diff --git a/_sound_generator_8h_source.html b/_sound_generator_8h_source.html
index 8e02350e4..3d2ab5b0c 100644
--- a/_sound_generator_8h_source.html
+++ b/_sound_generator_8h_source.html
@@ -304,448 +304,478 @@
-
-
-
-
-
+
+
+
+
+
+ 298 LOGD(
"SawToothGenerator" );
+
-
-
-
- 304 int random(
int min,
int max) {
return min + rand() % ((max + 1) - min); }
-
-
-
-
-
- 318 this->amplitude = amplitude;
-
-
- 321 for (
int i = 0; i < 5; i++)
- 322 white_values[i] = rand() % (amplitude / 5);
-
-
-
-
-
-
-
-
-
-
-
- 335 int diff = last_key ^ key;
-
- 337 for (
int i = 0; i < 5; i++) {
-
-
-
- 341 white_values[i] = rand() % (amplitude / 5);
- 342 sum += white_values[i];
-
-
-
-
-
-
-
- 350 unsigned int white_values[5];
- 351 unsigned int amplitude;
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
- 403 setChannels(channels);
-
-
-
-
- 409 void setChannels(
int channels) { this->channels = channels; }
-
-
-
-
- 415 if (p_stream !=
nullptr ) {
- 416 for (
int j = 0; j < channels; j++) {
- 417 p_stream->readBytes((uint8_t *)&data,
sizeof (T));
-
-
- 420 float avg = (total / channels) *
volume ();
- 421 if (avg > maxValue) {
-
- 423 }
else if (avg < -maxValue) {
-
-
-
-
-
-
-
-
-
- 433 Stream *p_stream =
nullptr ;
-
-
-
-
-
-
-
- 462 template <
size_t arrayLen>
-
- 464 bool setInactiveAtEnd =
false ,
size_t startIndex = 0) {
-
- 466 this->max_repeat = repeat;
- 467 this->inactive_at_end = setInactiveAtEnd;
- 468 this->sound_index = startIndex;
- 469 setArray(array, arrayLen);
-
-
-
-
-
-
-
-
- 478 template <
int arrayLen>
void setArray(T (&array)[arrayLen]) {
-
- 480 setArray(array, arrayLen);
-
-
- 483 void setArray(T *array,
size_t size) {
- 484 this->table_length = size;
-
- 486 LOGI(
"table_length: %d" , (
int )size);
-
-
- 489 virtual bool begin (AudioInfo info)
override {
- 490 return SoundGenerator<T>::begin(info);
-
-
-
-
-
-
-
-
-
-
-
-
-
- 506 if (sound_index >= table_length) {
-
-
-
-
- 511 if (max_repeat >= 1 && ++repeat_counter >= max_repeat) {
-
- 513 this->is_running =
false ;
- 514 if (inactive_at_end) {
- 515 this->active =
false ;
-
-
-
-
-
-
- 522 if (this->is_running) {
- 523 result = table[sound_index];
- 524 sound_index += index_increment;
-
-
-
-
-
-
- 531 void setIncrement(
int inc) { index_increment = inc; }
-
-
- 534 int setupSine(
int sampleRate,
float reqFrequency,
float amplitude = 1.0) {
-
- 536 static_cast< float > (sampleRate) /
-
- 538 float angle = 2.0 * PI / sample_count;
- 539 table =
new T[sample_count];
- 540 for (
int j = 0; j < sample_count; j++) {
- 541 table[j] = sinf(j * angle) * amplitude;
-
-
- 544 table_length = sample_count;
-
- 546 return sampleRate / sample_count;
-
-
-
- 550 bool isRunning() {
return is_running; }
-
-
-
-
- 555 int repeat_counter = 0;
- 556 bool inactive_at_end;
- 557 bool is_running =
false ;
- 558 bool owns_data =
false ;
-
- 560 size_t table_length = 0;
- 561 int index_increment = 1;
-
-
-
-
-
-
-
-
- 577 void setValue(T value) { value_set = value; }
-
-
-
-
-
- 584 value_return = value_set;
-
-
-
-
-
-
- 592 bool isRunning() {
return is_running; }
-
-
-
-
- 597 bool is_running =
false ;
-
-
-
-
-
- 610 this->amplitude = amplitude;
- 611 this->amplitude_to_be = amplitude;
-
+
+
+
+
+
+
+
+
+
+
+
+
+
+ 315 inline float saw (
float t) {
+ 316 float p = (t - (int)t) - 0.5f;
+
+
+
+
+
+
+
+
+
+
+
+
+
+ 339 int random(
int min,
int max) {
return min + rand() % ((max + 1) - min); }
+
+
+
+
+
+ 353 this->amplitude = amplitude;
+
+
+ 356 for (
int i = 0; i < 5; i++)
+ 357 white_values[i] = rand() % (amplitude / 5);
+
+
+
+
+
+
+
+
+
+
+
+ 370 int diff = last_key ^ key;
+
+ 372 for (
int i = 0; i < 5; i++) {
+
+
+
+ 376 white_values[i] = rand() % (amplitude / 5);
+ 377 sum += white_values[i];
+
+
+
+
+
+
+
+ 385 unsigned int white_values[5];
+ 386 unsigned int amplitude;
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ 438 setChannels(channels);
+
+
+
+
+ 444 void setChannels(
int channels) { this->channels = channels; }
+
+
+
+
+ 450 if (p_stream !=
nullptr ) {
+ 451 for (
int j = 0; j < channels; j++) {
+ 452 p_stream->readBytes((uint8_t *)&data,
sizeof (T));
+
+
+ 455 float avg = (total / channels) *
volume ();
+ 456 if (avg > maxValue) {
+
+ 458 }
else if (avg < -maxValue) {
+
+
+
+
+
+
+
+
+
+ 468 Stream *p_stream =
nullptr ;
+
+
+
+
+
+
+
+ 497 template <
size_t arrayLen>
+
+ 499 bool setInactiveAtEnd =
false ,
size_t startIndex = 0) {
+
+ 501 this->max_repeat = repeat;
+ 502 this->inactive_at_end = setInactiveAtEnd;
+ 503 this->sound_index = startIndex;
+ 504 setArray(array, arrayLen);
+
+
+
+
+
+
+
+
+ 513 template <
int arrayLen>
void setArray(T (&array)[arrayLen]) {
+
+ 515 setArray(array, arrayLen);
+
+
+ 518 void setArray(T *array,
size_t size) {
+ 519 this->table_length = size;
+
+ 521 LOGI(
"table_length: %d" , (
int )size);
+
+
+ 524 virtual bool begin (AudioInfo info)
override {
+ 525 return SoundGenerator<T>::begin(info);
+
+
+
+
+
+
+
+
+
+
+
+
+
+ 541 if (sound_index >= table_length) {
+
+
+
+
+ 546 if (max_repeat >= 1 && ++repeat_counter >= max_repeat) {
+
+ 548 this->is_running =
false ;
+ 549 if (inactive_at_end) {
+ 550 this->active =
false ;
+
+
+
+
+
+
+ 557 if (this->is_running) {
+ 558 result = table[sound_index];
+ 559 sound_index += index_increment;
+
+
+
+
+
+
+ 566 void setIncrement(
int inc) { index_increment = inc; }
+
+
+ 569 int setupSine(
int sampleRate,
float reqFrequency,
float amplitude = 1.0) {
+
+ 571 static_cast< float > (sampleRate) /
+
+ 573 float angle = 2.0 * PI / sample_count;
+ 574 table =
new T[sample_count];
+ 575 for (
int j = 0; j < sample_count; j++) {
+ 576 table[j] = sinf(j * angle) * amplitude;
+
+
+ 579 table_length = sample_count;
+
+ 581 return sampleRate / sample_count;
+
+
+
+ 585 bool isRunning() {
return is_running; }
+
+
+
+
+ 590 int repeat_counter = 0;
+ 591 bool inactive_at_end;
+ 592 bool is_running =
false ;
+ 593 bool owns_data =
false ;
+
+ 595 size_t table_length = 0;
+ 596 int index_increment = 1;
+
+
+
+
+
+
+
+
+ 612 void setValue(T value) { value_set = value; }
- 615 void setAmplitude (
float amplitude) { this->amplitude_to_be = amplitude; }
-
-
-
-
-
-
- 624 if (angle >= 360.0f) {
- 625 while (angle >= 360.0f) {
-
-
-
-
-
- 631 updateAmplitudeInSteps();
-
-
- 634 return interpolate(angle);
-
-
-
-
-
-
-
-
-
-
- 645 bool begin(AudioInfo info,
float frequency) {
- 646 SoundGenerator<T>::begin(info);
-
-
-
-
-
-
- 653 bool begin(
int channels,
int sample_rate, uint16_t frequency = 0) {
- 654 SoundGenerator<T>::info.channels = channels;
- 655 SoundGenerator<T>::info.sample_rate = sample_rate;
- 656 return begin(SoundGenerator<T>::info, frequency);
-
-
-
- 660 step_new = freq / base_frequency;
-
-
-
-
- 665 LOGD(
"step: %f" , step_new);
-
-
-
- 669 bool is_first =
true ;
-
- 671 float amplitude_to_be;
- 672 float max_amplitude_step = 50.0f;
- 673 float base_frequency = 1.0f;
-
- 675 float step_new = 1.0f;
-
-
- 678 const float values[181] = {
- 679 0, 0.0174524, 0.0348995, 0.052336, 0.0697565, 0.0871557,
- 680 0.104528, 0.121869, 0.139173, 0.156434, 0.173648, 0.190809,
- 681 0.207912, 0.224951, 0.241922, 0.258819, 0.275637, 0.292372,
- 682 0.309017, 0.325568, 0.34202, 0.358368, 0.374607, 0.390731,
- 683 0.406737, 0.422618, 0.438371, 0.45399, 0.469472, 0.48481,
- 684 0.5, 0.515038, 0.529919, 0.544639, 0.559193, 0.573576,
- 685 0.587785, 0.601815, 0.615661, 0.62932, 0.642788, 0.656059,
- 686 0.669131, 0.681998, 0.694658, 0.707107, 0.71934, 0.731354,
- 687 0.743145, 0.75471, 0.766044, 0.777146, 0.788011, 0.798636,
- 688 0.809017, 0.819152, 0.829038, 0.838671, 0.848048, 0.857167,
- 689 0.866025, 0.87462, 0.882948, 0.891007, 0.898794, 0.906308,
- 690 0.913545, 0.920505, 0.927184, 0.93358, 0.939693, 0.945519,
- 691 0.951057, 0.956305, 0.961262, 0.965926, 0.970296, 0.97437,
- 692 0.978148, 0.981627, 0.984808, 0.987688, 0.990268, 0.992546,
- 693 0.994522, 0.996195, 0.997564, 0.99863, 0.999391, 0.999848,
