baremetal/delay-effect-tutorial Howver the dry value and wet value were replace by the depth parameter + CDelayEffectBaseA Base class for delay based digital effects. Provides the basic methods that are shared amongst Flanger, Delay, Chorus and Phaser + CDistortionDistortion AudioEffect + CDriverPWMBaseBase Class for all PWM drivers + ►CDynamicMemoryStreamMemoryStream which is written and read using the internal RAM. For each write the data is allocated on the heap + CDataNode + CEffectSuiteBaseBase Class for Effects + CEncodedAudioOutputA more natural Print class to process encoded data (aac, wav, mp3...). Just define the output and the decoder and write the encoded data + CEncodedAudioStreamA more natural Stream class to process encoded data (aac, wav, mp3...) which also supports the decoding by calling readBytes() + CEncoderBase64EncoderBase64s - Encodes the input data into a Base64 string. By default each audio frame is followed by a new line, so that we can easily resynchronize the reading of a data stream. The generation of the new line can be configured with the setNewLine() method + CEncoderBasicEncoderBasic - supports mime type audio/basic. The content of the "audio/basic" subtype is single channel audio encoded using 8bit ISDN mu-law [PCM] at a sample rate of 8000 Hz. Requires https://github.com/pschatzmann/arduino-libg7xx + CEncoderFloatEncoderFloats - Encodes 16 bit PCM data stream to floats data + CEncoderL16EncoderL16s - Condenses 16 bit PCM data stream to 8 bits data. Most microcontrollers can not process 8 bit audio data directly. 8 bit data however is very memory efficient and helps if you need to store audio on constrained resources. This encoder translates 16bit data into 8bit data + CEncoderL8EncoderL8s - Condenses 16 bit PCM data stream to 8 bits data. Most microcontrollers can not process 8 bit audio data directly. 8 bit data however is very memory efficient and helps if you need to store audio on constrained resources. This encoder translates 16bit data into 8bit data. By default the encoded data is represented as uint8_t, so the values are from 0 to 255 + CEnvelopeFilterEnvelopeFilter + ►CEquilizer3Bands3 Band Equilizer inspired from https://www.musicdsp.org/en/latest/Filters/236-3-band-equaliser.html + CEQSTATE + CESP3288AudioOutputStream Adapter for ESP8288-Audio AudioOutput + CESPNowStreamESPNow as Arduino Stream + CESPNowStreamConfigConfiguration for ESP-NOW protocol + CExponentialVolumeControlSimple exponentional volume control using the formula pow(2.0, input) - 1.0; + CFadeFade In and Fade out in order to prevent popping sound when the audio is started or stopped. The fade in/out is performed over the length of the buffer + CFadeConverterConverter which does a fade out or fade in + CFadeStreamStream which can be used to manage fade in and fade out. Before you read or write data you need to call setAudioInfo() to provide the bits_per_sample and channels + CFastSineGeneratorSine wave which is based on a fast approximation function + CFaustStreamIntegration into Faust DSP see https://faust.grame.fr/ To generate code from faust, select src and cpp + CFECReedSolomonForward error correction using Reed-Solomon: write is encoding and readBytes does the decoding + CFFTDisplay + CFFTDriverAbstract Class which defines the basic FFT functionality + CFFTDriverCmsisFFTDriver for Cmsis-FFT see https://arm-software.github.io/CMSIS_5/DSP + CFFTDriverESP32FFTDriver for ESP32-FFT https://github.com/pschatzmann/esp32-fft
- CFFTDriverEspressifFFTFft Driver for espressif dsp library: https://espressif-docs.readthedocs-hosted.com/projects/esp-dsp/en/latest/esp-dsp-apis.html - CFFTDriverKissFFTDriver for RealFFT - CFFTDriverRealFFTDriver for RealFFT - CFileArduino File support using std::fstream - CFileLoopA simple class which implements a automatic looping file. The file needs to be of the class File from FS.h. The number of loops can be defined by calling setLoopCount(). You can also optinally limit the total looping file size by calling setSize(); - CFileLoopTA simple class which implements a automatic looping file. In order to support different file implementation the file class is a template parameter. The number of loops can be defined by calling setLoopCount(). You can also optinally limit the total looping file size by calling setSize(); - CFileOutputSimple layer for Print object to write to a c++ file - CFilterAbstract filter interface definition; - CFilterChainFilterChain - A Cascade of multiple filters - CFilteredDelayDelay effect that filters the repeat delay - CFilteredStreamStream to which we can apply Filters for each channel. The filter might change the result size! - CFilterEffectBaseA Base class for filter based effects including methods for simple high, low and band pass filtering - CFIRFIR Filter Converted from https://github.com/sebnil/FIR-filter-Arduino-Library/tree/master/src You can use https://www.arc.id.au/FilterDesign.html to design the filter - CFLACDecoderDecoder for FLAC. Depends on https://github.com/pschatzmann/arduino-libflac. We support an efficient streaming API and an very memory intensitiv standard interface. So you should prefer the streaming interface where you call setOutput() before the begin and copy() in the loop. Validated with http://www.2l.no/hires/ - CFLACEncoderFLACEncoder - CFlatTopFlatTop FFT Window function - Cfloat16Stores float values with 2 bytes - Cfloat32Stores float values as uint32_t so that we can use memory allocated with MALLOC_CAP_32BIT - CFormatConverterStreamConverter which converts bits_per_sample, channels and the sample_rate. The conversion is supported both on the input and output side - CFSEumlate FS using C++ or Posix functions - CFuzzFuzz AudioEffect - CG711_ALAWDecoder64 kbit/s g711 ALOW Decoder based on https://github.com/pschatzmann/arduino-libg7xx - CG711_ALAWEncoder64 kbit/s g711 ALOW Encoder based on https://github.com/pschatzmann/arduino-libg7xx - CG711_ULAWDecoder64 kbit/s g711 ULOW Decoder based on https://github.com/pschatzmann/arduino-libg7xx - CG711_ULAWEncoder64 kbit/s g711 ULOW Encoder based on https://github.com/pschatzmann/arduino-libg7xx - CG711Decoder64 kbit/s g711 ULOW Decoder based on https://github.com/pschatzmann/arduino-libg7xx Supported decoder parameters: alaw2linear, ulaw2linear - CG711Encoder64 kbit/s g711 ULOW Encoder based on https://github.com/pschatzmann/arduino-libg7xx Supported encoder parameters: linear2alaw2, linear2ulaw - CG721Decoder32Kbps G721 Decoder based on https://github.com/pschatzmann/arduino-libg7xx - CG721Encoder32Kbps G721 Encoder based on https://github.com/pschatzmann/arduino-libg7xx - CG722DecoderDecoder for G.722. Depends on https://github.com/pschatzmann/arduino-libg722 - CG722EncoderEncoder for G.722 - Depends on https://github.com/pschatzmann/arduino-libg722. Inspired by g722enc.c - CG723_24Decoder24Kbps G723 Decoder based on https://github.com/pschatzmann/arduino-libg7xx - CG723_24Encoder24Kbps G723 Encoder based on https://github.com/pschatzmann/arduino-libg7xx - CG723_40Decoder40Kbps G723 Decoder based on https://github.com/pschatzmann/arduino-libg7xx - CG723_40Encoder40Kbps G723 Encoder based on https://github.com/pschatzmann/arduino-libg7xx - CG7xxDecoderG723_24, g721, g723_40 Decoder based on https://github.com/pschatzmann/arduino-libg7xx - CG7xxEncoderG723_24, g721, g723_40 Encoder based on https://github.com/pschatzmann/arduino-libg7xx - CGeneratedSoundStreamSource for reading generated tones. Please note - CGeneratorFixedValueJust returns a constant value - CGeneratorFromArrayWe generate the samples from an array which is provided in the constructor - CGeneratorFromStreamAn Adapter Class which lets you use any Stream as a Generator - CGeneratorMixerMixer which combines multiple sound generators into one output - CGSMDecoderDecoder for GSM. Depends on https://github.com/pschatzmann/arduino-libgsm. Inspired by gsmdec.c - CGSMEncoderEncoder for GSM - Depends on https://github.com/pschatzmann/arduino-libgsm. Inspired by gsmenc.c - CHammingHamming FFT Window function - CHannHann FFT Window function - CHardwareSerial - CHexDumpOutputCreates a Hex Dump - CHLSParserSimple Parser for HLS data. We select the entry with min bandwidth - CHLSStreamHTTP Live Streaming using HLS: The result is a MPEG-TS data stream that must be decoded e.g. with a DecoderMTS - CHttpChunkReaderHttp might reply with chunks. So we need to dechunk the data. see https://en.wikipedia.org/wiki/Chunked_transfer_encoding - CHttpHeaderIn a http request and reply we need to process header information. With this API we can define and query the header information. The individual header lines are stored in a vector. This is the common functionality for the HttpRequest and HttpReplyHeader subclasses - CHttpHeaderLineA individual key - value header line - CHttpLineReaderWe read a single line. A terminating 0 is added to the string to make it compliant for c string functions - CHttpReplyHeaderReading and Writing of Http Replys - CHttpRequestSimple API to process get, put, post, del http requests I tried to use Arduino HttpClient, but I did not manage to extract the mime type from streaming get requests - CHttpRequestHeaderReading and writing of Http Requests - CI2SConfigConfiguration for all common i2s settings - CI2SDriverESP32Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels - ►CI2SDriverESP32V1Basic I2S API for the ESP32 (using the new API). https://docs.espressif.com/projects/esp-idf/en/v5.0.1/esp32/api-reference/peripherals/i2s.html#i2s-communication-mode - CDriverCommon - CDriverI2S - CI2SDriverESP8266Basic I2S API - for the ESP8266 Only 16 bits are supported ! - CI2SDriverNanoBLEBasic I2S API - for the Arduino Nano BLE Sense See https://content.arduino.cc/assets/Nano_BLE_MCU-nRF52840_PS_v1.1.pdf Douplex mode (RXTX_MODE) is currently not supported, but it should be quite easy to implement - CI2SDriverSAMDBasic I2S API - for the SAMD - CI2SDriverSTM32Basic I2S API - for the STM32 Depends on https://github.com/pschatzmann/stm32f411-i2s We just add a write and read buffer and pass some parameters to the STM32 API! - CI2SStreamWe support the Stream interface for the I2S access. In addition we allow a separate mute pin which might also be used to drive a LED.. - CICYStreamIcecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method - CICYStreamBufferedICYStream implementation for the ESP32 based on a FreeRTOS task This is a Icecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method - CICYUrlSetupResolve icy-metaint from HttpRequest and execute metadata callbacks - CID3v1 - CID3v1Enhanced - CID3v2 - CID3v2Frame - CID3v2FrameString - CIIRIIRFilter Converted from https://github.com/tttapa/Filters/blob/master/src/IIRFilter.h - CILBCDecoderDecoder for iLBC. Depends on https://github.com/pschatzmann/libilbc - CILBCEncoderEncoder for iLBC - Depends on https://github.com/pschatzmann/libopenilbc - CIMAState - CInputMergeMerges multiple input channels. The input must be mono! So if you provide 2 mono channels you get a stereo signal as result with the left channel from channel 0 and the right from channel 1 - CInputMixerMixerStream is mixing the input from Multiple Input Streams. All streams must have the same audo format (sample rate, channels, bits per sample) - Cint24_3bytes_t24bit integer which is used for I2S sound processing. The values are really using 3 bytes. It works only on little endian machines! - Cint24_4bytes_t24bit integer which is used for I2S sound processing. The values are represented as int32_t, but only 3 bytes are used. If you assign values which are too big, they are clipped - CJpegOpenCVDisplay image with opencv to be used on the desktop - CJpegTFTDisplay jpeg image using https://github.com/Bodmer/TFT_eSPI and https://github.com/Bodmer/JPEGDecoder - CJupyterAudioTOutput to Jupyter. We write the data just to a file from where we can load the data again for different representations - CLastSampleFaderIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0. Typless implementation - CLastSampleFaderTIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0 - CLC3DecoderDecoder for LC3. Depends on https://github.com/pschatzmann/arduino-liblc3 - CLC3EncoderEncoder for LC3 - Depends on https://github.com/pschatzmann/arduino-liblc3 - CLEDOutputLED output using the FastLED library - CLEDOutputConfig - CLEDOutputUnoR4LED output using the R4 LED matrix library - CLEDOutputUnoR4Config - CLinearVolumeControlThe simplest possible implementation of a VolumeControl: The input = output which describes a linear curve. You would use this implementation if you physically connect an audio pot! - ►CListDouble linked list - CIterator - CNode - CLockA simple RIA locking class for the ESP32 using _lock_t - CLogarithmicVolumeControlParametric Logarithmic volume control. Using the formula pow(b,input) * a - a, where b is b = pow(((1/ym)-1), 2) and a is a = 1.0 / (b - 1.0). The parameter ym is determining the steepness. See https://electronics.stackexchange.com/questions/304692/formula-for-logarithmic-audio-taper-pot - CMaximilianAudioTools integration with Maximilian - CMeasuringStreamClass which measures the truput - ►CMedianFilterAn embedded friendly, fast one-dimensional median filter algorithm implementation in C and C++ Useful for spike and noise removal from analog signals or other DSP Also known as "salt-and-pepper noise" or "impulse noise" filter - CsMedianFilter_t - CsMedianNode_t - CMemoryManagerMemoryManager which activates the use of external SPIRAM memory. When external memory is in use, the allocation strategy is to initially try to satisfy smaller allocation requests with internal memory and larger requests with external memory. This sets the limit between the two, as well as generally enabling allocation in external memory - CMemoryOutputWrites to a preallocated memory - CMemoryStreamA simple Stream implementation which is backed by allocated memory - ►CMetaDataFilterClass which filters out ID3v1 and ID3v2 Metadata and provides only the audio data to the decoder - CID3v2ID3 verion 2 TAG Header (10 bytes) - CMetaDataICYIcecast/Shoutcast Metadata Handling. Metadata class which splits the data into audio and metadata. The result is provided via callback methods. see https://www.codeproject.com/Articles/11308/SHOUTcast-Stream-Ripper - CMetaDataID3Simple ID3 Meta Data Parser which supports ID3 V1 and V2 and implements the Stream interface. You just need to set the callback(s) to receive the result and copy the audio data to this stream - CMetaDataID3BaseID3 Meta Data Common Functionality - CMetaDataID3V1Simple ID3 Meta Data API which supports ID3 V1 - CMetaDataID3V2Simple ID3 Meta Data API which supports ID3 V2: We only support the "TALB", "TOPE", "TIT2", "TCON" tags - CMetaDataOutputID3 and Icecast/Shoutcast metadata output support. Just write the audio data to an object of this class and receive the metadata via the callback - CMiniAudioConfigConfiguration for MiniAudio - CMiniAudioStreamMiniAudio: https://miniaud.io/ - CModulationBaseClassClass provides a wave table that can be populated with a number of preallocated waveforms. These can be used to generate audio in themselves or to modulate The parameters of another effect. Class initialised with sample rate - CMozziConfigMozzi Configuration for input or output stream - CMozziGeneratorSupport for https://sensorium.github.io/Mozzi/ Define your updateControl() method. Define your updateAudio() method. Start by calling begin(control_rate) do not call audioHook(); in the loop ! - CMozziStreamWe use the output functionality of Mozzi to output audio data. We expect the data as array of int16_t with one or two channels. Though we support the setting of a sample rate, we recommend to use the default sample rate from Mozzi which is available with the AUDIO_RATE define - CMP3DecoderHelixMP3 Decoder using libhelix: https://github.com/pschatzmann/arduino-libhelix This is basically just a simple wrapper to provide AudioInfo and AudioInfoSupport - CMP3DecoderMADMP3 Decoder using https://github.com/pschatzmann/arduino-libmad - CMP3DecoderMiniMP3 Decoder using https://github.com/pschatzmann/minimp3. This decoder does not provide any good results and it is not suited to decode any audio above 32000 on an ESP32. So the sample rate is limited by the MINIMP3_MAX_SAMPLE_RATE variable - CMP3EncoderLAMEEncodes PCM data to the MP3 format and writes the result to a stream This is basically just a wrapper using https://github.com/pschatzmann/arduino-liblame - CMP4AtomRepresents a single MPEG4 atom - CMP4ParseBuffer - CMTSDecoderMPEG-TS (MTS) decoder. Extracts the AAC audio data from a MPEG-TS (MTS) data stream. You can define the relevant stream types via the API. Required dependency: https://github.com/pschatzmann/arduino-tsdemux - CMultiConverterCombines multiple converters - CMultiOutputReplicates the output to multiple destinations - CMusicalNotesDetermination of the frequency of a music note - CNano_BLE_freq_infoMapping Frequency constants to available frequencies - CNano_BLE_ratio_infoMapping from Ratio Constants to frequency ratios - CNBufferA lock free N buffer. If count=2 we create a DoubleBuffer, if count=3 a TripleBuffer etc - CNoFilterNo change to the input - CNOPConverterDummy converter which does nothing - CNullStreamThe Arduino Stream which provides silence and simulates a null device when used as audio target or audio source - CNumberConverterConverts from a source to a target number with a different type - CNumberFormatConverterStreamConverter