AutioTools Configuration. More...

Classes
class	AdapterAudioOutputToAudioStream
	Wrapper which converts a AudioStream to a AudioOutput. More...
class	AdapterAudioStreamToAudioOutput
	Wrapper which converts a AudioStream to a AudioOutput. More...
class	AdapterPrintToAudioOutput
	Wrapper which converts a Print to a AudioOutput. More...
class	Allocator
	Memory allocateator which uses malloc. More...
class	AllocatorESP32
	ESP32 Memory allocateator which forces the allocation with the defined attributes (default MALLOC_CAP_8BIT \| MALLOC_CAP_INTERNAL). More...
class	AllocatorExt
	Memory allocateator which uses ps_malloc (on the ESP32) and if this fails it resorts to malloc. More...
class	AllocatorPSRAM
	Memory allocateator which uses ps_malloc to allocate the memory in PSRAM on the ESP32. More...
class	AllocatorZephyr
	Zephyr allocator using the kernel heap (k_malloc / k_free). This is the standard allocator for Zephyr builds. More...
struct	AppropriateSumType
	We reduce the number of samples in a datastream by summing (binning) or averaging. This will result in the same number of channels but binSize times less samples. If Average is true the sum is divided by binSize. More...
struct	AppropriateSumType< int16_t >
struct	AppropriateSumType< int24_t >
struct	AppropriateSumType< int32_t >
struct	AppropriateSumType< int8_t >
class	AudioDecoder
	Decoding of encoded audio into PCM data. More...
class	AudioDecoderExt
	Extended AudioDecoder interface to support block size configuration. More...
class	AudioEncoder
	Encoding of PCM data. More...
class	AudioEncoderExt
	Extended AudioEncoder interface to support block size configuration. More...
struct	AudioInfo
	Basic Audio information which drives e.g. I2S. More...
struct	AudioInfoFormat
	AudioInfo extended with a WAVEFORMATEX-style codec tag (the "wav code"): identifies the codec (PCM, AAC, ALAC, ...) - AudioFormat::UNKNOWN if not determined. mime() derives the corresponding mime type from format - not a separately stored field, so it can never go stale. More...
class	AudioInfoSource
	Supports the subscription to audio change notifications. More...
class	AudioInfoSupport
	Supports changes to the sampling rate, bits and channels. More...
class	AudioInputMonitor
	Monitors audio input and reports if the volume exceeds a specified limit within a given period. More...
class	AudioOutput
	Abstract Audio Ouptut class. More...
class	AudioOutputAdapter
	Base class for Output Adpapters. More...
class	AudioStream
	Base class for all Audio Streams. It support the boolean operator to test if the object is ready with data. More...
class	AudioStreamWrapper
	To be used to support implementations where the readBytes is not virtual. More...
class	AudioTime
	Tools for calculating timer values. More...
class	AudioTimer
	Common Interface definition for AudioTimer. More...
class	AudioTimerDriverAVR
	Repeating Timer functions for repeated execution: Plaease use the typedef AudioTimer. More...
class	AudioTimerDriverBase
class	AudioTimerDriverESP32
	Repeating Timer functions for simple scheduling of repeated execution using esp_timer_create(). More...
class	AudioTimerDriverESP32Legacy
	Repeating Timer functions for simple scheduling of repeated execution. The basic logic is taken from https://www.toptal.com/embedded/esp32-audio-sampling. Plaease use the typedef AudioTimer. More...
class	AudioTimerDriverESP8266
	Repeating Timer functions for repeated execution: Plaease use the typedef AudioTimer. More...
class	AudioTimerDriverLinux
	Repeating software timer driver for Desktop using std::thread. More...
class	AudioTimerDriverMBED
	Repeating Timer functions for repeated execution: Plaease use the typedef AudioTimer. More...
class	AudioTimerDriverRenesas
	Repeating Timer functions for repeated execution: Plaease use the typedef AudioTimer. By default we use a new GPT timer. You can also request 1 AGT timer by calling setTimer(1);. More...
class	AudioTimerDriverSTM32
	STM32 Repeating Timer functions for repeated execution: Please use the typedef AudioTimer. By default the TIM1 is used. More...
class	AudioTimerDriverZephyr
	Repeating Timer driver for Zephyr using k_timer. More...
class	AudioTimeSourceStream
	AudioTimeSourceStream: A stream that provides time information based on the audio data actually processed - implements the TimeSource interface. millis() (inherited, unmodified) is plain wall time; playbackTime() is the audio-derived progress, which is what a consumer that must stay tied to genuine playback (e.g. PacedVideoOutput) should use instead - see playbackTime()'s own comment for why they can diverge. More...
class	AudioWriter
	E.g. used by Encoders and Decoders. More...
class	AudioWriterToAudioOutput
	Adapter class which lets an AudioWriter behave like a Print. More...
class	BandPassFilter
	Second-order band-pass filter (BiQuad DF2). Passes frequencies near the center frequency and attenuates those further away. The bandwidth is controlled by the Q factor: higher Q produces a narrower passband. Coefficients are derived from the Audio EQ Cookbook. More...
class	BandPassFilterDF1
	Second-order band-pass filter (BiQuad DF1). Same coefficients as BandPassFilter; use this (not BandPassFilter) when T is a bounded fixed-point type like q1_14_t – see LowPassFilterDF1 for why. More...
class	BaseBuffer
	Shared functionality of all buffers. More...
class	BaseConverter
	Abstract Base class for Converters A converter is processing the data in the indicated array. More...
class	BaseStream
	Base class for all Streams. It relies on write(const uint8_t buffer, size_t size) and readBytes(uint8_t buffer, size_t length). More...
class	Bin
	Provides reduced sampling rates through binning. More...
class	BinT
struct	BiQuadCoeffs
	Biquad coefficients, always computed in float regardless of the filter's sample type T. Intermediate values like the angular cutoff frequency w0 range up to PI (~3.14) at Nyquist, which overflows bounded fixed-point sample types like q1_14_t (range ~[-2,2)); only the final, range-safe coefficients (typically within [-2,2) for a stable filter) get cast into T when assigned to a BiQuadDF1/BiQuadDF2 instance. More...
class	BiQuadDF1
	Second-order IIR filter in Direct Form I. Maintains separate input and output histories (x and y delay lines). Requires more memory than DF2 but is less susceptible to quantization issues with fixed-point arithmetic. More...
class	BiQuadDF2
	Second-order IIR filter in Direct Form II. Uses a single delay line, requiring less memory than DF1. This is the base class for the ready-to-use filter types (LowPassFilter, HighPassFilter, BandPassFilter, NotchFilter, LowShelfFilter, HighShelfFilter) which compute their coefficients from frequency, sample rate and Q/gain parameters. More...
class	BufferedArray
	Class which is usfull ot provide incremental data access e.g. for EdgeImpulse which request data with an offset and length starting from 0 up to the buffer length, restarting at 0 again. More...
class	BufferedStream
	The Arduino Stream supports operations on single characters. This is usually not the best way to push audio information, but we will support it anyway - by using a buffer. On reads: if the buffer is empty it gets refilled. More...
class	CallbackConverterT
	You can provide a lambda expression to convert the data. More...
class	CallbackStream
	CallbackStream: A Stream that allows to register callback methods for accessing and providing and updating data. The callbacks can be lambda expressions. The update callback can also be used just to read the data without changing it. Warning: this class does not propagate audio info changes to the target stream. You need to do this manually. More...
class	CatStream
	Provides data from a concatenation of Streams. Please note that the provided Streams can be played only once! You will need to reset them (e.g. moving the file pointer to the beginning) and read them back if you want to process them a second time. The default timeout on the available() method is set to 0. This might be too small if you use e.g. a URLStream. If you set beginReset to true in the constructor, the system will automatically reset the streams when begin() is called. More...
class	ChannelAvg
	We average pairs of channels in a datastream. E.g. if we have 4 channels we end up with 2 channels. The channels will be (channel_1 + channel_2)/2 (channel_3 - channel_4)/2. This is equivalent of stereo to mono conversion but will also work for quadric, sexic or octic audio. This will not work if you provide single channel data! More...
class	ChannelAvgT
	We average pairs of channels in a datastream. E.g. if we have 4 channels we end up with 2 channels. The channels will be (channel_1 + channel_2)/2 (channel_3 - channel_4)/2. This is equivalent of stereo to mono conversion but will also work for quadric, sexic or octic audio. This will not work if you provide single channel data! More...
class	ChannelBinDiff
	Provides combination of binning and subtracting channels. More...
class	ChannelBinDiffT
	We first bin the channels then we calculate the difference between pairs of channels in a datastream. E.g. For binning, if we bin 4 samples in each channel we will have 4 times less samples per channel E.g. For subtracting if we have 4 channels we end up with 2 channels. The channels will be channel_1 - channel_2 channel_3 - channel_4 This is the same as combining binning and subtracting channels. This will not work if you provide single channel data! More...
class	ChannelConverter
	Increasing or decreasing the number of channels. More...
class	ChannelDiff
class	ChannelDiffT
	We calculate the difference between pairs of channels in a datastream. E.g. if we have 4 channels we end up with 2 channels. The channels will be channel_1 - channel_2 channel_3 - channel_4 This is similar to background subtraction between two channels but will also work for quadric, sexic or octic audio. This will not work if you provide single channel data! More...
class	ChannelEnhancer
	Increases the channel count. More...
class	ChannelMixer
	We mix all input channels in a datastream. E.g. if we have stereo input data we end up with 2 identical mono channels. More...
class	ChannelReducer
	We combine a datastream which consists of multiple channels into less channels. E.g. 2 to 1 The last target channel will contain the combined values of the exceeding source channels. More...
class	ChannelReducerT
	We combine a datastream which consists of multiple channels into less channels. E.g. 2 to 1 The last target channel will contain the combined values of the exceeding source channels. More...