- 694 1, 0.999848, 0.999391, 0.99863, 0.997564, 0.996195,
- 695 0.994522, 0.992546, 0.990268, 0.987688, 0.984808, 0.981627,
- 696 0.978148, 0.97437, 0.970296, 0.965926, 0.961262, 0.956305,
- 697 0.951057, 0.945519, 0.939693, 0.93358, 0.927184, 0.920505,
- 698 0.913545, 0.906308, 0.898794, 0.891007, 0.882948, 0.87462,
- 699 0.866025, 0.857167, 0.848048, 0.838671, 0.829038, 0.819152,
- 700 0.809017, 0.798636, 0.788011, 0.777146, 0.766044, 0.75471,
- 701 0.743145, 0.731354, 0.71934, 0.707107, 0.694658, 0.681998,
- 702 0.669131, 0.656059, 0.642788, 0.62932, 0.615661, 0.601815,
- 703 0.587785, 0.573576, 0.559193, 0.544639, 0.529919, 0.515038,
- 704 0.5, 0.48481, 0.469472, 0.45399, 0.438371, 0.422618,
- 705 0.406737, 0.390731, 0.374607, 0.358368, 0.34202, 0.325568,
- 706 0.309017, 0.292372, 0.275637, 0.258819, 0.241922, 0.224951,
- 707 0.207912, 0.190809, 0.173648, 0.156434, 0.139173, 0.121869,
- 708 0.104528, 0.0871557, 0.0697565, 0.052336, 0.0348995, 0.0174524,
-
-
- 711 T interpolate(
float angle) {
- 712 bool positive = (angle <= 180.0f);
- 713 float angle_positive = positive ? angle : angle - 180.0f;
- 714 int angle_int1 = angle_positive;
- 715 int angle_int2 = angle_int1 + 1;
- 716 T v1 = values[angle_int1] * amplitude;
- 717 T v2 = values[angle_int2] * amplitude;
- 718 T result = v1 < v2 ? map(angle_positive, angle_int1, angle_int2, v1, v2)
- 719 : map(angle_positive, angle_int1, angle_int2, v2, v1);
-
- 721 return positive ? result : -result;
-
-
- 724 T map(T x, T in_min, T in_max, T out_min, T out_max) {
- 725 return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
-
-
- 728 void updateAmplitudeInSteps() {
- 729 float diff = amplitude_to_be - amplitude;
- 730 if (abs(diff) > max_amplitude_step) {
- 731 diff = (diff < 0) ? -max_amplitude_step : max_amplitude_step;
-
- 733 if (abs(diff) >= 1.0f) {
-
-
-
-
-
-
-
-
-
-
-
-
- 753 void clear() { vector.clear(); }
-
-
-
-
- 758 for (
auto &generator : vector) {
-
-
-
-
-
-
- 765 return count > 0.0f ? total / count : 0;
-
-
-
-
- 770 int actualChannel = 0;
-
-
-
+
+
+
+
+ 619 value_return = value_set;
+
+
+
+
+
+
+ 627 bool isRunning() {
return is_running; }
+
+
+
+
+ 632 bool is_running =
false ;
+
+
+
+
+
+ 645 this->amplitude = amplitude;
+ 646 this->amplitude_to_be = amplitude;
+
+
+ 650 void setAmplitude (
float amplitude) { this->amplitude_to_be = amplitude; }
+
+
+
+
+
+
+ 659 if (angle >= 360.0f) {
+ 660 while (angle >= 360.0f) {
+
+
+
+
+
+ 666 updateAmplitudeInSteps();
+
+
+ 669 return interpolate(angle);
+
+
+
+
+
+
+
+
+
+
+ 680 bool begin(AudioInfo info,
float frequency) {
+ 681 SoundGenerator<T>::begin(info);
+
+
+
+
+
+
+ 688 bool begin(
int channels,
int sample_rate, uint16_t frequency = 0) {
+ 689 SoundGenerator<T>::info.channels = channels;
+ 690 SoundGenerator<T>::info.sample_rate = sample_rate;
+ 691 return begin(SoundGenerator<T>::info, frequency);
+
+
+
+ 695 step_new = freq / base_frequency;
+
+
+
+
+ 700 LOGD(
"step: %f" , step_new);
+
+
+
+ 704 bool is_first =
true ;
+
+ 706 float amplitude_to_be;
+ 707 float max_amplitude_step = 50.0f;
+ 708 float base_frequency = 1.0f;
+
+ 710 float step_new = 1.0f;
+
+
+ 713 const float values[181] = {
+ 714 0, 0.0174524, 0.0348995, 0.052336, 0.0697565, 0.0871557,
+ 715 0.104528, 0.121869, 0.139173, 0.156434, 0.173648, 0.190809,
+ 716 0.207912, 0.224951, 0.241922, 0.258819, 0.275637, 0.292372,
+ 717 0.309017, 0.325568, 0.34202, 0.358368, 0.374607, 0.390731,
+ 718 0.406737, 0.422618, 0.438371, 0.45399, 0.469472, 0.48481,
+ 719 0.5, 0.515038, 0.529919, 0.544639, 0.559193, 0.573576,
+ 720 0.587785, 0.601815, 0.615661, 0.62932, 0.642788, 0.656059,
+ 721 0.669131, 0.681998, 0.694658, 0.707107, 0.71934, 0.731354,
+ 722 0.743145, 0.75471, 0.766044, 0.777146, 0.788011, 0.798636,
+ 723 0.809017, 0.819152, 0.829038, 0.838671, 0.848048, 0.857167,
+ 724 0.866025, 0.87462, 0.882948, 0.891007, 0.898794, 0.906308,
+ 725 0.913545, 0.920505, 0.927184, 0.93358, 0.939693, 0.945519,
+ 726 0.951057, 0.956305, 0.961262, 0.965926, 0.970296, 0.97437,
+ 727 0.978148, 0.981627, 0.984808, 0.987688, 0.990268, 0.992546,
+ 728 0.994522, 0.996195, 0.997564, 0.99863, 0.999391, 0.999848,
+ 729 1, 0.999848, 0.999391, 0.99863, 0.997564, 0.996195,
+ 730 0.994522, 0.992546, 0.990268, 0.987688, 0.984808, 0.981627,
+ 731 0.978148, 0.97437, 0.970296, 0.965926, 0.961262, 0.956305,
+ 732 0.951057, 0.945519, 0.939693, 0.93358, 0.927184, 0.920505,
+ 733 0.913545, 0.906308, 0.898794, 0.891007, 0.882948, 0.87462,
+ 734 0.866025, 0.857167, 0.848048, 0.838671, 0.829038, 0.819152,
+ 735 0.809017, 0.798636, 0.788011, 0.777146, 0.766044, 0.75471,
+ 736 0.743145, 0.731354, 0.71934, 0.707107, 0.694658, 0.681998,
+ 737 0.669131, 0.656059, 0.642788, 0.62932, 0.615661, 0.601815,
+ 738 0.587785, 0.573576, 0.559193, 0.544639, 0.529919, 0.515038,
+ 739 0.5, 0.48481, 0.469472, 0.45399, 0.438371, 0.422618,
+ 740 0.406737, 0.390731, 0.374607, 0.358368, 0.34202, 0.325568,
+ 741 0.309017, 0.292372, 0.275637, 0.258819, 0.241922, 0.224951,
+ 742 0.207912, 0.190809, 0.173648, 0.156434, 0.139173, 0.121869,
+ 743 0.104528, 0.0871557, 0.0697565, 0.052336, 0.0348995, 0.0174524,
+
+
+ 746 T interpolate(
float angle) {
+ 747 bool positive = (angle <= 180.0f);
+ 748 float angle_positive = positive ? angle : angle - 180.0f;
+ 749 int angle_int1 = angle_positive;
+ 750 int angle_int2 = angle_int1 + 1;
+ 751 T v1 = values[angle_int1] * amplitude;
+ 752 T v2 = values[angle_int2] * amplitude;
+ 753 T result = v1 < v2 ? map(angle_positive, angle_int1, angle_int2, v1, v2)
+ 754 : map(angle_positive, angle_int1, angle_int2, v2, v1);
+
+ 756 return positive ? result : -result;
+
+
+ 759 T map(T x, T in_min, T in_max, T out_min, T out_max) {
+ 760 return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
+
+
+ 763 void updateAmplitudeInSteps() {
+ 764 float diff = amplitude_to_be - amplitude;
+ 765 if (abs(diff) > max_amplitude_step) {
+ 766 diff = (diff < 0) ? -max_amplitude_step : max_amplitude_step;
+
+ 768 if (abs(diff) >= 1.0f) {
+
+
+
+
+
+
-
+
-
-
- 787 if (abs(value) >= max) {
-
-
-
-
-
-
-
-
-
-
-
-
-
+
+
+
+ 788 void clear() { vector.clear(); }
+
+
+
+
+ 793 for (
auto &generator : vector) {
+
+
+
+
+
+
+ 800 return count > 0.0f ? total / count : 0;
+
+
+
+
+ 805 int actualChannel = 0;
+
+
+
+
+
+
+
+
+ 822 if (abs(value) >= max) {
+
+
+
+
+
+
+
+
+
+
+
+
+
-
-
-
-
-
-
-
-
-
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-
+
+
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+
+
+
+
+
+
-
-
-
+
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+
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+
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+
+
@@ -763,15 +793,15 @@
-
-
+
+
-
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-
+
+
+
diff --git a/annotated.html b/annotated.html
index f18d558fb..66d77355d 100644
--- a/annotated.html
+++ b/annotated.html
@@ -534,127 +534,128 @@
C RTSPPCMAudioInfo PCMInfo subclass which provides the audio information from the AudioInfo parameter
C RTSPSourceFromAudioStream Simple 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
C RTSPSourceStream Simple 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
- C SBCDecoder Decoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcdec.c
- C SBCEncoder Encoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcenc.c
- C ScaledParameter Scales a dynamic parameter to the indicated range
- C SDDirect We 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 SD API
- C SDIndex We store all the relevant file names in an sequential index file. Form there we can access them via an index
- C SerialDAC A SimpleDAC which uses the Serial UART to output values. This implementation is not using any timers and therefore should work on any microcontroller
- C SilenceGenerator Provides 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