which converts between bits_per_sample and 16 bits. The templated NumberFormatConverterStreamT class is used based on the information provided by the bits_per_sample in the configuration - CNumberFormatConverterStreamTConverter which converts from source bits_per_sample to target bits_per_sample - CNumberReaderReads n numbers from an Arduino Stream - CNuttallNuttall FFT Window function - COggContainerDecoderDecoder for Ogg Container. Decodes a packet from an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass and overwrite the beginOfSegment() method to implement your own headers Dependency: https://github.com/pschatzmann/arduino-libopus - COggContainerEncoderEncoder for Ogg Container. Encodes a packet for an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass ond overwrite the writeHeader() method to implement your own header logic. When an optional encoder is specified in the constructor we package the encoded data. Dependency: https://github.com/pschatzmann/arduino-libopus - COggContainerOutputOutput class for the OggContainerEncoder. Each write is ending up as container entry - COnOffOutputSwitched Output class. Class which can be used to filter the output out based on a switch: If the switch is on the output is forwarded. If it is off it is just ignored - COpusAudioDecoderOpusAudioDecoder: Depends on https://github.com/pschatzmann/arduino-libopus.git - COpusAudioEncoderOpusAudioEncoder: Dependens on https://github.com/pschatzmann/arduino-libopus.git - COpusEncoderSettingsSetting for Opus Encoder where the following values are valid: -1 indicates that the default value should be used and that this codec is not setting the value - COpusOggDecoderOpus Decoder which uses the Ogg Container. See https://datatracker.ietf.org/doc/html/rfc7845. The audio data is transmitted in frames and the header information contains the sampler rate, channels and other critical info. Dependency: https://github.com/pschatzmann/arduino-libopus - COpusOggEncoderOpus Encoder which uses the Ogg Container: see https://datatracker.ietf.org/doc/html/rfc7845 Dependency: https://github.com/pschatzmann/arduino-libopus - COpusOggWriter - COpusSettingsSetting for Opus Decoder - COutputMixerMixing of multiple outputs to one final output - CParameterA constant value - CParseBufferWe try to keep the necessary buffer for parsing as small as possible, The data() method provides the start of the actual data and with consume we remove the processed data from the buffer to make space again - CParseObject - CPinInfoESP32Information for a PIN - CPinkNoiseGeneratorGenerates pink noise - CPitchShiftShifts the pitch by the indicated step size: e.g. 2 doubles the pitch - CPitchShiftInfoConfiguration for PitchShiftOutput: set the pitch_shift to define the shift - CPitchShiftOutputPitch Shift: Shifts the frequency up or down w/o impacting the length! We reduce the channels to 1 to calculate the pitch shift and provides the pitch shifted result in the correct number of channels. The pitch shifting is done with the help of a buffer that can have potentially multiple implementations - CPoppingSoundRemoverBig value gaps (at the beginning and the end of a recording) can lead to some popping sounds. We will try to set the values to 0 until the first transition thru 0 of the audio curve - CPortAudioConfigPortAudio information - CPortAudioStreamArduino Audio Stream using PortAudio - CPrint - CProcessedResult - CProgressStreamGeneric calss to measure the the total bytes which were processed in order to calculate the progress as a percentage of the total size - CProgressStreamInfoConfiguration for ProgressStream - CPWMAudioOutputCommon functionality for PWM output. Please use the PWMAudioOutput typedef instead which references the implementation - CPWMConfigConfiguration data for PWM audio output - CPWMDriverAVRExperimental: Audio output to PWM pins for the AVR. The AVR supports only up to 2 channels - CPWMDriverESP32Audio output to PWM pins for the ESP32. The ESP32 supports up to 16 channels - CPWMDriverMBEDAudio output to PWM pins for MBED based Arduino implementations - CPWMDriverRenesasAudio output to PWM pins for Renesas based Arduino implementations - CPWMDriverSTM32Audio output to PWM pins for STM32. We use one timer to generate the sample rate and one timer for the PWM signal - CQueueFIFO Queue which is based on a List - ►CQueueLockFreeLock Free Queue - Cnode - CQueueStreamAudioStream class which stores the data in a temporary queue buffer. The queue can be consumed e.g. by a callback function by calling readBytes(); - CRECT - CRectangeRectange FFT Window function - CReformatBaseStreamBase class for chained converting streams - CResampleConfigOptional Configuration object. The critical information is the channels and the step_size. All other information is not used - CResampleStreamDynamic Resampling. We can use a variable factor to speed up or slow down the playback - CRingBufferImplements a typed Ringbuffer - CRingBufferFileAn File backed Ring Buffer that we can use to receive streaming audio. We expect an open p_file as parameter - CRingBufferStreamA Stream backed by a Ringbuffer. We can write to the end and read from the beginning of the stream - CRTSPFormatAbtXAbtX format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatAudioToolsRTSPFormat which supports the AudioInfo class - CRTSPFormatG711G711 μ-Law format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats Packet intervall: 20, frame size: any - CRTSPFormatGSMGSM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatOpusOpus format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatPCMPCM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatPCM8L8 format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPOutputWe can write PCM data to the RTSPOutput. This is encoded by the indicated encoder (e.g. SBCEncoder) and can be consumed by a RTSPServer. You have to make sure that the codec supports the provided audio format: e.g. GSM support only 8000 samples per second with one channel. Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPOutputPCMInfoPCMInfo subclass which provides the audio information from the related AudioStream Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPPCMAudioInfoPCMInfo subclass which provides the audio information from the AudioInfo parameter - CRTSPSourceFromAudioStreamSimple Facade which can turn AudioStream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPSourceStreamSimple Facade which can turn any Stream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CSBCDecoderDecoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcdec.c - CSBCEncoderEncoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcenc.c - CScaledParameterScales a dynamic parameter to the indicated range - CSDDirectWe access the files directy with an index. The index is determined by a recurseve tree walk thru the directory. Unfortunatly the SDTFAT library has it's own API which is incompatible with the SDT API - CSDIndexWe store all the relevant file names in an sequential index file. Form there we can access them via an index - CSilenceGeneratorProvides a fixed value (e.g. 0) as sound data. This can be used e.g. to test the output functionality which should optimally just output silence and no artifacts - CSilenceRemovalConverterRemoves any silence from the buffer that is longer then n samples with a amplitude below the indicated threshhold. If you process multiple channels you need to multiply the channels with the number of samples to indicate n - CSimpleChorusSimple Chorus effect with a single delay voice and mono output Chorus is effective between 15 and 20 miliseconds delay of original audio. Requires the sample rate when initialising - CSimpleContainerConfig - CSimpleContainerDataHeader - CSimpleContainerMetaDataHeader - CSimpleDelaySimple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples - CSimpleFlangerSimple Flanger Effect Consistig of a single voice flanger The flanger has an effective range between 0 and 15 miliseconds in this case dleay buffer should be set to sampleRate*3/200 Constructor requires internal delay in samples - CSimpleLPFSimpleLPF - CSimulatedAudioPotSimple simulated audio pot volume control inspired by https://eepower.com/resistor-guide/resistor-types/potentiometer-taper/# We split up the input/output curve into 2 linear pieces with a slow and a fast raising part. The slow raising part goes from (0,0) to (x,y). The fast raising part goes from (x,y) to (1,1) - CSineFromTableA sine generator based on a table. The table is created based using degrees where one full wave is 360 degrees - CSineWaveGeneratorGenerates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead - CSingleBufferA simple Buffer implementation which just uses a (dynamically sized) array - CSliceHelps to split up a big memory array into smaller slices. There are no additinal heap allocations! Example: if we have an array with 9 entries (1,2,3,4,5,6,7,8,9): slices(5) gives 2. slice(5,0) returns size 5 with 1,2,3,4,5 and slice(5,1) returns size 4 with 6,7,8,9! - CSmoothTransitionChanges the samples at the beginning or at the end to slowly ramp up the volume - CSOSFilterSecond Order Filter: Instead of manually cascading BiQuad filters, you can use a Second Order Sections filter (SOS). converted from https://github.com/tttapa/Filters/blob/master/src/SOSFilter.h Use float or double (and not a integer type) as type parameter - CSoundGeneratorBase class to define the abstract interface for the sound generating classes - CSoundGeneratorModulationSoundGenerator using the ModulationBaseClass to generate the samples - CSPDIFConfigSPDIF configuration - CSPDIFOutputOutput as 16 bit stereo SPDIF on the I2S data output pin - CSquareWaveGeneratorGenerates a square wave sound - CStackLIFO Stack which is based on a List - CStdioStream - CSTKChorusChorus Effect - CSTKChowningReverbJohn Chowning's reverberator class - CSTKEchoEcho Effect - CSTKEffectUse any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK - CSTKFreeVerbJezar at Dreampoint's FreeVerb, implemented in STK - CSTKGeneratorThe Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language.
+ CFFTDriverEspressifFFTFft Driver for espressif dsp library: https://espressif-docs.readthedocs-hosted.com/projects/esp-dsp/en/latest/esp-dsp-apis.html + CFFTDriverKissFFTDriver for RealFFT + CFFTDriverRealFFTDriver for RealFFT + CFileArduino File support using std::fstream + CFileLoopA simple class which implements a automatic looping file. The file needs to be of the class File from FS.h. The number of loops can be defined by calling setLoopCount(). You can also optinally limit the total looping file size by calling setSize(); + CFileLoopTA simple class which implements a automatic looping file. In order to support different file implementation the file class is a template parameter. The number of loops can be defined by calling setLoopCount(). You can also optinally limit the total looping file size by calling setSize(); + CFileOutputSimple layer for Print object to write to a c++ file + CFilterAbstract filter interface definition; + CFilterChainFilterChain - A Cascade of multiple filters + CFilteredDelayDelay effect that filters the repeat delay + CFilteredStreamStream to which we can apply Filters for each channel. The filter might change the result size! + CFilterEffectBaseA Base class for filter based effects including methods for simple high, low and band pass filtering + CFIRFIR Filter Converted from https://github.com/sebnil/FIR-filter-Arduino-Library/tree/master/src You can use https://www.arc.id.au/FilterDesign.html to design the filter + CFLACDecoderDecoder for FLAC. Depends on https://github.com/pschatzmann/arduino-libflac. We support an efficient streaming API and an very memory intensitiv standard interface. So you should prefer the streaming interface where you call setOutput() before the begin and copy() in the loop. Validated with http://www.2l.no/hires/ + CFLACEncoderFLACEncoder + CFlatTopFlatTop FFT Window function + Cfloat16Stores float values with 2 bytes + Cfloat32Stores float values as uint32_t so that we can use memory allocated with MALLOC_CAP_32BIT + CFormatConverterStreamConverter which converts bits_per_sample, channels and the sample_rate. The conversion is supported both on the input and output side + CFSEumlate FS using C++ or Posix functions + CFuzzFuzz AudioEffect + CG711_ALAWDecoder64 kbit/s g711 ALOW Decoder based on https://github.com/pschatzmann/arduino-libg7xx + CG711_ALAWEncoder64 kbit/s g711 ALOW Encoder based on https://github.com/pschatzmann/arduino-libg7xx + CG711_ULAWDecoder64 kbit/s g711 ULOW Decoder based on https://github.com/pschatzmann/arduino-libg7xx + CG711_ULAWEncoder64 kbit/s g711 ULOW Encoder based on https://github.com/pschatzmann/arduino-libg7xx + CG711Decoder64 kbit/s g711 ULOW Decoder based on https://github.com/pschatzmann/arduino-libg7xx Supported decoder parameters: alaw2linear, ulaw2linear + CG711Encoder64 kbit/s g711 ULOW Encoder based on https://github.com/pschatzmann/arduino-libg7xx Supported encoder parameters: linear2alaw2, linear2ulaw + CG721Decoder32Kbps G721 Decoder based on https://github.com/pschatzmann/arduino-libg7xx + CG721Encoder32Kbps G721 Encoder based on https://github.com/pschatzmann/arduino-libg7xx + CG722DecoderDecoder for G.722. Depends on https://github.com/pschatzmann/arduino-libg722 + CG722EncoderEncoder for G.722 - Depends on https://github.com/pschatzmann/arduino-libg722. Inspired by g722enc.c + CG723_24Decoder24Kbps G723 Decoder based on https://github.com/pschatzmann/arduino-libg7xx + CG723_24Encoder24Kbps G723 Encoder based on https://github.com/pschatzmann/arduino-libg7xx + CG723_40Decoder40Kbps G723 Decoder based on https://github.com/pschatzmann/arduino-libg7xx + CG723_40Encoder40Kbps G723 Encoder based on https://github.com/pschatzmann/arduino-libg7xx + CG7xxDecoderG723_24, g721, g723_40 Decoder based on https://github.com/pschatzmann/arduino-libg7xx + CG7xxEncoderG723_24, g721, g723_40 Encoder based on https://github.com/pschatzmann/arduino-libg7xx + CGeneratedSoundStreamSource for reading generated tones. Please note + CGeneratorFixedValueJust returns a constant value + CGeneratorFromArrayWe generate the samples from an array which is provided in the constructor + CGeneratorFromStreamAn Adapter Class which lets you use any Stream as a Generator + CGeneratorMixerMixer which combines multiple sound generators into one output + CGSMDecoderDecoder for GSM. Depends on https://github.com/pschatzmann/arduino-libgsm. Inspired by gsmdec.c + CGSMEncoderEncoder for GSM - Depends on https://github.com/pschatzmann/arduino-libgsm. Inspired by gsmenc.c + CHammingHamming FFT Window function + CHannHann FFT Window function + CHardwareSerial + CHexDumpOutputCreates a Hex Dump + CHLSParserSimple Parser for HLS data. We select the entry with min bandwidth + CHLSStreamHTTP Live Streaming using HLS: The result is a MPEG-TS data stream that must be decoded e.g. with a DecoderMTS + CHttpChunkReaderHttp might reply with chunks. So we need to dechunk the data. see https://en.wikipedia.org/wiki/Chunked_transfer_encoding + CHttpHeaderIn a http request and reply we need to process header information. With this API we can define and query the header information. The individual header lines are stored in a vector. This is the common functionality for the HttpRequest and HttpReplyHeader subclasses + CHttpHeaderLineA individual key - value header line + CHttpLineReaderWe read a single line. A terminating 0 is added to the string to make it compliant for c string functions + CHttpReplyHeaderReading and Writing of Http Replys + CHttpRequestSimple API to process get, put, post, del http requests I tried to use Arduino HttpClient, but I did not manage to extract the mime type from streaming get requests + CHttpRequestHeaderReading and writing of Http Requests + CI2SConfigConfiguration for all common i2s settings + CI2SDriverESP32Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels + ►CI2SDriverESP32V1Basic I2S API for the ESP32 (using the new API). https://docs.espressif.com/projects/esp-idf/en/v5.0.1/esp32/api-reference/peripherals/i2s.html#i2s-communication-mode + CDriverCommon + CDriverI2S + CI2SDriverESP8266Basic I2S API - for the ESP8266 Only 16 bits are supported ! + CI2SDriverNanoBLEBasic I2S API - for the Arduino Nano BLE Sense See https://content.arduino.cc/assets/Nano_BLE_MCU-nRF52840_PS_v1.1.pdf Douplex mode (RXTX_MODE) is currently not supported, but it should be quite easy to implement + CI2SDriverSAMDBasic I2S API - for the SAMD + CI2SDriverSTM32Basic I2S API - for the STM32 Depends on https://github.com/pschatzmann/stm32f411-i2s We just add a write and read buffer and pass some parameters to the STM32 API! + CI2SStreamWe support the Stream interface for the I2S access. In addition we allow a separate mute pin which might also be used to drive a LED.. + CICYStreamIcecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method + CICYStreamBufferedICYStream implementation for the ESP32 based on a FreeRTOS task This is a Icecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method + CICYUrlSetupResolve icy-metaint from HttpRequest and execute metadata callbacks + CID3v1 + CID3v1Enhanced + CID3v2 + CID3v2Frame + CID3v2FrameString + CIIRIIRFilter Converted from https://github.com/tttapa/Filters/blob/master/src/IIRFilter.h + CILBCDecoderDecoder for iLBC. Depends on https://github.com/pschatzmann/libilbc + CILBCEncoderEncoder for iLBC - Depends on https://github.com/pschatzmann/libopenilbc + CIMAState + CInputMergeMerges multiple input channels. The input must be mono! So if you provide 2 mono channels you get a stereo signal as result with the left channel from channel 0 and the right from channel 1 + CInputMixerMixerStream is mixing the input from Multiple Input Streams. All streams must have the same audo format (sample rate, channels, bits per sample) + Cint24_3bytes_t24bit integer which is used for I2S sound processing. The values are really using 3 bytes. It works only on little endian machines! + Cint24_4bytes_t24bit integer which is used for I2S sound processing. The values are represented as int32_t, but only 3 bytes are used. If you assign values which are too big, they are clipped + CJpegOpenCVDisplay image with opencv to be used on the desktop + CJpegTFTDisplay jpeg image using https://github.com/Bodmer/TFT_eSPI and https://github.com/Bodmer/JPEGDecoder + CJupyterAudioTOutput to Jupyter. We write the data just to a file from where we can load the data again for different representations + CLastSampleFaderIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0. Typless implementation + CLastSampleFaderTIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0 + CLC3DecoderDecoder for LC3. Depends on https://github.com/pschatzmann/arduino-liblc3 + CLC3EncoderEncoder for LC3 - Depends on https://github.com/pschatzmann/arduino-liblc3 + CLEDOutputLED output using the FastLED library + CLEDOutputConfig + CLEDOutputUnoR4LED output using the R4 LED matrix library + CLEDOutputUnoR4Config + CLinearVolumeControlThe simplest possible implementation of a VolumeControl: The input = output which describes a linear curve. You would use this implementation if you physically connect an audio pot! + ►CListDouble linked list + CIterator + CNode + CLockA simple RIA locking class for the ESP32 using _lock_t + CLogarithmicVolumeControlParametric Logarithmic volume control. Using the formula pow(b,input) * a - a, where b is b = pow(((1/ym)-1), 2) and a is a = 1.0 / (b - 1.0). The parameter ym is determining the steepness. See https://electronics.stackexchange.com/questions/304692/formula-for-logarithmic-audio-taper-pot + CMaximilianAudioTools integration with Maximilian + CMeasuringStreamClass which measures the truput + ►CMedianFilterAn embedded friendly, fast one-dimensional median filter algorithm implementation in C and C++ Useful for spike and noise removal from analog signals or other DSP Also known as "salt-and-pepper noise" or "impulse noise" filter + CsMedianFilter_t + CsMedianNode_t + CMemoryManagerMemoryManager which activates the use of external SPIRAM memory. When external memory is in use, the allocation strategy is to initially try to satisfy smaller allocation requests with internal memory and larger requests with external memory. This sets the limit between the two, as well as generally enabling allocation in external memory + CMemoryOutputWrites to a preallocated memory + CMemoryStreamA simple Stream implementation which is backed by allocated memory + ►CMetaDataFilterClass which filters out ID3v1 and ID3v2 Metadata and provides only the audio data to the decoder + CID3v2ID3 verion 2 TAG Header (10 bytes) + CMetaDataICYIcecast/Shoutcast Metadata Handling. Metadata class which splits the data into audio and metadata. The result is provided via callback methods. see https://www.codeproject.com/Articles/11308/SHOUTcast-Stream-Ripper + CMetaDataID3Simple ID3 Meta Data Parser which supports ID3 V1 and V2 and implements the Stream interface. You just need to set the callback(s) to receive the result and copy the audio data to this stream + CMetaDataID3BaseID3 Meta Data Common Functionality + CMetaDataID3V1Simple ID3 Meta Data API which supports ID3 V1 + CMetaDataID3V2Simple ID3 Meta Data API which supports ID3 V2: We only support the "TALB", "TOPE", "TIT2", "TCON" tags + CMetaDataOutputID3 and Icecast/Shoutcast metadata output support. Just write the audio data to an object of this class and receive the metadata via the callback + CMiniAudioConfigConfiguration for MiniAudio + CMiniAudioStreamMiniAudio: https://miniaud.io/ + CModulationBaseClassClass provides a wave table that can be populated with a number of preallocated waveforms. These can be used to generate audio in themselves or to modulate The parameters of another effect. Class initialised with sample rate + CMozziConfigMozzi Configuration for input or output stream + CMozziGeneratorSupport for https://sensorium.github.io/Mozzi/ Define your updateControl() method. Define your updateAudio() method. Start by calling begin(control_rate) do not call audioHook(); in the loop ! + CMozziStreamWe use the output functionality of Mozzi to output audio data. We expect the data as array of int16_t with one or two channels. Though we support the setting of a sample rate, we recommend to use the default sample rate from Mozzi which is available with the AUDIO_RATE define + CMP3DecoderHelixMP3 Decoder using libhelix: https://github.com/pschatzmann/arduino-libhelix This is basically just a simple wrapper to provide AudioInfo and AudioInfoSupport + CMP3DecoderMADMP3 Decoder using https://github.com/pschatzmann/arduino-libmad + CMP3DecoderMiniMP3 Decoder using https://github.com/pschatzmann/minimp3. This decoder does not provide any good results and it is not suited to decode any audio above 32000 on an ESP32. So the sample rate is limited by the MINIMP3_MAX_SAMPLE_RATE variable + CMP3EncoderLAMEEncodes PCM data to the MP3 format and writes the result to a stream This is basically just a wrapper using https://github.com/pschatzmann/arduino-liblame + CMP4AtomRepresents a single MPEG4 atom + CMP4ParseBuffer + CMTSDecoderMPEG-TS (MTS) decoder. Extracts the AAC audio data from a MPEG-TS (MTS) data stream. You can define the relevant stream types via the API. Required dependency: https://github.com/pschatzmann/arduino-tsdemux + CMultiConverterCombines multiple converters + CMultiOutputReplicates the output to multiple destinations + CMusicalNotesDetermination of the frequency of a music note + CNano_BLE_freq_infoMapping Frequency constants to available frequencies + CNano_BLE_ratio_infoMapping from Ratio Constants to frequency ratios + CNBufferA lock free N buffer. If count=2 we create a DoubleBuffer, if count=3 a TripleBuffer etc + CNoFilterNo change to the input + CNOPConverterDummy converter which does nothing + CNullStreamThe Arduino Stream which provides silence and simulates a null device when used as audio target or audio source + CNumberConverterConverts from a source to a target number with a different type + CNumberFormatConverterStreamConverter which converts between bits_per_sample and 16 bits. The templated NumberFormatConverterStreamT class is used based on the information provided by the bits_per_sample in the configuration + CNumberFormatConverterStreamTConverter which converts from source bits_per_sample to target bits_per_sample + CNumberReaderReads n numbers from an Arduino Stream + CNuttallNuttall FFT Window function + COggContainerDecoderDecoder for Ogg Container. Decodes a packet from an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass and overwrite the beginOfSegment() method to implement your own headers Dependency: https://github.com/pschatzmann/arduino-libopus + COggContainerEncoderEncoder for Ogg Container. Encodes a packet for an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass ond overwrite the writeHeader() method to implement your own header logic. When an optional encoder is specified in the constructor we package the encoded data. Dependency: https://github.com/pschatzmann/arduino-libopus + COggContainerOutputOutput class for the OggContainerEncoder. Each write is ending up as container entry + COnOffOutputSwitched Output class. Class which can be used to filter the output out based on a switch: If the switch is on the output is forwarded. If it is off it is just ignored + COpusAudioDecoderOpusAudioDecoder: Depends on https://github.com/pschatzmann/arduino-libopus.git + COpusAudioEncoderOpusAudioEncoder: Dependens on https://github.com/pschatzmann/arduino-libopus.git + COpusEncoderSettingsSetting for Opus Encoder where the following values are valid: -1 indicates that the default value should be used and that this codec is not setting the value + COpusOggDecoderOpus Decoder which uses the Ogg Container. See https://datatracker.ietf.org/doc/html/rfc7845. The audio data is transmitted in frames and the header information contains the sampler rate, channels and other critical info. Dependency: https://github.com/pschatzmann/arduino-libopus + COpusOggEncoderOpus Encoder which uses the Ogg Container: see https://datatracker.ietf.org/doc/html/rfc7845 Dependency: https://github.com/pschatzmann/arduino-libopus + COpusOggWriter + COpusSettingsSetting for Opus Decoder + COutputMixerMixing of multiple outputs to one final output + CParameterA constant value + CParseBufferWe try to keep the necessary buffer for parsing as small as possible, The data() method provides the start of the actual data and with consume we remove the processed data from the buffer to make space again + CParseObject + CPinInfoESP32Information for a PIN + CPinkNoiseGeneratorGenerates pink noise + CPitchShiftShifts the pitch by the indicated step size: e.g. 2 doubles the pitch + CPitchShiftInfoConfiguration for PitchShiftOutput: set the pitch_shift to define the shift + CPitchShiftOutputPitch Shift: Shifts the frequency up or down w/o impacting the length! We reduce the channels to 1 to calculate the pitch shift and provides the pitch shifted result in the correct number of channels. The pitch shifting is done with the help of a buffer that can have potentially multiple implementations + CPoppingSoundRemoverBig value gaps (at the beginning and the end of a recording) can lead to some popping sounds. We will try to set the values to 0 until the first transition thru 0 of the audio curve + CPortAudioConfigPortAudio information + CPortAudioStreamArduino Audio Stream using PortAudio + CPrint + CProcessedResult + CProgressStreamGeneric calss to measure the the total bytes which were processed in order to calculate the progress as a percentage of the total size + CProgressStreamInfoConfiguration for ProgressStream + CPWMAudioOutputCommon functionality for PWM output. Please use the PWMAudioOutput typedef instead which references the implementation + CPWMConfigConfiguration data for PWM audio output + CPWMDriverAVRExperimental: Audio output to PWM pins for the AVR. The AVR supports only up to 2 channels + CPWMDriverESP32Audio output to PWM pins for the ESP32. The ESP32 supports up to 16 channels + CPWMDriverMBEDAudio output to PWM pins for MBED based Arduino implementations + CPWMDriverRenesasAudio output to PWM pins for Renesas based Arduino implementations + CPWMDriverSTM32Audio output to PWM pins for STM32. We use one timer to generate the sample rate and one timer for the PWM signal + CQueueFIFO Queue which is based on a List + ►CQueueLockFreeLock Free Queue + Cnode + CQueueStreamAudioStream class which stores the data in a temporary queue buffer. The queue can be consumed e.g. by a callback function by calling readBytes(); + CRECT + CRectangeRectange FFT Window function + CReformatBaseStreamBase class for chained converting streams + CResampleConfigOptional Configuration object. The critical information is the channels and the step_size. All other information is not used + CResampleStreamDynamic Resampling. We can use a variable factor to speed up or slow down the playback + CRingBufferImplements a typed Ringbuffer + CRingBufferFileAn File backed Ring Buffer that we can use to receive streaming audio. We expect an open p_file as parameter + CRingBufferStreamA Stream backed by a Ringbuffer. We can write to the end and read from the beginning of the stream + CRTSPFormatAbtXAbtX format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatAudioToolsRTSPFormat which supports the AudioInfo class + CRTSPFormatG711G711 μ-Law format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats Packet intervall: 20, frame size: any + CRTSPFormatGSMGSM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatOpusOpus format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatPCMPCM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatPCM8L8 format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPOutputWe can write PCM data to the RTSPOutput. This is encoded by the indicated encoder (e.g. SBCEncoder) and can be consumed by a RTSPServer. You have to make sure that the codec supports the provided audio format: e.g. GSM support only 8000 samples per second with one channel. Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPOutputPCMInfoPCMInfo subclass which provides the audio information from the related AudioStream Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPPCMAudioInfoPCMInfo subclass which provides the audio information from the AudioInfo parameter + CRTSPSourceFromAudioStreamSimple Facade which can turn AudioStream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPSourceStreamSimple Facade which can turn any Stream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CSBCDecoderDecoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcdec.c + CSBCEncoderEncoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcenc.c + CScaledParameterScales a dynamic parameter to the indicated range + CSDDirectWe access the files directy with an index. The index is determined by a recurseve tree walk thru the directory. Unfortunatly the SDTFAT library has it's own API which is incompatible with the SDT API + CSDIndexWe store all the relevant file names in an sequential index file. Form there we can access them via an index + CSilenceGeneratorProvides a fixed value (e.g. 0) as sound data. This can be used e.g. to test the output functionality which should optimally just output silence and no artifacts + CSilenceRemovalConverterRemoves any silence from the buffer that is longer then n samples with a amplitude below the indicated threshhold. If you process multiple channels you need to multiply the channels with the number of samples to indicate n + CSimpleChorusSimple Chorus effect with a single delay voice and mono output Chorus is effective between 15 and 20 miliseconds delay of original audio. Requires the sample rate when initialising + CSimpleContainerConfig + CSimpleContainerDataHeader + CSimpleContainerMetaDataHeader + CSimpleDelaySimple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples + CSimpleFlangerSimple Flanger Effect Consistig of a single voice flanger The flanger has an effective range between 0 and 15 miliseconds in this case dleay buffer should be set to sampleRate*3/200 Constructor requires internal delay in samples + CSimpleLPFSimpleLPF + CSimulatedAudioPotSimple simulated audio pot volume control inspired by https://eepower.com/resistor-guide/resistor-types/potentiometer-taper/# We split up the input/output curve into 2 linear pieces with a slow and a fast raising part. The slow raising part goes from (0,0) to (x,y). The fast raising part goes from (x,y) to (1,1) + CSineFromTableA sine generator based on a table. The table is created based using degrees where one full wave is 360 degrees + CSineWaveGeneratorGenerates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead + CSingleBufferA simple Buffer implementation which just uses a (dynamically sized) array + CSliceHelps to split up a big memory array into smaller slices. There are no additinal heap allocations! Example: if we have an array with 9 entries (1,2,3,4,5,6,7,8,9): slices(5) gives 2. slice(5,0) returns size 5 with 1,2,3,4,5 and slice(5,1) returns size 4 with 6,7,8,9! + CSmoothTransitionChanges the samples at the beginning or at the end to slowly ramp up the volume + CSOSFilterSecond Order Filter: Instead of manually cascading BiQuad filters, you can use a Second Order Sections filter (SOS). converted from https://github.com/tttapa/Filters/blob/master/src/SOSFilter.h Use float or double (and not a integer type) as type parameter + CSoundGeneratorBase class to define the abstract interface for the sound generating classes + CSoundGeneratorModulationSoundGenerator using the ModulationBaseClass to generate the samples + CSPDIFConfigSPDIF configuration + CSPDIFOutputOutput as 16 bit stereo SPDIF on the I2S data output pin + CSquareWaveGeneratorGenerates a square wave sound + CStackLIFO Stack which is based on a List + CStdioStream + CSTKChorusChorus Effect + CSTKChowningReverbJohn Chowning's reverberator class + CSTKEchoEcho Effect + CSTKEffectUse any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK + CSTKFreeVerbJezar at Dreampoint's FreeVerb, implemented in STK + CSTKGeneratorThe Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language.