class	ChannelSplitOutput
	Simple functionality to extract mono streams from a multichannel (e.g. stereo) signal. More...
class	ChannelsSelectOutput
	Flexible functionality to extract one or more channels from a multichannel signal. Warning: the destinatios added with addOutput are not automatically notified about audio changes. More...
class	CodecNOP
	Dummy no implmentation Codec. This is used so that we can initialize some pointers to decoders and encoders to make sure that they do not point to null. More...
class	ContainerDecoder
	Parent class for all container formats. More...
class	ContainerTarget
	ContainerTarget: forwards requests to both the output and the encoder/decoder and sets up the output chain for Containers. We also manage the proper sequence of the output classes. More...
class	ContainerTargetPrint
class	Converter1Channel
	Converter for 1 Channel which applies the indicated Filter. More...
class	ConverterAutoCenter
	Makes sure that the avg of the signal is set to 0. More...
class	ConverterAutoCenterT
	Makes sure that the avg of the signal is set to 0. More...
class	ConverterFillLeftAndRight
	Make sure that both channels contain any data. We copy the data from the non-empty channel to the empty one. More...
class	ConverterInvert
	Inverts the signal (multiplies every sample by -1). More...
class	ConverterNChannels
	Converter for n Channels which applies the indicated Filter. More...
class	ConverterScaler
	Multiplies the values with the indicated factor adds the offset and clips at maxValue. To mute use a factor of 0.0! More...
class	ConverterStream
	Both the data of the read or write operations will be converted with the help of the indicated converter. More...
class	ConverterSwitchLeftAndRight
	Switches the left and right channel. More...
class	ConverterToInternalDACFormat
	special case for internal DAC output for the ESP32. The incomming PCM buffer needs to be converted from signed 16bit to unsigned More...
class	CopyChannels
	Copy channel Cx value of type T shifted by S bits to all Cn channels. More...
class	CsvOutput
	Stream Wrapper which can be used to print the values as readable ASCII to the screen to be analyzed in the Serial Plotter The frames are separated by a new line. The channels in one frame are separated by a ,. More...
struct	DataNode
class	Decimate
	Provides a reduced sampling rate by taking a sample at every factor location (ingoring factor-1 samples). More...
class	DecimateT
	Provides reduced sampling rates. More...
class	DynamicMemoryStream
	MemoryStream which is written and read using the internal RAM. For each write the data is allocated on the heap. More...
class	EncodedAudioOutput
	A more natural Print class to process encoded data (aac, wav, mp3...). Just define the output and the decoder and write the encoded data. More...
class	EncodedAudioStream
	A more natural Stream class to process encoded data (aac, wav, mp3...) which also supports the decoding by calling readBytes(). More...
class	FastIntSineGenerator
	Sine wave generator that does not use any floating point operations in the readSample() hot path: it advances a 32 bit phase accumulator with an integer increment and looks up the result in a Q15 fixed point table. All floating point math (converting frequency/sample_rate into the phase increment) is done once in setFrequency()/setAudioInfo(), never per sample. This makes it a good fit for MCUs without a hardware FPU - e.g. the RP2040 - where sinf() is comparatively expensive. More...
class	FastSineGenerator
	Sine wave which is based on a fast approximation function using floating point math. More...
class	Filter
	Abstract filter interface definition. Subclasses implement process() to transform audio samples one at a time. Use reset() to clear internal state (e.g. delay lines) without changing the filter coefficients. More...
class	FilterChain
	A cascade of N arbitrary filters applied in series. Each sample is passed through all filters in order. The caller owns the filter pointers and must ensure they outlive the chain. More...
class	FilteredStream
	Stream to which we can apply Filters for each channel. The filter might change the result size! More...
struct	FilterSampleConverter
	Converts a raw sample (type T) to/from a filter's arithmetic type (FT) for ConverterNChannels. The default just casts the raw sample straight through – this matches the traditional convention used here, where FT=float and filter coefficients are calibrated against the sample's raw magnitude (e.g. up to +-32767 for 16-bit audio), so no rescaling is wanted. Specialized below for FT=q1_14_t, whose bounded ~[-2,2) range means a plain (q1_14_t)sample cast would misinterpret the raw PCM magnitude as a literal Q1.14 value and saturate almost immediately; q1_14_t::fromIntNN()/toIntNN() do the correct bit-shift scaling instead. More...
struct	FilterSampleConverter< int16_t, q1_14_t >
struct	FilterSampleConverter< int24_t, q1_14_t >
struct	FilterSampleConverter< int32_t, q1_14_t >
class	FIR
	Finite Impulse Response (FIR) filter. Performs convolution of the input signal with a set of feedforward coefficients b[]. For integer types, an optional scaling factor is applied to preserve precision. You can use https://www.arc.id.au/FilterDesign.html to design the coefficients. More...
class	FloatSineGenerator
	Generates a Sound with the help of sin() function. If performance is of concern, I suggest to use theSineFromTable, the FastSineGenerator or the FastIntSineGenerator. More...
class	FrameBuffer
	A FrameBuffer reads multiple values for array of 2 dimensional frames. More...
class	GeneratedSoundStream
	Source for reading generated tones. Please note. More...
class	GeneratorFixedValue
	Just returns a constant value. More...
class	GeneratorFromArray
	We generate the samples from an array which is provided in the constructor. More...
class	GeneratorFromStream
	An Adapter Class which lets you use any Stream as a Generator. More...
class	GeneratorMixer
	Generator which combines (mixes) multiple sound generators into one output. More...
class	GeneratorQueue
	Queues SoundGenerators and plays them sequentially. Queued SoundGenerators must be limited (eg: setPlayTime()) or no later Generator will ever be played. More...
class	HexDumpOutput
	Creates a Hex Dump. More...
class	HighPassFilter
	Second-order high-pass filter (BiQuad DF2). Attenuates frequencies below the cutoff frequency. Coefficients are derived from the Audio EQ Cookbook. More...
class	HighPassFilterDF1
	Second-order high-pass filter (BiQuad DF1). Same coefficients as HighPassFilter; use this (not HighPassFilter) when T is a bounded fixed-point type like q1_14_t – see LowPassFilterDF1 for why. More...
class	HighShelfFilter
	Second-order high-shelf filter (BiQuad DF2). Boosts or cuts frequencies above the shelf frequency by the specified gain (in dB) while leaving lower frequencies unchanged. Commonly used in tone controls and equalization. Coefficients are derived from the Audio EQ Cookbook. More...
class	HighShelfFilterDF1
	Second-order high-shelf filter (BiQuad DF1). Same coefficients as HighShelfFilter. NOTE: unlike the other DF1 filters here, this is NOT a working fix for q1_14_t (or similar bounded fixed-point types) – see the. More...
class	IIR
	Infinite Impulse Response (IIR) filter. Uses both feedforward (b[]) and feedback (a[]) coefficients. The a[0] coefficient is used to normalize all other coefficients. For integer types, an optional scaling factor is applied to preserve precision. More...
class	InputMerge
	Merges multiple input streams. 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. More...
class	InputMixer
	MixerStream is mixing the input from Multiple Input Streams. All streams must have the same audo format (sample rate, channels, bits per sample). More...
class	int24_3bytes_t
	24bit integer which is used for I2S sound processing. The values are really using 3 bytes. It works only on little endian machines! More...
class	int24_4bytes_t
	24bit 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. More...
class	List
	Double linked list. More...
class	LowPassFilter
	Second-order low-pass filter (BiQuad DF2). Attenuates frequencies above the cutoff frequency. Coefficients are derived from the Audio EQ Cookbook. More...
class	LowPassFilterDF1
	Second-order low-pass filter (BiQuad DF1). Same coefficients as LowPassFilter, but built on BiQuadDF1, whose separate x[]/y[] delay lines stay bounded to the actual signal range – unlike DF2's single delay line, which can need far more dynamic range than the signal for low cutoff/ sampleRate ratios. Use this (not LowPassFilter) when T is a bounded fixed-point type like q1_14_t. More...
class	LowShelfFilter
	Second-order low-shelf filter (BiQuad DF2). Boosts or cuts frequencies below the shelf frequency by the specified gain (in dB) while leaving higher frequencies unchanged. Commonly used in tone controls and equalization. Coefficients are derived from the Audio EQ Cookbook. More...
class	LowShelfFilterDF1
	Second-order low-shelf filter (BiQuad DF1). Same coefficients as LowShelfFilter. NOTE: switching to DF1 does NOT make this reliable for bounded fixed-point types like q1_14_t – shelf coefficients scale with the linear gain factor and can exceed the type's range even at modest gain settings; this is a coefficient headroom problem, not the DF2 state-headroom problem DF1 fixes elsewhere. Watch for the LOGE warning from checkCoeffRange() in begin(). More...
class	MemoryOutput
	Writes to a preallocated memory. More...
class	MemoryStream
	A simple Stream implementation which is backed by allocated memory. More...
class	MimeSource
	Abstract interface for classes that can provide MIME type information. More...
struct	MixGain
	Mixing of multiple audio input streams into a single output stream. More...
class	ModifyingOutput
	Abstract class: Objects can be put into a pipleline. More...
class	ModifyingStream
	Abstract class: Objects can be put into a pipleline. More...
class	MultiConverter
	Combines multiple converters. More...
class	MultiOutput
	Replicates the output to multiple destinations. More...
class	NBuffer
	A N buffer. If count=2 we create a DoubleBuffer, if count=3 a TripleBuffer etc. More...
class	NBufferExt
	A NBufferExt is a subclass of NBuffer which allows to use a direct access API to the BaseBuffer. More...
class	NBufferFile
	A File backed buffer which uses the provided files for buffering with the indicated max size. A file is made available for reading as soon as it reached the size limit. You must provide the files opened in "Write" mode with the addFile() method! More...