- C SilenceRemovalConverter Removes 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
- C SimpleChorus Simple 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
- C SimpleContainerConfig
- C SimpleContainerDataHeader
- C SimpleContainerMetaDataHeader
- C SimpleDAC Software Implementation of a Simple DAC - We quantize a digital int16_t sample my mapping the value to the range of 0b0 to info.output_bits number of 1, where the intensity is represented by the number of ones. This is very similar to PWM!
- C SimpleDelay Simple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples
- C SimpleFlanger Simple 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
- C SimpleLPF SimpleLPF
- C SimulatedAudioPot Simple 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)
- C SineFromTable A sine generator based on a table. The table is created using degrees where one full wave is 360 degrees
- C SineWaveGenerator Generates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead
- C SingleBuffer A simple Buffer implementation which just uses a (dynamically sized) array
- C Slice Helps 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!
- C SmoothTransition Changes the samples at the beginning or at the end to slowly ramp up the volume
- C SOSFilter Second 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 float (and not a integer type) as type parameter
- C SoundGenerator Base class to define the abstract interface for the sound generating classes
- C SoundGeneratorModulation SoundGenerator using the ModulationBaseClass to generate the samples
- C SPDIFConfig SPDIF configuration
- C SPDIFOutput Output as 16 bit stereo SPDIF on the I2S data output pin
- C SquareWaveGenerator Generates a square wave sound
- C Stack LIFO Stack which is based on a List
- C StdioStream Direct binary Audio Output to stdout. On linux you can hear the audio e.g. with ./generator | aplay -f cd or reading data from stdin
- C STKChorus Chorus Effect
- C STKChowningReverb John Chowning's reverberator class
- C STKEcho Echo Effect
- C STKEffect Use any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK
- C STKFreeVerb Jezar at Dreampoint's FreeVerb, implemented in STK
- C STKGenerator The Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language. You need to install https://github.com/pschatzmann/Arduino-STK
+ C SawToothGenerator Sine wave which is based on a fast approximation function
+ C SBCDecoder Decoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcdec.c
+ C SBCEncoder Encoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcenc.c
+ C ScaledParameter Scales a dynamic parameter to the indicated range
+ C SDDirect We 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 SD API
+ C SDIndex We store all the relevant file names in an sequential index file. Form there we can access them via an index
+ C SerialDAC A SimpleDAC which uses the Serial UART to output values. This implementation is not using any timers and therefore should work on any microcontroller
+ C SilenceGenerator Provides 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
+ C SilenceRemovalConverter Removes 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
+ C SimpleChorus Simple 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
+ C SimpleContainerConfig
+ C SimpleContainerDataHeader
+ C SimpleContainerMetaDataHeader
+ C SimpleDAC Software Implementation of a Simple DAC - We quantize a digital int16_t sample my mapping the value to the range of 0b0 to info.output_bits number of 1, where the intensity is represented by the number of ones. This is very similar to PWM!
+ C SimpleDelay Simple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples
+ C SimpleFlanger Simple 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
+ C SimpleLPF SimpleLPF
+ C SimulatedAudioPot Simple 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)
+ C SineFromTable A sine generator based on a table. The table is created using degrees where one full wave is 360 degrees
+ C SineWaveGenerator Generates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead
+ C SingleBuffer A simple Buffer implementation which just uses a (dynamically sized) array
+ C Slice Helps 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!
+ C SmoothTransition Changes the samples at the beginning or at the end to slowly ramp up the volume
+ C SOSFilter Second 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 float (and not a integer type) as type parameter
+ C SoundGenerator Base class to define the abstract interface for the sound generating classes
+ C SoundGeneratorModulation SoundGenerator using the ModulationBaseClass to generate the samples
+ C SPDIFConfig SPDIF configuration
+ C SPDIFOutput Output as 16 bit stereo SPDIF on the I2S data output pin
+ C SquareWaveGenerator Generates a square wave sound
+ C Stack LIFO Stack which is based on a List
+ C StdioStream Direct binary Audio Output to stdout. On linux you can hear the audio e.g. with ./generator | aplay -f cd or reading data from stdin
+ C STKChorus Chorus Effect
+ C STKChowningReverb John Chowning's reverberator class
+ C STKEcho Echo Effect
+ C STKEffect Use any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK
+ C STKFreeVerb Jezar at Dreampoint's FreeVerb, implemented in STK
+ C STKGenerator The Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language. You need to install https://github.com/pschatzmann/Arduino-STK
- C STKLentPitShift Pitch shifter effect class based on the Lent algorithm
- C STKNReverb CCRMA's NRev reverberator class
- C STKPerryReverb Perry's simple reverberator class
- C STKPitShift Simple Pitch shifter effect class: This class implements a simple pitch shifter using a delay line
- C STKStream STK Stream for Instrument or Voicer
- C Str A 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!
- C Stream
- C StreamCopyT Typed Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples
- C StreamingDecoder A Streaming Decoder where we provide both the input and output as streams
- C StreamingDecoderAdapter Converts any AudioDecoder to a StreamingDecoder
- C StrExt Str which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting
- C SynchronizedBuffer Wrapper class that can turn any Buffer into a thread save implementation
- C SynchronizedNBuffer NBuffer which uses some RTOS queues to manage the available and filled buffers
- ► C Synthesizer A simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels
- C KeyParameter
- C SynthesizerKey Arduino GPIO pin to note assossiation
- C Task FreeRTOS task
- C TestGenerator Generates a test signal which is easy to check because the values are incremented or decremented by 1
- C TfLiteAbstractRecognizeCommands Base 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
- C TfLiteAudioErrorReporter Error Reporter using the Audio Tools Logger
- C TfLiteAudioStream TfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data)
- C TfLiteAudioStreamBase Astract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers
- C TfLiteConfig Configuration settings for TfLiteAudioStream
- C TfLiteMicroSpeachWriter TfLiteMicroSpeachWriter for Audio Data
- ► C TfLiteMicroSpeechRecognizeCommands This 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
- C Result
- C TfLiteQuantizer Quantizer that helps to quantize and dequantize between float and int8
- C TfLiteReader Input class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce
- C TfLiteSineReader Generate a sine output from a model that was trained on the sine method. (=hello_world)
- C TfLiteWriter Output class which interprets audio data if TfLiteAudioStream is treated as audio sink
- C Throttle Throttle the sending or receiving of the audio data to limit it to the indicated sample rate
- C ThrottleConfig Configure Throttle setting
- C TimedStream AudioStream 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!