- CSTKLentPitShiftPitch shifter effect class based on the Lent algorithm - CSTKNReverbCCRMA's NRev reverberator class - CSTKPerryReverbPerry's simple reverberator class - CSTKPitShiftSimple Pitch shifter effect class: This class implements a simple pitch shifter using a delay line - CSTKStreamSTK Stream for Instrument or Voicer - CStrA simple wrapper to provide string functions on char*. If the underlying char* is a const we do not allow any updates; The ownership of the chr* must be managed externally! - CStream - CStreamCopyWe provide the typeless StreamCopy as a subclass of StreamCopyT - CStreamCopyTTyped Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples - CStreamingDecoderA Streaming Decoder where we provide both the input and output as streams - CStrExtStr which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting - ►CSynthesizerA simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels - CKeyParameter - CSynthesizerKeyArduino GPIO pin to note assossiation - CTestGeneratorGenerates a test signal which is easy to check because the values are incremented or decremented by 1 - CTfLiteAbstractRecognizeCommandsBase class for implementing different primitive decoding models on top of the instantaneous results from running an audio recognition model on a single window of samples - CTfLiteAudioErrorReporterError Reporter using the Audio Tools Logger - CTfLiteAudioStreamTfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data) - CTfLiteAudioStreamBaseAstract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers - CTfLiteConfigConfiguration settings for TfLiteAudioStream - CTfLiteMicroSpeachWriterTfLiteMicroSpeachWriter for Audio Data - ►CTfLiteMicroSpeechRecognizeCommandsThis class is designed to apply a very primitive decoding model on top of the instantaneous results from running an audio recognition model on a single window of samples. It applies smoothing over time so that noisy individual label scores are averaged, increasing the confidence that apparent matches are real. To use it, you should create a class object with the configuration you want, and then feed results from running a TensorFlow model into the processing method. The timestamp for each subsequent call should be increasing from the previous, since the class is designed to process a stream of data over time - CResult - CTfLiteQuantizerQuantizer that helps to quantize and dequantize between float and int8 - CTfLiteReaderInput class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce - CTfLiteSineReaderGenerate a sine output from a model that was trained on the sine method. (=hello_world) - CTfLiteWriterOutput class which interprets audio data if TfLiteAudioStream is treated as audio sink - CThrottleThrottle the sending of the audio data to limit it to the indicated sample rate - CThrottleConfigConfigure Throttle setting - CTimedStreamAudioStream class that can define a start and (an optional) stop time Usually it is used to wrap an Audio Sink (e.g. I2SStream), but wrapping an Audio Source is supported as well. Only wrap classes which represent PCM data! - CTimerAlarmRepeatingCommon Interface definition for TimerAlarmRepeating - CTimerAlarmRepeatingDriverAVRRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverBase - CTimerAlarmRepeatingDriverESP8266Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverMBEDRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverRenesasRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating. By default we use a new GPT timer. You can also request 1 AGT timer by calling setTimer(1); - CTimerAlarmRepeatingDriverSTM32STM32 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerCallbackAudioStreamCallback driven Audio Source (rx_tx_mode==RX_MODE) or Audio Sink (rx_tx_mode==TX_MODE). This class allows to to integrate external libraries in order to consume or generate a data stream which is based on a timer - CTimerCallbackAudioStreamInfoTimerCallbackAudioStream Configuration - ►CTransformationReaderConverterStream Helper class which implements the converting readBytes with the help of write - CAdapterPrintToArray - CTremoloTremolo AudioEffect - CTriangleTriangle FFT Window function - CUDPStream - CUrlURL parser which breaks a full url string up into its individual parts - CURLHistory - CURLLoader - CURLStreamRepresents the content of a URL as Stream. We use the WiFi.h API - CURLStreamBufferedURLStream implementation for the ESP32 based on a separate FreeRTOS task - CVariableSpeedRingBufferOptimized Buffer implementation for Pitch Shift. We try to interpolate the samples and restore the phase when the read pointer and write pointer overtake each other - CVariableSpeedRingBuffer180Varialbe speed ring buffer where we read with 0 and 180 degree and blend the result to prevent overrun artifacts. See https://github.com/YetAnotherElectronicsChannel/STM32_DSP_PitchShift - CVariableSpeedRingBufferSimpleVery Simple Buffer implementation for Pitch Shift. We write in constant speed, but reading can be done in a variable speed. We will hear some noise when the buffer read and write pointers overrun each other - CVBANConfig - CVBANStreamVBAN Audio Source and Sink for the ESP32. For further details please see https://vb-audio.com/Voicemeeter/vban.htm . Inspired by https://github.com/rkinnett/ESP32-VBAN-Audio-Source/tree/master and https://github.com/rkinnett/ESP32-VBAN-Network-Audio-Player - ►CVectorVector implementation which provides the most important methods as defined by std::vector. This class it is quite handy to have and most of the times quite better then dealing with raw c arrays - CiteratorIterator for the Vector class - CVideoAudioBufferedSyncLogic to Synchronize video and audio output: we use a buffer to store the audio and instead of delaying the frames with delay() we play audio. The bufferSize defines the audio buffer in bytes. The correctionMs is used to slow down or speed up the playback of the video to prevent any audio buffer underflows - CVideoAudioSyncLogic to Synchronize video and audio output: This is the minimum implementatin which actually does not synchronize, but directly processes the data. No additinal memory is used! Provide your own optimized platform specific implementation - CVideoOutputAbstract class for video playback. This class is used to assemble a complete video frame in memory - CVolumeControlAbstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal - CVolumeOutputA simple class to determine the volume - CVolumeStreamAdjust the volume of the related input or output: To work properly the class needs to know the bits per sample and number of channels! AudioChanges are forwareded to the related Print or Stream class - CVolumeStreamConfigConfig for VolumeStream - CVorbisDecoderVorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-idec https://github.com/pschatzmann/arduino-libopus.git - CVS1053ConfigConfiguration for VS1053Stream - CVS1053StreamVS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration; - CWAVAudioInfoSound information which is available in the WAV header - CWAVDecoderA simple WAVDecoder: We parse the header data on the first record to determine the format. If no AudioDecoderExt is specified we just write the PCM data to the output that is defined by calling setOutput(). You can define a ADPCM decoder to decode WAV files that contain ADPCM data - CWAVEncoderA simple WAV file encoder. If no AudioEncoderExt is specified the WAV file contains PCM data, otherwise it is encoded as ADPCM. The WAV header is written with the first writing of audio data. Calling begin() is making sure that the header is written again - CWAVFormatX - CWAVHeaderParser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE - CWavIMAAudioInfoSound information which is available in the WAV header - adjusted for IMA ADPCM - CWavIMADecoderObsolete: WavIMADecoder - based on WAVDecoder - We parse the header data as we receive it and send the sound data to the stream which was indicated in the constructor. Only WAV files with WAVE_FORMAT_IMA_ADPCM are supported by this codec! - CWavIMAHeader - CWelchWelch FFT Window function - CWhiteNoiseGeneratorGenerates a random noise sound with the help of rand() function - CWindowFunctionFFT Window Function - CWM8960ConfigConfiguration for WM8960 - CWM8960StreamStream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960 + CSTKLentPitShiftPitch shifter effect class based on the Lent algorithm + CSTKNReverbCCRMA's NRev reverberator class + CSTKPerryReverbPerry's simple reverberator class + CSTKPitShiftSimple Pitch shifter effect class: This class implements a simple pitch shifter using a delay line + CSTKStreamSTK Stream for Instrument or Voicer + CStrA simple wrapper to provide string functions on char*. If the underlying char* is a const we do not allow any updates; The ownership of the chr* must be managed externally! + CStream + CStreamCopyWe provide the typeless StreamCopy as a subclass of StreamCopyT + CStreamCopyTTyped Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples + CStreamingDecoderA Streaming Decoder where we provide both the input and output as streams + CStrExtStr which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting + ►CSynthesizerA simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels + CKeyParameter + CSynthesizerKeyArduino GPIO pin to note assossiation + CTestGeneratorGenerates a test signal which is easy to check because the values are incremented or decremented by 1 + CTfLiteAbstractRecognizeCommandsBase class for implementing different primitive decoding models on top of the instantaneous results from running an audio recognition model on a single window of samples + CTfLiteAudioErrorReporterError Reporter using the Audio Tools Logger + CTfLiteAudioStreamTfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data) + CTfLiteAudioStreamBaseAstract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers + CTfLiteConfigConfiguration settings for TfLiteAudioStream + CTfLiteMicroSpeachWriterTfLiteMicroSpeachWriter for Audio Data + ►CTfLiteMicroSpeechRecognizeCommandsThis class is designed to apply a very primitive decoding model on top of the instantaneous results from running an audio recognition model on a single window of samples. It applies smoothing over time so that noisy individual label scores are averaged, increasing the confidence that apparent matches are real. To use it, you should create a class object with the configuration you want, and then feed results from running a TensorFlow model into the processing method. The timestamp for each subsequent call should be increasing from the previous, since the class is designed to process a stream of data over time + CResult + CTfLiteQuantizerQuantizer that helps to quantize and dequantize between float and int8 + CTfLiteReaderInput class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce + CTfLiteSineReaderGenerate a sine output from a model that was trained on the sine method. (=hello_world) + CTfLiteWriterOutput class which interprets audio data if TfLiteAudioStream is treated as audio sink + CThrottleThrottle the sending of the audio data to limit it to the indicated sample rate + CThrottleConfigConfigure Throttle setting + CTimedStreamAudioStream class that can define a start and (an optional) stop time Usually it is used to wrap an Audio Sink (e.g. I2SStream), but wrapping an Audio Source is supported as well. Only wrap classes which represent PCM data! + CTimerAlarmRepeatingCommon Interface definition for TimerAlarmRepeating + CTimerAlarmRepeatingDriverAVRRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverBase + CTimerAlarmRepeatingDriverESP8266Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverMBEDRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverRenesasRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating. By default we use a new GPT timer. You can also request 1 AGT timer by calling setTimer(1); + CTimerAlarmRepeatingDriverSTM32STM32 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerCallbackAudioStreamCallback driven Audio Source (rx_tx_mode==RX_MODE) or Audio Sink (rx_tx_mode==TX_MODE). This class allows to to integrate external libraries in order to consume or generate a data stream which is based on a timer + CTimerCallbackAudioStreamInfoTimerCallbackAudioStream Configuration + ►CTransformationReaderConverterStream Helper class which implements the converting readBytes with the help of write + CAdapterPrintToArray + CTremoloTremolo AudioEffect + CTriangleTriangle FFT Window function + CUDPStream + CUrlURL parser which breaks a full url string up into its individual parts + CURLHistory + CURLLoader + CURLStreamRepresents the content of a URL as Stream. We use the WiFi.h API + CURLStreamBufferedURLStream implementation for the ESP32 based on a separate FreeRTOS task + CVariableSpeedRingBufferOptimized Buffer implementation for Pitch Shift. We try to interpolate the samples and restore the phase when the read pointer and write pointer overtake each other + CVariableSpeedRingBuffer180Varialbe speed ring buffer where we read with 0 and 180 degree and blend the result to prevent overrun artifacts. See https://github.com/YetAnotherElectronicsChannel/STM32_DSP_PitchShift + CVariableSpeedRingBufferSimpleVery Simple Buffer implementation for Pitch Shift. We write in constant speed, but reading can be done in a variable speed. We will hear some noise when the buffer read and write pointers overrun each other + CVBANConfig + CVBANStreamVBAN Audio Source and Sink for the ESP32. For further details please see https://vb-audio.com/Voicemeeter/vban.htm . Inspired by https://github.com/rkinnett/ESP32-VBAN-Audio-Source/tree/master and https://github.com/rkinnett/ESP32-VBAN-Network-Audio-Player + ►CVectorVector implementation which provides the most important methods as defined by std::vector. This class it is quite handy to have and most of the times quite better then dealing with raw c arrays + CiteratorIterator for the Vector class + CVideoAudioBufferedSyncLogic to Synchronize video and audio output: we use a buffer to store the audio and instead of delaying the frames with delay() we play audio. The bufferSize defines the audio buffer in bytes. The correctionMs is used to slow down or speed up the playback of the video to prevent any audio buffer underflows + CVideoAudioSyncLogic to Synchronize video and audio output: This is the minimum implementatin which actually does not synchronize, but directly processes the data. No additinal memory is used! Provide your own optimized platform specific implementation + CVideoOutputAbstract class for video playback. This class is used to assemble a complete video frame in memory + CVolumeControlAbstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal + CVolumeOutputA simple class to determine the volume + CVolumeStreamAdjust the volume of the related input or output: To work properly the class needs to know the bits per sample and number of channels! AudioChanges are forwareded to the related Print or Stream class + CVolumeStreamConfigConfig for VolumeStream + CVorbisDecoderVorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-idec https://github.com/pschatzmann/arduino-libopus.git + CVS1053ConfigConfiguration for VS1053Stream + CVS1053StreamVS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration; + CWAVAudioInfoSound information which is available in the WAV header + CWAVDecoderA simple WAVDecoder: We parse the header data on the first record to determine the format. If no AudioDecoderExt is specified we just write the PCM data to the output that is defined by calling setOutput(). You can define a ADPCM decoder to decode WAV files that contain ADPCM data + CWAVEncoderA simple WAV file encoder. If no AudioEncoderExt is specified the WAV file contains PCM data, otherwise it is encoded as ADPCM. The WAV header is written with the first writing of audio data. Calling begin() is making sure that the header is written again + CWAVFormatX + CWAVHeaderParser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE + CWavIMAAudioInfoSound information which is available in the WAV header - adjusted for IMA ADPCM + CWavIMADecoderObsolete: WavIMADecoder - based on WAVDecoder - We parse the header data as we receive it and send the sound data to the stream which was indicated in the constructor. Only WAV files with WAVE_FORMAT_IMA_ADPCM are supported by this codec! + CWavIMAHeader + CWelchWelch FFT Window function + CWhiteNoiseGeneratorGenerates a random noise sound with the help of rand() function + CWindowFunctionFFT Window Function + CWM8960ConfigConfiguration for WM8960 + CWM8960StreamStream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960  ►Naudiotools_rtsp  COurRTSPClient  COurSink diff --git a/classaudio__tools_1_1_audio_effect.html b/classaudio__tools_1_1_audio_effect.html index 58c7dc1ea5..1ad9b4cc69 100644 --- a/classaudio__tools_1_1_audio_effect.html +++ b/classaudio__tools_1_1_audio_effect.html @@ -87,21 +87,22 @@ ADSRGain Boost -Delay -Distortion -EffectSuiteBase -Fuzz -PitchShift -STKChorus -STKChowningReverb -STKEcho -STKEffect -STKFreeVerb -STKLentPitShift -STKNReverb -STKPerryReverb -STKPitShift -Tremolo +Compressor +Delay +Distortion +EffectSuiteBase +Fuzz +PitchShift +STKChorus +STKChowningReverb +STKEcho +STKEffect +STKFreeVerb +STKLentPitShift +STKNReverb +STKPerryReverb +STKPitShift +Tremolo diff --git a/classaudio__tools_1_1_audio_effect.png b/classaudio__tools_1_1_audio_effect.png index 49c8359131..bd9ca99e1a 100644 Binary files a/classaudio__tools_1_1_audio_effect.png and b/classaudio__tools_1_1_audio_effect.png differ diff --git a/classaudio__tools_1_1_compressor-members.html b/classaudio__tools_1_1_compressor-members.html new file mode 100644 index 0000000000..8313b819a1 --- /dev/null +++ b/classaudio__tools_1_1_compressor-members.html @@ -0,0 +1,108 @@ + + + + + + + +arduino-audio-tools: Member List + + + + + + + + + +
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Compressor Member List
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This is the complete list of members for Compressor, including all inherited members.