class	NoFilter
	Passes the input through unchanged. Useful as a placeholder when a Filter is required but no processing is desired. More...
class	NOPConverter
	Dummy converter which does nothing. More...
class	NotchFilter
	Second-order notch (band-reject) filter (BiQuad DF2). Rejects frequencies near the center frequency and passes those further away. Useful for removing a specific unwanted frequency (e.g. mains hum at 50/60 Hz). Coefficients are derived from the Audio EQ Cookbook. More...
class	NotchFilterDF1
	Second-order notch (band-reject) filter (BiQuad DF1). Same coefficients as NotchFilter; use this (not NotchFilter) when T is a bounded fixed-point type like q1_14_t – see LowPassFilterDF1 for why. More...
class	NullStream
	The Arduino Stream which provides silence and simulates a null device when used as audio target or audio source. More...
class	NumberConverter
	Converts from a source to a target number with a different type. More...
class	NumberReader
	Reads n numbers from an Arduino Stream. More...
class	OutputMixer
class	PinkNoiseGenerator
	Generates pink noise. More...
class	PoppingSoundRemover
	Big 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. More...
class	PriorityQueue
	Priority Queue which is based on a List. The order of the elements is defined by a compare function which is provided in the constructor. If the function returns > 0 if v1 > v2, the data will be provided in increasing order. More...
class	ProgressStream
	Generic calss to measure the the total bytes which were processed in order to calculate the progress as a percentage of the total size. More...
class	ProgressStreamInfo
	Configuration for ProgressStream. More...
class	q1_14_t
	Fixed-point Q1.14 number: a plain 16-bit signed integer holding 1 integer bit and 14 fractional bits (value = raw / 16384.0). Covers the range [-2.0, 1.99993896484375] with a resolution of ~6.1e-5, using only integer add/sub/mul/div – no hardware floating-point instructions. More...
class	Queue
	FIFO Queue which is based on a List. More...
class	QueueFromVector
	FIFO Queue which is based on a Vector. More...
class	QueueStream
	Stream class which stores the data in a temporary queue buffer. The queue can be consumed e.g. by a callback function by calling readBytes();. More...
class	ReformatBaseStream
	Base class for chained converting streams. More...
class	RingBuffer
	Implements a typed Ringbuffer. More...
class	RingBufferFile
	An File backed Ring Buffer that we can use to receive streaming audio. We expect an open file as parameter. More...
class	RingBufferSPSC
	Lock-free Single-Producer Single-Consumer ring buffer. More...
class	RingBufferStream
	An AudioStream backed by a Ringbuffer. We can write to the end and read from the beginning of the stream. More...
class	RTSPFormat
	Audio Format Definition - Base class for RTSP audio formats. More...
class	RTSPFormatAAC
	AAC format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats See RFC 3640 for details. More...
class	RTSPFormatAbtX
	abtX format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats More...
class	RTSPFormatADPCM
	RTSP/RTP formatter for mono IMA ADPCM (DVI4). More...
class	RTSPFormatG711
	G711 μ-Law format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats Packet intervall: 20, frame size: any. More...
class	RTSPFormatGSM
	GSM format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats. More...
class	RTSPFormatH264
	H.264 (AVC) format for RTSP video streaming, per RFC 6184. More...
class	RTSPFormatMJPEG
	MJPEG (Motion JPEG) format for RTSP video streaming. More...
class	RTSPFormatMP3
	MP3 format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats. More...
class	RTSPFormatMTS
	MPEG-2 Transport Stream (MP2T) format for RTSP streaming of a pre-muxed audio+video container, per RFC 2250. More...
class	RTSPFormatOpus
	Opus format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats. More...
class	RTSPFormatPCM
	Linear PCM Format for RTSP Streaming. More...
class	RTSPFormatPCM8
	L8 format for RTSP https://en.wikipedia.org/wiki/RTP_payload_formats. More...
class	RTSPPlatform
	Template-based platform abstraction for RTSP networking. More...
struct	RTSPVideoInfo
	Video Information Structure. More...
class	RTSPVideoInfoSupport
	Base class for video information support. More...
class	SawToothGenerator
	Generates a saw tooth wave sound. Uses a 32 bit phase accumulator with an integer increment, just like FastIntSineGenerator: no floating point operations or divisions in the readSample() hot path. Unlike a sine, a saw tooth needs no lookup table either - a wrapping unsigned accumulator reinterpreted as signed already is a linear ramp - so this is even cheaper than FastIntSineGenerator and a good fit for MCUs without a hardware FPU (e.g. RP2040), where the previous sinf()/float based implementation (inherited from SineGenerator) was comparatively expensive. More...
class	SilenceGenerator
	Provides a fixed value (e.g. 0) as sound data. This can be used e.g. to test the output functionality which should optimally just output silence and no artifacts. More...
class	SilenceRemovalConverter
	Removes any silence from the buffer that is longer then n samples with a amplitude below the indicated threshhold. If you process multiple channels you need to multiply the channels with the number of samples to indicate n. More...
class	SineFromTable
	A sine generator based on a table. The table is created using degrees where one full wave is 360 degrees. More...
class	SingleBuffer
	A simple Buffer implementation which just uses a (dynamically sized) array. More...
class	Slice
	Helps to split up a big memory array into smaller slices. There are no additinal heap allocations! Example: if we have an array with 9 entries (1,2,3,4,5,6,7,8,9): slices(5) gives 2. slice(5,0) returns size 5 with 1,2,3,4,5 and slice(5,1) returns size 4 with 6,7,8,9! More...
class	SmoothTransition
	Changes the samples at the beginning or at the end to slowly ramp up the volume. More...
class	soft_float_t
	Software "pseudo-float": a real number represented as a signed 16-bit mantissa plus a 16-bit exponent (value = mantissa * 2^exponent), computed with integer multiply/shift/compare only – no hardware floating-point instructions. This generalizes speexdsp's classic fixed-point trick of tracking a handful of wide-dynamic-range control variables as a normalized mantissa+exponent pair into a full, operator-overloaded numeric type usable as a drop-in replacement for float (see MDFFixedPoint in MDFEchoCancellation.h, which uses this class as the element type for the MDF echo canceller's sample/spectrum arrays and internal state, so the whole algorithm runs without native float arithmetic). More...
class	SOSFilter
	Second Order Sections (SOS) filter — a cascade of N BiQuad DF2 stages. Higher-order filters should be decomposed into second-order sections to avoid numerical instability. Each section has its own b[3]/a[3] coefficients and optional gain. Tools like scipy.signal.butter(..., output='sos') or MATLAB's tf2sos produce the required coefficient arrays. More...
class	SoundGenerator
	Base class to define the abstract interface for the sound generating classes. More...
class	SquareWaveGenerator
	Generates a square wave sound. Uses the same 32 bit phase accumulator as SawToothGenerator/FastIntSineGenerator: no floating point operations or sin() calls in the readSample() hot path. A square wave needs neither a sine approximation nor a lookup table - just the sign bit of the ramp - so readSample() is a single comparison, making this a good fit for MCUs without a hardware FPU (e.g. RP2040), where the previous FastSineGenerator based implementation still did a float polynomial evaluation per sample just to keep its sign. More...
class	Stack
	LIFO Stack which is based on a List. More...
class	Str
	Str which keeps the data on the heap. We grow the allocated memory only if the copy source is not fitting. More...
class	StrView
	A simple wrapper to provide string functions on existing allocated char. If the underlying char is a const we do not allow any updates; The ownership of the char* must be managed externally! More...
class	TestGenerator
	Generates a test signal which is easy to check because the values are incremented or decremented by 1. More...
class	Throttle
	Throttle the sending or receiving of the audio data to limit it to the indicated sample rate. More...
struct	ThrottleConfig
	Configure Throttle setting. More...
class	TimedStream
	AudioStream class that can define a start and (an optional) stop time Usually it is used to wrap an Audio Sink (e.g. I2SStream), but wrapping an Audio Source is supported as well. Only wrap classes which represent PCM data! More...
class	TimerCallback
	Internal class to manage the different timer callbacks for the 4 hardware timers. More...
class	TimerCallbackAudioStream
	Callback driven Audio Source (rx_tx_mode==RX_MODE) or Audio Sink (rx_tx_mode==TX_MODE). This class allows to to integrate external libraries in order to consume or generate a data stream which is based on a timer. More...
struct	TimerCallbackAudioStreamInfo
	TimerCallbackAudioStream Configuration. More...
class	TimeSource
	Interface for classes that can provide time information - two distinct notions of "now", not interchangeable: millis() is real wall time; playbackTime() is however far actual processing has gotten, which is not the same thing whenever the two can diverge (a consumer blocked/starved, a paused source, ...). A consumer that must stay tied to genuine progress (e.g. PacedVideoOutput scheduling video against actual decoded-audio playback, not merely time passing) should use playbackTime(), not millis(). More...
class	TransformationReader
	ConverterStream Helper class which implements the readBytes with the help of write. More...
class	UserCallback
	Internal class to manage User callbacks. An optinal parameter can be passed to the callback method We support 4 timers. More...
class	Vector
	Vector implementation which provides the most important methods as defined by std::vector. This class it is quite handy to have and most of the times quite better then dealing with raw c arrays. More...
class	VolumeMeter
	A simple class to determine the volume. You can use it as final output or as output or input in your audio chain. More...
class	VolumeSupport
	Supports the setting and getting of the volume. More...
class	WhiteNoiseGenerator
	Generates a random noise sound with the help of rand() function. More...

