diff --git a/_i2_s_e_s_p32_8h_source.html b/_i2_s_e_s_p32_8h_source.html index 4defa4ec92..62802e5506 100644 --- a/_i2_s_e_s_p32_8h_source.html +++ b/_i2_s_e_s_p32_8h_source.html @@ -72,7 +72,7 @@
1 #pragma once
2 
3 #include "AudioConfig.h"
-
4 #if defined(ESP32) && defined(USE_I2S) && ESP_IDF_VERSION < ESP_IDF_VERSION_VAL(5, 0 , 0)
+
4 #if defined(ESP32) && defined(USE_I2S) && ESP_IDF_VERSION < ESP_IDF_VERSION_VAL(5, 0 , 0) || defined(DOXYGEN)
5 
6 #include "AudioI2S/I2SConfig.h"
7 #include "driver/i2s.h"
@@ -84,60 +84,60 @@
13 
14 namespace audio_tools {
15 
-
22 class I2SDriverESP32 {
+
22 class I2SDriverESP32 {
23 
24  friend class AnalogAudio;
-
25  friend class AudioKitStream;
+
25  friend class AudioKitStream;
26 
27  public:
28 
-
30  I2SConfig defaultConfig(RxTxMode mode) {
-
31  I2SConfig c(mode);
+
30  I2SConfig defaultConfig(RxTxMode mode) {
+
31  I2SConfig c(mode);
32  return c;
33  }
34 
-
36  bool begin(RxTxMode mode) {
-
37  return begin(defaultConfig(mode));
+
36  bool begin(RxTxMode mode) {
+
37  return begin(defaultConfig(mode));
38  }
39 
-
41  bool begin() {
-
42  return !is_started ? begin(cfg) : true;
+
41  bool begin() {
+
42  return !is_started ? begin(cfg) : true;
43  }
44 
-
46  bool begin(I2SConfig cfg) {
+
46  bool begin(I2SConfig cfg) {
47  TRACED();
48  this->cfg = cfg;
-
49  switch(cfg.rx_tx_mode){
+
49  switch(cfg.rx_tx_mode){
50  case TX_MODE:
-
51  return begin(cfg, cfg.pin_data, I2S_PIN_NO_CHANGE);
+
51  return begin(cfg, cfg.pin_data, I2S_PIN_NO_CHANGE);
52  case RX_MODE:
53  // usually we expet cfg.pin_data but if the used assinged rx we might consider this one
-
54  return begin(cfg, I2S_PIN_NO_CHANGE, cfg.pin_data!=I2S_PIN_NO_CHANGE?cfg.pin_data:cfg.pin_data_rx );
+
54  return begin(cfg, I2S_PIN_NO_CHANGE, cfg.pin_data!=I2S_PIN_NO_CHANGE?cfg.pin_data:cfg.pin_data_rx );
55  default:
-
56  return begin(cfg, cfg.pin_data, cfg.pin_data_rx);
+
56  return begin(cfg, cfg.pin_data, cfg.pin_data_rx);
57  }
58  LOGE("Did not expect go get here");
59  }
60 
-
62  int available() {
+
62  int available() {
63  return I2S_BUFFER_COUNT*I2S_BUFFER_SIZE;
64  }
65 
-
67  int availableForWrite() {
+
67  int availableForWrite() {
68  return I2S_BUFFER_COUNT*I2S_BUFFER_SIZE;
69  }
70 
-
72  void end(){
+
72  void end(){
73  TRACED();
74  i2s_driver_uninstall(i2s_num);
75  is_started = false;
76  }
77 
-
79  I2SConfig config() {
+
79  I2SConfig config() {
80  return cfg;
81  }
82 
-
84  size_t writeBytes(const void *src, size_t size_bytes){
+
84  size_t writeBytes(const void *src, size_t size_bytes){
85  TRACED();
86 
87  size_t result = 0;
@@ -147,7 +147,7 @@
91  }
92  LOGD("i2s_write %d -> %d bytes", size_bytes, result);
93  } else {
-
94  result = I2SDriverESP32::writeExpandChannel(i2s_num, cfg.bits_per_sample, src, size_bytes);
+
94  result = I2SDriverESP32::writeExpandChannel(i2s_num, cfg.bits_per_sample, src, size_bytes);
95  }
96  return result;
97  }
@@ -189,7 +189,7 @@
133  }
134 
135  protected:
-
136  I2SConfig cfg = defaultConfig(RX_MODE);
+
136  I2SConfig cfg = defaultConfig(RX_MODE);
137  i2s_port_t i2s_num;
138  i2s_config_t i2s_config;
139  bool is_started = false;
@@ -203,7 +203,7 @@
147  return false;
148  }
149 
-
151  bool begin(I2SConfig cfg, int txPin, int rxPin) {
+
151  bool begin(I2SConfig cfg, int txPin, int rxPin) {
152  TRACED();
153  cfg.logInfo();
154  this->cfg = cfg;
@@ -231,7 +231,7 @@
176 
177  // We make sure that we can reconfigure
178  if (is_started) {
-
179  end();
+
179  end();
180  LOGD("%s", "I2S restarting");
181  }
182 
@@ -278,7 +278,7 @@
223  }
224 
225 
-
227  static size_t writeExpandChannel(i2s_port_t i2s_num, const int bits_per_sample, const void *src, size_t size_bytes){
+
227  static size_t writeExpandChannel(i2s_port_t i2s_num, const int bits_per_sample, const void *src, size_t size_bytes){
228  size_t result = 0;
229  int j;
230  switch(bits_per_sample){
@@ -425,6 +425,21 @@
371 }
372 
373 #endif
+
audio_tools::AudioKitStream
AudioKit Stream which uses the https://github.com/pschatzmann/arduino-audiokit library.
Definition: AudioKit.h:189
+
audio_tools::I2SConfig
configuration for all common i2s settings
Definition: I2SConfig.h:52
+
audio_tools::I2SConfig::rx_tx_mode
RxTxMode rx_tx_mode
public settings
Definition: I2SConfig.h:94
+
audio_tools::I2SDriverESP32
Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels.
Definition: I2SESP32.h:22
+
audio_tools::I2SDriverESP32::writeExpandChannel
static size_t writeExpandChannel(i2s_port_t i2s_num, const int bits_per_sample, const void *src, size_t size_bytes)
writes the data by making sure that we send 2 channels
Definition: I2SESP32.h:227
+
audio_tools::I2SDriverESP32::available
int available()
we assume the data is already available in the buffer
Definition: I2SESP32.h:62
+
audio_tools::I2SDriverESP32::begin
bool begin()
starts the DAC with the current config - if not started yet. If I2S has been started there is no acti...
Definition: I2SESP32.h:41
+
audio_tools::I2SDriverESP32::defaultConfig
I2SConfig defaultConfig(RxTxMode mode)
Provides the default configuration.
Definition: I2SESP32.h:30
+
audio_tools::I2SDriverESP32::availableForWrite
int availableForWrite()
We limit the write size to the buffer size.
Definition: I2SESP32.h:67
+
audio_tools::I2SDriverESP32::config
I2SConfig config()
provides the actual configuration
Definition: I2SESP32.h:79
+
audio_tools::I2SDriverESP32::end
void end()
stops the I2C and unistalls the driver
Definition: I2SESP32.h:72
+
audio_tools::I2SDriverESP32::begin
bool begin(I2SConfig cfg)
starts the DAC
Definition: I2SESP32.h:46
+
audio_tools::I2SDriverESP32::begin
bool begin(RxTxMode mode)
starts the DAC with the default config
Definition: I2SESP32.h:36
+
audio_tools::I2SDriverESP32::writeBytes
size_t writeBytes(const void *src, size_t size_bytes)
writes the data to the I2S interface
Definition: I2SESP32.h:84
+
audio_tools::I2SDriverESP32::begin
bool begin(I2SConfig cfg, int txPin, int rxPin)
starts the DAC
Definition: I2SESP32.h:151
audio_tools::int24_4bytes_t
24bit integer which is used for I2S sound processing. The values are represented as int32_t,...
Definition: Int24_4bytes_t.h:16
audio_tools::RxTxMode
RxTxMode
Audio Source (TX_MODE) or Audio Sink (RX_MODE). RXTX_MODE is Source and Sink at the same time!
Definition: AudioTypes.h:24
audio_tools
Generic Implementation of sound input and output for desktop environments using portaudio.
Definition: AnalogAudio.h:10
diff --git a/_i2_s_e_s_p32_v1_8h_source.html b/_i2_s_e_s_p32_v1_8h_source.html index 3dd780a248..37a8e29e88 100644 --- a/_i2_s_e_s_p32_v1_8h_source.html +++ b/_i2_s_e_s_p32_v1_8h_source.html @@ -72,7 +72,7 @@
1 #pragma once
2 
3 #include "AudioConfig.h"
-
4 #if defined(ESP32) && defined(USE_I2S) && ESP_IDF_VERSION >= ESP_IDF_VERSION_VAL(5, 0, 0)
+
4 #if defined(ESP32) && defined(USE_I2S) && ESP_IDF_VERSION >= ESP_IDF_VERSION_VAL(5, 0, 0) || defined(DOXYGEN)
5 
6 #include "AudioI2S/I2SConfig.h"
7 #include "driver/i2s_pdm.h"
diff --git a/_i2_s_stream_8h_source.html b/_i2_s_stream_8h_source.html index 0d81875de3..552ce95d7c 100644 --- a/_i2_s_stream_8h_source.html +++ b/_i2_s_stream_8h_source.html @@ -108,16 +108,16 @@
46 #endif
47 
49  I2SConfig defaultConfig(RxTxMode mode=TX_MODE) {
-
50  return i2s.defaultConfig(mode);
+
50  return i2s.defaultConfig(mode);
51  }
52 
53  bool begin() {
-
54  return i2s.begin();
+
54  return i2s.begin();
55  }
56 
58  bool begin(I2SConfig cfg) {
59  TRACED();
-
60  bool result = i2s.begin(cfg);
+
60  bool result = i2s.begin(cfg);
61  // unmute
62  mute(false);
63  return result;
@@ -126,13 +126,13 @@
67  void end() {
68  TRACED();
69  mute(true);
-
70  i2s.end();
+
70  i2s.end();
71  }
72 
74  virtual void setAudioInfo(AudioInfo info) {
75  TRACEI();
76  AudioStream::setAudioInfo(info);
-
77  I2SConfig cfg = i2s.config();
+
77  I2SConfig cfg = i2s.config();
78  if (cfg.sample_rate != info.sample_rate
79  || cfg.channels != info.channels
80  || cfg.bits_per_sample != info.bits_per_sample) {
@@ -141,14 +141,14 @@
83  cfg.channels = info.channels;
84  cfg.logInfo("I2SStream");
85 
-
86  i2s.end();
-
87  i2s.begin(cfg);
+
86  i2s.end();
+
87  i2s.begin(cfg);
88  }
89  }
90 
92  virtual size_t write(const uint8_t *buffer, size_t size) {
93  LOGD("I2SStream::write: %d", size);
-
94  return i2s.writeBytes(buffer, size);
+
94  return i2s.writeBytes(buffer, size);
95  }
96 
98  virtual size_t readBytes( uint8_t *data, size_t length) override {
@@ -156,21 +156,21 @@
100  }
101 
103  virtual int available() override {
-
104  return i2s.available();
+
104  return i2s.available();
105  }
106 
108  virtual int availableForWrite() override {
-
109  return i2s.availableForWrite();
+
109  return i2s.availableForWrite();
110  }
111 
112  void flush() override {}
113 
-
115  I2SDriver* driver() {
+
115  I2SDriver* driver() {
116  return &i2s;
117  }
118 
119  protected:
-
120  I2SDriver i2s;
+
120  I2SDriver i2s;
121  int mute_pin;
122 
124  void mute(bool is_mute){
@@ -188,14 +188,14 @@
136 #endif
audio_tools::AudioStream
Base class for all Audio Streams. It support the boolean operator to test if the object is ready with...
Definition: AudioStreams.h:47
audio_tools::I2SConfig
configuration for all common i2s settings
Definition: I2SConfig.h:52
-
audio_tools::I2SDriverESP32V1
Basic I2S API for the ESP32 (using the new API). https://docs.espressif.com/projects/esp-idf/en/v5....
Definition: I2SESP32V1.h:21
-
audio_tools::I2SDriverESP32V1::available
int available()
we assume the data is already available in the buffer
Definition: I2SESP32V1.h:57
-
audio_tools::I2SDriverESP32V1::defaultConfig
I2SConfig defaultConfig(RxTxMode mode)
Provides the default configuration.
Definition: I2SESP32V1.h:25
-
audio_tools::I2SDriverESP32V1::availableForWrite
int availableForWrite()
We limit the write size to the buffer size.
Definition: I2SESP32V1.h:60
-
audio_tools::I2SDriverESP32V1::config
I2SConfig config()
provides the actual configuration
Definition: I2SESP32V1.h:78
-
audio_tools::I2SDriverESP32V1::end
void end()
stops the I2C and unistalls the driver
Definition: I2SESP32V1.h:63
-
audio_tools::I2SDriverESP32V1::begin
bool begin(RxTxMode mode)
starts the DAC with the default config
Definition: I2SESP32V1.h:31
-
audio_tools::I2SDriverESP32V1::writeBytes
size_t writeBytes(const void *src, size_t size_bytes)
writes the data to the I2S interface
Definition: I2SESP32V1.h:81
+
audio_tools::I2SDriverESP32
Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels.
Definition: I2SESP32.h:22
+
audio_tools::I2SDriverESP32::available
int available()
we assume the data is already available in the buffer
Definition: I2SESP32.h:62
+
audio_tools::I2SDriverESP32::defaultConfig
I2SConfig defaultConfig(RxTxMode mode)
Provides the default configuration.
Definition: I2SESP32.h:30
+
audio_tools::I2SDriverESP32::availableForWrite
int availableForWrite()
We limit the write size to the buffer size.
Definition: I2SESP32.h:67
+
audio_tools::I2SDriverESP32::config
I2SConfig config()
provides the actual configuration
Definition: I2SESP32.h:79
+
audio_tools::I2SDriverESP32::end
void end()
stops the I2C and unistalls the driver
Definition: I2SESP32.h:72
+
audio_tools::I2SDriverESP32::begin
bool begin(RxTxMode mode)
starts the DAC with the default config
Definition: I2SESP32.h:36
+
audio_tools::I2SDriverESP32::writeBytes
size_t writeBytes(const void *src, size_t size_bytes)
writes the data to the I2S interface
Definition: I2SESP32.h:84
audio_tools::I2SStream
We support the Stream interface for the I2S access. In addition we allow a separate mute pin which mi...
Definition: I2SStream.h:32
audio_tools::I2SStream::readBytes
virtual size_t readBytes(uint8_t *data, size_t length) override
Reads the audio data.
Definition: I2SStream.h:98
audio_tools::I2SStream::write
virtual size_t write(const uint8_t *buffer, size_t size)
Writes the audio data to I2S.
Definition: I2SStream.h:92
diff --git a/annotated.html b/annotated.html index 86fe383799..a17997645b 100644 --- a/annotated.html +++ b/annotated.html @@ -313,262 +313,263 @@  CHttpRequestSimple API to process get, put, post, del http requests I tried to use Arduino HttpClient, but I did not manage to extract the mime type from streaming get requests  CHttpRequestHeaderReading and writing of Http Requests  CI2SConfigConfiguration for all common i2s settings - ►CI2SDriverESP32V1Basic I2S API for the ESP32 (using the new API). https://docs.espressif.com/projects/esp-idf/en/v5.0.1/esp32/api-reference/peripherals/i2s.html#i2s-communication-mode - CDriverCommon - CDriverI2S - CI2SDriverESP8266Basic I2S API - for the ESP8266 Only 16 bits are supported ! - CI2SDriverNanoBLEBasic I2S API - for the Arduino Nano BLE Sense See https://content.arduino.cc/assets/Nano_BLE_MCU-nRF52840_PS_v1.1.pdf Douplex mode (RXTX_MODE) is currently not supported, but it should be quite easy to implement - CI2SDriverSAMDBasic I2S API - for the SAMD - CI2SDriverSTM32Basic I2S API - for the STM32 Depends on https://github.com/pschatzmann/stm32f411-i2s We just add a write and read buffer and pass some parameters to the STM32 API! - CI2SStreamWe support the Stream interface for the I2S access. In addition we allow a separate mute pin which might also be used to drive a LED.. - CICYStreamIcecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method - CICYStreamBufferedICYStream implementation for the ESP32 based on a FreeRTOS task This is a Icecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method - CICYUrlSetupResolve icy-metaint from HttpRequest and execute metadata callbacks - CID3v1 - CID3v1Enhanced - CID3v2 - CID3v2Frame - CID3v2FrameString - CIIRIIRFilter Converted from https://github.com/tttapa/Filters/blob/master/src/IIRFilter.h - CILBCDecoderDecoder for iLBC. Depends on https://github.com/pschatzmann/libilbc - CILBCEncoderEncoder for iLBC - Depends on https://github.com/pschatzmann/libopenilbc - CIMAState - CInputMergeMerges multiple input channels. The input must be mono! So if you provide 2 mono channels you get a stereo signal as result with the left channel from channel 0 and the right from channel 1 - CInputMixerMixerStream is mixing the input from Multiple Input Streams. All streams must have the same audo format (sample rate, channels, bits per sample) - Cint24_3bytes_t24bit integer which is used for I2S sound processing. The values are really using 3 bytes. It works only on little endian machines! - Cint24_4bytes_t24bit integer which is used for I2S sound processing. The values are represented as int32_t, but only 3 bytes are used. If you assign values which are too big, they are clipped - CJpegOpenCVDisplay image with opencv to be used on the desktop - CJpegTFTDisplay jpeg image using https://github.com/Bodmer/TFT_eSPI and https://github.com/Bodmer/JPEGDecoder - CJupyterAudioTOutput to Jupyter. We write the data just to a file from where we can load the data again for different representations - CLastSampleFaderIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0. Typless implementation - CLastSampleFaderTIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0 - CLC3DecoderDecoder for LC3. Depends on https://github.com/pschatzmann/arduino-liblc3 - CLC3EncoderEncoder for LC3 - Depends on https://github.com/pschatzmann/arduino-liblc3 - CLEDOutputLED output using the FastLED library - CLEDOutputConfig - CLEDOutputUnoR4LED output using the R4 LED matrix library - CLEDOutputUnoR4Config - CLinearVolumeControlThe simplest possible implementation of a VolumeControl: The input = output which describes a linear curve. You would use this implementation if you physically connect an audio pot! - ►CListDouble linked list - CIterator - CNode - CLockA simple RIA locking class for the ESP32 using _lock_t - CLogarithmicVolumeControlParametric Logarithmic volume control. Using the formula pow(b,input) * a - a, where b is b = pow(((1/ym)-1), 2) and a is a = 1.0 / (b - 1.0). The parameter ym is determining the steepness. See https://electronics.stackexchange.com/questions/304692/formula-for-logarithmic-audio-taper-pot - CMaximilianAudioTools integration with Maximilian - CMeasuringStreamClass which measures the truput - ►CMedianFilterAn embedded friendly, fast one-dimensional median filter algorithm implementation in C and C++ Useful for spike and noise removal from analog signals or other DSP Also known as "salt-and-pepper noise" or "impulse noise" filter - CsMedianFilter_t - CsMedianNode_t - CMemoryManagerMemoryManager which activates the use of external SPIRAM memory. When external memory is in use, the allocation strategy is to initially try to satisfy smaller allocation requests with internal memory and larger requests with external memory. This sets the limit between the two, as well as generally enabling allocation in external memory - CMemoryOutputWrites to a preallocated memory - CMemoryStreamA simple Stream implementation which is backed by allocated memory - ►CMetaDataFilterClass which filters out ID3v1 and ID3v2 Metadata and provides only the audio data to the decoder - CID3v2ID3 verion 2 TAG Header (10 bytes) - CMetaDataICYIcecast/Shoutcast Metadata Handling. Metadata class which splits the data into audio and metadata. The result is provided via callback methods. see https://www.codeproject.com/Articles/11308/SHOUTcast-Stream-Ripper - CMetaDataID3Simple ID3 Meta Data Parser which supports ID3 V1 and V2 and implements the Stream interface. You just need to set the callback(s) to receive the result and copy the audio data to this stream - CMetaDataID3BaseID3 Meta Data Common Functionality - CMetaDataID3V1Simple ID3 Meta Data API which supports ID3 V1 - CMetaDataID3V2Simple ID3 Meta Data API which supports ID3 V2: We only support the "TALB", "TOPE", "TIT2", "TCON" tags - CMetaDataOutputID3 and Icecast/Shoutcast metadata output support. Just write the audio data to an object of this class and receive the metadata via the callback - CMiniAudioConfigConfiguration for MiniAudio - CMiniAudioStreamMiniAudio: https://miniaud.io/ - CModulationBaseClassClass provides a wave table that can be populated with a number of preallocated waveforms. These can be used to generate audio in themselves or to modulate The parameters of another effect. Class initialised with sample rate - CMozziConfigMozzi Configuration for input or output stream - CMozziGeneratorSupport for https://sensorium.github.io/Mozzi/ Define your updateControl() method. Define your updateAudio() method. Start by calling begin(control_rate) do not call audioHook(); in the loop ! - CMozziStreamWe use the output functionality of Mozzi to output audio data. We expect the data as array of int16_t with one or two channels. Though we support the setting of a sample rate, we recommend to use the default sample rate from Mozzi which is available with the AUDIO_RATE define - CMP3DecoderHelixMP3 Decoder using libhelix: https://github.com/pschatzmann/arduino-libhelix This is basically just a simple wrapper to provide AudioInfo and AudioInfoSupport - CMP3DecoderMADMP3 Decoder using https://github.com/pschatzmann/arduino-libmad - CMP3DecoderMiniMP3 Decoder using https://github.com/pschatzmann/minimp3. This decoder does not provide any good results and it is not suited to decode any audio above 32000 on an ESP32. So the sample rate is limited by the MINIMP3_MAX_SAMPLE_RATE variable - CMP3EncoderLAMEEncodes PCM data to the MP3 format and writes the result to a stream This is basically just a wrapper using https://github.com/pschatzmann/arduino-liblame - CMP4AtomRepresents a single MPEG4 atom - CMP4ParseBuffer - CMTSDecoderMPEG-TS (MTS) decoder. Extracts the AAC audio data from a MPEG-TS (MTS) data stream. You can define the relevant stream types via the API. Required dependency: https://github.com/pschatzmann/arduino-tsdemux - CMultiConverterCombines multiple converters - CMultiOutputReplicates the output to multiple destinations - CMusicalNotesDetermination of the frequency of a music note - CNano_BLE_freq_infoMapping Frequency constants to available frequencies - CNano_BLE_ratio_infoMapping from Ratio Constants to frequency ratios - CNBufferA lock free N buffer. If count=2 we create a DoubleBuffer, if count=3 a TripleBuffer etc - CNoFilterNo change to the input - CNOPConverterDummy converter which does nothing - CNullStreamThe Arduino Stream which provides silence and simulates a null device when used as audio target or audio source - CNumberConverterConverts from a source to a target number with a different type - CNumberFormatConverterStreamConverter which converts between bits_per_sample and 16 bits. The templated NumberFormatConverterStreamT class is used based on the information provided by the bits_per_sample in the configuration - CNumberFormatConverterStreamTConverter which converts from source bits_per_sample to target bits_per_sample - CNumberReaderReads n numbers from an Arduino Stream - CNuttallNuttall FFT Window function - COggContainerDecoderDecoder for Ogg Container. Decodes a packet from an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass and overwrite the beginOfSegment() method to implement your own headers Dependency: https://github.com/pschatzmann/arduino-libopus - COggContainerEncoderEncoder for Ogg Container. Encodes a packet for an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass ond overwrite the writeHeader() method to implement your own header logic. When an optional encoder is specified in the constructor we package the encoded data. Dependency: https://github.com/pschatzmann/arduino-libopus - COggContainerOutputOutput class for the OggContainerEncoder. Each write is ending up as container entry - COnOffOutputSwitched Output class. Class which can be used to filter the output out based on a switch: If the switch is on the output is forwarded. If it is off it is just ignored - COpusAudioDecoderOpusAudioDecoder: Depends on https://github.com/pschatzmann/arduino-libopus.git - COpusAudioEncoderOpusAudioEncoder: Dependens on https://github.com/pschatzmann/arduino-libopus.git - COpusEncoderSettingsSetting for Opus Encoder where the following