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active()AudioEffectinlinevirtual
active_flag (defined in AudioEffect)AudioEffectprotected
attack_count (defined in Compressor)Compressorprotected
AudioEffect()=default (defined in AudioEffect)AudioEffect
clip(int32_t in, int16_t clipLimit=32767, int16_t resultLimit=32767)AudioEffectinlineprotected
CompStates enum name (defined in Compressor)Compressorprotected
copyParent(AudioEffect *copy) (defined in AudioEffect)AudioEffectinlineprotected
gain (defined in Compressor)Compressorprotected
gain_step_attack (defined in Compressor)Compressorprotected
gain_step_release (defined in Compressor)Compressorprotected
gainreduce (defined in Compressor)Compressorprotected
hold_count (defined in Compressor)Compressorprotected
id()AudioEffectinline
id_value (defined in AudioEffect)AudioEffectprotected
recalculate() (defined in Compressor)Compressorinlineprotected
release_count (defined in Compressor)Compressorprotected
S_Attack enum value (defined in Compressor)Compressorprotected
S_GainReduction enum value (defined in Compressor)Compressorprotected
S_NoOperation enum value (defined in Compressor)Compressorprotected
S_Release enum value (defined in Compressor)Compressorprotected
sample_rate (defined in Compressor)Compressorprotected
setActive(bool value)AudioEffectinlinevirtual
setId(int id)AudioEffectinline
State (defined in Compressor)Compressorprotected
threshold (defined in Compressor)Compressorprotected
timeout (defined in Compressor)Compressorprotected
~AudioEffect()=default (defined in AudioEffect)AudioEffectvirtual
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+ + + + + + + + +
+
+ + +
+ +
+ + +
+
+
+Public Member Functions | +Protected Types | +Protected Member Functions | +Protected Attributes | +List of all members
+
+
Compressor Class Reference
Arduino Audio Tools » DSP » Effects
+
+
+ +