Typedefs
using	ActivityCallback = void (*)(bool isActive)
	Callback function type for activity state changes.
using	AudioTimerDriver = AudioTimerDriverAVR
	use AudioTimer!
using	CallbackConverter = CallbackConverterT<int16_t>
	CallbackConverter with int16_t as default type.
using	digital_pin_t = gpio_dt_spec
	Zephyr GPIO spec as digital_pin_t.
using	EncodedAudioPrint = EncodedAudioOutput
	Legacy alias for EncodedAudioOutput.
using	int24_t = int24_4bytes_t
typedef void(*	my_repeating_timer_callback_t) (void *obj)
using	Pins = Vector<int>
	Type alias for a collection of pin numbers.
using	PseudoFloat = soft_float_t
typedef void(*	repeating_timer_callback_t) (void *obj)
using	sample_rate_t = uint32_t
	Type alias for sample rate values.
template<class T = int16_t>
using	SineGenerator = FloatSineGenerator<T>
template<class T = int16_t>
using	SineWaveGenerator = SineGenerator<T>
	Alias for SineGenerator.
using	TAllocatorExt = AllocatorZephyr
using	TAllocatorSTD = AllocatorZephyr
using	TimerAlarmRepeating = AudioTimer
using	VolumeOutput = VolumeMeter
	Legacy alias for VolumeMeter.
using	VolumePrint = VolumeMeter
	Legacy alias for VolumeMeter.