values are valid: -1 indicates that the default value should be used and that this codec is not setting the value - COpusOggDecoderOpus Decoder which uses the Ogg Container. See https://datatracker.ietf.org/doc/html/rfc7845. The audio data is transmitted in frames and the header information contains the sampler rate, channels and other critical info. Dependency: https://github.com/pschatzmann/arduino-libopus - COpusOggEncoderOpus Encoder which uses the Ogg Container: see https://datatracker.ietf.org/doc/html/rfc7845 Dependency: https://github.com/pschatzmann/arduino-libopus - COpusOggWriter - COpusSettingsSetting for Opus Decoder - COutputMixerMixing of multiple outputs to one final output - CParameterA constant value - CParseBufferWe try to keep the necessary buffer for parsing as small as possible, The data() method provides the start of the actual data and with consume we remove the processed data from the buffer to make space again - CParseObject - CPinInfoESP32Information for a PIN - CPinkNoiseGeneratorGenerates pink noise - CPitchShiftShifts the pitch by the indicated step size: e.g. 2 doubles the pitch - CPitchShiftInfoConfiguration for PitchShiftOutput: set the pitch_shift to define the shift - CPitchShiftOutputPitch Shift: Shifts the frequency up or down w/o impacting the length! We reduce the channels to 1 to calculate the pitch shift and provides the pitch shifted result in the correct number of channels. The pitch shifting is done with the help of a buffer that can have potentially multiple implementations - CPoppingSoundRemoverBig value gaps (at the beginning and the end of a recording) can lead to some popping sounds. We will try to set the values to 0 until the first transition thru 0 of the audio curve - CPortAudioConfigPortAudio information - CPortAudioStreamArduino Audio Stream using PortAudio - CPrint - CProcessedResult - CProgressStreamGeneric calss to measure the the total bytes which were processed in order to calculate the progress as a percentage of the total size - CProgressStreamInfoConfiguration for ProgressStream - CPWMAudioOutputCommon functionality for PWM output. Please use the PWMAudioOutput typedef instead which references the implementation - CPWMConfigConfiguration data for PWM audio output - CPWMDriverAVRExperimental: Audio output to PWM pins for the AVR. The AVR supports only up to 2 channels - CPWMDriverESP32Audio output to PWM pins for the ESP32. The ESP32 supports up to 16 channels - CPWMDriverMBEDAudio output to PWM pins for MBED based Arduino implementations - CPWMDriverRenesasAudio output to PWM pins for Renesas based Arduino implementations - CPWMDriverSTM32Audio output to PWM pins for STM32. We use one timer to generate the sample rate and one timer for the PWM signal - CQueueFIFO Queue which is based on a List - ►CQueueLockFreeLock Free Queue - Cnode - CQueueStreamAudioStream class which stores the data in a temporary queue buffer. The queue can be consumed e.g. by a callback function by calling readBytes(); - CRECT - CRectangeRectange FFT Window function - CReformatBaseStreamBase class for chained converting streams - CResampleConfigOptional Configuration object. The critical information is the channels and the step_size. All other information is not used - CResampleStreamDynamic Resampling. We can use a variable factor to speed up or slow down the playback - CRingBufferImplements a typed Ringbuffer - CRingBufferFileAn File backed Ring Buffer that we can use to receive streaming audio. We expect an open p_file as parameter - CRingBufferStreamA Stream backed by a Ringbuffer. We can write to the end and read from the beginning of the stream - CRTSPFormatAbtXAbtX format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatAudioToolsRTSPFormat which supports the AudioInfo class - CRTSPFormatG711G711 μ-Law format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats Packet intervall: 20, frame size: any - CRTSPFormatGSMGSM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatOpusOpus format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatPCMPCM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatPCM8L8 format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPOutputWe can write PCM data to the RTSPOutput. This is encoded by the indicated encoder (e.g. SBCEncoder) and can be consumed by a RTSPServer. You have to make sure that the codec supports the provided audio format: e.g. GSM support only 8000 samples per second with one channel. Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPOutputPCMInfoPCMInfo subclass which provides the audio information from the related AudioStream Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPPCMAudioInfoPCMInfo subclass which provides the audio information from the AudioInfo parameter - CRTSPSourceFromAudioStreamSimple Facade which can turn AudioStream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPSourceStreamSimple Facade which can turn any Stream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CSBCDecoderDecoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcdec.c - CSBCEncoderEncoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcenc.c - CScaledParameterScales a dynamic parameter to the indicated range - CSDDirectWe access the files directy with an index. The index is determined by a recurseve tree walk thru the directory. Unfortunatly the SDTFAT library has it's own API which is incompatible with the SDT API - CSDIndexWe store all the relevant file names in an sequential index file. Form there we can access them via an index - CSilenceGeneratorProvides a fixed value (e.g. 0) as sound data. This can be used e.g. to test the output functionality which should optimally just output silence and no artifacts - CSilenceRemovalConverterRemoves any silence from the buffer that is longer then n samples with a amplitude below the indicated threshhold. If you process multiple channels you need to multiply the channels with the number of samples to indicate n - CSimpleChorusSimple Chorus effect with a single delay voice and mono output Chorus is effective between 15 and 20 miliseconds delay of original audio. Requires the sample rate when initialising - CSimpleContainerConfig - CSimpleContainerDataHeader - CSimpleContainerMetaDataHeader - CSimpleDelaySimple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples - CSimpleFlangerSimple Flanger Effect Consistig of a single voice flanger The flanger has an effective range between 0 and 15 miliseconds in this case dleay buffer should be set to sampleRate*3/200 Constructor requires internal delay in samples - CSimpleLPFSimpleLPF - CSimulatedAudioPotSimple simulated audio pot volume control inspired by https://eepower.com/resistor-guide/resistor-types/potentiometer-taper/# We split up the input/output curve into 2 linear pieces with a slow and a fast raising part. The slow raising part goes from (0,0) to (x,y). The fast raising part goes from (x,y) to (1,1) - CSineFromTableA sine generator based on a table. The table is created based using degrees where one full wave is 360 degrees - CSineWaveGeneratorGenerates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead - CSingleBufferA simple Buffer implementation which just uses a (dynamically sized) array - CSliceHelps to split up a big memory array into smaller slices. There are no additinal heap allocations! Example: if we have an array with 9 entries (1,2,3,4,5,6,7,8,9): slices(5) gives 2. slice(5,0) returns size 5 with 1,2,3,4,5 and slice(5,1) returns size 4 with 6,7,8,9! - CSmoothTransitionChanges the samples at the beginning or at the end to slowly ramp up the volume - CSOSFilterSecond Order Filter: Instead of manually cascading BiQuad filters, you can use a Second Order Sections filter (SOS). converted from https://github.com/tttapa/Filters/blob/master/src/SOSFilter.h Use float or double (and not a integer type) as type parameter - CSoundGeneratorBase class to define the abstract interface for the sound generating classes - CSoundGeneratorModulationSoundGenerator using the ModulationBaseClass to generate the samples - CSPDIFConfigSPDIF configuration - CSPDIFOutputOutput as 16 bit stereo SPDIF on the I2S data output pin - CSquareWaveGeneratorGenerates a square wave sound - CStackLIFO Stack which is based on a List - CStdioStream - CSTKChorusChorus Effect - CSTKChowningReverbJohn Chowning's reverberator class - CSTKEchoEcho Effect - CSTKEffectUse any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK - CSTKFreeVerbJezar at Dreampoint's FreeVerb, implemented in STK - CSTKGeneratorThe Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language.
+ CI2SDriverESP32Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels + ►CI2SDriverESP32V1Basic I2S API for the ESP32 (using the new API). https://docs.espressif.com/projects/esp-idf/en/v5.0.1/esp32/api-reference/peripherals/i2s.html#i2s-communication-mode + CDriverCommon + CDriverI2S + CI2SDriverESP8266Basic I2S API - for the ESP8266 Only 16 bits are supported ! + CI2SDriverNanoBLEBasic I2S API - for the Arduino Nano BLE Sense See https://content.arduino.cc/assets/Nano_BLE_MCU-nRF52840_PS_v1.1.pdf Douplex mode (RXTX_MODE) is currently not supported, but it should be quite easy to implement + CI2SDriverSAMDBasic I2S API - for the SAMD + CI2SDriverSTM32Basic I2S API - for the STM32 Depends on https://github.com/pschatzmann/stm32f411-i2s We just add a write and read buffer and pass some parameters to the STM32 API! + CI2SStreamWe support the Stream interface for the I2S access. In addition we allow a separate mute pin which might also be used to drive a LED.. + CICYStreamIcecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method + CICYStreamBufferedICYStream implementation for the ESP32 based on a FreeRTOS task This is a Icecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method + CICYUrlSetupResolve icy-metaint from HttpRequest and execute metadata callbacks + CID3v1 + CID3v1Enhanced + CID3v2 + CID3v2Frame + CID3v2FrameString + CIIRIIRFilter Converted from https://github.com/tttapa/Filters/blob/master/src/IIRFilter.h + CILBCDecoderDecoder for iLBC. Depends on https://github.com/pschatzmann/libilbc + CILBCEncoderEncoder for iLBC - Depends on https://github.com/pschatzmann/libopenilbc + CIMAState + CInputMergeMerges multiple input channels. The input must be mono! So if you provide 2 mono channels you get a stereo signal as result with the left channel from channel 0 and the right from channel 1 + CInputMixerMixerStream is mixing the input from Multiple Input Streams. All streams must have the same audo format (sample rate, channels, bits per sample) + Cint24_3bytes_t24bit integer which is used for I2S sound processing. The values are really using 3 bytes. It works only on little endian machines! + Cint24_4bytes_t24bit integer which is used for I2S sound processing. The values are represented as int32_t, but only 3 bytes are used. If you assign values which are too big, they are clipped + CJpegOpenCVDisplay image with opencv to be used on the desktop + CJpegTFTDisplay jpeg image using https://github.com/Bodmer/TFT_eSPI and https://github.com/Bodmer/JPEGDecoder + CJupyterAudioTOutput to Jupyter. We write the data just to a file from where we can load the data again for different representations + CLastSampleFaderIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0. Typless implementation + CLastSampleFaderTIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0 + CLC3DecoderDecoder for LC3. Depends on https://github.com/pschatzmann/arduino-liblc3 + CLC3EncoderEncoder for LC3 - Depends on https://github.com/pschatzmann/arduino-liblc3 + CLEDOutputLED output using the FastLED library + CLEDOutputConfig + CLEDOutputUnoR4LED output using the R4 LED matrix library + CLEDOutputUnoR4Config + CLinearVolumeControlThe simplest possible implementation of a VolumeControl: The input = output which describes a linear curve. You would use this implementation if you physically connect an audio pot! + ►CListDouble linked list + CIterator + CNode + CLockA simple RIA locking class for the ESP32 using _lock_t + CLogarithmicVolumeControlParametric Logarithmic volume control. Using the formula pow(b,input) * a - a, where b is b = pow(((1/ym)-1), 2) and a is a = 1.0 / (b - 1.0). The parameter ym is determining the steepness. See https://electronics.stackexchange.com/questions/304692/formula-for-logarithmic-audio-taper-pot + CMaximilianAudioTools integration with Maximilian + CMeasuringStreamClass which measures the truput + ►CMedianFilterAn embedded friendly, fast one-dimensional median filter algorithm implementation in C and C++ Useful for spike and noise removal from analog signals or other DSP Also known as "salt-and-pepper noise" or "impulse noise" filter + CsMedianFilter_t + CsMedianNode_t + CMemoryManagerMemoryManager which activates the use of external SPIRAM memory. When external memory is in use, the allocation strategy is to initially try to satisfy smaller allocation requests with internal memory and larger requests with external memory. This sets the limit between the two, as well as generally enabling allocation in external memory + CMemoryOutputWrites to a preallocated memory + CMemoryStreamA simple Stream implementation which is backed by allocated memory + ►CMetaDataFilterClass which filters out ID3v1 and ID3v2 Metadata and provides only the audio data to the decoder + CID3v2ID3 verion 2 TAG Header (10 bytes) + CMetaDataICYIcecast/Shoutcast Metadata Handling. Metadata class which splits the data into audio and metadata. The result is provided via callback methods. see https://www.codeproject.com/Articles/11308/SHOUTcast-Stream-Ripper + CMetaDataID3Simple ID3 Meta Data Parser which supports ID3 V1 and V2 and implements the Stream interface. You just need to set the callback(s) to receive the result and copy the audio data to this stream + CMetaDataID3BaseID3 Meta Data Common Functionality + CMetaDataID3V1Simple ID3 Meta Data API which supports ID3 V1 + CMetaDataID3V2Simple ID3 Meta Data API which supports ID3 V2: We only support the "TALB", "TOPE", "TIT2", "TCON" tags + CMetaDataOutputID3 and Icecast/Shoutcast metadata output support. Just write the audio data to an object of this class and receive the metadata via the callback + CMiniAudioConfigConfiguration for MiniAudio + CMiniAudioStreamMiniAudio: https://miniaud.io/ + CModulationBaseClassClass provides a wave table that can be populated with a number of preallocated waveforms. These can be used to generate audio in themselves or to modulate The parameters of another effect. Class initialised with sample rate + CMozziConfigMozzi Configuration for input or output stream + CMozziGeneratorSupport for https://sensorium.github.io/Mozzi/ Define your updateControl() method. Define your updateAudio() method. Start by calling begin(control_rate) do not call audioHook(); in the loop ! + CMozziStreamWe use the output functionality of Mozzi to output audio data. We expect the data as array of int16_t with one or two channels. Though we support the setting of a sample rate, we recommend to use the default sample rate from Mozzi which is available with the AUDIO_RATE define + CMP3DecoderHelixMP3 Decoder using libhelix: https://github.com/pschatzmann/arduino-libhelix This is basically just a simple wrapper to provide AudioInfo and AudioInfoSupport + CMP3DecoderMADMP3 Decoder using https://github.com/pschatzmann/arduino-libmad + CMP3DecoderMiniMP3 Decoder using https://github.com/pschatzmann/minimp3. This decoder does not provide any good results and it is not suited to decode any audio above 32000 on an ESP32. So the sample rate is limited by the MINIMP3_MAX_SAMPLE_RATE variable + CMP3EncoderLAMEEncodes PCM data to the MP3 format and writes the result to a stream This is basically just a wrapper using https://github.com/pschatzmann/arduino-liblame + CMP4AtomRepresents a single MPEG4 atom + CMP4ParseBuffer + CMTSDecoderMPEG-TS (MTS) decoder. Extracts the AAC audio data from a MPEG-TS (MTS) data stream. You can define the relevant stream types via the API. Required dependency: https://github.com/pschatzmann/arduino-tsdemux + CMultiConverterCombines multiple converters + CMultiOutputReplicates the output to multiple destinations + CMusicalNotesDetermination of the frequency of a music note + CNano_BLE_freq_infoMapping Frequency constants to available frequencies + CNano_BLE_ratio_infoMapping from Ratio Constants to frequency ratios + CNBufferA lock free N buffer. If count=2 we create a DoubleBuffer, if count=3 a TripleBuffer etc + CNoFilterNo change to the input + CNOPConverterDummy converter which does nothing + CNullStreamThe Arduino Stream which provides silence and simulates a null device when used as audio target or audio source + CNumberConverterConverts from a source to a target number with a different type + CNumberFormatConverterStreamConverter which converts between bits_per_sample and 16 bits. The templated NumberFormatConverterStreamT class is used based on the information provided by the bits_per_sample in the configuration + CNumberFormatConverterStreamTConverter which converts from source bits_per_sample to target bits_per_sample + CNumberReaderReads n numbers from an Arduino Stream + CNuttallNuttall FFT Window function + COggContainerDecoderDecoder for Ogg Container. Decodes a packet from an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass and overwrite the beginOfSegment() method to implement your own headers Dependency: https://github.com/pschatzmann/arduino-libopus + COggContainerEncoderEncoder for Ogg Container. Encodes a packet for an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass ond overwrite the writeHeader() method to implement your own header logic. When an optional encoder is specified in the constructor we package the encoded data. Dependency: https://github.com/pschatzmann/arduino-libopus + COggContainerOutputOutput class for the OggContainerEncoder. Each write is ending up as container entry + COnOffOutputSwitched Output class. Class which can be used to filter the output out based on a switch: If the switch is on the output is forwarded. If it is off it is just ignored + COpusAudioDecoderOpusAudioDecoder: Depends on https://github.com/pschatzmann/arduino-libopus.git + COpusAudioEncoderOpusAudioEncoder: Dependens on https://github.com/pschatzmann/arduino-libopus.git + COpusEncoderSettingsSetting for Opus Encoder where the following values are valid: -1 indicates that the default value should be used and that this codec is not setting the value + COpusOggDecoderOpus Decoder which uses the Ogg Container. See https://datatracker.ietf.org/doc/html/rfc7845. The audio data is transmitted in frames and the header information contains the sampler rate, channels and other critical info. Dependency: https://github.com/pschatzmann/arduino-libopus + COpusOggEncoderOpus Encoder which uses the Ogg Container: see https://datatracker.ietf.org/doc/html/rfc7845 Dependency: https://github.com/pschatzmann/arduino-libopus + COpusOggWriter + COpusSettingsSetting for Opus Decoder + COutputMixerMixing of multiple outputs to one final output + CParameterA constant value + CParseBufferWe try to keep the necessary buffer for parsing as small as possible, The data() method provides the start of the actual data and with consume we remove the processed data from the buffer to make space again + CParseObject + CPinInfoESP32Information for a PIN + CPinkNoiseGeneratorGenerates pink noise + CPitchShiftShifts the pitch by the indicated step size: e.g. 2 doubles the pitch + CPitchShiftInfoConfiguration for PitchShiftOutput: set the pitch_shift to define the shift + CPitchShiftOutputPitch Shift: Shifts the frequency up or down w/o impacting the length! We reduce the channels to 1 to calculate the pitch shift and provides the pitch shifted result in the correct number of channels. The pitch shifting is done with the help of a buffer that can have potentially multiple implementations + CPoppingSoundRemoverBig value gaps (at the beginning and the end of a recording) can lead to some popping sounds. We will try to set the values to 0 until the first transition thru 0 of the audio curve + CPortAudioConfigPortAudio information + CPortAudioStreamArduino Audio Stream using PortAudio + CPrint + CProcessedResult + CProgressStreamGeneric calss to measure the the total bytes which were processed in order to calculate the progress as a percentage of the total size + CProgressStreamInfoConfiguration for ProgressStream + CPWMAudioOutputCommon functionality for PWM output. Please use the PWMAudioOutput typedef instead which references the implementation + CPWMConfigConfiguration data for PWM audio output + CPWMDriverAVRExperimental: Audio output to PWM pins for the AVR. The AVR supports only up to 2 channels + CPWMDriverESP32Audio output to PWM pins for the ESP32. The ESP32 supports up to 16 channels + CPWMDriverMBEDAudio output to PWM pins for MBED based Arduino implementations + CPWMDriverRenesasAudio output to PWM pins for Renesas based Arduino implementations + CPWMDriverSTM32Audio output to PWM pins for STM32. We use one timer to generate the sample rate and one timer for the PWM signal + CQueueFIFO Queue which is based on a List + ►CQueueLockFreeLock Free Queue + Cnode + CQueueStreamAudioStream class which stores the data in a temporary queue buffer. The queue can be consumed e.g. by a callback function by calling readBytes(); + CRECT + CRectangeRectange FFT Window function + CReformatBaseStreamBase class for chained converting streams + CResampleConfigOptional Configuration object. The critical information is the channels and the step_size. All other information is not used + CResampleStreamDynamic Resampling. We can use a variable factor to speed up or slow down the playback + CRingBufferImplements a typed Ringbuffer + CRingBufferFileAn File backed Ring Buffer that we can use to receive streaming audio. We expect an open p_file as parameter + CRingBufferStreamA Stream backed by a Ringbuffer. We can write to the end and read from the beginning of the stream + CRTSPFormatAbtXAbtX format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatAudioToolsRTSPFormat which supports the AudioInfo class + CRTSPFormatG711G711 μ-Law format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats Packet intervall: 20, frame size: any + CRTSPFormatGSMGSM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatOpusOpus format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatPCMPCM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatPCM8L8 format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPOutputWe can write PCM data to the RTSPOutput. This is encoded by the indicated encoder (e.g. SBCEncoder) and can be consumed by a RTSPServer. You have to make sure that the codec supports the provided audio format: e.g. GSM support only 8000 samples per second with one channel. Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPOutputPCMInfoPCMInfo subclass which provides the audio information from the related AudioStream Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPPCMAudioInfoPCMInfo subclass which provides the audio information from the AudioInfo parameter + CRTSPSourceFromAudioStreamSimple Facade which can turn AudioStream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPSourceStreamSimple Facade which can turn any Stream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CSBCDecoderDecoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcdec.c + CSBCEncoderEncoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcenc.c + CScaledParameterScales a dynamic parameter to the indicated range + CSDDirectWe access the files directy with an index. The index is determined by a recurseve tree walk thru the directory. Unfortunatly the SDTFAT library has it's own API which is incompatible with the SDT API + CSDIndexWe store all the relevant file names in an sequential index file. Form there we can access them via an index + CSilenceGeneratorProvides a fixed value (e.g. 0) as sound data. This can be used e.g. to test the output functionality which should optimally just output silence and no artifacts + CSilenceRemovalConverterRemoves any silence from the buffer that is longer then n samples with a amplitude below the indicated threshhold. If you process multiple channels you need to multiply the channels with the number of samples to indicate n + CSimpleChorusSimple Chorus effect with a single delay voice and mono output Chorus is effective between 15 and 20 miliseconds delay of original audio. Requires the sample rate when initialising + CSimpleContainerConfig + CSimpleContainerDataHeader + CSimpleContainerMetaDataHeader + CSimpleDelaySimple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples + CSimpleFlangerSimple Flanger Effect Consistig of a single voice flanger The flanger has an effective range between 0 and 15 miliseconds in this case dleay buffer should be set to sampleRate*3/200 Constructor requires internal delay in samples + CSimpleLPFSimpleLPF + CSimulatedAudioPotSimple simulated audio pot volume control inspired by https://eepower.com/resistor-guide/resistor-types/potentiometer-taper/# We split up the input/output curve into 2 linear pieces with a slow and a fast raising part. The slow raising part goes from (0,0) to (x,y). The fast raising part goes from (x,y) to (1,1) + CSineFromTableA sine generator based on a table. The table is created based using degrees where one full wave is 360 degrees + CSineWaveGeneratorGenerates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead + CSingleBufferA simple Buffer implementation which just uses a (dynamically sized) array + CSliceHelps to split up a big memory array into smaller slices. There are no additinal heap allocations! Example: if we have an array with 9 entries (1,2,3,4,5,6,7,8,9): slices(5) gives 2. slice(5,0) returns size 5 with 1,2,3,4,5 and slice(5,1) returns size 4 with 6,7,8,9! + CSmoothTransitionChanges the samples at the beginning or at the end to slowly ramp up the volume + CSOSFilterSecond Order Filter: Instead of manually cascading BiQuad filters, you can use a Second Order Sections filter (SOS). converted from https://github.com/tttapa/Filters/blob/master/src/SOSFilter.h Use float or double (and not a integer type) as type parameter + CSoundGeneratorBase class to define the abstract interface for the sound generating classes + CSoundGeneratorModulationSoundGenerator using the ModulationBaseClass to generate the samples + CSPDIFConfigSPDIF configuration + CSPDIFOutputOutput as 16 bit stereo SPDIF on the I2S data output pin + CSquareWaveGeneratorGenerates a square wave sound + CStackLIFO Stack which is based on a List + CStdioStream + CSTKChorusChorus Effect + CSTKChowningReverbJohn Chowning's reverberator class + CSTKEchoEcho Effect + CSTKEffectUse any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK + CSTKFreeVerbJezar at Dreampoint's FreeVerb, implemented in STK + CSTKGeneratorThe Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language.
- CSTKLentPitShiftPitch shifter effect class based on the Lent algorithm - CSTKNReverbCCRMA's NRev reverberator class - CSTKPerryReverbPerry's simple reverberator class - CSTKPitShiftPitch shifter effect class based on the Lent algorithm - CSTKStreamSTK Stream for Instrument or Voicer - CStrA simple wrapper to provide string functions on char*. If the underlying char* is a const we do not allow any updates; The ownership of the chr* must be managed externally! - CStream - CStreamCopyWe provide the typeless StreamCopy as a subclass of StreamCopyT - CStreamCopyTTyped Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples - CStreamingDecoderA Streaming Decoder where we provide both the input and output as streams - CStrExtStr which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting - ►CSynthesizerA simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels - CKeyParameter - CSynthesizerKeyArduino GPIO pin to note assossiation - CTestGeneratorGenerates a test signal which is easy to check because the values are incremented or decremented by 1 - CTfLiteAbstractRecognizeCommandsBase class for implementing different primitive decoding models on top of the instantaneous results from running an audio recognition model on a single window of samples - CTfLiteAudioErrorReporterError Reporter using the Audio Tools Logger - CTfLiteAudioStreamTfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data) - CTfLiteAudioStreamBaseAstract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers - CTfLiteConfigConfiguration settings for TfLiteAudioStream - CTfLiteMicroSpeachWriterTfLiteMicroSpeachWriter for Audio Data - ►CTfLiteMicroSpeechRecognizeCommandsThis class is designed to apply a very primitive decoding model on top of the instantaneous results from running an audio recognition model on a single window of samples. It applies smoothing over time so that noisy individual label scores are averaged, increasing the confidence that apparent matches are real. To use it, you should create a class object with the configuration you want, and then feed results from running a TensorFlow model into the processing method. The timestamp for each subsequent call should be increasing from the previous, since the class is designed to process a stream of data over time - CResult - CTfLiteQuantizerQuantizer that helps to quantize and dequantize between float and int8 - CTfLiteReaderInput class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce - CTfLiteSineReaderGenerate a sine output from a model that was trained on the sine method. (=hello_world) - CTfLiteWriterOutput class which interprets audio data if TfLiteAudioStream is treated as audio sink - CThrottleThrottle the sending of the audio data to limit it to the indicated sample rate - CThrottleConfigConfigure Throttle setting - CTimedStreamAudioStream class that can define a start and (an optional) stop time Usually it is used to wrap an Audio Sink (e.g. I2SStream), but wrapping an Audio Source is supported as well. Only wrap classes which represent PCM data! - CTimerAlarmRepeatingCommon Interface definition for TimerAlarmRepeating - CTimerAlarmRepeatingDriverAVRRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverBase - CTimerAlarmRepeatingDriverESP8266Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverMBEDRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverRenesasRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating. By default we use a new GPT timer. You can also request 1 AGT timer by calling setTimer(1); - CTimerAlarmRepeatingDriverSTM32STM32 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerCallbackAudioStreamCallback driven Audio Source (rx_tx_mode==RX_MODE) or Audio Sink (rx_tx_mode==TX_MODE). This class allows to to integrate external libraries in order to consume or generate a data stream which is based on a timer - CTimerCallbackAudioStreamInfoTimerCallbackAudioStream Configuration - ►CTransformationReaderConverterStream Helper class which implements the converting readBytes with the help of write - CAdapterPrintToArray - CTremoloTremolo AudioEffect - CTriangleTriangle FFT Window function - CUDPStream - CUrlURL parser which breaks a full url string up into its individual parts - CURLHistory - CURLLoader - CURLStreamRepresents the content of a URL as Stream. We use the WiFi.h API - CURLStreamBufferedURLStream implementation for the ESP32 based on a separate FreeRTOS task - CVariableSpeedRingBufferOptimized Buffer implementation for Pitch Shift. We try to interpolate the samples and restore the phase when the read pointer and write pointer overtake each other - CVariableSpeedRingBuffer180Varialbe speed ring buffer where we read with 0 and 180 degree and blend the result to prevent overrun artifacts. See https://github.com/YetAnotherElectronicsChannel/STM32_DSP_PitchShift - CVariableSpeedRingBufferSimpleVery Simple Buffer implementation for Pitch Shift. We write in constant speed, but reading can be done in a variable speed. We will hear some noise when the buffer read and write pointers overrun each other - CVBANConfig - CVBANStreamVBAN Audio Source and Sink for the ESP32. For further details please see https://vb-audio.com/Voicemeeter/vban.htm . Inspired by https://github.com/rkinnett/ESP32-VBAN-Audio-Source/tree/master and https://github.com/rkinnett/ESP32-VBAN-Network-Audio-Player - ►CVectorVector implementation which provides the most important methods as defined by std::vector. This class it is quite handy to have and most of the times quite better then dealing with raw c arrays - CiteratorIterator for the Vector class - CVideoAudioBufferedSyncLogic to Synchronize video and audio output: we use a buffer to store the audio and instead of delaying the frames with delay() we play audio. The bufferSize defines the audio buffer in bytes. The correctionMs is used to slow down or speed up the playback of the video to prevent any audio buffer underflows - CVideoAudioSyncLogic to Synchronize video and audio output: This is the minimum implementatin which actually does not synchronize, but directly processes the data. No additinal memory is used! Provide your own optimized platform specific implementation - CVideoOutputAbstract class for video playback. This class is used to assemble a complete video frame in memory - CVolumeControlAbstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal - CVolumeOutputA simple class to determine the volume - CVolumeStreamAdjust the volume of the related input or output: To work properly the class needs to know the bits per sample and number of channels! AudioChanges are forwareded to the related Print or Stream class - CVolumeStreamConfigConfig for VolumeStream - CVorbisDecoderVorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-idec https://github.com/pschatzmann/arduino-libopus.git - CVS1053ConfigConfiguration for VS1053Stream - CVS1053StreamVS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration; - CWAVAudioInfoSound information which is available in the WAV header - CWAVDecoderA simple WAVDecoder: We parse the header data on the first record to determine the format. If no AudioDecoderExt is specified we just write the PCM data to the output that is defined by calling setOutput(). You can define a ADPCM decoder to decode WAV files that contain ADPCM data - CWAVEncoderA simple WAV file encoder. If no AudioEncoderExt is specified the WAV file contains PCM data, otherwise it is encoded as ADPCM. The WAV header is written with the first writing of audio data. Calling begin() is making sure that the header is written again - CWAVFormatX - CWAVHeaderParser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE - CWavIMAAudioInfoSound information which is available in the WAV header - adjusted for IMA ADPCM - CWavIMADecoderObsolete: WavIMADecoder - based on WAVDecoder - We parse the header data as we receive it and send the sound data to the stream which was indicated in the constructor. Only WAV files with WAVE_FORMAT_IMA_ADPCM are supported by this codec! - CWavIMAHeader - CWelchWelch FFT Window function - CWhiteNoiseGeneratorGenerates a random noise sound with the help of rand() function - CWindowFunctionFFT Window Function - CWM8960ConfigConfiguration for WM8960 - CWM8960StreamStream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960 + CSTKLentPitShiftPitch shifter effect class based on the Lent algorithm + CSTKNReverbCCRMA's NRev reverberator class + CSTKPerryReverbPerry's simple reverberator class + CSTKPitShiftPitch shifter effect class based on the Lent algorithm + CSTKStreamSTK Stream for Instrument or Voicer + CStrA simple wrapper to provide string functions on char*. If the underlying char* is a const we do not allow any updates; The ownership of the chr* must be managed externally! + CStream + CStreamCopyWe provide the typeless StreamCopy as a subclass of StreamCopyT + CStreamCopyTTyped Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples + CStreamingDecoderA Streaming Decoder where we provide both the input and output as streams + CStrExtStr which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting + ►CSynthesizerA simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels + CKeyParameter + CSynthesizerKeyArduino GPIO pin to note assossiation + CTestGeneratorGenerates a test signal which is easy to check because the values are incremented or decremented by 1 + CTfLiteAbstractRecognizeCommandsBase class for implementing different primitive decoding models on top of the instantaneous results from running an audio recognition model on a single window of samples + CTfLiteAudioErrorReporterError Reporter using the Audio Tools Logger + CTfLiteAudioStreamTfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data) + CTfLiteAudioStreamBaseAstract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers + CTfLiteConfigConfiguration settings for TfLiteAudioStream + CTfLiteMicroSpeachWriterTfLiteMicroSpeachWriter for Audio Data + ►CTfLiteMicroSpeechRecognizeCommandsThis class is designed to apply a very primitive decoding model on top of the instantaneous results from running an audio recognition model on a single window of samples. It applies smoothing over time so that noisy individual label scores are averaged, increasing the confidence that apparent matches are real. To use it, you should create a class object with the configuration you want, and then feed results from running a TensorFlow model into the processing method. The timestamp for each subsequent call should be increasing from the previous, since the class is designed to process a stream of data over time + CResult + CTfLiteQuantizerQuantizer that helps to quantize and dequantize between float and int8 + CTfLiteReaderInput class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce + CTfLiteSineReaderGenerate a sine output from a model that was trained on the sine method. (=hello_world) + CTfLiteWriterOutput class which interprets audio data if TfLiteAudioStream is treated as audio sink + CThrottleThrottle the sending of the audio data to limit it to the indicated sample rate + CThrottleConfigConfigure Throttle setting + CTimedStreamAudioStream class that can define a start and (an optional) stop time Usually it is used to wrap an Audio Sink (e.g. I2SStream), but wrapping an Audio Source is supported as well. Only wrap classes which represent PCM data! + CTimerAlarmRepeatingCommon Interface definition for TimerAlarmRepeating + CTimerAlarmRepeatingDriverAVRRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverBase + CTimerAlarmRepeatingDriverESP8266Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverMBEDRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverRenesasRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating. By default we use a new GPT timer. You can also request 1 AGT timer by calling setTimer(1); + CTimerAlarmRepeatingDriverSTM32STM32 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerCallbackAudioStreamCallback driven Audio Source (rx_tx_mode==RX_MODE) or Audio Sink (rx_tx_mode==TX_MODE). This class allows to to integrate external libraries in order to consume or generate a data stream which is based on a timer + CTimerCallbackAudioStreamInfoTimerCallbackAudioStream Configuration + ►CTransformationReaderConverterStream Helper class which implements the converting readBytes with the help of write + CAdapterPrintToArray + CTremoloTremolo AudioEffect + CTriangleTriangle FFT Window function + CUDPStream + CUrlURL parser which breaks a full url string up into its individual parts + CURLHistory + CURLLoader + CURLStreamRepresents the content of a URL as Stream. We use the WiFi.h API + CURLStreamBufferedURLStream implementation for the ESP32 based on a separate FreeRTOS task + CVariableSpeedRingBufferOptimized Buffer implementation for Pitch Shift. We try to interpolate the samples and restore the phase when the read pointer and write pointer overtake each other + CVariableSpeedRingBuffer180Varialbe speed ring buffer where we read with 0 and 180 degree and blend the result to prevent overrun artifacts. See https://github.com/YetAnotherElectronicsChannel/STM32_DSP_PitchShift + CVariableSpeedRingBufferSimpleVery Simple Buffer implementation for Pitch Shift. We write in constant speed, but reading can be done in a variable speed. We will hear some noise when the buffer read and write pointers overrun each other + CVBANConfig + CVBANStreamVBAN Audio Source and Sink for the ESP32. For further details please see https://vb-audio.com/Voicemeeter/vban.htm . Inspired by https://github.com/rkinnett/ESP32-VBAN-Audio-Source/tree/master and https://github.com/rkinnett/ESP32-VBAN-Network-Audio-Player + ►CVectorVector implementation which provides the most important methods as defined by std::vector. This class it is quite handy to have and most of the times quite better then dealing with raw c arrays + CiteratorIterator for the Vector class + CVideoAudioBufferedSyncLogic to Synchronize video and audio output: we use a buffer to store the audio and instead of delaying the frames with delay() we play audio. The bufferSize defines the audio buffer in bytes. The correctionMs is used to slow down or speed up the playback of the video to prevent any audio buffer underflows + CVideoAudioSyncLogic to Synchronize video and audio output: This is the minimum implementatin which actually does not synchronize, but directly processes the data. No additinal memory is used! Provide your own optimized platform specific implementation + CVideoOutputAbstract class for video playback. This class is used to assemble a complete video frame in memory + CVolumeControlAbstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal + CVolumeOutputA simple class to determine the volume + CVolumeStreamAdjust the volume of the related input or output: To work properly the class needs to know the bits per sample and number of channels! AudioChanges are forwareded to the related Print or Stream class + CVolumeStreamConfigConfig for VolumeStream + CVorbisDecoderVorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-idec https://github.com/pschatzmann/arduino-libopus.git + CVS1053ConfigConfiguration for VS1053Stream + CVS1053StreamVS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration; + CWAVAudioInfoSound information which is available in the WAV header + CWAVDecoderA simple WAVDecoder: We parse the header data on the first record to determine the format. If no AudioDecoderExt is specified we just write the PCM data to the output that is defined by calling setOutput(). You can define a ADPCM decoder to decode WAV files that contain ADPCM data + CWAVEncoderA simple WAV file encoder. If no AudioEncoderExt is specified the WAV file contains PCM data, otherwise it is encoded as ADPCM. The WAV header is written with the first writing of audio data. Calling begin() is making sure that the header is written again + CWAVFormatX + CWAVHeaderParser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE + CWavIMAAudioInfoSound information which is available in the WAV header - adjusted for IMA ADPCM + CWavIMADecoderObsolete: WavIMADecoder - based on WAVDecoder - We parse the header data as we receive it and send the sound data to the stream which was indicated in the constructor. Only WAV files with WAVE_FORMAT_IMA_ADPCM are supported by this codec! + CWavIMAHeader + CWelchWelch FFT Window function + CWhiteNoiseGeneratorGenerates a random noise sound with the help of rand() function + CWindowFunctionFFT Window Function + CWM8960ConfigConfiguration for WM8960 + CWM8960StreamStream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960  ►Nffft  CArray  CDynArray diff --git a/classaudio__tools_1_1_i2_s_driver_e_s_p32-members.html b/classaudio__tools_1_1_i2_s_driver_e_s_p32-members.html new file mode 100644 index 0000000000..dd43e3dea2 --- /dev/null +++ b/classaudio__tools_1_1_i2_s_driver_e_s_p32-members.html @@ -0,0 +1,104 @@ + + + + + + + +arduino-audio-tools: Member List + + + + + + + + + +
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I2SDriverESP32 Member List
+
+
+ +