- C TimerAlarmRepeating Common Interface definition for TimerAlarmRepeating
- C TimerAlarmRepeatingDriverAVR Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
- C TimerAlarmRepeatingDriverBase
- C TimerAlarmRepeatingDriverESP8266 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
- C TimerAlarmRepeatingDriverMBED Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
- C TimerAlarmRepeatingDriverRenesas Repeating 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);
- C TimerAlarmRepeatingDriverSTM32 STM32 Repeating Timer functions for repeated execution: Please use the typedef TimerAlarmRepeating . By default the TIM1 is used
- C TimerCallbackAudioStream Callback 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
- C TimerCallbackAudioStreamInfo TimerCallbackAudioStream Configuration
- C TransformationReader ConverterStream Helper class which implements the readBytes with the help of write
- C Tremolo Tremolo AudioEffect
- C Triangle Triangle FFT Window function
- C UDPStream A UDP class which makes sure that we can use UDP as AudioSource and AudioSink. By default the WiFiUDP object is used and we login to wifi if the ssid and password is provided and we are not already connected
- C Url URL parser which breaks a full url string up into its individual parts
- C URLHistory
- C URLLoaderHLS
- C URLLoaderHLSBase Abstract API for URLLoaderHLS
- C URLLoaderHLSOutput URLLoader which saves the HLS segments to the indicated output
- C URLStream Represents the content of a URL as Stream . We use the WiFi.h API
- C URLStreamBuffered URLStream implementation for the ESP32 based on a separate FreeRTOS task
- C VariableSpeedRingBuffer Optimized 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
- C VariableSpeedRingBuffer180 Varialbe 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
- C VariableSpeedRingBufferSimple Very 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
- C VBANConfig
- C VBANStream VBAN 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
- ► C Vector Vector 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
- C iterator Iterator for the Vector class
- C VideoAudioBufferedSync Logic 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
- C VideoAudioSync Logic 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
- C VideoOutput Abstract class for video playback. This class is used to assemble a complete video frame in memory
- C VolumeControl Abstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal
- C VolumeMeter A simple class to determine the volume. You can use it as final output or as output or input in your audio chain
- C VolumeStream Adjust 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
- C VolumeStreamConfig Config for VolumeStream
- C VolumeSupport Supports the setting and getting of the volume
- C VorbisDecoder Vorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-tremor
- C VS1053Config Configuration for VS1053Stream
- C VS1053Stream VS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration;
- C WAVAudioInfo Sound information which is available in the WAV header
- C WAVDecoder A 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
- C WAVEncoder A 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
- C WAVFormatX
- C WAVHeader Parser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE
- C WavIMAAudioInfo Sound information which is available in the WAV header - adjusted for IMA ADPCM
- C WavIMADecoder Obsolete: 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!
- C WavIMAHeader
- C Welch Welch FFT Window function
- C WhiteNoiseGenerator Generates a random noise sound with the help of rand() function
- C WindowFunction FFT Window Function
- C WM8960Config Configuration for WM8960
- C WM8960Stream Stream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960
+ C STKLentPitShift Pitch shifter effect class based on the Lent algorithm
+ C STKNReverb CCRMA's NRev reverberator class
+ C STKPerryReverb Perry's simple reverberator class
+ C STKPitShift Simple Pitch shifter effect class: This class implements a simple pitch shifter using a delay line
+ C STKStream STK Stream for Instrument or Voicer
+ C Str A 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!
+ C Stream
+ C StreamCopyT Typed Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples
+ C StreamingDecoder A Streaming Decoder where we provide both the input and output as streams
+ C StreamingDecoderAdapter Converts any AudioDecoder to a StreamingDecoder
+ C StrExt Str which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting
+ C SynchronizedBuffer Wrapper class that can turn any Buffer into a thread save implementation
+ C SynchronizedNBuffer NBuffer which uses some RTOS queues to manage the available and filled buffers
+ ► C Synthesizer A simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels
+ C KeyParameter
+ C SynthesizerKey Arduino GPIO pin to note assossiation
+ C Task FreeRTOS task
+ C TestGenerator Generates a test signal which is easy to check because the values are incremented or decremented by 1
+ C TfLiteAbstractRecognizeCommands Base 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
+ C TfLiteAudioErrorReporter Error Reporter using the Audio Tools Logger
+ C TfLiteAudioStream TfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data)
+ C TfLiteAudioStreamBase Astract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers
+ C TfLiteConfig Configuration settings for TfLiteAudioStream
+ C TfLiteMicroSpeachWriter TfLiteMicroSpeachWriter for Audio Data
+ ► C TfLiteMicroSpeechRecognizeCommands This 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
+ C Result
+ C TfLiteQuantizer Quantizer that helps to quantize and dequantize between float and int8
+ C TfLiteReader Input class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce
+ C TfLiteSineReader Generate a sine output from a model that was trained on the sine method. (=hello_world)
+ C TfLiteWriter Output class which interprets audio data if TfLiteAudioStream is treated as audio sink
+ C Throttle Throttle the sending or receiving of the audio data to limit it to the indicated sample rate
+ C ThrottleConfig Configure Throttle setting
+ C TimedStream AudioStream 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!
+ C TimerAlarmRepeating Common Interface definition for TimerAlarmRepeating
+ C TimerAlarmRepeatingDriverAVR Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
+ C TimerAlarmRepeatingDriverBase
+ C TimerAlarmRepeatingDriverESP8266 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
+ C TimerAlarmRepeatingDriverMBED Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
+ C TimerAlarmRepeatingDriverRenesas Repeating 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);
+ C TimerAlarmRepeatingDriverSTM32 STM32 Repeating Timer functions for repeated execution: Please use the typedef TimerAlarmRepeating . By default the TIM1 is used
+ C TimerCallbackAudioStream Callback 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
+ C TimerCallbackAudioStreamInfo TimerCallbackAudioStream Configuration
+ C TransformationReader ConverterStream Helper class which implements the readBytes with the help of write
+ C Tremolo Tremolo AudioEffect
+ C Triangle Triangle FFT Window function
+ C UDPStream A UDP class which makes sure that we can use UDP as AudioSource and AudioSink. By default the WiFiUDP object is used and we login to wifi if the ssid and password is provided and we are not already connected
+ C Url URL parser which breaks a full url string up into its individual parts
+ C URLHistory
+ C URLLoaderHLS
+ C URLLoaderHLSBase Abstract API for URLLoaderHLS
+ C URLLoaderHLSOutput URLLoader which saves the HLS segments to the indicated output
+ C URLStream Represents the content of a URL as Stream . We use the WiFi.h API
+ C URLStreamBuffered URLStream implementation for the ESP32 based on a separate FreeRTOS task
+ C VariableSpeedRingBuffer Optimized 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
+ C VariableSpeedRingBuffer180 Varialbe 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
+ C VariableSpeedRingBufferSimple Very 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
+ C VBANConfig
+ C VBANStream VBAN 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
+ ► C Vector Vector 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
+ C iterator Iterator for the Vector class
+ C VideoAudioBufferedSync Logic 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
+ C VideoAudioSync Logic 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
+ C VideoOutput Abstract class for video playback. This class is used to assemble a complete video frame in memory
+ C VolumeControl Abstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal
+ C VolumeMeter A simple class to determine the volume. You can use it as final output or as output or input in your audio chain
+ C VolumeStream Adjust 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
+ C VolumeStreamConfig Config for VolumeStream
+ C VolumeSupport Supports the setting and getting of the volume
+ C VorbisDecoder Vorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-tremor
+ C VS1053Config Configuration for VS1053Stream
+ C VS1053Stream VS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration;
+ C WAVAudioInfo Sound information which is available in the WAV header
+ C WAVDecoder A 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
+ C WAVEncoder A 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
+ C WAVFormatX
+ C WAVHeader Parser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE
+ C WavIMAAudioInfo Sound information which is available in the WAV header - adjusted for IMA ADPCM
+ C WavIMADecoder Obsolete: 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!
+ C WavIMAHeader
+ C Welch Welch FFT Window function
+ C WhiteNoiseGenerator Generates a random noise sound with the help of rand() function
+ C WindowFunction FFT Window Function
+ C WM8960Config Configuration for WM8960
+ C WM8960Stream Stream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960
► N audiotools_rtsp AudioTools internal: rtsp
C OurRTSPClient
C OurSink
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This is the complete list of members for SawToothGenerator< T > , including all inherited members.