Compressor inspired by https://github.com/YetAnotherElectronicsChannel/STM32_DSP_COMPRESSOR/blob/master/code/Src/main.c. + More...

+ +

#include <AudioEffect.h>

+
+Inheritance diagram for Compressor:
+
+
+ + +AudioEffect + +
+ + + + + + + + + + + + + + +

+Public Member Functions

+virtual bool active ()
 determines if the effect is active
 
+int id ()
 Allows to identify an effect.
 
+virtual void setActive (bool value)
 sets the effect active/inactive
 
+void setId (int id)
 Allows to identify an effect.
 
+ + + +

+Protected Types

enum  CompStates { S_NoOperation +, S_Attack +, S_GainReduction +, S_Release + }
 
+ + + + + + + + +

+Protected Member Functions

+int16_t clip (int32_t in, int16_t clipLimit=32767, int16_t resultLimit=32767)
 generic clipping method
 
+void copyParent (AudioEffect *copy)
 
+void recalculate ()
 
+ + + + + + + + + + + + + + + + + + + + + + + + + + + +

+Protected Attributes

+bool active_flag = true
 
+int32_t attack_count
 
+float gain
 
+float gain_step_attack
 
+float gain_step_release
 
+float gainreduce
 
+int32_t hold_count
 
+int id_value = -1
 
+int32_t release_count
 
+uint32_t sample_rate
 
+enum CompStates State = S_NoOperation
 
+float threshold
 
+int32_t timeout
 
+

Detailed Description

+

Compressor inspired by https://github.com/YetAnotherElectronicsChannel/STM32_DSP_COMPRESSOR/blob/master/code/Src/main.c.

+
Author
Phil Schatzmann
+
Copyright
GPLv3
+
The documentation for this class was generated from the following file: +
+ +
+Generated by doxygen 1.9.1 +
+ + diff --git a/classaudio__tools_1_1_compressor.png b/classaudio__tools_1_1_compressor.png new file mode 100644 index 0000000000..0737c18b2f Binary files /dev/null and b/classaudio__tools_1_1_compressor.png differ diff --git a/classaudio__tools_1_1_f_e_c_reed_solomon-members.html b/classaudio__tools_1_1_f_e_c_reed_solomon-members.html index e73fab0cdf..f84b1c396d 100644 --- a/classaudio__tools_1_1_f_e_c_reed_solomon-members.html +++ b/classaudio__tools_1_1_f_e_c_reed_solomon-members.html @@ -66,26 +66,29 @@
-
FECReedSolomon< msglen, ecclen > Member List
+
FECReedSolomon< bytecount, additional_bytes > Member List
-

This is the complete list of members for FECReedSolomon< msglen, ecclen >, including all inherited members.

+

This is the complete list of members for FECReedSolomon< bytecount, additional_bytes >, including all inherited members.