This is the complete list of members for I2SDriverESP32, including all inherited members.

+ + + + + + + + + + + + + + + + + + + + + + + + +
AnalogAudio (defined in I2SDriverESP32)I2SDriverESP32friend
AudioKitStream (defined in I2SDriverESP32)I2SDriverESP32friend
available()I2SDriverESP32inline
availableForWrite()I2SDriverESP32inline
begin(RxTxMode mode)I2SDriverESP32inline
begin()I2SDriverESP32inline
begin(I2SConfig cfg)I2SDriverESP32inline
begin(I2SConfig cfg, int txPin, int rxPin)I2SDriverESP32inlineprotected
cfg (defined in I2SDriverESP32)I2SDriverESP32protected
config()I2SDriverESP32inline
defaultConfig(RxTxMode mode)I2SDriverESP32inline
end()I2SDriverESP32inline
getModeDigital(I2SConfig &cfg) (defined in I2SDriverESP32)I2SDriverESP32inlineprotected
i2s_config (defined in I2SDriverESP32)I2SDriverESP32protected
i2s_num (defined in I2SDriverESP32)I2SDriverESP32protected
is_started (defined in I2SDriverESP32)I2SDriverESP32protected
isNoChannelConversion(I2SConfig cfg) (defined in I2SDriverESP32)I2SDriverESP32inlineprotected
readBytes(void *dest, size_t size_bytes) (defined in I2SDriverESP32)I2SDriverESP32inline
setChannels(int channels) (defined in I2SDriverESP32)I2SDriverESP32inlineprotected
toCommFormat(I2SFormat mode) (defined in I2SDriverESP32)I2SDriverESP32inlineprotected
toMode(I2SConfig &cfg) (defined in I2SDriverESP32)I2SDriverESP32inlineprotected
writeBytes(const void *src, size_t size_bytes)I2SDriverESP32inline
writeExpandChannel(i2s_port_t i2s_num, const int bits_per_sample, const void *src, size_t size_bytes)I2SDriverESP32inlineprotectedstatic
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+Public Member Functions | +Protected Member Functions | +Static Protected Member Functions | +Protected Attributes | +Friends | +List of all members
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+
I2SDriverESP32 Class Reference
Arduino Audio Tools » Platform
+
+
+ +

Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels. + More...

+ +

#include <I2SESP32.h>

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+Public Member Functions

+int available ()
 we assume the data is already available in the buffer
 
+int availableForWrite ()
 We limit the write size to the buffer size.
 
+bool begin ()
 starts the DAC with the current config - if not started yet. If I2S has been started there is no action and we return true
 
+bool begin (I2SConfig cfg)
 starts the DAC
 
+bool begin (RxTxMode mode)
 starts the DAC with the default config
 
+I2SConfig config ()
 provides the actual configuration
 
+I2SConfig defaultConfig (RxTxMode mode)
 Provides the default configuration.
 
+void end ()
 stops the I2C and unistalls the driver
 
+size_t readBytes (void *dest, size_t size_bytes)
 
+size_t writeBytes (const void *src, size_t size_bytes)
 writes the data to the I2S interface
 
+ + + + + + + + + + + + + + +

+Protected Member Functions

+bool begin (I2SConfig cfg, int txPin, int rxPin)
 starts the DAC
 
+int getModeDigital (I2SConfig &cfg)
 
+bool isNoChannelConversion (I2SConfig cfg)
 
+void setChannels (int channels)
 
+i2s_comm_format_t toCommFormat (I2SFormat mode)
 
+i2s_mode_t toMode (I2SConfig &cfg)
 
+ + + + +

+Static Protected Member Functions

+static size_t writeExpandChannel (i2s_port_t i2s_num, const int bits_per_sample, const void *src, size_t size_bytes)
 writes the data by making sure that we send 2 channels
 
+ + + + + + + + + +

+Protected Attributes

+I2SConfig cfg = defaultConfig(RX_MODE)
 
+i2s_config_t i2s_config
 
+i2s_port_t i2s_num
 
+bool is_started = false
 
+ + + + + +

+Friends

+class AnalogAudio
 
+class AudioKitStream
 
+

Detailed Description

+

Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels.

+
Author
Phil Schatzmann
+
Copyright
GPLv3
+
The documentation for this class was generated from the following file: +
+ +
+Generated by doxygen 1.9.1 +
+ + diff --git a/classaudio__tools_1_1_i2_s_stream.html b/classaudio__tools_1_1_i2_s_stream.html index 7f98050526..de04dab3d8 100644 --- a/classaudio__tools_1_1_i2_s_stream.html +++ b/classaudio__tools_1_1_i2_s_stream.html @@ -118,7 +118,7 @@  Provides the default configuration.
  -I2SDriver * driver () +I2SDriver * driver ()  Provides access to the driver.
  @@ -180,7 +180,7 @@ int _timeout = 10   -I2SDriver i2s +I2SDriver i2s   AudioInfo info diff --git a/classes.html b/classes.html index 55765a9403..c9eedd8601 100644 --- a/classes.html +++ b/classes.html @@ -93,7 +93,7 @@
Hamming (audio_tools)
Hann (audio_tools)
HardwareSerial (audio_tools)
HexDumpOutput (audio_tools)
HLSParser (audio_tools)
HLSStream (audio_tools)
HttpChunkReader (audio_tools)
HttpHeader (audio_tools)
HttpHeaderLine (audio_tools)
HttpLineReader (audio_tools)
HttpReplyHeader (audio_tools)
HttpRequest (audio_tools)
HttpRequestHeader (audio_tools)
I
-
I2SConfig (audio_tools)
I2SDriverESP32V1 (audio_tools)
I2SDriverESP8266 (audio_tools)
I2SDriverNanoBLE (audio_tools)
I2SDriverSAMD (audio_tools)
I2SDriverSTM32 (audio_tools)
I2SStream (audio_tools)
ICYStream (audio_tools)
ICYStreamBuffered (audio_tools)
ICYUrlSetup (audio_tools)
ID3v1 (audio_tools)
ID3v1Enhanced (audio_tools)
ID3v2 (audio_tools)
MetaDataFilter::ID3v2 (audio_tools)
ID3v2Frame (audio_tools)
ID3v2FrameString (audio_tools)
IIR (audio_tools)
ILBCDecoder (audio_tools)
ILBCEncoder (audio_tools)
IMAState (audio_tools)
initializer_list (std)
InputMerge (audio_tools)
InputMixer (audio_tools)
int24_3bytes_t (audio_tools)
int24_4bytes_t (audio_tools)
AnalogDriverESP32V1::IO16Bit (audio_tools)
List::Iterator (audio_tools)
Vector::iterator (audio_tools)
+
I2SConfig (audio_tools)
I2SDriverESP32 (audio_tools)
I2SDriverESP32V1 (audio_tools)
I2SDriverESP8266 (audio_tools)
I2SDriverNanoBLE (audio_tools)
I2SDriverSAMD (audio_tools)
I2SDriverSTM32 (audio_tools)
I2SStream (audio_tools)
ICYStream (audio_tools)
ICYStreamBuffered (audio_tools)
ICYUrlSetup (audio_tools)
ID3v1 (audio_tools)
ID3v1Enhanced (audio_tools)
MetaDataFilter::ID3v2 (audio_tools)
ID3v2 (audio_tools)
ID3v2Frame (audio_tools)
ID3v2FrameString (audio_tools)
IIR (audio_tools)
ILBCDecoder (audio_tools)
ILBCEncoder (audio_tools)
IMAState (audio_tools)
initializer_list (std)
InputMerge (audio_tools)
InputMixer (audio_tools)
int24_3bytes_t (audio_tools)
int24_4bytes_t (audio_tools)
AnalogDriverESP32V1::IO16Bit (audio_tools)
List::Iterator (audio_tools)
Vector::iterator (audio_tools)
J
JpegOpenCV (audio_tools)
JpegTFT (audio_tools)
JupyterAudioT (audio_tools)
diff --git a/functions.html b/functions.html index 1847b2e8bf..c344c1f3ce 100644 --- a/functions.html +++ b/functions.html @@ -99,7 +99,7 @@