+
+ active (defined in SoundGenerator< T > )SoundGenerator< T > protected
+ activeWarningIssued (defined in SoundGenerator< T > )SoundGenerator< T > protected
+ audioInfo ()SoundGenerator< T > inline virtual
+ begin () override (defined in SineWaveGenerator< T > )SineWaveGenerator< T > inline virtual
+ begin (AudioInfo info) override (defined in SineWaveGenerator< T > )SineWaveGenerator< T > inline virtual
+ begin (AudioInfo info, float frequency) (defined in SineWaveGenerator< T > )SineWaveGenerator< T > inline
+ begin (int channels, int sample_rate, float frequency) (defined in SineWaveGenerator< T > )SineWaveGenerator< T > inline
+ defaultConfig () overrideSineWaveGenerator< T > inline virtual
+ double_Pi (defined in SineWaveGenerator< T > )SineWaveGenerator< T > protected
+ end ()SoundGenerator< T > inline virtual
+ info (defined in SoundGenerator< T > )SoundGenerator< T > protected
+ isActive ()SoundGenerator< T > inline virtual
+ logStatus () (defined in SineWaveGenerator< T > )SineWaveGenerator< T > inline protected
+ m_amplitude (defined in SineWaveGenerator< T > )SineWaveGenerator< T > protected
+ m_cycles (defined in SineWaveGenerator< T > )SineWaveGenerator< T > protected
+ m_deltaTime (defined in SineWaveGenerator< T > )SineWaveGenerator< T > protected
+ m_frequency (defined in SineWaveGenerator< T > )SineWaveGenerator< T > protected
+ m_phase (defined in SineWaveGenerator< T > )SineWaveGenerator< T > protected
+ readBytes (uint8_t *data, size_t len)SoundGenerator< T > inline virtual
+ readBytesFrames (uint8_t *buffer, size_t lengthBytes, int frames, int channels) (defined in SoundGenerator< T > )SoundGenerator< T > inline protected
+ readBytesFromBuffer (uint8_t *buffer, size_t lengthBytes, int frame_size, int channels) (defined in SoundGenerator< T > )SoundGenerator< T > inline protected
+ readSample () overrideSawToothGenerator< T > inline virtual
+ ring_buffer (defined in SoundGenerator< T > )SoundGenerator< T > protected
+ saw (float t)SawToothGenerator< T > inline protected
+ SawToothGenerator (float amplitude=32767.0, float phase=0.0) (defined in SawToothGenerator< T > )SawToothGenerator< T > inline
+ setAmplitude (float amp) (defined in SineWaveGenerator< T > )SineWaveGenerator< T > inline
+ setAudioInfo (AudioInfo info) overrideSineWaveGenerator< T > inline virtual
+ setFrequency (float frequency) overrideSineWaveGenerator< T > inline virtual
+ SineWaveGenerator (float amplitude=0.9f *NumberConverter::maxValueT< T >(), float phase=0.0f) (defined in SineWaveGenerator< T > )SineWaveGenerator< T > inline
+ SoundGenerator () (defined in SoundGenerator< T > )SoundGenerator< T > inline
+ ~SoundGenerator () (defined in SoundGenerator< T > )SoundGenerator< T > inline virtual
+
+
+
+
+
diff --git a/classaudio__tools_1_1_saw_tooth_generator.html b/classaudio__tools_1_1_saw_tooth_generator.html
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+arduino-audio-tools: SawToothGenerator< T > Class Template Reference
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+ arduino-audio-tools
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+
Sine wave which is based on a fast approximation function.
+ More...
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+
#include <SoundGenerator.h >
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+
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+
+
+
+
+ SawToothGenerator (float amplitude=32767.0, float phase=0.0)
+
+
+virtual AudioInfo audioInfo ()
+ Provides the AudioInfo .
+
+
+bool begin () override
+
+
+bool begin (AudioInfo info) override
+
+
+bool begin (AudioInfo info, float frequency)
+
+
+bool begin (int channels, int sample_rate, float frequency)
+
+
+virtual AudioInfo defaultConfig () override
+ Provides the default configuration.
+
+
+virtual void end ()
+ ends the processing
+
+virtual bool isActive ()
+
+
+virtual size_t readBytes (uint8_t *data, size_t len)
+ Provides the data as byte array with the requested number of channels.
+
+
+virtual T readSample () override
+ Provides a single sample.
+
+
+void setAmplitude (float amp)
+
+
+virtual void setAudioInfo (AudioInfo info) override
+ Defines/updates the AudioInfo .
+
+
+void setFrequency (float frequency) override
+ Defines the frequency - after the processing has been started.
+
+
+
+
+void logStatus ()
+
+
+size_t readBytesFrames (uint8_t *buffer, size_t lengthBytes, int frames, int channels)
+
+
+size_t readBytesFromBuffer (uint8_t *buffer, size_t lengthBytes, int frame_size, int channels)
+
+
+float saw (float t)
+ sine approximation.
+
+
+
+
+bool active = false
+
+
+bool activeWarningIssued = false
+
+
+const float double_Pi = PI * 2.0f
+
+
+AudioInfo info
+
+
+float m_amplitude = 1.0f
+
+
+float m_cycles = 0.0f
+
+
+float m_deltaTime = 0.0f
+
+
+volatile float m_frequency = 0.0f
+
+
+float m_phase = 0.0f
+
+
+RingBuffer < uint8_t > ring_buffer {0}
+
+
+
+
template<class T>
+class audio_tools::SawToothGenerator< T >
+
+
Sine wave which is based on a fast approximation function.
+
Author Vivian Leigh Stewart
+
Copyright GPLv3
+
Template Parameters
+
+
+
+
+
+
◆ isActive()
+
+
+
+
+
+
+
+
+ virtual bool isActive
+ (
+ )
+
+
+
+
+
+inline virtual inherited
+
+
+
+
Checks if the begin method has been called - after end() isActive returns false
+
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+
The documentation for this class was generated from the following file:
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+
diff --git a/classaudio__tools_1_1_saw_tooth_generator.png b/classaudio__tools_1_1_saw_tooth_generator.png
new file mode 100644
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diff --git a/classaudio__tools_1_1_sine_wave_generator.html b/classaudio__tools_1_1_sine_wave_generator.html
index c86fe8a6c..ff921baa5 100644
--- a/classaudio__tools_1_1_sine_wave_generator.html
+++ b/classaudio__tools_1_1_sine_wave_generator.html
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diff --git a/classaudio__tools_1_1_sine_wave_generator.png b/classaudio__tools_1_1_sine_wave_generator.png
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diff --git a/classes.html b/classes.html
index a504b3256..3af9f24c6 100644
--- a/classes.html
+++ b/classes.html
@@ -123,7 +123,7 @@
R2RConfig (audio_tools )R2RDriver (audio_tools )R2RDriverBase (audio_tools )R2ROutput (audio_tools )ReadioHeadStream (audio_tools )ReasampleLinearInterpolation (audio_tools )RECT (audio_tools )Rectange (audio_tools )ReedSolomon (RS )ReedSolomonFEC (audio_tools )ReformatBaseStream (audio_tools )Resample2Point3Order (audio_tools )Resample4Point2Order (audio_tools )ResampleBSpline (audio_tools )ResampleConfig (audio_tools )ResampleLagrange (audio_tools )ResampleParabolic (audio_tools )ResampleStream (audio_tools )ResmpleHermite (audio_tools )TfLiteMicroSpeechRecognizeCommands::Result (audio_tools )RingBuffer (audio_tools )RingBufferFile (audio_tools )RingBufferStream (audio_tools )RTSPFormatAbtX (audio_tools )RTSPFormatAudioTools (audio_tools )RTSPFormatG711 (audio_tools )RTSPFormatGSM (audio_tools )RTSPFormatOpus (audio_tools )RTSPFormatPCM (audio_tools )RTSPFormatPCM8 (audio_tools )RTSPOutput (audio_tools )RTSPOutputPCMInfo (audio_tools )RTSPPCMAudioInfo (audio_tools )RTSPSourceFromAudioStream (audio_tools )RTSPSourceStream (audio_tools )
S