Enumerations
enum class	AudioFormat : uint16_t { UNKNOWN = 0x0000 , PCM = 0x0001 , ADPCM = 0x0002 , IEEE_FLOAT = 0x0003 , ALAW = 0x0006 , MULAW = 0x0007 , OKI_ADPCM = 0x0010 , DVI_ADPCM = 0x0011 , IMA_ADPCM = DVI_ADPCM , MEDIASPACE_ADPCM = 0x0012 , SIERRA_ADPCM = 0x0013 , G723_ADPCM = 0x0014 , DIALOGIC_OKI_ADPCM = 0x0017 , MEDIAVISION_ADPCM = 0x0018 , YAMAHA_ADPCM = 0x0020 , ANTEX_ADPCME = 0x0033 , DIGIADPCM = 0x0036 , NMS_VBXADPCM = 0x0038 , CS_IMAADPCM = 0x0039 , ROCKWELL_ADPCM = 0x003B , G721_ADPCM = 0x0040 , G726_ADPCM = 0x0064 , G722_ADPCM = 0x0065 , INFOCOM_ITS_G721_ADPCM = 0x008B , AC3 = 0x0092 , ZYXEL_ADPCM = 0x0097 , RHETOREX_ADPCM = 0x0100 , SANYO_LD_ADPCM = 0x0125 , G726ADPCM = 0x0140 , UNISYS_NAP_ADPCM = 0x0170 , MP3 = 0x0055 , ALAC = 0x6C61 , AAC = 0xA106 , UNKNOWN = 0x0000 , PCM = 0x0001 , ADPCM = 0x0002 , IEEE_FLOAT = 0x0003 , ALAW = 0x0006 , MULAW = 0x0007 , OKI_ADPCM = 0x0010 , DVI_ADPCM = 0x0011 , IMA_ADPCM = DVI_ADPCM , MEDIASPACE_ADPCM = 0x0012 , SIERRA_ADPCM = 0x0013 , G723_ADPCM = 0x0014 , DIALOGIC_OKI_ADPCM = 0x0017 , MEDIAVISION_ADPCM = 0x0018 , YAMAHA_ADPCM = 0x0020 , ANTEX_ADPCME = 0x0033 , DIGIADPCM = 0x0036 , NMS_VBXADPCM = 0x0038 , CS_IMAADPCM = 0x0039 , ROCKWELL_ADPCM = 0x003B , G721_ADPCM = 0x0040 , G726_ADPCM = 0x0064 , G722_ADPCM = 0x0065 , INFOCOM_ITS_G721_ADPCM = 0x008B , AC3 = 0x0092 , ZYXEL_ADPCM = 0x0097 , RHETOREX_ADPCM = 0x0100 , SANYO_LD_ADPCM = 0x0125 , G726ADPCM = 0x0140 , UNISYS_NAP_ADPCM = 0x0170 , MP3 = 0x0055 , ALAC = 0x6C61 , AAC = 0xA106 }
	Audio format codes used by Microsoft e.g. in avi or wav files. More...
enum class	AudioFormat : uint16_t { UNKNOWN = 0x0000 , PCM = 0x0001 , ADPCM = 0x0002 , IEEE_FLOAT = 0x0003 , ALAW = 0x0006 , MULAW = 0x0007 , OKI_ADPCM = 0x0010 , DVI_ADPCM = 0x0011 , IMA_ADPCM = DVI_ADPCM , MEDIASPACE_ADPCM = 0x0012 , SIERRA_ADPCM = 0x0013 , G723_ADPCM = 0x0014 , DIALOGIC_OKI_ADPCM = 0x0017 , MEDIAVISION_ADPCM = 0x0018 , YAMAHA_ADPCM = 0x0020 , ANTEX_ADPCME = 0x0033 , DIGIADPCM = 0x0036 , NMS_VBXADPCM = 0x0038 , CS_IMAADPCM = 0x0039 , ROCKWELL_ADPCM = 0x003B , G721_ADPCM = 0x0040 , G726_ADPCM = 0x0064 , G722_ADPCM = 0x0065 , INFOCOM_ITS_G721_ADPCM = 0x008B , AC3 = 0x0092 , ZYXEL_ADPCM = 0x0097 , RHETOREX_ADPCM = 0x0100 , SANYO_LD_ADPCM = 0x0125 , G726ADPCM = 0x0140 , UNISYS_NAP_ADPCM = 0x0170 , MP3 = 0x0055 , ALAC = 0x6C61 , AAC = 0xA106 , UNKNOWN = 0x0000 , PCM = 0x0001 , ADPCM = 0x0002 , IEEE_FLOAT = 0x0003 , ALAW = 0x0006 , MULAW = 0x0007 , OKI_ADPCM = 0x0010 , DVI_ADPCM = 0x0011 , IMA_ADPCM = DVI_ADPCM , MEDIASPACE_ADPCM = 0x0012 , SIERRA_ADPCM = 0x0013 , G723_ADPCM = 0x0014 , DIALOGIC_OKI_ADPCM = 0x0017 , MEDIAVISION_ADPCM = 0x0018 , YAMAHA_ADPCM = 0x0020 , ANTEX_ADPCME = 0x0033 , DIGIADPCM = 0x0036 , NMS_VBXADPCM = 0x0038 , CS_IMAADPCM = 0x0039 , ROCKWELL_ADPCM = 0x003B , G721_ADPCM = 0x0040 , G726_ADPCM = 0x0064 , G722_ADPCM = 0x0065 , INFOCOM_ITS_G721_ADPCM = 0x008B , AC3 = 0x0092 , ZYXEL_ADPCM = 0x0097 , RHETOREX_ADPCM = 0x0100 , SANYO_LD_ADPCM = 0x0125 , G726ADPCM = 0x0140 , UNISYS_NAP_ADPCM = 0x0170 , MP3 = 0x0055 , ALAC = 0x6C61 , AAC = 0xA106 }
	Audio format codes used by Microsoft e.g. in avi or wav files. More...
enum	FillLeftAndRightStatus { Auto , LeftIsEmpty , RightIsEmpty , Auto , LeftIsEmpty , RightIsEmpty }
	Configure ConverterFillLeftAndRight. More...
enum	FillLeftAndRightStatus { Auto , LeftIsEmpty , RightIsEmpty , Auto , LeftIsEmpty , RightIsEmpty }
	Configure ConverterFillLeftAndRight. More...
enum	I2SFormat { I2S_STD_FORMAT , I2S_LSB_FORMAT , I2S_MSB_FORMAT , I2S_PHILIPS_FORMAT , I2S_RIGHT_JUSTIFIED_FORMAT , I2S_LEFT_JUSTIFIED_FORMAT , I2S_PCM , I2S_STD_FORMAT , I2S_LSB_FORMAT , I2S_MSB_FORMAT , I2S_PHILIPS_FORMAT , I2S_RIGHT_JUSTIFIED_FORMAT , I2S_LEFT_JUSTIFIED_FORMAT , I2S_PCM }
	I2S Formats. More...
enum	I2SFormat { I2S_STD_FORMAT , I2S_LSB_FORMAT , I2S_MSB_FORMAT , I2S_PHILIPS_FORMAT , I2S_RIGHT_JUSTIFIED_FORMAT , I2S_LEFT_JUSTIFIED_FORMAT , I2S_PCM , I2S_STD_FORMAT , I2S_LSB_FORMAT , I2S_MSB_FORMAT , I2S_PHILIPS_FORMAT , I2S_RIGHT_JUSTIFIED_FORMAT , I2S_LEFT_JUSTIFIED_FORMAT , I2S_PCM }
	I2S Formats. More...
enum	I2SSignalType { Digital , Analog , PDM , TDM , Digital , Analog , PDM , TDM }
	I2S Signal Types: Digital, Analog, PDM. More...
enum	I2SSignalType { Digital , Analog , PDM , TDM , Digital , Analog , PDM , TDM }
	I2S Signal Types: Digital, Analog, PDM. More...
enum	MediaType { MEDIA_AUDIO , MEDIA_VIDEO }
	Media type enumeration for RTSP streams. More...
enum	MemoryType { RAM , PS_RAM , FLASH_RAM , RAM , PS_RAM , FLASH_RAM }
	Memory types. More...
enum	MemoryType { RAM , PS_RAM , FLASH_RAM , RAM , PS_RAM , FLASH_RAM }
	Memory types. More...
enum class	RTSPVideoFormat { RGB24 , RGB565 , YUV420 , JPEG , MJPEG , GRAYSCALE , H264 }
	Video format enumeration for different pixel formats. More...
enum	RxTxMode { UNDEFINED_MODE = 0 , TX_MODE = 1 , RX_MODE = 2 , RXTX_MODE = 3 , UNDEFINED_MODE = 0 , TX_MODE = 1 , RX_MODE = 2 , RXTX_MODE = 3 }
	The Microcontroller is the Audio Source (TX_MODE) or Audio Sink (RX_MODE). RXTX_MODE is Source and Sink at the same time! More...
enum	RxTxMode { UNDEFINED_MODE = 0 , TX_MODE = 1 , RX_MODE = 2 , RXTX_MODE = 3 , UNDEFINED_MODE = 0 , TX_MODE = 1 , RX_MODE = 2 , RXTX_MODE = 3 }
	The Microcontroller is the Audio Source (TX_MODE) or Audio Sink (RX_MODE). RXTX_MODE is Source and Sink at the same time! More...
enum	TimerFunction { DirectTimerCallback , TimerCallbackInThread , SimpleThreadLoop , DirectTimerCallback , TimerCallbackInThread , SimpleThreadLoop }
enum	TimerFunction { DirectTimerCallback , TimerCallbackInThread , SimpleThreadLoop , DirectTimerCallback , TimerCallbackInThread , SimpleThreadLoop }
enum	TimeUnit { MS , US , HZ , MS , US , HZ }
	Time Units. More...
enum	TimeUnit { MS , US , HZ , MS , US , HZ }
	Time Units. More...