- a -

diff --git a/namespaceaudio__tools.html b/namespaceaudio__tools.html index f378204751..d9d2dc7d82 100644 --- a/namespaceaudio__tools.html +++ b/namespaceaudio__tools.html @@ -781,6 +781,9 @@ class  I2SConfig  configuration for all common i2s settings More...
  +class  I2SDriverESP32 + Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels. More...
+  class  I2SDriverESP32V1  Basic I2S API for the ESP32 (using the new API). https://docs.espressif.com/projects/esp-idf/en/v5.0.1/esp32/api-reference/peripherals/i2s.html#i2s-communication-mode. More...
  @@ -1538,7 +1541,7 @@ using HexDumpStream = HexDumpOutput   -using I2SDriver = I2SDriverESP32V1 +using I2SDriver = I2SDriverESP32   using JupyterAudio = JupyterAudioT< int16_t > diff --git a/namespaces.html b/namespaces.html index 6c19a1b2e8..7099be23d4 100644 --- a/namespaces.html +++ b/namespaces.html @@ -313,262 +313,263 @@  CHttpRequestSimple API to process get, put, post, del http requests I tried to use Arduino HttpClient, but I did not manage to extract the mime type from streaming get requests  CHttpRequestHeaderReading and writing of Http Requests  CI2SConfigConfiguration for all common i2s settings - ►CI2SDriverESP32V1Basic I2S API for the ESP32 (using the new API). https://docs.espressif.com/projects/esp-idf/en/v5.0.1/esp32/api-reference/peripherals/i2s.html#i2s-communication-mode - CDriverCommon - CDriverI2S - CI2SDriverESP8266Basic I2S API - for the ESP8266 Only 16 bits are supported ! - CI2SDriverNanoBLEBasic I2S API - for the Arduino Nano BLE Sense See https://content.arduino.cc/assets/Nano_BLE_MCU-nRF52840_PS_v1.1.pdf Douplex mode (RXTX_MODE) is currently not supported, but it should be quite easy to implement - CI2SDriverSAMDBasic I2S API - for the SAMD - CI2SDriverSTM32Basic I2S API - for the STM32 Depends on https://github.com/pschatzmann/stm32f411-i2s We just add a write and read buffer and pass some parameters to the STM32 API! - CI2SStreamWe support the Stream interface for the I2S access. In addition we allow a separate mute pin which might also be used to drive a LED.. - CICYStreamIcecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method - CICYStreamBufferedICYStream implementation for the ESP32 based on a FreeRTOS task This is a Icecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method - CICYUrlSetupResolve icy-metaint from HttpRequest and execute metadata callbacks - CID3v1 - CID3v1Enhanced - CID3v2 - CID3v2Frame - CID3v2FrameString - CIIRIIRFilter Converted from https://github.com/tttapa/Filters/blob/master/src/IIRFilter.h - CILBCDecoderDecoder for iLBC. Depends on https://github.com/pschatzmann/libilbc - CILBCEncoderEncoder for iLBC - Depends on https://github.com/pschatzmann/libopenilbc - CIMAState - CInputMergeMerges multiple input channels. The input must be mono! So if you provide 2 mono channels you get a stereo signal as result with the left channel from channel 0 and the right from channel 1 - CInputMixerMixerStream is mixing the input from Multiple Input Streams. All streams must have the same audo format (sample rate, channels, bits per sample) - Cint24_3bytes_t24bit integer which is used for I2S sound processing. The values are really using 3 bytes. It works only on little endian machines! - Cint24_4bytes_t24bit integer which is used for I2S sound processing. The values are represented as int32_t, but only 3 bytes are used. If you assign values which are too big, they are clipped - CJpegOpenCVDisplay image with opencv to be used on the desktop - CJpegTFTDisplay jpeg image using https://github.com/Bodmer/TFT_eSPI and https://github.com/Bodmer/JPEGDecoder - CJupyterAudioTOutput to Jupyter. We write the data just to a file from where we can load the data again for different representations - CLastSampleFaderIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0. Typless implementation - CLastSampleFaderTIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0 - CLC3DecoderDecoder for LC3. Depends on https://github.com/pschatzmann/arduino-liblc3 - CLC3EncoderEncoder for LC3 - Depends on https://github.com/pschatzmann/arduino-liblc3 - CLEDOutputLED output using the FastLED library - CLEDOutputConfig - CLEDOutputUnoR4LED output using the R4 LED matrix library - CLEDOutputUnoR4Config - CLinearVolumeControlThe simplest possible implementation of a VolumeControl: The input = output which describes a linear curve. You would use this implementation if you physically connect an audio pot! - ►CListDouble linked list - CIterator - CNode - CLockA simple RIA locking class for the ESP32 using _lock_t - CLogarithmicVolumeControlParametric Logarithmic volume control. Using the formula pow(b,input) * a - a, where b is b = pow(((1/ym)-1), 2) and a is a = 1.0 / (b - 1.0). The parameter ym is determining the steepness. See https://electronics.stackexchange.com/questions/304692/formula-for-logarithmic-audio-taper-pot - CMaximilianAudioTools integration with Maximilian - CMeasuringStreamClass which measures the truput - ►CMedianFilterAn embedded friendly, fast one-dimensional median filter algorithm implementation in C and C++ Useful for spike and noise removal from analog signals or other DSP Also known as "salt-and-pepper noise" or "impulse noise" filter - CsMedianFilter_t - CsMedianNode_t - CMemoryManagerMemoryManager which activates the use of external SPIRAM memory. When external memory is in use, the allocation strategy is to initially try to satisfy smaller allocation requests with internal memory and larger requests with external memory. This sets the limit between the two, as well as generally enabling allocation in external memory - CMemoryOutputWrites to a preallocated memory - CMemoryStreamA simple Stream implementation which is backed by allocated memory - ►CMetaDataFilterClass which filters out ID3v1 and ID3v2 Metadata and provides only the audio data to the decoder - CID3v2ID3 verion 2 TAG Header (10 bytes) - CMetaDataICYIcecast/Shoutcast Metadata Handling. Metadata class which splits the data into audio and metadata. The result is provided via callback methods. see https://www.codeproject.com/Articles/11308/SHOUTcast-Stream-Ripper - CMetaDataID3Simple ID3 Meta Data Parser which supports ID3 V1 and V2 and implements the Stream interface. You just need to set the callback(s) to receive the result and copy the audio data to this stream - CMetaDataID3BaseID3 Meta Data Common Functionality - CMetaDataID3V1Simple ID3 Meta Data API which supports ID3 V1 - CMetaDataID3V2Simple ID3 Meta Data API which supports ID3 V2: We only support the "TALB", "TOPE", "TIT2", "TCON" tags - CMetaDataOutputID3 and Icecast/Shoutcast metadata output support. Just write the audio data to an object of this class and receive the metadata via the callback - CMiniAudioConfigConfiguration for MiniAudio - CMiniAudioStreamMiniAudio: https://miniaud.io/ - CModulationBaseClassClass provides a wave table that can be populated with a number of preallocated waveforms. These can be used to generate audio in themselves or to modulate The parameters of another effect. Class initialised with sample rate - CMozziConfigMozzi Configuration for input or output stream - CMozziGeneratorSupport for https://sensorium.github.io/Mozzi/ Define your updateControl() method. Define your updateAudio() method. Start by calling begin(control_rate) do not call audioHook(); in the loop ! - CMozziStreamWe use the output functionality of Mozzi to output audio data. We expect the data as array of int16_t with one or two channels. Though we support the setting of a sample rate, we recommend to use the default sample rate from Mozzi which is available with the AUDIO_RATE define - CMP3DecoderHelixMP3 Decoder using libhelix: https://github.com/pschatzmann/arduino-libhelix This is basically just a simple wrapper to provide AudioInfo and AudioInfoSupport - CMP3DecoderMADMP3 Decoder using https://github.com/pschatzmann/arduino-libmad - CMP3DecoderMiniMP3 Decoder using https://github.com/pschatzmann/minimp3. This decoder does not provide any good results and it is not suited to decode any audio above 32000 on an ESP32. So the sample rate is limited by the MINIMP3_MAX_SAMPLE_RATE variable - CMP3EncoderLAMEEncodes PCM data to the MP3 format and writes the result to a stream This is basically just a wrapper using https://github.com/pschatzmann/arduino-liblame - CMP4AtomRepresents a single MPEG4 atom - CMP4ParseBuffer - CMTSDecoderMPEG-TS (MTS) decoder. Extracts the AAC audio data from a MPEG-TS (MTS) data stream. You can define the relevant stream types via the API. Required dependency: https://github.com/pschatzmann/arduino-tsdemux - CMultiConverterCombines multiple converters - CMultiOutputReplicates the output to multiple destinations - CMusicalNotesDetermination of the frequency of a music note - CNano_BLE_freq_infoMapping Frequency constants to available frequencies - CNano_BLE_ratio_infoMapping from Ratio Constants to frequency ratios - CNBufferA lock free N buffer. If count=2 we create a DoubleBuffer, if count=3 a TripleBuffer etc - CNoFilterNo change to the input - CNOPConverterDummy converter which does nothing - CNullStreamThe Arduino Stream which provides silence and simulates a null device when used as audio target or audio source - CNumberConverterConverts from a source to a target number with a different type - CNumberFormatConverterStreamConverter which converts between bits_per_sample and 16 bits. The templated NumberFormatConverterStreamT class is used based on the information provided by the bits_per_sample in the configuration - CNumberFormatConverterStreamTConverter which converts from source bits_per_sample to target bits_per_sample - CNumberReaderReads n numbers from an Arduino Stream - CNuttallNuttall FFT Window function - COggContainerDecoderDecoder for Ogg Container. Decodes a packet from an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass and overwrite the beginOfSegment() method to implement your own headers Dependency: https://github.com/pschatzmann/arduino-libopus - COggContainerEncoderEncoder for Ogg Container. Encodes a packet for an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass ond overwrite the writeHeader() method to implement your own header logic. When an optional encoder is specified in the constructor we package the encoded data. Dependency: https://github.com/pschatzmann/arduino-libopus - COggContainerOutputOutput class for the OggContainerEncoder. Each write is ending up as container entry - COnOffOutputSwitched Output class. Class which can be used to filter the output out based on a switch: If the switch is on the output is forwarded. If it is off it is just ignored - COpusAudioDecoderOpusAudioDecoder: Depends on https://github.com/pschatzmann/arduino-libopus.git - COpusAudioEncoderOpusAudioEncoder: Dependens on https://github.com/pschatzmann/arduino-libopus.git - COpusEncoderSettingsSetting for Opus Encoder where the following values are valid: -1 indicates that the default value should be used and that this codec is not setting the value - COpusOggDecoderOpus Decoder which uses the Ogg Container. See https://datatracker.ietf.org/doc/html/rfc7845. The audio data is transmitted in frames and the header information contains the sampler rate, channels and other critical info. Dependency: https://github.com/pschatzmann/arduino-libopus - COpusOggEncoderOpus Encoder which uses the Ogg Container: see https://datatracker.ietf.org/doc/html/rfc7845 Dependency: https://github.com/pschatzmann/arduino-libopus - COpusOggWriter - COpusSettingsSetting for Opus Decoder - COutputMixerMixing of multiple outputs to one final output - CParameterA constant value - CParseBufferWe try to keep the necessary buffer for parsing as small as possible, The data() method provides the start of the actual data and with consume we remove the processed data from the buffer to make space again - CParseObject - CPinInfoESP32Information for a PIN - CPinkNoiseGeneratorGenerates pink noise - CPitchShiftShifts the pitch by the indicated step size: e.g. 2 doubles the pitch - CPitchShiftInfoConfiguration for PitchShiftOutput: set the pitch_shift to define the shift - CPitchShiftOutputPitch Shift: Shifts the frequency up or down w/o impacting the length! We reduce the channels to 1 to calculate the pitch shift and provides the pitch shifted result in the correct number of channels. The pitch shifting is done with the help of a buffer that can have potentially multiple implementations - CPoppingSoundRemoverBig value gaps (at the beginning and the end of a recording) can lead to some popping sounds. We will try to set the values to 0 until the first transition thru 0 of the audio curve - CPortAudioConfigPortAudio information - CPortAudioStreamArduino Audio Stream using PortAudio - CPrint - CProcessedResult - CProgressStreamGeneric calss to measure the the total bytes which were processed in order to calculate the progress as a percentage of the total size - CProgressStreamInfoConfiguration for ProgressStream - CPWMAudioOutputCommon functionality for PWM output. Please use the PWMAudioOutput typedef instead which references the implementation - CPWMConfigConfiguration data for PWM audio output - CPWMDriverAVRExperimental: Audio output to PWM pins for the AVR. The AVR supports only up to 2 channels - CPWMDriverESP32Audio output to PWM pins for the ESP32. The ESP32 supports up to 16 channels - CPWMDriverMBEDAudio output to PWM pins for MBED based Arduino implementations - CPWMDriverRenesasAudio output to PWM pins for Renesas based Arduino implementations - CPWMDriverSTM32Audio output to PWM pins for STM32. We use one timer to generate the sample rate and one timer for the PWM signal - CQueueFIFO Queue which is based on a List - ►CQueueLockFreeLock Free Queue - Cnode - CQueueStreamAudioStream class which stores the data in a temporary queue buffer. The queue can be consumed e.g. by a callback function by calling readBytes(); - CRECT - CRectangeRectange FFT Window function - CReformatBaseStreamBase class for chained converting streams - CResampleConfigOptional Configuration object. The critical information is the channels and the step_size. All other information is not used - CResampleStreamDynamic Resampling. We can use a variable factor to speed up or slow down the playback - CRingBufferImplements a typed Ringbuffer - CRingBufferFileAn File backed Ring Buffer that we can use to receive streaming audio. We expect an open p_file as parameter - CRingBufferStreamA Stream backed by a Ringbuffer. We can write to the end and read from the beginning of the stream - CRTSPFormatAbtXAbtX format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatAudioToolsRTSPFormat which supports the AudioInfo class - CRTSPFormatG711G711 μ-Law format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats Packet intervall: 20, frame size: any - CRTSPFormatGSMGSM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatOpusOpus format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatPCMPCM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPFormatPCM8L8 format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats - CRTSPOutputWe can write PCM data to the RTSPOutput. This is encoded by the indicated encoder (e.g. SBCEncoder) and can be consumed by a RTSPServer. You have to make sure that the codec supports the provided audio format: e.g. GSM support only 8000 samples per second with one channel. Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPOutputPCMInfoPCMInfo subclass which provides the audio information from the related AudioStream Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPPCMAudioInfoPCMInfo subclass which provides the audio information from the AudioInfo parameter - CRTSPSourceFromAudioStreamSimple Facade which can turn AudioStream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CRTSPSourceStreamSimple Facade which can turn any Stream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library - CSBCDecoderDecoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcdec.c - CSBCEncoderEncoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcenc.c - CScaledParameterScales a dynamic parameter to the indicated range - CSDDirectWe access the files directy with an index. The index is determined by a recurseve tree walk thru the directory. Unfortunatly the SDTFAT library has it's own API which is incompatible with the SDT API - CSDIndexWe store all the relevant file names in an sequential index file. Form there we can access them via an index - CSilenceGeneratorProvides a fixed value (e.g. 0) as sound data. This can be used e.g. to test the output functionality which should optimally just output silence and no artifacts - CSilenceRemovalConverterRemoves any silence from the buffer that is longer then n samples with a amplitude below the indicated threshhold. If you process multiple channels you need to multiply the channels with the number of samples to indicate n - CSimpleChorusSimple Chorus effect with a single delay voice and mono output Chorus is effective between 15 and 20 miliseconds delay of original audio. Requires the sample rate when initialising - CSimpleContainerConfig - CSimpleContainerDataHeader - CSimpleContainerMetaDataHeader - CSimpleDelaySimple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples - CSimpleFlangerSimple Flanger Effect Consistig of a single voice flanger The flanger has an effective range between 0 and 15 miliseconds in this case dleay buffer should be set to sampleRate*3/200 Constructor requires internal delay in samples - CSimpleLPFSimpleLPF - CSimulatedAudioPotSimple simulated audio pot volume control inspired by https://eepower.com/resistor-guide/resistor-types/potentiometer-taper/# We split up the input/output curve into 2 linear pieces with a slow and a fast raising part. The slow raising part goes from (0,0) to (x,y). The fast raising part goes from (x,y) to (1,1) - CSineFromTableA sine generator based on a table. The table is created based using degrees where one full wave is 360 degrees - CSineWaveGeneratorGenerates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead - CSingleBufferA simple Buffer implementation which just uses a (dynamically sized) array - CSliceHelps to split up a big memory array into smaller slices. There are no additinal heap allocations! Example: if we have an array with 9 entries (1,2,3,4,5,6,7,8,9): slices(5) gives 2. slice(5,0) returns size 5 with 1,2,3,4,5 and slice(5,1) returns size 4 with 6,7,8,9! - CSmoothTransitionChanges the samples at the beginning or at the end to slowly ramp up the volume - CSOSFilterSecond Order Filter: Instead of manually cascading BiQuad filters, you can use a Second Order Sections filter (SOS). converted from https://github.com/tttapa/Filters/blob/master/src/SOSFilter.h Use float or double (and not a integer type) as type parameter - CSoundGeneratorBase class to define the abstract interface for the sound generating classes - CSoundGeneratorModulationSoundGenerator using the ModulationBaseClass to generate the samples - CSPDIFConfigSPDIF configuration - CSPDIFOutputOutput as 16 bit stereo SPDIF on the I2S data output pin - CSquareWaveGeneratorGenerates a square wave sound - CStackLIFO Stack which is based on a List - CStdioStream - CSTKChorusChorus Effect - CSTKChowningReverbJohn Chowning's reverberator class - CSTKEchoEcho Effect - CSTKEffectUse any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK - CSTKFreeVerbJezar at Dreampoint's FreeVerb, implemented in STK - CSTKGeneratorThe Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language.