-SBCDecoder (audio_tools )SBCEncoder (audio_tools )ScaledParameter (audio_tools )SDDirect (audio_tools )SDIndex (audio_tools )SerialDAC (audio_tools )SilenceGenerator (audio_tools )SilenceRemovalConverter (audio_tools )SimpleChorus (audio_tools )SimpleContainerConfig (audio_tools )SimpleContainerDataHeader (audio_tools )SimpleContainerMetaDataHeader (audio_tools )SimpleDAC (audio_tools )SimpleDelay (audio_tools )SimpleFlanger (audio_tools )SimpleLPF (audio_tools )SimulatedAudioPot (audio_tools )SineFromTable (audio_tools )SineWaveGenerator (audio_tools )SingleBuffer (audio_tools )Slice (audio_tools )SmoothTransition (audio_tools )SOSFilter (audio_tools )SoundGenerator (audio_tools )SoundGeneratorModulation (audio_tools )SPDIFConfig (audio_tools )SPDIFOutput (audio_tools )SquareWaveGenerator (audio_tools )Stack (audio_tools )StdioStream (audio_tools )STKChorus (audio_tools )STKChowningReverb (audio_tools )STKEcho (audio_tools )STKEffect (audio_tools )STKFreeVerb (audio_tools )STKGenerator (audio_tools )STKLentPitShift (audio_tools )STKNReverb (audio_tools )STKPerryReverb (audio_tools )STKPitShift (audio_tools )STKStream (audio_tools )Str (audio_tools )Stream (audio_tools )StreamClientState (audiotools_rtsp )StreamCopyT (audio_tools )StreamingDecoder (audio_tools )StreamingDecoderAdapter (audio_tools )StrExt (audio_tools )SynchronizedBuffer (audio_tools )SynchronizedNBuffer (audio_tools )Synthesizer (audio_tools )SynthesizerKey (audio_tools )
+SawToothGenerator (audio_tools )SBCDecoder (audio_tools )SBCEncoder (audio_tools )ScaledParameter (audio_tools )SDDirect (audio_tools )SDIndex (audio_tools )SerialDAC (audio_tools )SilenceGenerator (audio_tools )SilenceRemovalConverter (audio_tools )SimpleChorus (audio_tools )SimpleContainerConfig (audio_tools )SimpleContainerDataHeader (audio_tools )SimpleContainerMetaDataHeader (audio_tools )SimpleDAC (audio_tools )SimpleDelay (audio_tools )SimpleFlanger (audio_tools )SimpleLPF (audio_tools )SimulatedAudioPot (audio_tools )SineFromTable (audio_tools )SineWaveGenerator (audio_tools )SingleBuffer (audio_tools )Slice (audio_tools )SmoothTransition (audio_tools )SOSFilter (audio_tools )SoundGenerator (audio_tools )SoundGeneratorModulation (audio_tools )SPDIFConfig (audio_tools )SPDIFOutput (audio_tools )SquareWaveGenerator (audio_tools )Stack (audio_tools )StdioStream (audio_tools )STKChorus (audio_tools )STKChowningReverb (audio_tools )STKEcho (audio_tools )STKEffect (audio_tools )STKFreeVerb (audio_tools )STKGenerator (audio_tools )STKLentPitShift (audio_tools )STKNReverb (audio_tools )STKPerryReverb (audio_tools )STKPitShift (audio_tools )STKStream (audio_tools )Str (audio_tools )Stream (audio_tools )StreamClientState (audiotools_rtsp )StreamCopyT (audio_tools )StreamingDecoder (audio_tools )StreamingDecoderAdapter (audio_tools )StrExt (audio_tools )SynchronizedBuffer (audio_tools )SynchronizedNBuffer (audio_tools )Synthesizer (audio_tools )SynthesizerKey (audio_tools )
T
Task (audio_tools )TestGenerator (audio_tools )TfLiteAbstractRecognizeCommands (audio_tools )TfLiteAudioErrorReporter (audio_tools )TfLiteAudioStream (audio_tools )TfLiteAudioStreamBase (audio_tools )TfLiteConfig (audio_tools )TfLiteMicroSpeachWriter (audio_tools )TfLiteMicroSpeechRecognizeCommands (audio_tools )TfLiteQuantizer (audio_tools )TfLiteReader (audio_tools )TfLiteSineReader (audio_tools )TfLiteWriter (audio_tools )Throttle (audio_tools )ThrottleConfig (audio_tools )TimedStream (audio_tools )TimerAlarmRepeating (audio_tools )TimerAlarmRepeatingDriverAVR (audio_tools )TimerAlarmRepeatingDriverBase (audio_tools )TimerAlarmRepeatingDriverESP8266 (audio_tools )TimerAlarmRepeatingDriverMBED (audio_tools )TimerAlarmRepeatingDriverRenesas (audio_tools )TimerAlarmRepeatingDriverSTM32 (audio_tools )TimerCallbackAudioStream (audio_tools )TimerCallbackAudioStreamInfo (audio_tools )TransformationReader (audio_tools )Tremolo (audio_tools )Triangle (audio_tools )
diff --git a/functions.html b/functions.html
index d12eaa9d3..e21e6a56e 100644
--- a/functions.html
+++ b/functions.html
@@ -147,10 +147,10 @@ - a -
addOutput()
: ChannelSplitOutput
-, ChannelsSelectOutput
+, ChannelsSelectOutput
addPeer()
-: ESPNowStream
+: ESPNowStream
addPeers()
: ESPNowStream
@@ -197,7 +197,7 @@ - a -
AnalogAudioStream()
-: AnalogAudioStream
+: AnalogAudioStream
AnalogConfigESP32()
: AnalogConfigESP32
@@ -253,7 +253,7 @@ - a -
AudioEffects()
-: AudioEffects< GeneratorT >
+: AudioEffects< GeneratorT >
AudioFFTBase()
: AudioFFTBase
@@ -334,7 +334,7 @@ - a -
AudioSourceSDFAT()
-: AudioSourceSDFAT
+: AudioSourceSDFAT
AudioSourceSDMMC()
: AudioSourceSDMMC
@@ -401,7 +401,7 @@ - a -
, I2SStream
, MeasuringStream
, NBuffer< T >
-, OutputMixer< T >
+, OutputMixer< T >
, ProgressStream
, RingBuffer< T >
, RingBufferFile< File, T >
diff --git a/functions_func.html b/functions_func.html
index 9e45ff52c..10206337d 100644
--- a/functions_func.html
+++ b/functions_func.html
@@ -103,7 +103,7 @@ - a -
addAction()
: AudioBoardStream
@@ -147,7 +147,7 @@ - a -
addOutput()
: ChannelSplitOutput
-, ChannelsSelectOutput
+, ChannelsSelectOutput
addPeer()
: ESPNowStream
@@ -244,7 +244,7 @@ - a -
AudioEffects()
-: AudioEffects< GeneratorT >
+: AudioEffects< GeneratorT >
AudioFFTBase()
: AudioFFTBase
@@ -259,7 +259,7 @@ - a -
AudioInfo()
-: AudioInfo
+: AudioInfo
audioInfo()
: AudioInfoSupport
@@ -313,7 +313,7 @@ - a -
AudioSourceIdxSDFAT()
-: AudioSourceIdxSDFAT
+: AudioSourceIdxSDFAT
AudioSourceIdxSDMMC()
: AudioSourceIdxSDMMC
diff --git a/functions_func_r.html b/functions_func_r.html
index e746ad066..3849494ea 100644
--- a/functions_func_r.html
+++ b/functions_func_r.html
@@ -143,6 +143,7 @@ - r -
ResampleStream()
-: ResampleStream
+: ResampleStream
reset()
: AudioFFTBase
diff --git a/functions_func_s.html b/functions_func_s.html
index de1914191..a3c796f2a 100644
--- a/functions_func_s.html
+++ b/functions_func_s.html
@@ -70,6 +70,9 @@ - s -
samples()
: WindowFunction
+saw()
+: SawToothGenerator< T >
+
scale()
: NumberReader
@@ -90,19 +93,19 @@ - s -
, AudioSourceCallback
, AudioSourceIdxSDFAT
, AudioSourceIdxSDMMC
-, AudioSourceLittleFS
+, AudioSourceLittleFS
, AudioSourceSD
, AudioSourceSDFAT
, AudioSourceSDMMC
, AudioSourceSPIFFS
-, AudioSourceSTD
+, AudioSourceSTD
, AudioSourceURL
set()
: AudioInfo
, InputMixer< T >
, int24_4bytes_t
-, Str
+, Str
setActionVolumeIncrementValue()
: AudioBoardStream
@@ -276,7 +279,7 @@ - s -
setBoard()
-: I2SCodecStream
+: I2SCodecStream
setBuffer()
: DriverPWMBase
@@ -432,7 +435,7 @@ - s -
setFilter()
: ConverterNChannels< T, FT >
-, FilteredStream< T, TF >
+, FilteredStream< T, TF >
setFilterMetaData()
: MP3DecoderHelix
@@ -471,7 +474,7 @@ - s -
setInput()
: AudioEffects< GeneratorT >
-, Pipeline
+, Pipeline
, StreamingDecoder
setInputBufferSize()
diff --git a/functions_r.html b/functions_r.html
index 9fd6369b4..b12684702 100644
--- a/functions_r.html
+++ b/functions_r.html
@@ -146,6 +146,7 @@ - r -
selectStream()
-: AudioSource
-, AudioSourceCallback
+: AudioSource
+, AudioSourceCallback
, AudioSourceIdxSDFAT
, AudioSourceIdxSDMMC
, AudioSourceLittleFS
-, AudioSourceSD
+, AudioSourceSD
, AudioSourceSDFAT
, AudioSourceSDMMC
, AudioSourceSPIFFS
-, AudioSourceSTD
+, AudioSourceSTD
, AudioSourceURL
set()
@@ -191,7 +194,7 @@ - s -
, MeasuringStream
, MiniAudioStream
, MozziStream
-, MP3EncoderLAME
+, MP3EncoderLAME
, MultiOutput
, NumberFormatConverterStream
, NumberFormatConverterStreamT< TFrom, TTo >
@@ -216,7 +219,7 @@ - s -
@@ -256,7 +259,7 @@ - s -
: SimpleChorus
setBin()
-: AudioFFTBase
+: AudioFFTBase
, FFTDriver
, FFTDriverCmsisFFT
, FFTDriverESP32FFT
diff --git a/group__generator.html b/group__generator.html
index d76056f71..a8aff5939 100644
--- a/group__generator.html
+++ b/group__generator.html
@@ -91,6 +91,9 @@
class PinkNoiseGenerator< T >
Generates pink noise. More...