Functions
BiQuadCoeffs	calculateBandPassCoeffs (float frequency, float sampleRate, float q)
	Computes the b0/b1/b2/a1/a2 biquad coefficients for a band-pass filter.
BiQuadCoeffs	calculateHighPassCoeffs (float frequency, float sampleRate, float q)
	Computes the b0/b1/b2/a1/a2 biquad coefficients for a high-pass filter.
BiQuadCoeffs	calculateHighShelfCoeffs (float frequency, float sampleRate, float gain, float slope)
	Computes the b0/b1/b2/a1/a2 biquad coefficients for a high-shelf filter.
BiQuadCoeffs	calculateLowPassCoeffs (float frequency, float sampleRate, float q)
	Computes the b0/b1/b2/a1/a2 biquad coefficients for a low-pass filter.
BiQuadCoeffs	calculateLowShelfCoeffs (float frequency, float sampleRate, float gain, float slope)
	Computes the b0/b1/b2/a1/a2 biquad coefficients for a low-shelf filter.
BiQuadCoeffs	calculateNotchCoeffs (float frequency, float sampleRate, float q)
template<typename T>
void	checkCoeffRange (const BiQuadCoeffs &c)
	Warns (once per call, via LOGE) about any coefficient that doesn't survive being stored in T. This is expected/harmless for float or double (T can represent the value, roundtrip is exact); it's a real problem for a bounded fixed-point type like q1_14_t (range ~[-2,2)), where a coefficient outside that range silently saturates instead of erroring, quietly producing a wrong filter – e.g. shelf filters, whose coefficients scale with the linear gain factor and can exceed +-2 even at modest gains (a limitation of q1_14_t's 1-bit-of-headroom range, not of DF1 vs DF2; see LowShelfFilterDF1/HighShelfFilterDF1).
static void	checkMemory (bool printMemory=false)
	Executes heap_caps_check_integrity_all().
void	delay (uint32_t ms)
void	delayMicroseconds (uint32_t us)
int	digitalRead (digital_pin_t pin)
void	digitalWrite (digital_pin_t pin, bool value)
AudioFormat	fromMime (const char *mime)
	Best-effort inverse of toMime(): maps a mime type (e.g. an AudioEncoder's mime()) back to the AudioFormat wav code it came from - AudioFormat::UNKNOWN if mime is null or doesn't match any of toMime()'s mappings (most codecs - Opus, FLAC, GSM, ... - have no wav code at all, so this can never be exhaustive; "audio/wav" itself maps back to PCM specifically, even though toMime() also uses it for every ADPCM variant, since PCM is the common case).
bool	isWavFormat (AudioFormat format)
	True if the wav code is handled via the WAV decoder (i.e. toMime() maps it to "audio/wav": PCM and all ADPCM variants).
template<typename T>
T	mapT (T x, T in_min, T in_max, T out_min, T out_max)
	Similar to Arduino map function but using floats.
uint64_t	micros ()
uint32_t	millis ()
	Returns the milliseconds since the start.
static bool	operator!= (audio_tools::digital_pin_t &a, audio_tools::digital_pin_t &b)
	Support for pin compare.
bool	operator!= (double a, soft_float_t b)
bool	operator!= (float a, q1_14_t b)
bool	operator!= (float a, soft_float_t b)
bool	operator!= (int a, soft_float_t b)
soft_float_t	operator* (double a, soft_float_t b)
q1_14_t	operator* (float a, q1_14_t b)
soft_float_t	operator* (float a, soft_float_t b)
soft_float_t	operator* (int a, soft_float_t b)
soft_float_t	operator+ (double a, soft_float_t b)
q1_14_t	operator+ (float a, q1_14_t b)
soft_float_t	operator+ (float a, soft_float_t b)
soft_float_t	operator+ (int a, soft_float_t b)
soft_float_t	operator- (double a, soft_float_t b)
q1_14_t	operator- (float a, q1_14_t b)
soft_float_t	operator- (float a, soft_float_t b)
soft_float_t	operator- (int a, soft_float_t b)
soft_float_t	operator/ (double a, soft_float_t b)
q1_14_t	operator/ (float a, q1_14_t b)
soft_float_t	operator/ (float a, soft_float_t b)
soft_float_t	operator/ (int a, soft_float_t b)
bool	operator< (double a, soft_float_t b)
bool	operator< (float a, q1_14_t b)
bool	operator< (float a, soft_float_t b)
bool	operator< (int a, soft_float_t b)
bool	operator<= (double a, soft_float_t b)
bool	operator<= (float a, q1_14_t b)
bool	operator<= (float a, soft_float_t b)
bool	operator<= (int a, soft_float_t b)
static bool	operator== (audio_tools::digital_pin_t &a, audio_tools::digital_pin_t &b)
	Support for pin compare.
bool	operator== (double a, soft_float_t b)
bool	operator== (float a, q1_14_t b)
bool	operator== (float a, soft_float_t b)
bool	operator== (int a, soft_float_t b)
bool	operator> (double a, soft_float_t b)
bool	operator> (float a, q1_14_t b)
bool	operator> (float a, soft_float_t b)
bool	operator> (int a, soft_float_t b)
bool	operator>= (double a, soft_float_t b)
bool	operator>= (float a, q1_14_t b)
bool	operator>= (float a, soft_float_t b)
bool	operator>= (int a, soft_float_t b)
void	pinMode (digital_pin_t pin, int mode)
void	printNChar (char ch, int n)
	prints n times the character ch and a new line
void	printVersionInfo ()
	prints the available version information
template<typename T>
T	readSample (Stream *p_stream)
	guaranteed to return the requested data
template<typename T>
size_t	readSamples (Stream p_stream, T data, int samples, int retryCount=-1)
	guaranteed to return the requested data
void	stop ()
	stops any further processing by spinning in an endless loop
static void	timerCallback (void *obj)
static void	timerCallback0 ()
static void	timerCallback1 ()
static void	timerCallback2 ()
static void	timerCallback3 ()
const char *	toMime (AudioFormat format)
	Provides the mime type for a AudioFormat wav code, or nullptr if not known/mapped.
static void	userCallback0 ()
static void	userCallback1 ()
static void	userCallback2 ()
static void	userCallback3 ()
void	waitFor (bool &flag)
	wait for flag to be active
template<typename T>
size_t	writeData (Print p_out, T data, int samples, int maxSamples=512)
template<typename T, class P>
size_t	writeDataT (P p_out, T data, int samples, int maxSamples=512)
	guaranteed to return the requested data
void	yield ()