- - - - - + + + + + - - - - - - + + + + + + + + +
_timeout (defined in Print)Printprotected
available() (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >inlinevirtual
availableForWrite() (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >inlinevirtual
encoded (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >protected
FECReedSolomon(Stream &stream) (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >inline
FECReedSolomon(Print &print) (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >inline
available() override (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >inlinevirtual
availableForWrite() override (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >inlinevirtual
encoded (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >protected
FECReedSolomon(Stream &stream) (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >inline
FECReedSolomon(Print &print) (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >inline
flush() (defined in Print)Printinlinevirtual
operator bool() (defined in Stream)Streaminline
p_print (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >protected
p_stream (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >protected
raw (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >protected
readBytes(uint8_t *data, size_t len) (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >inlinevirtual
rs (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >protected
write(const uint8_t *data, size_t len) (defined in FECReedSolomon< msglen, ecclen >)FECReedSolomon< msglen, ecclen >inlinevirtual
p_print (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >protected
p_stream (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >protected
peek() overrideFECReedSolomon< bytecount, additional_bytes >inline
raw (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >protected
read() overrideFECReedSolomon< bytecount, additional_bytes >inline
readBytes(uint8_t *data, size_t len) override (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >inlinevirtual
rs (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >protected
write(const uint8_t *data, size_t len) override (defined in FECReedSolomon< bytecount, additional_bytes >)FECReedSolomon< bytecount, additional_bytes >inlinevirtual
write(uint8_t c) overrideFECReedSolomon< bytecount, additional_bytes >inline
diff --git a/classaudio__tools_1_1_f_e_c_reed_solomon.html b/classaudio__tools_1_1_f_e_c_reed_solomon.html index f4576bedff..580ccb024b 100644 --- a/classaudio__tools_1_1_f_e_c_reed_solomon.html +++ b/classaudio__tools_1_1_f_e_c_reed_solomon.html @@ -5,7 +5,7 @@ -arduino-audio-tools: FECReedSolomon< msglen, ecclen > Class Template Reference +arduino-audio-tools: FECReedSolomon< bytecount, additional_bytes > Class Template Reference @@ -70,7 +70,7 @@ Protected Attributes | List of all members
-
FECReedSolomon< msglen, ecclen > Class Template Reference
+
FECReedSolomon< bytecount, additional_bytes > Class Template Reference
@@ -79,13 +79,13 @@

#include <FECReedSolomon.h>

-Inheritance diagram for FECReedSolomon< msglen, ecclen >:
+Inheritance diagram for FECReedSolomon< bytecount, additional_bytes >:
- - -Stream -Print + + +Stream +Print
@@ -97,24 +97,36 @@ - - - - + + + + - - - - + + + + + + + + + + + + +
 FECReedSolomon (Stream &stream)
 
-int available ()
 
-int availableForWrite ()
 
+int available () override
 
+int availableForWrite () override
 
virtual void flush ()
 
 operator bool ()
 
-size_t readBytes (uint8_t *data, size_t len)
 
-size_t write (const uint8_t *data, size_t len)
 
+int peek () override
 To be avoided.
 
+int read () override
 To be avoided.
 
+size_t readBytes (uint8_t *data, size_t len) override
 
+size_t write (const uint8_t *data, size_t len) override
 
+size_t write (uint8_t c) override
 To be avoided.
 
@@ -122,7 +134,7 @@ int  +SingleBuffer< uint8_t >  @@ -131,15 +143,15 @@ Stream *  +SingleBuffer< uint8_t >  - - + +

Protected Attributes

_timeout = 10
 
-SingleBuffer< uint8_t > encoded {msglen+ecclen}
encoded {bytecount+additional_bytes}
 
Print * p_print = nullptr
p_stream = nullptr
 
-SingleBuffer< uint8_t > raw {msglen}
raw {bytecount}
 
-RS::ReedSolomon< msglen, ecclen > rs
 
+RS::ReedSolomon< bytecount, additional_bytes > rs
 

Detailed Description

-

template<int msglen, int ecclen>
-class audio_tools::FECReedSolomon< msglen, ecclen >

+

template<int bytecount, int additional_bytes>
+class audio_tools::FECReedSolomon< bytecount, additional_bytes >

Forward error correction using Reed-Solomon: write is encoding and readBytes does the decoding.

Author
Phil Schatzmann
diff --git a/classaudio__tools_1_1_f_e_c_reed_solomon.png b/classaudio__tools_1_1_f_e_c_reed_solomon.png index 4ad5eab21d..88c1216308 100644 Binary files a/classaudio__tools_1_1_f_e_c_reed_solomon.png and b/classaudio__tools_1_1_f_e_c_reed_solomon.png differ diff --git a/classaudio__tools_1_1_hamming-members.html b/classaudio__tools_1_1_hamming-members.html index 3f511543e1..262446ad02 100644 --- a/classaudio__tools_1_1_hamming-members.html +++ b/classaudio__tools_1_1_hamming-members.html @@ -66,25 +66,52 @@
-
Hamming Member List
+
Hamming< bytecount, block_t > Member List
-

This is the complete list of members for Hamming, including all inherited members.

+

This is the complete list of members for Hamming< bytecount, block_t >, including all inherited members.

- - - + + + + + + + + + + + - - - - + + + + + + + + + + + + + + + + + + + + - - + + + + +
begin(int samples)WindowFunctioninlinevirtual
factor(int idx)WindowFunctioninline
factor_internal(int idx) (defined in Hamming)Hamminginlinevirtual
_timeout (defined in Print)Printprotected
available() override (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlinevirtual
availableForWrite() override (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlinevirtual
begin(int samples)WindowFunctioninlinevirtual
decode(block_t input[], int len) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
encode(uint8_t *input, int len) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
encoded (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >protected
encodedSize() (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
factor(int idx)WindowFunctioninline
factor_internal(int idx) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlinevirtual
flush() (defined in Print)Printinlinevirtual
fourPi (defined in WindowFunction)WindowFunctionprotected
Hamming()=default (defined in Hamming)Hamming
i_half_samples (defined in WindowFunction)WindowFunctionprotected
i_samples (defined in WindowFunction)WindowFunctionprotected
ratio(int idx) (defined in WindowFunction)WindowFunctioninlineprotected
getBit(block_t b, int i) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
getBlocks() (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
getCharBit(uint8_t b, int i) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
Hamming()=default (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >
Hamming(Stream &stream) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inline
Hamming(Print &print) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inline
i_half_samples (defined in WindowFunction)WindowFunctionprotected
i_samples (defined in WindowFunction)WindowFunctionprotected
inList(uint8_t **list, uint8_t *testString, int listLen) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
modifyBit(block_t n, int p, bool b) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
modifyCharBit(uint8_t n, int p, bool b) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
multipleXor(int *indicies, int len) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
operator bool() (defined in Stream)Streaminline
p_print (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >protected
p_stream (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >protected
peek() overrideHamming< bytecount, block_t >inline
ratio(int idx) (defined in WindowFunction)WindowFunctioninlineprotected
raw (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >protected
read() overrideHamming< bytecount, block_t >inline
readBytes(uint8_t *data, size_t len) override (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlinevirtual
samples()WindowFunctioninline
samples_minus_1 (defined in WindowFunction)WindowFunctionprotected
sixPi (defined in WindowFunction)WindowFunctionprotected
twoPi (defined in WindowFunction)WindowFunctionprotected
WindowFunction()=default (defined in WindowFunction)WindowFunction
toggleBit(block_t b, int i) (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlineprotected
twoPi (defined in WindowFunction)WindowFunctionprotected
WindowFunction()=default (defined in WindowFunction)WindowFunction
write(const uint8_t *data, size_t len) override (defined in Hamming< bytecount, block_t >)Hamming< bytecount, block_t >inlinevirtual
write(uint8_t c) overrideHamming< bytecount, block_t >inline
diff --git a/classaudio__tools_1_1_hamming.html b/classaudio__tools_1_1_hamming.html index 7aea0aaff4..98438f64f8 100644 --- a/classaudio__tools_1_1_hamming.html +++ b/classaudio__tools_1_1_hamming.html @@ -5,7 +5,7 @@ -arduino-audio-tools: Hamming Class Reference +arduino-audio-tools: Hamming< bytecount, block_t > Class Template Reference @@ -71,7 +71,7 @@ Protected Attributes | List of all members
-
Hamming Class Reference
+
Hamming< bytecount, block_t > Class Template Reference
@@ -80,17 +80,31 @@

#include <FFTWindows.h>

-Inheritance diagram for Hamming:
+Inheritance diagram for Hamming< bytecount, block_t >:
- - -WindowFunction + + +WindowFunction +Stream +Print
+ + + + + + + + @@ -102,19 +116,82 @@ + + + + + + + + + + + + + + + + +

Public Member Functions

+ Hamming (Print &print)
 
+ Hamming (Stream &stream)
 
+int available () override
 
+int availableForWrite () override
 
virtual void begin (int samples)
 Setup the window function providing the fft length.
float factor_internal (int idx)
 
+virtual void flush ()
 
+ operator bool ()
 
+int peek () override
 To be avoided.
 
+int read () override
 To be avoided.
 
+size_t readBytes (uint8_t *data, size_t len) override
 
int samples ()
 Provides the number of samples (fft length)
 
+size_t write (const uint8_t *data, size_t len) override
 
+size_t write (uint8_t c) override
 To be avoided.
 
+ + + + + + + + + + + + + + + + + + + + + +

Protected Member Functions

+bool decode (block_t input[], int len)
 
+int encode (uint8_t *input, int len)
 
+int encodedSize ()
 
+bool getBit (block_t b, int i)
 
+int getBlocks ()
 
+bool getCharBit (uint8_t b, int i)
 
+int inList (uint8_t **list, uint8_t *testString, int listLen)
 
+block_t modifyBit (block_t n, int p, bool b)
 
+uint8_t modifyCharBit (uint8_t n, int p, bool b)
 
+int multipleXor (int *indicies, int len)
 
float ratio (int idx)
 
+block_t toggleBit (block_t b, int i)
 
+ + + + @@ -124,6 +201,15 @@ + + + + + + @@ -135,11 +221,18 @@

Protected Attributes

+int _timeout = 10
 
+SingleBuffer< uint8_t > encoded {encodedSize()}
 
const float fourPi = 12.56637061f
 
int i_samples = 0
 
+Print * p_print = nullptr
 
+Stream * p_stream = nullptr
 
+SingleBuffer< uint8_t > raw {bytecount}
 
float samples_minus_1 = 0.0f
 
 

Detailed Description

-

Hamming FFT Window function.

+

template<int bytecount, class block_t>
+class audio_tools::Hamming< bytecount, block_t >

+ +

Hamming FFT Window function.