+class SawToothGenerator< T >
+ Sine wave which is based on a fast approximation function. More...
+
class SilenceGenerator< T >
Provides 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. More...
diff --git a/hierarchy.html b/hierarchy.html
index 8adb54cc5..beecdefff 100644
--- a/hierarchy.html
+++ b/hierarchy.html
@@ -700,7 +700,8 @@
► C SineWaveGenerator< T > Generates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead
C FastSineGenerator< int16_t >
C FastSineGenerator< T > Sine wave which is based on a fast approximation function
- C SquareWaveGenerator< T > Generates a square wave sound
+ C SawToothGenerator< T > Sine wave which is based on a fast approximation function
+ C SquareWaveGenerator< T > Generates a square wave sound
C SoundGeneratorModulation< T > SoundGenerator using the ModulationBaseClass to generate the samples
C TestGenerator< T > Generates a test signal which is easy to check because the values are incremented or decremented by 1
C WhiteNoiseGenerator< T > Generates a random noise sound with the help of rand() function
diff --git a/namespaceaudio__tools.html b/namespaceaudio__tools.html
index 34f2e0b2c..b30401631 100644
--- a/namespaceaudio__tools.html
+++ b/namespaceaudio__tools.html
@@ -1383,6 +1383,9 @@
class RTSPSourceStream
Simple 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. More...
+class SawToothGenerator
+ Sine wave which is based on a fast approximation function. More...
+
class SBCDecoder
Decoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcdec.c. More...
diff --git a/namespaces.html b/namespaces.html
index 4a23e456c..59babaa83 100644
--- a/namespaces.html
+++ b/namespaces.html
@@ -534,127 +534,128 @@
C RTSPPCMAudioInfo PCMInfo subclass which provides the audio information from the AudioInfo parameter
C RTSPSourceFromAudioStream Simple 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
C RTSPSourceStream Simple 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
- C SBCDecoder Decoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcdec.c
- C SBCEncoder Encoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcenc.c
- C ScaledParameter Scales a dynamic parameter to the indicated range
- C SDDirect We 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 SD API
- C SDIndex We store all the relevant file names in an sequential index file. Form there we can access them via an index
- C SerialDAC A SimpleDAC which uses the Serial UART to output values. This implementation is not using any timers and therefore should work on any microcontroller
- C SilenceGenerator Provides 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
- C SilenceRemovalConverter Removes 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
- C SimpleChorus Simple 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
- C SimpleContainerConfig
- C SimpleContainerDataHeader
- C SimpleContainerMetaDataHeader
- C SimpleDAC Software Implementation of a Simple DAC - We quantize a digital int16_t sample my mapping the value to the range of 0b0 to info.output_bits number of 1, where the intensity is represented by the number of ones. This is very similar to PWM!
- C SimpleDelay Simple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples
- C SimpleFlanger Simple 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
- C SimpleLPF SimpleLPF
- C SimulatedAudioPot Simple 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)
- C SineFromTable A sine generator based on a table. The table is created using degrees where one full wave is 360 degrees
- C SineWaveGenerator Generates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead
- C SingleBuffer A simple Buffer implementation which just uses a (dynamically sized) array
- C Slice Helps 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!
- C SmoothTransition Changes the samples at the beginning or at the end to slowly ramp up the volume
- C SOSFilter Second 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 float (and not a integer type) as type parameter
- C SoundGenerator Base class to define the abstract interface for the sound generating classes
- C SoundGeneratorModulation SoundGenerator using the ModulationBaseClass to generate the samples
- C SPDIFConfig SPDIF configuration
- C SPDIFOutput Output as 16 bit stereo SPDIF on the I2S data output pin
- C SquareWaveGenerator Generates a square wave sound
- C Stack LIFO Stack which is based on a List
- C StdioStream Direct binary Audio Output to stdout. On linux you can hear the audio e.g. with ./generator | aplay -f cd or reading data from stdin
- C STKChorus Chorus Effect
- C STKChowningReverb John Chowning's reverberator class
- C STKEcho Echo Effect
- C STKEffect Use any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK
- C STKFreeVerb Jezar at Dreampoint's FreeVerb, implemented in STK
- C STKGenerator The Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language. You need to install https://github.com/pschatzmann/Arduino-STK
+ C SawToothGenerator Sine wave which is based on a fast approximation function
+ C SBCDecoder Decoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcdec.c
+ C SBCEncoder Encoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc . Inspired by sbcenc.c
+ C ScaledParameter Scales a dynamic parameter to the indicated range
+ C SDDirect We 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 SD API
+ C SDIndex We store all the relevant file names in an sequential index file. Form there we can access them via an index
+ C SerialDAC A SimpleDAC which uses the Serial UART to output values. This implementation is not using any timers and therefore should work on any microcontroller
+ C SilenceGenerator Provides 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
+ C SilenceRemovalConverter Removes 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
+ C SimpleChorus Simple 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
+ C SimpleContainerConfig
+ C SimpleContainerDataHeader
+ C SimpleContainerMetaDataHeader
+ C SimpleDAC Software Implementation of a Simple DAC - We quantize a digital int16_t sample my mapping the value to the range of 0b0 to info.output_bits number of 1, where the intensity is represented by the number of ones. This is very similar to PWM!
+ C SimpleDelay Simple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples
+ C SimpleFlanger Simple 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
+ C SimpleLPF SimpleLPF
+ C SimulatedAudioPot Simple 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)
+ C SineFromTable A sine generator based on a table. The table is created using degrees where one full wave is 360 degrees
+ C SineWaveGenerator Generates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead
+ C SingleBuffer A simple Buffer implementation which just uses a (dynamically sized) array
+ C Slice Helps 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!
+ C SmoothTransition Changes the samples at the beginning or at the end to slowly ramp up the volume
+ C SOSFilter Second 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 float (and not a integer type) as type parameter
+ C SoundGenerator Base class to define the abstract interface for the sound generating classes
+ C SoundGeneratorModulation SoundGenerator using the ModulationBaseClass to generate the samples
+ C SPDIFConfig SPDIF configuration
+ C SPDIFOutput Output as 16 bit stereo SPDIF on the I2S data output pin
+ C SquareWaveGenerator Generates a square wave sound
+ C Stack LIFO Stack which is based on a List
+ C StdioStream Direct binary Audio Output to stdout. On linux you can hear the audio e.g. with ./generator | aplay -f cd or reading data from stdin
+ C STKChorus Chorus Effect
+ C STKChowningReverb John Chowning's reverberator class
+ C STKEcho Echo Effect
+ C STKEffect Use any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK
+ C STKFreeVerb Jezar at Dreampoint's FreeVerb, implemented in STK
+ C STKGenerator The Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language. You need to install https://github.com/pschatzmann/Arduino-STK
- C STKLentPitShift Pitch shifter effect class based on the Lent algorithm
- C STKNReverb CCRMA's NRev reverberator class
- C STKPerryReverb Perry's simple reverberator class
- C STKPitShift Simple Pitch shifter effect class: This class implements a simple pitch shifter using a delay line
- C STKStream STK Stream for Instrument or Voicer
- C Str A 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!
- C Stream
- C StreamCopyT Typed Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples
- C StreamingDecoder A Streaming Decoder where we provide both the input and output as streams
- C StreamingDecoderAdapter Converts any AudioDecoder to a StreamingDecoder
- C StrExt Str which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting
- C SynchronizedBuffer Wrapper class that can turn any Buffer into a thread save implementation
- C SynchronizedNBuffer NBuffer which uses some RTOS queues to manage the available and filled buffers
- ► C Synthesizer A simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels
- C KeyParameter
- C SynthesizerKey Arduino GPIO pin to note assossiation
- C Task FreeRTOS task
- C TestGenerator Generates a test signal which is easy to check because the values are incremented or decremented by 1
- C TfLiteAbstractRecognizeCommands Base 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
- C TfLiteAudioErrorReporter Error Reporter using the Audio Tools Logger
- C TfLiteAudioStream TfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data)
- C TfLiteAudioStreamBase Astract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers
- C TfLiteConfig Configuration settings for TfLiteAudioStream
- C TfLiteMicroSpeachWriter TfLiteMicroSpeachWriter for Audio Data
- ► C TfLiteMicroSpeechRecognizeCommands This 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
- C Result
- C TfLiteQuantizer Quantizer that helps to quantize and dequantize between float and int8
- C TfLiteReader Input class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce
- C TfLiteSineReader Generate a sine output from a model that was trained on the sine method. (=hello_world)
- C TfLiteWriter Output class which interprets audio data if TfLiteAudioStream is treated as audio sink
- C Throttle Throttle the sending or receiving of the audio data to limit it to the indicated sample rate
- C ThrottleConfig Configure Throttle setting
- C TimedStream AudioStream 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!