Variables
static TAllocatorExt	DefaultAllocator
static TAllocatorSTD	DefaultAllocatorRAM
static AllocatorESP32	DefaultESP32AllocatorRAM
template<class T>
constexpr int16_t	FastIntSineGenerator< T >::sine_table []
static gpio_dt_spec	GPIO_NONE = {nullptr, 0, 0}
	GPIO_NONE is no pin defined.
static const char *	i2s_formats []
static const char *	i2s_signal_types [] = {"Digital", "Analog", "PDM", "TDM"}
static const char *	mime_pcm = "audio/pcm"
	Mime type for PCM.
static const char *	RxTxModeNames [4]
	Text string (description) for RxTxMode.
class audio_tools::audio_tools::UserCallback *	simpleUserCallback = nullptr
static AudioTimerDriverMBED *	timerAlarmRepeating = nullptr
static AudioTimerDriverSTM32 *	timerAlarmRepeating = nullptr
static AudioTimerDriverAVR *	timerAlarmRepeatingRef = nullptr
class audio_tools::audio_tools::TimerCallback *	timerCallbackArray = nullptr
static const char *	TimeUnitStr [3] {"MS", "US", "HZ"}

Detailed Description

AutioTools Configuration.

Author: Phil Schatzmann

Copyright: GPLv3

Author: Phil Schatzmann

AutioTools Configuration

Copyright: GPLv3

Author: Phil Schatzmann

AutioTools Configuration

Copyright: GPLv3

Author: Phil Schatzmann

AutioTools Configuration

Copyright: GPLv3

Author: Phil Schatzmann

Copyright: GPLv3

Author: Phil Schatzmann

Copyright: GPLv3

Author: Phil Schatzmann

AutioTools Configuration

Copyright: GPLv3

Typedef Documentation

◆ ActivityCallback

typedef void(* ActivityCallback)(bool isActive) = void (*)(bool isActive)

Callback function type for activity state changes.

◆ AudioTimerDriver

typedef AudioTimerDriverZephyr AudioTimerDriver = AudioTimerDriverAVR

use AudioTimer!

Default timer implementation for Zephyr.

◆ CallbackConverter

typedef CallbackConverterT< int16_t > CallbackConverter = CallbackConverterT<int16_t>

CallbackConverter with int16_t as default type.

◆ digital_pin_t

typedef int digital_pin_t = gpio_dt_spec

Zephyr GPIO spec as digital_pin_t.

◆ int24_t

using int24_t = int24_4bytes_t

◆ my_repeating_timer_callback_t

typedef void(* my_repeating_timer_callback_t)(void *obj)

◆ PseudoFloat

using PseudoFloat = soft_float_t

Deprecated: use soft_float_t

Deprecated: use soft_float_t

◆ repeating_timer_callback_t

typedef void(* repeating_timer_callback_t)(void *obj)

◆ SineGenerator

template<class T = int16_t>

using SineGenerator = FloatSineGenerator<T>

Default sine generator: FastIntSineGenerator (no FPU needed) on platforms that set PREFER_FIXEDPOINT, FloatSineGenerator (sinf() based) otherwise.

◆ SineWaveGenerator

template<class T = int16_t>

using SineWaveGenerator = SineGenerator<T>

Alias for SineGenerator.

◆ TAllocatorExt

using TAllocatorExt = AllocatorZephyr

◆ TAllocatorSTD

using TAllocatorSTD = AllocatorZephyr

◆ TimerAlarmRepeating

using TimerAlarmRepeating = AudioTimer

Enumeration Type Documentation

◆ I2SFormat [1/2]

enum I2SFormat

I2S Formats.

Enumerator
I2S_STD_FORMAT
I2S_LSB_FORMAT
I2S_MSB_FORMAT
I2S_PHILIPS_FORMAT
I2S_RIGHT_JUSTIFIED_FORMAT
I2S_LEFT_JUSTIFIED_FORMAT
I2S_PCM
I2S_STD_FORMAT
I2S_LSB_FORMAT
I2S_MSB_FORMAT
I2S_PHILIPS_FORMAT
I2S_RIGHT_JUSTIFIED_FORMAT
I2S_LEFT_JUSTIFIED_FORMAT
I2S_PCM

◆ I2SFormat [2/2]

enum I2SFormat

I2S Formats.

Enumerator
I2S_STD_FORMAT
I2S_LSB_FORMAT
I2S_MSB_FORMAT
I2S_PHILIPS_FORMAT
I2S_RIGHT_JUSTIFIED_FORMAT
I2S_LEFT_JUSTIFIED_FORMAT
I2S_PCM
I2S_STD_FORMAT
I2S_LSB_FORMAT
I2S_MSB_FORMAT
I2S_PHILIPS_FORMAT
I2S_RIGHT_JUSTIFIED_FORMAT
I2S_LEFT_JUSTIFIED_FORMAT
I2S_PCM

◆ I2SSignalType [1/2]

enum I2SSignalType

I2S Signal Types: Digital, Analog, PDM.

Enumerator
Digital
Analog
PDM
TDM
Digital
Analog
PDM
TDM

◆ I2SSignalType [2/2]

enum I2SSignalType

I2S Signal Types: Digital, Analog, PDM.

Enumerator
Digital
Analog
PDM
TDM
Digital
Analog
PDM
TDM

◆ MediaType

enum MediaType

Media type enumeration for RTSP streams.

Enumerator
MEDIA_AUDIO	Audio stream.
MEDIA_VIDEO	Video stream.

◆ RTSPVideoFormat

enum class RTSPVideoFormat

strong

Video format enumeration for different pixel formats.

Note: Scoped (and named RTSPVideoFormat, not VideoFormat) to avoid colliding with the unrelated audio_tools::VideoFormat enum in AudioTools/Video/VideoOutput.h (used by the container muxers/demuxers) - both live in the audio_tools namespace, and a translation unit that needs both RTSP.h and a container header (e.g. to mux audio+video into MPEG-TS before streaming it over RTSP) would otherwise hit a redefinition error.

Enumerator
RGB24	24-bit RGB (8 bits per channel)
RGB565	16-bit RGB (5-6-5 bits per channel)
YUV420	YUV 4:2:0 planar format.
JPEG	JPEG compressed format.
MJPEG	Motion JPEG format.
GRAYSCALE	8-bit grayscale
H264	H.264/AVC compressed format.

◆ TimerFunction [1/2]

enum TimerFunction

Enumerator
DirectTimerCallback
TimerCallbackInThread
SimpleThreadLoop
DirectTimerCallback
TimerCallbackInThread
SimpleThreadLoop

◆ TimerFunction [2/2]

enum TimerFunction

Enumerator
DirectTimerCallback
TimerCallbackInThread
SimpleThreadLoop
DirectTimerCallback
TimerCallbackInThread
SimpleThreadLoop

Function Documentation

◆ calculateBandPassCoeffs()

BiQuadCoeffs calculateBandPassCoeffs	(	float	frequency,
		float	sampleRate,
		float	q )

inline

Computes the b0/b1/b2/a1/a2 biquad coefficients for a band-pass filter.