+ CI2SDriverESP32Basic I2S API - for the ESP32. If we receive 1 channel, we expand the result to 2 channels + ►CI2SDriverESP32V1Basic I2S API for the ESP32 (using the new API). https://docs.espressif.com/projects/esp-idf/en/v5.0.1/esp32/api-reference/peripherals/i2s.html#i2s-communication-mode + CDriverCommon + CDriverI2S + CI2SDriverESP8266Basic I2S API - for the ESP8266 Only 16 bits are supported ! + CI2SDriverNanoBLEBasic I2S API - for the Arduino Nano BLE Sense See https://content.arduino.cc/assets/Nano_BLE_MCU-nRF52840_PS_v1.1.pdf Douplex mode (RXTX_MODE) is currently not supported, but it should be quite easy to implement + CI2SDriverSAMDBasic I2S API - for the SAMD + CI2SDriverSTM32Basic I2S API - for the STM32 Depends on https://github.com/pschatzmann/stm32f411-i2s We just add a write and read buffer and pass some parameters to the STM32 API! + CI2SStreamWe support the Stream interface for the I2S access. In addition we allow a separate mute pin which might also be used to drive a LED.. + CICYStreamIcecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method + CICYStreamBufferedICYStream implementation for the ESP32 based on a FreeRTOS task This is a Icecast/Shoutcast Audio Stream which splits the data into metadata and audio data. The Audio data is provided via the regular stream functions. The metadata is handled with the help of the MetaDataICY state machine and provided via a callback method + CICYUrlSetupResolve icy-metaint from HttpRequest and execute metadata callbacks + CID3v1 + CID3v1Enhanced + CID3v2 + CID3v2Frame + CID3v2FrameString + CIIRIIRFilter Converted from https://github.com/tttapa/Filters/blob/master/src/IIRFilter.h + CILBCDecoderDecoder for iLBC. Depends on https://github.com/pschatzmann/libilbc + CILBCEncoderEncoder for iLBC - Depends on https://github.com/pschatzmann/libopenilbc + CIMAState + CInputMergeMerges multiple input channels. The input must be mono! So if you provide 2 mono channels you get a stereo signal as result with the left channel from channel 0 and the right from channel 1 + CInputMixerMixerStream is mixing the input from Multiple Input Streams. All streams must have the same audo format (sample rate, channels, bits per sample) + Cint24_3bytes_t24bit integer which is used for I2S sound processing. The values are really using 3 bytes. It works only on little endian machines! + Cint24_4bytes_t24bit integer which is used for I2S sound processing. The values are represented as int32_t, but only 3 bytes are used. If you assign values which are too big, they are clipped + CJpegOpenCVDisplay image with opencv to be used on the desktop + CJpegTFTDisplay jpeg image using https://github.com/Bodmer/TFT_eSPI and https://github.com/Bodmer/JPEGDecoder + CJupyterAudioTOutput to Jupyter. We write the data just to a file from where we can load the data again for different representations + CLastSampleFaderIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0. Typless implementation + CLastSampleFaderTIf we end audio and the last sample is not close to 0 we can hear a popping noise. This functionality brings the last value slowly to 0 + CLC3DecoderDecoder for LC3. Depends on https://github.com/pschatzmann/arduino-liblc3 + CLC3EncoderEncoder for LC3 - Depends on https://github.com/pschatzmann/arduino-liblc3 + CLEDOutputLED output using the FastLED library + CLEDOutputConfig + CLEDOutputUnoR4LED output using the R4 LED matrix library + CLEDOutputUnoR4Config + CLinearVolumeControlThe simplest possible implementation of a VolumeControl: The input = output which describes a linear curve. You would use this implementation if you physically connect an audio pot! + ►CListDouble linked list + CIterator + CNode + CLockA simple RIA locking class for the ESP32 using _lock_t + CLogarithmicVolumeControlParametric Logarithmic volume control. Using the formula pow(b,input) * a - a, where b is b = pow(((1/ym)-1), 2) and a is a = 1.0 / (b - 1.0). The parameter ym is determining the steepness. See https://electronics.stackexchange.com/questions/304692/formula-for-logarithmic-audio-taper-pot + CMaximilianAudioTools integration with Maximilian + CMeasuringStreamClass which measures the truput + ►CMedianFilterAn embedded friendly, fast one-dimensional median filter algorithm implementation in C and C++ Useful for spike and noise removal from analog signals or other DSP Also known as "salt-and-pepper noise" or "impulse noise" filter + CsMedianFilter_t + CsMedianNode_t + CMemoryManagerMemoryManager which activates the use of external SPIRAM memory. When external memory is in use, the allocation strategy is to initially try to satisfy smaller allocation requests with internal memory and larger requests with external memory. This sets the limit between the two, as well as generally enabling allocation in external memory + CMemoryOutputWrites to a preallocated memory + CMemoryStreamA simple Stream implementation which is backed by allocated memory + ►CMetaDataFilterClass which filters out ID3v1 and ID3v2 Metadata and provides only the audio data to the decoder + CID3v2ID3 verion 2 TAG Header (10 bytes) + CMetaDataICYIcecast/Shoutcast Metadata Handling. Metadata class which splits the data into audio and metadata. The result is provided via callback methods. see https://www.codeproject.com/Articles/11308/SHOUTcast-Stream-Ripper + CMetaDataID3Simple ID3 Meta Data Parser which supports ID3 V1 and V2 and implements the Stream interface. You just need to set the callback(s) to receive the result and copy the audio data to this stream + CMetaDataID3BaseID3 Meta Data Common Functionality + CMetaDataID3V1Simple ID3 Meta Data API which supports ID3 V1 + CMetaDataID3V2Simple ID3 Meta Data API which supports ID3 V2: We only support the "TALB", "TOPE", "TIT2", "TCON" tags + CMetaDataOutputID3 and Icecast/Shoutcast metadata output support. Just write the audio data to an object of this class and receive the metadata via the callback + CMiniAudioConfigConfiguration for MiniAudio + CMiniAudioStreamMiniAudio: https://miniaud.io/ + CModulationBaseClassClass provides a wave table that can be populated with a number of preallocated waveforms. These can be used to generate audio in themselves or to modulate The parameters of another effect. Class initialised with sample rate + CMozziConfigMozzi Configuration for input or output stream + CMozziGeneratorSupport for https://sensorium.github.io/Mozzi/ Define your updateControl() method. Define your updateAudio() method. Start by calling begin(control_rate) do not call audioHook(); in the loop ! + CMozziStreamWe use the output functionality of Mozzi to output audio data. We expect the data as array of int16_t with one or two channels. Though we support the setting of a sample rate, we recommend to use the default sample rate from Mozzi which is available with the AUDIO_RATE define + CMP3DecoderHelixMP3 Decoder using libhelix: https://github.com/pschatzmann/arduino-libhelix This is basically just a simple wrapper to provide AudioInfo and AudioInfoSupport + CMP3DecoderMADMP3 Decoder using https://github.com/pschatzmann/arduino-libmad + CMP3DecoderMiniMP3 Decoder using https://github.com/pschatzmann/minimp3. This decoder does not provide any good results and it is not suited to decode any audio above 32000 on an ESP32. So the sample rate is limited by the MINIMP3_MAX_SAMPLE_RATE variable + CMP3EncoderLAMEEncodes PCM data to the MP3 format and writes the result to a stream This is basically just a wrapper using https://github.com/pschatzmann/arduino-liblame + CMP4AtomRepresents a single MPEG4 atom + CMP4ParseBuffer + CMTSDecoderMPEG-TS (MTS) decoder. Extracts the AAC audio data from a MPEG-TS (MTS) data stream. You can define the relevant stream types via the API. Required dependency: https://github.com/pschatzmann/arduino-tsdemux + CMultiConverterCombines multiple converters + CMultiOutputReplicates the output to multiple destinations + CMusicalNotesDetermination of the frequency of a music note + CNano_BLE_freq_infoMapping Frequency constants to available frequencies + CNano_BLE_ratio_infoMapping from Ratio Constants to frequency ratios + CNBufferA lock free N buffer. If count=2 we create a DoubleBuffer, if count=3 a TripleBuffer etc + CNoFilterNo change to the input + CNOPConverterDummy converter which does nothing + CNullStreamThe Arduino Stream which provides silence and simulates a null device when used as audio target or audio source + CNumberConverterConverts from a source to a target number with a different type + CNumberFormatConverterStreamConverter which converts between bits_per_sample and 16 bits. The templated NumberFormatConverterStreamT class is used based on the information provided by the bits_per_sample in the configuration + CNumberFormatConverterStreamTConverter which converts from source bits_per_sample to target bits_per_sample + CNumberReaderReads n numbers from an Arduino Stream + CNuttallNuttall FFT Window function + COggContainerDecoderDecoder for Ogg Container. Decodes a packet from an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass and overwrite the beginOfSegment() method to implement your own headers Dependency: https://github.com/pschatzmann/arduino-libopus + COggContainerEncoderEncoder for Ogg Container. Encodes a packet for an Ogg container. The Ogg begin segment contains the AudioInfo structure. You can subclass ond overwrite the writeHeader() method to implement your own header logic. When an optional encoder is specified in the constructor we package the encoded data. Dependency: https://github.com/pschatzmann/arduino-libopus + COggContainerOutputOutput class for the OggContainerEncoder. Each write is ending up as container entry + COnOffOutputSwitched Output class. Class which can be used to filter the output out based on a switch: If the switch is on the output is forwarded. If it is off it is just ignored + COpusAudioDecoderOpusAudioDecoder: Depends on https://github.com/pschatzmann/arduino-libopus.git + COpusAudioEncoderOpusAudioEncoder: Dependens on https://github.com/pschatzmann/arduino-libopus.git + COpusEncoderSettingsSetting for Opus Encoder where the following values are valid: -1 indicates that the default value should be used and that this codec is not setting the value + COpusOggDecoderOpus Decoder which uses the Ogg Container. See https://datatracker.ietf.org/doc/html/rfc7845. The audio data is transmitted in frames and the header information contains the sampler rate, channels and other critical info. Dependency: https://github.com/pschatzmann/arduino-libopus + COpusOggEncoderOpus Encoder which uses the Ogg Container: see https://datatracker.ietf.org/doc/html/rfc7845 Dependency: https://github.com/pschatzmann/arduino-libopus + COpusOggWriter + COpusSettingsSetting for Opus Decoder + COutputMixerMixing of multiple outputs to one final output + CParameterA constant value + CParseBufferWe try to keep the necessary buffer for parsing as small as possible, The data() method provides the start of the actual data and with consume we remove the processed data from the buffer to make space again + CParseObject + CPinInfoESP32Information for a PIN + CPinkNoiseGeneratorGenerates pink noise + CPitchShiftShifts the pitch by the indicated step size: e.g. 2 doubles the pitch + CPitchShiftInfoConfiguration for PitchShiftOutput: set the pitch_shift to define the shift + CPitchShiftOutputPitch Shift: Shifts the frequency up or down w/o impacting the length! We reduce the channels to 1 to calculate the pitch shift and provides the pitch shifted result in the correct number of channels. The pitch shifting is done with the help of a buffer that can have potentially multiple implementations + CPoppingSoundRemoverBig value gaps (at the beginning and the end of a recording) can lead to some popping sounds. We will try to set the values to 0 until the first transition thru 0 of the audio curve + CPortAudioConfigPortAudio information + CPortAudioStreamArduino Audio Stream using PortAudio + CPrint + CProcessedResult + CProgressStreamGeneric calss to measure the the total bytes which were processed in order to calculate the progress as a percentage of the total size + CProgressStreamInfoConfiguration for ProgressStream + CPWMAudioOutputCommon functionality for PWM output. Please use the PWMAudioOutput typedef instead which references the implementation + CPWMConfigConfiguration data for PWM audio output + CPWMDriverAVRExperimental: Audio output to PWM pins for the AVR. The AVR supports only up to 2 channels + CPWMDriverESP32Audio output to PWM pins for the ESP32. The ESP32 supports up to 16 channels + CPWMDriverMBEDAudio output to PWM pins for MBED based Arduino implementations + CPWMDriverRenesasAudio output to PWM pins for Renesas based Arduino implementations + CPWMDriverSTM32Audio output to PWM pins for STM32. We use one timer to generate the sample rate and one timer for the PWM signal + CQueueFIFO Queue which is based on a List + ►CQueueLockFreeLock Free Queue + Cnode + CQueueStreamAudioStream class which stores the data in a temporary queue buffer. The queue can be consumed e.g. by a callback function by calling readBytes(); + CRECT + CRectangeRectange FFT Window function + CReformatBaseStreamBase class for chained converting streams + CResampleConfigOptional Configuration object. The critical information is the channels and the step_size. All other information is not used + CResampleStreamDynamic Resampling. We can use a variable factor to speed up or slow down the playback + CRingBufferImplements a typed Ringbuffer + CRingBufferFileAn File backed Ring Buffer that we can use to receive streaming audio. We expect an open p_file as parameter + CRingBufferStreamA Stream backed by a Ringbuffer. We can write to the end and read from the beginning of the stream + CRTSPFormatAbtXAbtX format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatAudioToolsRTSPFormat which supports the AudioInfo class + CRTSPFormatG711G711 μ-Law format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats Packet intervall: 20, frame size: any + CRTSPFormatGSMGSM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatOpusOpus format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatPCMPCM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPFormatPCM8L8 format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats + CRTSPOutputWe can write PCM data to the RTSPOutput. This is encoded by the indicated encoder (e.g. SBCEncoder) and can be consumed by a RTSPServer. You have to make sure that the codec supports the provided audio format: e.g. GSM support only 8000 samples per second with one channel. Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPOutputPCMInfoPCMInfo subclass which provides the audio information from the related AudioStream Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPPCMAudioInfoPCMInfo subclass which provides the audio information from the AudioInfo parameter + CRTSPSourceFromAudioStreamSimple Facade which can turn AudioStream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CRTSPSourceStreamSimple Facade which can turn any Stream into a IAudioSource. This way we can e.g. use an I2SStream as source to stream data Depends on the https://github.com/pschatzmann/Micro-RTSP-Audio/ library + CSBCDecoderDecoder for SBC. Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcdec.c + CSBCEncoderEncoder for SBC - Depends on https://github.com/pschatzmann/arduino-libsbc. Inspired by sbcenc.c + CScaledParameterScales a dynamic parameter to the indicated range + CSDDirectWe access the files directy with an index. The index is determined by a recurseve tree walk thru the directory. Unfortunatly the SDTFAT library has it's own API which is incompatible with the SDT API + CSDIndexWe store all the relevant file names in an sequential index file. Form there we can access them via an index + CSilenceGeneratorProvides a fixed value (e.g. 0) as sound data. This can be used e.g. to test the output functionality which should optimally just output silence and no artifacts + CSilenceRemovalConverterRemoves any silence from the buffer that is longer then n samples with a amplitude below the indicated threshhold. If you process multiple channels you need to multiply the channels with the number of samples to indicate n + CSimpleChorusSimple Chorus effect with a single delay voice and mono output Chorus is effective between 15 and 20 miliseconds delay of original audio. Requires the sample rate when initialising + CSimpleContainerConfig + CSimpleContainerDataHeader + CSimpleContainerMetaDataHeader + CSimpleDelaySimple Delay effect consiting of a single tap delay with Effect Gain and feed back controls Constructor requires internal delay in samples + CSimpleFlangerSimple Flanger Effect Consistig of a single voice flanger The flanger has an effective range between 0 and 15 miliseconds in this case dleay buffer should be set to sampleRate*3/200 Constructor requires internal delay in samples + CSimpleLPFSimpleLPF + CSimulatedAudioPotSimple simulated audio pot volume control inspired by https://eepower.com/resistor-guide/resistor-types/potentiometer-taper/# We split up the input/output curve into 2 linear pieces with a slow and a fast raising part. The slow raising part goes from (0,0) to (x,y). The fast raising part goes from (x,y) to (1,1) + CSineFromTableA sine generator based on a table. The table is created based using degrees where one full wave is 360 degrees + CSineWaveGeneratorGenerates a Sound with the help of sin() function. If you plan to change the amplitude or frequency (incrementally), I suggest to use SineFromTable instead + CSingleBufferA simple Buffer implementation which just uses a (dynamically sized) array + CSliceHelps to split up a big memory array into smaller slices. There are no additinal heap allocations! Example: if we have an array with 9 entries (1,2,3,4,5,6,7,8,9): slices(5) gives 2. slice(5,0) returns size 5 with 1,2,3,4,5 and slice(5,1) returns size 4 with 6,7,8,9! + CSmoothTransitionChanges the samples at the beginning or at the end to slowly ramp up the volume + CSOSFilterSecond Order Filter: Instead of manually cascading BiQuad filters, you can use a Second Order Sections filter (SOS). converted from https://github.com/tttapa/Filters/blob/master/src/SOSFilter.h Use float or double (and not a integer type) as type parameter + CSoundGeneratorBase class to define the abstract interface for the sound generating classes + CSoundGeneratorModulationSoundGenerator using the ModulationBaseClass to generate the samples + CSPDIFConfigSPDIF configuration + CSPDIFOutputOutput as 16 bit stereo SPDIF on the I2S data output pin + CSquareWaveGeneratorGenerates a square wave sound + CStackLIFO Stack which is based on a List + CStdioStream + CSTKChorusChorus Effect + CSTKChowningReverbJohn Chowning's reverberator class + CSTKEchoEcho Effect + CSTKEffectUse any effect from the STK framework: e.g. Chorus, Echo, FreeVerb, JCRev, PitShift... https://github.com/pschatzmann/Arduino-STK + CSTKFreeVerbJezar at Dreampoint's FreeVerb, implemented in STK + CSTKGeneratorThe Synthesis ToolKit in C++ (STK) is a set of open source audio signal processing and algorithmic synthesis classes written in the C++ programming language.