- C TimerAlarmRepeating Common Interface definition for TimerAlarmRepeating
- C TimerAlarmRepeatingDriverAVR Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
- C TimerAlarmRepeatingDriverBase
- C TimerAlarmRepeatingDriverESP8266 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
- C TimerAlarmRepeatingDriverMBED Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
- C TimerAlarmRepeatingDriverRenesas Repeating 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);
- C TimerAlarmRepeatingDriverSTM32 STM32 Repeating Timer functions for repeated execution: Please use the typedef TimerAlarmRepeating . By default the TIM1 is used
- C TimerCallbackAudioStream Callback 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
- C TimerCallbackAudioStreamInfo TimerCallbackAudioStream Configuration
- C TransformationReader ConverterStream Helper class which implements the readBytes with the help of write
- C Tremolo Tremolo AudioEffect
- C Triangle Triangle FFT Window function
- C UDPStream A UDP class which makes sure that we can use UDP as AudioSource and AudioSink. By default the WiFiUDP object is used and we login to wifi if the ssid and password is provided and we are not already connected
- C Url URL parser which breaks a full url string up into its individual parts
- C URLHistory
- C URLLoaderHLS
- C URLLoaderHLSBase Abstract API for URLLoaderHLS
- C URLLoaderHLSOutput URLLoader which saves the HLS segments to the indicated output
- C URLStream Represents the content of a URL as Stream . We use the WiFi.h API
- C URLStreamBuffered URLStream implementation for the ESP32 based on a separate FreeRTOS task
- C VariableSpeedRingBuffer Optimized 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
- C VariableSpeedRingBuffer180 Varialbe 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
- C VariableSpeedRingBufferSimple Very 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
- C VBANConfig
- C VBANStream VBAN 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
- ► C Vector Vector 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
- C iterator Iterator for the Vector class
- C VideoAudioBufferedSync Logic 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
- C VideoAudioSync Logic 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
- C VideoOutput Abstract class for video playback. This class is used to assemble a complete video frame in memory
- C VolumeControl Abstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal
- C VolumeMeter A simple class to determine the volume. You can use it as final output or as output or input in your audio chain
- C VolumeStream Adjust 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
- C VolumeStreamConfig Config for VolumeStream
- C VolumeSupport Supports the setting and getting of the volume
- C VorbisDecoder Vorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-tremor
- C VS1053Config Configuration for VS1053Stream
- C VS1053Stream VS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration;
- C WAVAudioInfo Sound information which is available in the WAV header
- C WAVDecoder A 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
- C WAVEncoder A 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
- C WAVFormatX
- C WAVHeader Parser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE
- C WavIMAAudioInfo Sound information which is available in the WAV header - adjusted for IMA ADPCM
- C WavIMADecoder Obsolete: 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!
- C WavIMAHeader
- C Welch Welch FFT Window function
- C WhiteNoiseGenerator Generates a random noise sound with the help of rand() function
- C WindowFunction FFT Window Function
- C WM8960Config Configuration for WM8960
- C WM8960Stream Stream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960
+ C STKLentPitShift Pitch shifter effect class based on the Lent algorithm
+ C STKNReverb CCRMA's NRev reverberator class
+ C STKPerryReverb Perry's simple reverberator class
+ C STKPitShift Simple Pitch shifter effect class: This class implements a simple pitch shifter using a delay line
+ C STKStream STK Stream for Instrument or Voicer
+ C Str A 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!
+ C Stream
+ C StreamCopyT Typed Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples
+ C StreamingDecoder A Streaming Decoder where we provide both the input and output as streams
+ C StreamingDecoderAdapter Converts any AudioDecoder to a StreamingDecoder
+ C StrExt Str which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting
+ C SynchronizedBuffer Wrapper class that can turn any Buffer into a thread save implementation
+ C SynchronizedNBuffer NBuffer which uses some RTOS queues to manage the available and filled buffers
+ ► C Synthesizer A simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels
+ C KeyParameter
+ C SynthesizerKey Arduino GPIO pin to note assossiation
+ C Task FreeRTOS task
+ C TestGenerator Generates a test signal which is easy to check because the values are incremented or decremented by 1
+ C TfLiteAbstractRecognizeCommands Base 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
+ C TfLiteAudioErrorReporter Error Reporter using the Audio Tools Logger
+ C TfLiteAudioStream TfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data)
+ C TfLiteAudioStreamBase Astract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers
+ C TfLiteConfig Configuration settings for TfLiteAudioStream
+ C TfLiteMicroSpeachWriter TfLiteMicroSpeachWriter for Audio Data
+ ► C TfLiteMicroSpeechRecognizeCommands This 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
+ C Result
+ C TfLiteQuantizer Quantizer that helps to quantize and dequantize between float and int8
+ C TfLiteReader Input class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce
+ C TfLiteSineReader Generate a sine output from a model that was trained on the sine method. (=hello_world)
+ C TfLiteWriter Output class which interprets audio data if TfLiteAudioStream is treated as audio sink
+ C Throttle Throttle the sending or receiving of the audio data to limit it to the indicated sample rate
+ C ThrottleConfig Configure Throttle setting
+ C TimedStream AudioStream 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!
+ C TimerAlarmRepeating Common Interface definition for TimerAlarmRepeating
+ C TimerAlarmRepeatingDriverAVR Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
+ C TimerAlarmRepeatingDriverBase
+ C TimerAlarmRepeatingDriverESP8266 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
+ C TimerAlarmRepeatingDriverMBED Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating
+ C TimerAlarmRepeatingDriverRenesas Repeating 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);
+ C TimerAlarmRepeatingDriverSTM32 STM32 Repeating Timer functions for repeated execution: Please use the typedef TimerAlarmRepeating . By default the TIM1 is used
+ C TimerCallbackAudioStream Callback 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
+ C TimerCallbackAudioStreamInfo TimerCallbackAudioStream Configuration
+ C TransformationReader ConverterStream Helper class which implements the readBytes with the help of write
+ C Tremolo Tremolo AudioEffect
+ C Triangle Triangle FFT Window function
+ C UDPStream A UDP class which makes sure that we can use UDP as AudioSource and AudioSink. By default the WiFiUDP object is used and we login to wifi if the ssid and password is provided and we are not already connected
+ C Url URL parser which breaks a full url string up into its individual parts
+ C URLHistory
+ C URLLoaderHLS
+ C URLLoaderHLSBase Abstract API for URLLoaderHLS
+ C URLLoaderHLSOutput URLLoader which saves the HLS segments to the indicated output
+ C URLStream Represents the content of a URL as Stream . We use the WiFi.h API
+ C URLStreamBuffered URLStream implementation for the ESP32 based on a separate FreeRTOS task
+ C VariableSpeedRingBuffer Optimized 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
+ C VariableSpeedRingBuffer180 Varialbe 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
+ C VariableSpeedRingBufferSimple Very 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
+ C VBANConfig
+ C VBANStream VBAN 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
+ ► C Vector Vector 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
+ C iterator Iterator for the Vector class
+ C VideoAudioBufferedSync Logic 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
+ C VideoAudioSync Logic 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
+ C VideoOutput Abstract class for video playback. This class is used to assemble a complete video frame in memory
+ C VolumeControl Abstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal
+ C VolumeMeter A simple class to determine the volume. You can use it as final output or as output or input in your audio chain
+ C VolumeStream Adjust 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
+ C VolumeStreamConfig Config for VolumeStream
+ C VolumeSupport Supports the setting and getting of the volume
+ C VorbisDecoder Vorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-tremor
+ C VS1053Config Configuration for VS1053Stream
+ C VS1053Stream VS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration;
+ C WAVAudioInfo Sound information which is available in the WAV header
+ C WAVDecoder A 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
+ C WAVEncoder A 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
+ C WAVFormatX
+ C WAVHeader Parser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE
+ C WavIMAAudioInfo Sound information which is available in the WAV header - adjusted for IMA ADPCM
+ C WavIMADecoder Obsolete: 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!
+ C WavIMAHeader
+ C Welch Welch FFT Window function
+ C WhiteNoiseGenerator Generates a random noise sound with the help of rand() function
+ C WindowFunction FFT Window Function
+ C WM8960Config Configuration for WM8960
+ C WM8960Stream Stream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960
► N audiotools_rtsp AudioTools internal: rtsp
C OurRTSPClient
C OurSink
diff --git a/search/all_1.js b/search/all_1.js
index 12ff63ddd..bb9a105fd 100644
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- ['addurl_42',['addURL',['../classaudio__tools_1_1_audio_source_dynamic_u_r_l.html#a91e2dc3e370b5eaa7d2758abba2b1d31',1,'audio_tools::AudioSourceDynamicURL']]],
+ ['addurl_41',['addURL',['../classaudio__tools_1_1_audio_source_dynamic_u_r_l.html#a91e2dc3e370b5eaa7d2758abba2b1d31',1,'audio_tools::AudioSourceDynamicURL']]],
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['addvolumeactions_43',['addVolumeActions',['../classaudio__tools_1_1_audio_board_stream.html#a9a6c8645266db32c2f6fd8257cc51f06',1,'audio_tools::AudioBoardStream']]],
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index e357ac564..0650875ba 100644
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