Author
Phil Schatzmann
-
Copyright
GPLv3
-
The documentation for this class was generated from the following file:
    +
    Copyright
    GPLv3
    +

    Hamming forware error correction. Inspired by https://github.com/nasserkessas/hamming-codes

    +

    Hamming<1024,uint16_t> hamming; // block_ts of 1k with block_tsize 16bits = 31.25% redundency

    +

    Block size (bits) Redundant bits Redundancy percentage 4 3 75% 8 4 50% 16 5 31.25% 32 6 18.75% 64 7 10.94%

    +
The documentation for this class was generated from the following files: diff --git a/classaudio__tools_1_1_hamming.png b/classaudio__tools_1_1_hamming.png index 61c8716769..5bdc726d75 100644 Binary files a/classaudio__tools_1_1_hamming.png and b/classaudio__tools_1_1_hamming.png differ diff --git a/classaudio__tools_1_1_print.html b/classaudio__tools_1_1_print.html index 5609995831..ab2274da9c 100644 --- a/classaudio__tools_1_1_print.html +++ b/classaudio__tools_1_1_print.html @@ -79,33 +79,34 @@
-AudioOutput -FileOutput -OutputMixer< T > -Stream -AudioFFTBase -AudioOutputAdapter -AudioOutputWithCallback -AudioServerEx -AudioSyncWriter -ChannelSplitOutput< T > -CsvOutput< T > -HexDumpOutput -MemoryOutput -MetaDataOutput -MultiOutput -OggContainerOutput -OnOffOutput -PWMAudioOutput -PitchShiftOutput< T, BufferT > -RTSPOutput -Throttle -VolumeOutput -AudioStream -Client -FECReedSolomon< msglen, ecclen > -File -HardwareSerial +AudioOutput +FileOutput +OutputMixer< T > +Stream +AudioFFTBase +AudioOutputAdapter +AudioOutputWithCallback +AudioServerEx +AudioSyncWriter +ChannelSplitOutput< T > +CsvOutput< T > +HexDumpOutput +MemoryOutput +MetaDataOutput +MultiOutput +OggContainerOutput +OnOffOutput +PWMAudioOutput +PitchShiftOutput< T, BufferT > +RTSPOutput +Throttle +VolumeOutput +AudioStream +Client +FECReedSolomon< bytecount, additional_bytes > +File +Hamming< bytecount, block_t > +HardwareSerial
diff --git a/classaudio__tools_1_1_print.png b/classaudio__tools_1_1_print.png index 960ac59d7b..1506184b98 100644 Binary files a/classaudio__tools_1_1_print.png and b/classaudio__tools_1_1_print.png differ diff --git a/classaudio__tools_1_1_stream.html b/classaudio__tools_1_1_stream.html index 8a55f41979..d2f25b8021 100644 --- a/classaudio__tools_1_1_stream.html +++ b/classaudio__tools_1_1_stream.html @@ -79,63 +79,64 @@
-Print -AudioStream -Client -FECReedSolomon< msglen, ecclen > -File -HardwareSerial -FileLoopT< File > -GeneratedSoundStream< int16_t > -QueueStream< uint8_t > -A2DPStream -AbstractURLStream -AnalogAudioStream -AnalogDriverESP32V1::IO16Bit -AudioEffectStream -AudioEffectStreamT< T > -AudioKitStream -AudioMP34DT05 -AudioStreamWrapper -AudioSyncReader -BufferedStream -BufferedTaskStream -CallbackStream -CatStream -ConverterStream< T > -DynamicMemoryStream -ESP3288AudioOutput -ESPNowStream -EncodedAudioOutput -Equilizer3Bands -FadeStream -FaustStream< DSP > -FileLoopT< FileType > -FilteredStream< T, TF > -GeneratedSoundStream< T > -HLSStream -I2SStream -InputMerge< T > -InputMixer< T > -JupyterAudioT< T > -MeasuringStream -MemoryStream -MiniAudioStream -MozziStream -NullStream -PortAudioStream -ProgressStream -QueueStream< T > -ReformatBaseStream -RingBufferStream -SPDIFOutput -StdioStream -TfLiteAudioStreamBase -TimedStream -VBANStream -VS1053Stream -VolumeStream -WM8960Stream +Print +AudioStream +Client +FECReedSolomon< bytecount, additional_bytes > +File +Hamming< bytecount, block_t > +HardwareSerial +FileLoopT< File > +GeneratedSoundStream< int16_t > +QueueStream< uint8_t > +A2DPStream +AbstractURLStream +AnalogAudioStream +AnalogDriverESP32V1::IO16Bit +AudioEffectStream +AudioEffectStreamT< T > +AudioKitStream +AudioMP34DT05 +AudioStreamWrapper +AudioSyncReader +BufferedStream +BufferedTaskStream +CallbackStream +CatStream +ConverterStream< T > +DynamicMemoryStream +ESP3288AudioOutput +ESPNowStream +EncodedAudioOutput +Equilizer3Bands +FadeStream +FaustStream< DSP > +FileLoopT< FileType > +FilteredStream< T, TF > +GeneratedSoundStream< T > +HLSStream +I2SStream +InputMerge< T > +InputMixer< T > +JupyterAudioT< T > +MeasuringStream +MemoryStream +MiniAudioStream +MozziStream +NullStream +PortAudioStream +ProgressStream +QueueStream< T > +ReformatBaseStream +RingBufferStream +SPDIFOutput +StdioStream +TfLiteAudioStreamBase +TimedStream +VBANStream +VS1053Stream +VolumeStream +WM8960Stream
diff --git a/classaudio__tools_1_1_stream.png b/classaudio__tools_1_1_stream.png index 45e59c2803..34a186f66a 100644 Binary files a/classaudio__tools_1_1_stream.png and b/classaudio__tools_1_1_stream.png differ diff --git a/classaudio__tools_1_1_window_function.html b/classaudio__tools_1_1_window_function.html index 4124df108a..8797dfffec 100644 --- a/classaudio__tools_1_1_window_function.html +++ b/classaudio__tools_1_1_window_function.html @@ -85,17 +85,17 @@
-Blackman -BlackmanHarris -BlackmanNuttall -BufferedWindow -FlatTop -Hamming -Hann -Nuttall -Rectange -Triangle -Welch +Blackman +BlackmanHarris +BlackmanNuttall +BufferedWindow +FlatTop +Hamming< bytecount, block_t > +Hann +Nuttall +Rectange +Triangle +Welch
diff --git a/classaudio__tools_1_1_window_function.png b/classaudio__tools_1_1_window_function.png index ea45e57dca..c711330c8b 100644 Binary files a/classaudio__tools_1_1_window_function.png and b/classaudio__tools_1_1_window_function.png differ diff --git a/classes.html b/classes.html index c30e50329c..70480d8068 100644 --- a/classes.html +++ b/classes.html @@ -75,7 +75,7 @@
BaseBuffer (audio_tools)
BaseConverter (audio_tools)
BinaryContainerDecoder (audio_tools)
BinaryContainerEncoder (audio_tools)
BiQuadDF1 (audio_tools)
BiQuadDF2 (audio_tools)
BitmapInfoHeader (audio_tools)
Blackman (audio_tools)
BlackmanHarris (audio_tools)
BlackmanNuttall (audio_tools)
Boost (audio_tools)
BufferedArray (audio_tools)
BufferedStream (audio_tools)
BufferedTaskStream (audio_tools)
BufferedWindow (audio_tools)
C
-
CachedVolumeControl (audio_tools)
CallbackStream (audio_tools)
CallbackVolumeControl (audio_tools)
CatStream (audio_tools)
ChannelConverter (audio_tools)
ChannelEnhancer (audio_tools)
ChannelFormatConverterStream (audio_tools)
ChannelFormatConverterStreamT (audio_tools)
ChannelReducer (audio_tools)
ChannelReducerT (audio_tools)
ChannelSplitOutput::ChannelSelectionOutputDef (audio_tools)
ChannelSplitOutput (audio_tools)
ChartT (audio_tools)
Client (audio_tools)
Codec2Decoder (audio_tools)
Codec2Encoder (audio_tools)
CodecNOP (audio_tools)
CommonHeader (audio_tools)
ConfigEquilizer3Bands (audio_tools)
ContainerMP4 (audio_tools)
ContainerTarget (audio_tools)
ContainerTargetPrint (audio_tools)
Converter1Channel (audio_tools)
ConverterAutoCenter (audio_tools)
ConverterAutoCenterT (audio_tools)
ConverterFillLeftAndRight (audio_tools)
ConverterNChannels (audio_tools)
ConverterScaler (audio_tools)
ConverterStream (audio_tools)
ConverterSwitchLeftAndRight (audio_tools)
ConverterToInternalDACFormat (audio_tools)
CopyDecoder (audio_tools)
CopyEncoder (audio_tools)
CsvOutput (audio_tools)
+
CachedVolumeControl (audio_tools)
CallbackStream (audio_tools)
CallbackVolumeControl (audio_tools)
CatStream (audio_tools)
ChannelConverter (audio_tools)
ChannelEnhancer (audio_tools)
ChannelFormatConverterStream (audio_tools)
ChannelFormatConverterStreamT (audio_tools)
ChannelReducer (audio_tools)
ChannelReducerT (audio_tools)
ChannelSplitOutput::ChannelSelectionOutputDef (audio_tools)
ChannelSplitOutput (audio_tools)
ChartT (audio_tools)
Client (audio_tools)
Codec2Decoder (audio_tools)
Codec2Encoder (audio_tools)
CodecNOP (audio_tools)
CommonHeader (audio_tools)
Compressor (audio_tools)
ConfigEquilizer3Bands (audio_tools)
ContainerMP4 (audio_tools)
ContainerTarget (audio_tools)
ContainerTargetPrint (audio_tools)
Converter1Channel (audio_tools)
ConverterAutoCenter (audio_tools)
ConverterAutoCenterT (audio_tools)
ConverterFillLeftAndRight (audio_tools)
ConverterNChannels (audio_tools)
ConverterScaler (audio_tools)
ConverterStream (audio_tools)
ConverterSwitchLeftAndRight (audio_tools)
ConverterToInternalDACFormat (audio_tools)
CopyDecoder (audio_tools)
CopyEncoder (audio_tools)
CsvOutput (audio_tools)
D
DynamicMemoryStream::DataNode (audio_tools)
Debouncer (audio_tools)
Decimate (audio_tools)
DecimateT (audio_tools)
DecoderBase64 (audio_tools)
DecoderBasic (audio_tools)
DecoderFloat (audio_tools)
DecoderHelix (audio_tools)
DecoderL16 (audio_tools)
DecoderL8 (audio_tools)
DefaultSynthesizerChannel (audio_tools)
Delay (audio_tools)
DelayEffectBase (audio_tools)
Distortion (audio_tools)
I2SDriverESP32V1::DriverCommon (audio_tools)
I2SDriverESP32V1::DriverI2S (audio_tools)
DriverPWMBase (audio_tools)
dsp
dsp_memory_manager
DynamicMemoryStream (audio_tools)
DynArray (ffft)
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class  Boost
 Boost AudioEffect. More...
 
class  Compressor
 Compressor inspired by https://github.com/YetAnotherElectronicsChannel/STM32_DSP_COMPRESSOR/blob/master/code/Src/main.c. More...
 
class  Delay
 Delay/Echo AudioEffect. See https://wiki.analog.com/resources/tools-software/sharc-audio-module/baremetal/delay-effect-tutorial Howver the dry value and wet value were replace by the depth parameter. More...
 
 ►CAudioEffectAbstract Base class for Sound Effects
 CAudioEffectCommon
 CAudioFFTResultResult of the FFT
 ►CAudioHeaderCommon Header for all records
 CPrintable
 CProcessedResult
 CQueue< T >FIFO Queue which is based on a List
 CQueueLockFree< T >Lock Free Queue
 CRECT
 CReedSolomon< msg_length, ecc_length >
 CReedSolomon< msglen, ecclen >
 CReedSolomon< bytecount, additional_bytes >
 CTfLiteMicroSpeechRecognizeCommands::Result
 ►CRTSPClient
 
struct  CommonHeader
 
class  Compressor
 Compressor inspired by https://github.com/YetAnotherElectronicsChannel/STM32_DSP_COMPRESSOR/blob/master/code/Src/main.c. More...
 
struct  ConfigEquilizer3Bands
 Configuration for 3 Band Equilizer: Set channels,bits_per_sample,sample_rate. Set and update gain_low, gain_medium and gain_high to value between 0 and 1.0. More...
 
 ►Naudiotools_rtsp