- CSTKLentPitShiftPitch shifter effect class based on the Lent algorithm - CSTKNReverbCCRMA's NRev reverberator class - CSTKPerryReverbPerry's simple reverberator class - CSTKPitShiftPitch shifter effect class based on the Lent algorithm - CSTKStreamSTK Stream for Instrument or Voicer - CStrA simple wrapper to provide string functions on char*. If the underlying char* is a const we do not allow any updates; The ownership of the chr* must be managed externally! - CStream - CStreamCopyWe provide the typeless StreamCopy as a subclass of StreamCopyT - CStreamCopyTTyped Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples - CStreamingDecoderA Streaming Decoder where we provide both the input and output as streams - CStrExtStr which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting - ►CSynthesizerA simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels - CKeyParameter - CSynthesizerKeyArduino GPIO pin to note assossiation - CTestGeneratorGenerates a test signal which is easy to check because the values are incremented or decremented by 1 - CTfLiteAbstractRecognizeCommandsBase class for implementing different primitive decoding models on top of the instantaneous results from running an audio recognition model on a single window of samples - CTfLiteAudioErrorReporterError Reporter using the Audio Tools Logger - CTfLiteAudioStreamTfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data) - CTfLiteAudioStreamBaseAstract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers - CTfLiteConfigConfiguration settings for TfLiteAudioStream - CTfLiteMicroSpeachWriterTfLiteMicroSpeachWriter for Audio Data - ►CTfLiteMicroSpeechRecognizeCommandsThis class is designed to apply a very primitive decoding model on top of the instantaneous results from running an audio recognition model on a single window of samples. It applies smoothing over time so that noisy individual label scores are averaged, increasing the confidence that apparent matches are real. To use it, you should create a class object with the configuration you want, and then feed results from running a TensorFlow model into the processing method. The timestamp for each subsequent call should be increasing from the previous, since the class is designed to process a stream of data over time - CResult - CTfLiteQuantizerQuantizer that helps to quantize and dequantize between float and int8 - CTfLiteReaderInput class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce - CTfLiteSineReaderGenerate a sine output from a model that was trained on the sine method. (=hello_world) - CTfLiteWriterOutput class which interprets audio data if TfLiteAudioStream is treated as audio sink - CThrottleThrottle the sending of the audio data to limit it to the indicated sample rate - CThrottleConfigConfigure Throttle setting - CTimedStreamAudioStream class that can define a start and (an optional) stop time Usually it is used to wrap an Audio Sink (e.g. I2SStream), but wrapping an Audio Source is supported as well. Only wrap classes which represent PCM data! - CTimerAlarmRepeatingCommon Interface definition for TimerAlarmRepeating - CTimerAlarmRepeatingDriverAVRRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverBase - CTimerAlarmRepeatingDriverESP8266Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverMBEDRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerAlarmRepeatingDriverRenesasRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating. By default we use a new GPT timer. You can also request 1 AGT timer by calling setTimer(1); - CTimerAlarmRepeatingDriverSTM32STM32 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating - CTimerCallbackAudioStreamCallback driven Audio Source (rx_tx_mode==RX_MODE) or Audio Sink (rx_tx_mode==TX_MODE). This class allows to to integrate external libraries in order to consume or generate a data stream which is based on a timer - CTimerCallbackAudioStreamInfoTimerCallbackAudioStream Configuration - ►CTransformationReaderConverterStream Helper class which implements the converting readBytes with the help of write - CAdapterPrintToArray - CTremoloTremolo AudioEffect - CTriangleTriangle FFT Window function - CUDPStream - CUrlURL parser which breaks a full url string up into its individual parts - CURLHistory - CURLLoader - CURLStreamRepresents the content of a URL as Stream. We use the WiFi.h API - CURLStreamBufferedURLStream implementation for the ESP32 based on a separate FreeRTOS task - CVariableSpeedRingBufferOptimized Buffer implementation for Pitch Shift. We try to interpolate the samples and restore the phase when the read pointer and write pointer overtake each other - CVariableSpeedRingBuffer180Varialbe speed ring buffer where we read with 0 and 180 degree and blend the result to prevent overrun artifacts. See https://github.com/YetAnotherElectronicsChannel/STM32_DSP_PitchShift - CVariableSpeedRingBufferSimpleVery Simple Buffer implementation for Pitch Shift. We write in constant speed, but reading can be done in a variable speed. We will hear some noise when the buffer read and write pointers overrun each other - CVBANConfig - CVBANStreamVBAN Audio Source and Sink for the ESP32. For further details please see https://vb-audio.com/Voicemeeter/vban.htm . Inspired by https://github.com/rkinnett/ESP32-VBAN-Audio-Source/tree/master and https://github.com/rkinnett/ESP32-VBAN-Network-Audio-Player - ►CVectorVector implementation which provides the most important methods as defined by std::vector. This class it is quite handy to have and most of the times quite better then dealing with raw c arrays - CiteratorIterator for the Vector class - CVideoAudioBufferedSyncLogic to Synchronize video and audio output: we use a buffer to store the audio and instead of delaying the frames with delay() we play audio. The bufferSize defines the audio buffer in bytes. The correctionMs is used to slow down or speed up the playback of the video to prevent any audio buffer underflows - CVideoAudioSyncLogic to Synchronize video and audio output: This is the minimum implementatin which actually does not synchronize, but directly processes the data. No additinal memory is used! Provide your own optimized platform specific implementation - CVideoOutputAbstract class for video playback. This class is used to assemble a complete video frame in memory - CVolumeControlAbstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal - CVolumeOutputA simple class to determine the volume - CVolumeStreamAdjust the volume of the related input or output: To work properly the class needs to know the bits per sample and number of channels! AudioChanges are forwareded to the related Print or Stream class - CVolumeStreamConfigConfig for VolumeStream - CVorbisDecoderVorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-idec https://github.com/pschatzmann/arduino-libopus.git - CVS1053ConfigConfiguration for VS1053Stream - CVS1053StreamVS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration; - CWAVAudioInfoSound information which is available in the WAV header - CWAVDecoderA simple WAVDecoder: We parse the header data on the first record to determine the format. If no AudioDecoderExt is specified we just write the PCM data to the output that is defined by calling setOutput(). You can define a ADPCM decoder to decode WAV files that contain ADPCM data - CWAVEncoderA simple WAV file encoder. If no AudioEncoderExt is specified the WAV file contains PCM data, otherwise it is encoded as ADPCM. The WAV header is written with the first writing of audio data. Calling begin() is making sure that the header is written again - CWAVFormatX - CWAVHeaderParser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE - CWavIMAAudioInfoSound information which is available in the WAV header - adjusted for IMA ADPCM - CWavIMADecoderObsolete: WavIMADecoder - based on WAVDecoder - We parse the header data as we receive it and send the sound data to the stream which was indicated in the constructor. Only WAV files with WAVE_FORMAT_IMA_ADPCM are supported by this codec! - CWavIMAHeader - CWelchWelch FFT Window function - CWhiteNoiseGeneratorGenerates a random noise sound with the help of rand() function - CWindowFunctionFFT Window Function - CWM8960ConfigConfiguration for WM8960 - CWM8960StreamStream for reading and writing audio data using the WM8960 Codec Chip You need to install https://github.com/pschatzmann/arduino-wm8960 + CSTKLentPitShiftPitch shifter effect class based on the Lent algorithm + CSTKNReverbCCRMA's NRev reverberator class + CSTKPerryReverbPerry's simple reverberator class + CSTKPitShiftPitch shifter effect class based on the Lent algorithm + CSTKStreamSTK Stream for Instrument or Voicer + CStrA simple wrapper to provide string functions on char*. If the underlying char* is a const we do not allow any updates; The ownership of the chr* must be managed externally! + CStream + CStreamCopyWe provide the typeless StreamCopy as a subclass of StreamCopyT + CStreamCopyTTyped Stream Copy which supports the conversion from channel to 2 channels. We make sure that we allways copy full samples + CStreamingDecoderA Streaming Decoder where we provide both the input and output as streams + CStrExtStr which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting + ►CSynthesizerA simple Synthesizer which can generate sound having multiple keys pressed. The main purpose of this class is managing the synthezizer channels + CKeyParameter + CSynthesizerKeyArduino GPIO pin to note assossiation + CTestGeneratorGenerates a test signal which is easy to check because the values are incremented or decremented by 1 + CTfLiteAbstractRecognizeCommandsBase class for implementing different primitive decoding models on top of the instantaneous results from running an audio recognition model on a single window of samples + CTfLiteAudioErrorReporterError Reporter using the Audio Tools Logger + CTfLiteAudioStreamTfLiteAudioStream which uses Tensorflow Light to analyze the data. If it is used as a generator (where we read audio data) + CTfLiteAudioStreamBaseAstract TfLiteAudioStream to provide access to TfLiteAudioStream for Reader and Writers + CTfLiteConfigConfiguration settings for TfLiteAudioStream + CTfLiteMicroSpeachWriterTfLiteMicroSpeachWriter for Audio Data + ►CTfLiteMicroSpeechRecognizeCommandsThis class is designed to apply a very primitive decoding model on top of the instantaneous results from running an audio recognition model on a single window of samples. It applies smoothing over time so that noisy individual label scores are averaged, increasing the confidence that apparent matches are real. To use it, you should create a class object with the configuration you want, and then feed results from running a TensorFlow model into the processing method. The timestamp for each subsequent call should be increasing from the previous, since the class is designed to process a stream of data over time + CResult + CTfLiteQuantizerQuantizer that helps to quantize and dequantize between float and int8 + CTfLiteReaderInput class which provides the next value if the TfLiteAudioStream is treated as an audio sourcce + CTfLiteSineReaderGenerate a sine output from a model that was trained on the sine method. (=hello_world) + CTfLiteWriterOutput class which interprets audio data if TfLiteAudioStream is treated as audio sink + CThrottleThrottle the sending of the audio data to limit it to the indicated sample rate + CThrottleConfigConfigure Throttle setting + CTimedStreamAudioStream class that can define a start and (an optional) stop time Usually it is used to wrap an Audio Sink (e.g. I2SStream), but wrapping an Audio Source is supported as well. Only wrap classes which represent PCM data! + CTimerAlarmRepeatingCommon Interface definition for TimerAlarmRepeating + CTimerAlarmRepeatingDriverAVRRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverBase + CTimerAlarmRepeatingDriverESP8266Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverMBEDRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerAlarmRepeatingDriverRenesasRepeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating. By default we use a new GPT timer. You can also request 1 AGT timer by calling setTimer(1); + CTimerAlarmRepeatingDriverSTM32STM32 Repeating Timer functions for repeated execution: Plaease use the typedef TimerAlarmRepeating + CTimerCallbackAudioStreamCallback driven Audio Source (rx_tx_mode==RX_MODE) or Audio Sink (rx_tx_mode==TX_MODE). This class allows to to integrate external libraries in order to consume or generate a data stream which is based on a timer + CTimerCallbackAudioStreamInfoTimerCallbackAudioStream Configuration + ►CTransformationReaderConverterStream Helper class which implements the converting readBytes with the help of write + CAdapterPrintToArray + CTremoloTremolo AudioEffect + CTriangleTriangle FFT Window function + CUDPStream + CUrlURL parser which breaks a full url string up into its individual parts + CURLHistory + CURLLoader + CURLStreamRepresents the content of a URL as Stream. We use the WiFi.h API + CURLStreamBufferedURLStream implementation for the ESP32 based on a separate FreeRTOS task + CVariableSpeedRingBufferOptimized Buffer implementation for Pitch Shift. We try to interpolate the samples and restore the phase when the read pointer and write pointer overtake each other + CVariableSpeedRingBuffer180Varialbe speed ring buffer where we read with 0 and 180 degree and blend the result to prevent overrun artifacts. See https://github.com/YetAnotherElectronicsChannel/STM32_DSP_PitchShift + CVariableSpeedRingBufferSimpleVery Simple Buffer implementation for Pitch Shift. We write in constant speed, but reading can be done in a variable speed. We will hear some noise when the buffer read and write pointers overrun each other + CVBANConfig + CVBANStreamVBAN Audio Source and Sink for the ESP32. For further details please see https://vb-audio.com/Voicemeeter/vban.htm . Inspired by https://github.com/rkinnett/ESP32-VBAN-Audio-Source/tree/master and https://github.com/rkinnett/ESP32-VBAN-Network-Audio-Player + ►CVectorVector implementation which provides the most important methods as defined by std::vector. This class it is quite handy to have and most of the times quite better then dealing with raw c arrays + CiteratorIterator for the Vector class + CVideoAudioBufferedSyncLogic to Synchronize video and audio output: we use a buffer to store the audio and instead of delaying the frames with delay() we play audio. The bufferSize defines the audio buffer in bytes. The correctionMs is used to slow down or speed up the playback of the video to prevent any audio buffer underflows + CVideoAudioSyncLogic to Synchronize video and audio output: This is the minimum implementatin which actually does not synchronize, but directly processes the data. No additinal memory is used! Provide your own optimized platform specific implementation + CVideoOutputAbstract class for video playback. This class is used to assemble a complete video frame in memory + CVolumeControlAbstract class for handling of the linear input volume to determine the multiplication factor which should be applied to the audio signal + CVolumeOutputA simple class to determine the volume + CVolumeStreamAdjust the volume of the related input or output: To work properly the class needs to know the bits per sample and number of channels! AudioChanges are forwareded to the related Print or Stream class + CVolumeStreamConfigConfig for VolumeStream + CVorbisDecoderVorbis Streaming Decoder using https://github.com/pschatzmann/arduino-libvorbis-idec https://github.com/pschatzmann/arduino-libopus.git + CVS1053ConfigConfiguration for VS1053Stream + CVS1053StreamVS1053 Output Interface which processes PCM data by default. If you want to write encoded data set is_encoded_data = true in the configuration; + CWAVAudioInfoSound information which is available in the WAV header + CWAVDecoderA simple WAVDecoder: We parse the header data on the first record to determine the format. If no AudioDecoderExt is specified we just write the PCM data to the output that is defined by calling setOutput(). You can define a ADPCM decoder to decode WAV files that contain ADPCM data + CWAVEncoderA simple WAV file encoder. If no AudioEncoderExt is specified the WAV file contains PCM data, otherwise it is encoded as ADPCM. The WAV header is written with the first writing of audio data. Calling begin() is making sure that the header is written again + CWAVFormatX + CWAVHeaderParser for Wav header data for details see https://de.wikipedia.org/wiki/RIFF_WAVE + CWavIMAAudioInfoSound information which is available in the WAV header - adjusted for IMA ADPCM + CWavIMADecoderObsolete: WavIMADecoder - based on WAVDecoder - We parse the header data as we receive it and send the sound data to the stream which was indicated in the constructor. 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