◆ calculateHighPassCoeffs()

BiQuadCoeffs calculateHighPassCoeffs	(	float	frequency,
		float	sampleRate,
		float	q )

inline

Computes the b0/b1/b2/a1/a2 biquad coefficients for a high-pass filter.

◆ calculateHighShelfCoeffs()

BiQuadCoeffs calculateHighShelfCoeffs	(	float	frequency,
		float	sampleRate,
		float	gain,
		float	slope )

inline

Computes the b0/b1/b2/a1/a2 biquad coefficients for a high-shelf filter.

◆ calculateLowPassCoeffs()

BiQuadCoeffs calculateLowPassCoeffs	(	float	frequency,
		float	sampleRate,
		float	q )

inline

Computes the b0/b1/b2/a1/a2 biquad coefficients for a low-pass filter.

◆ calculateLowShelfCoeffs()

BiQuadCoeffs calculateLowShelfCoeffs	(	float	frequency,
		float	sampleRate,
		float	gain,
		float	slope )

inline

Computes the b0/b1/b2/a1/a2 biquad coefficients for a low-shelf filter.

◆ calculateNotchCoeffs()

BiQuadCoeffs calculateNotchCoeffs	(	float	frequency,
		float	sampleRate,
		float	q )

inline

Computes the b0/b1/b2/a1/a2 biquad coefficients for a notch (band-reject) filter.

◆ checkCoeffRange()

template<typename T>

void checkCoeffRange ( const BiQuadCoeffs & c )

Warns (once per call, via LOGE) about any coefficient that doesn't survive being stored in T. This is expected/harmless for float or double (T can represent the value, roundtrip is exact); it's a real problem for a bounded fixed-point type like q1_14_t (range ~[-2,2)), where a coefficient outside that range silently saturates instead of erroring, quietly producing a wrong filter – e.g. shelf filters, whose coefficients scale with the linear gain factor and can exceed +-2 even at modest gains (a limitation of q1_14_t's 1-bit-of-headroom range, not of DF1 vs DF2; see LowShelfFilterDF1/HighShelfFilterDF1).

◆ delay()

void delay ( uint32_t ms )

inline

◆ delayMicroseconds()

void delayMicroseconds ( uint32_t us )

inline

◆ digitalRead()

int digitalRead ( digital_pin_t pin )

◆ digitalWrite()

void digitalWrite	(	digital_pin_t	pin,
		bool	value )

◆ mapT()

template<typename T>

T mapT	(	T	x,
		T	in_min,
		T	in_max,
		T	out_min,
		T	out_max )

inline

Similar to Arduino map function but using floats.

Parameters

x
in_min
in_max
out_min
out_max

Returns

◆ micros()

uint64_t micros ( void )

inline

◆ millis()

uint32_t millis ( )

inline

Returns the milliseconds since the start.

◆ operator!=() [1/5]

bool operator!=	(	audio_tools::digital_pin_t &	a,
		audio_tools::digital_pin_t &	b )

inlinestatic

Support for pin compare.

◆ operator!=() [2/5]

bool operator!=	(	double	a,
		soft_float_t	b )

inline

◆ operator!=() [3/5]

bool operator!=	(	float	a,
		q1_14_t	b )

inline

◆ operator!=() [4/5]

bool operator!=	(	float	a,
		soft_float_t	b )

inline

◆ operator!=() [5/5]

bool operator!=	(	int	a,
		soft_float_t	b )

inline

◆ operator*() [1/4]

soft_float_t operator*	(	double	a,
		soft_float_t	b )

inline

◆ operator*() [2/4]

q1_14_t operator*	(	float	a,
		q1_14_t	b )

inline

◆ operator*() [3/4]

soft_float_t operator*	(	float	a,
		soft_float_t	b )

inline

◆ operator*() [4/4]

soft_float_t operator*	(	int	a,
		soft_float_t	b )

inline

◆ operator+() [1/4]

soft_float_t operator+	(	double	a,
		soft_float_t	b )

inline

◆ operator+() [2/4]

q1_14_t operator+	(	float	a,
		q1_14_t	b )

inline

◆ operator+() [3/4]

soft_float_t operator+	(	float	a,
		soft_float_t	b )

inline

◆ operator+() [4/4]

soft_float_t operator+	(	int	a,
		soft_float_t	b )

inline

◆ operator-() [1/4]

soft_float_t operator-	(	double	a,
		soft_float_t	b )

inline

◆ operator-() [2/4]

q1_14_t operator-	(	float	a,
		q1_14_t	b )

inline

◆ operator-() [3/4]

soft_float_t operator-	(	float	a,
		soft_float_t	b )

inline

◆ operator-() [4/4]

soft_float_t operator-	(	int	a,
		soft_float_t	b )

inline

◆ operator/() [1/4]

soft_float_t operator/	(	double	a,
		soft_float_t	b )

inline

◆ operator/() [2/4]

q1_14_t operator/	(	float	a,
		q1_14_t	b )

inline

◆ operator/() [3/4]

soft_float_t operator/	(	float	a,
		soft_float_t	b )

inline

◆ operator/() [4/4]

soft_float_t operator/	(	int	a,
		soft_float_t	b )

inline

◆ operator<() [1/4]

bool operator<	(	double	a,
		soft_float_t	b )

inline

◆ operator<() [2/4]

bool operator<	(	float	a,
		q1_14_t	b )

inline

◆ operator<() [3/4]

bool operator<	(	float	a,
		soft_float_t	b )

inline

◆ operator<() [4/4]

bool operator<	(	int	a,
		soft_float_t	b )

inline

◆ operator<=() [1/4]

bool operator<=	(	double	a,
		soft_float_t	b )

inline

◆ operator<=() [2/4]

bool operator<=	(	float	a,
		q1_14_t	b )

inline

◆ operator<=() [3/4]

bool operator<=	(	float	a,
		soft_float_t	b )

inline

◆ operator<=() [4/4]

bool operator<=	(	int	a,
		soft_float_t	b )

inline

◆ operator==() [1/5]

bool operator==	(	audio_tools::digital_pin_t &	a,
		audio_tools::digital_pin_t &	b )

inlinestatic

Support for pin compare.

◆ operator==() [2/5]

bool operator==	(	double	a,
		soft_float_t	b )

inline

◆ operator==() [3/5]

bool operator==	(	float	a,
		q1_14_t	b )

inline

◆ operator==() [4/5]

bool operator==	(	float	a,
		soft_float_t	b )

inline

◆ operator==() [5/5]

bool operator==	(	int	a,
		soft_float_t	b )

inline

◆ operator>() [1/4]

bool operator>	(	double	a,
		soft_float_t	b )

inline

◆ operator>() [2/4]

bool operator>	(	float	a,
		q1_14_t	b )

inline

◆ operator>() [3/4]

bool operator>	(	float	a,
		soft_float_t	b )

inline

◆ operator>() [4/4]

bool operator>	(	int	a,
		soft_float_t	b )

inline

◆ operator>=() [1/4]

bool operator>=	(	double	a,
		soft_float_t	b )

inline

◆ operator>=() [2/4]

bool operator>=	(	float	a,
		q1_14_t	b )

inline

◆ operator>=() [3/4]

bool operator>=	(	float	a,
		soft_float_t	b )

inline

◆ operator>=() [4/4]

bool operator>=	(	int	a,
		soft_float_t	b )

inline

◆ pinMode()

void pinMode	(	digital_pin_t	pin,
		int	mode )

◆ printVersionInfo()

void printVersionInfo ( )

inline

prints the available version information

◆ readSample()

template<typename T>

T readSample ( Stream * p_stream )

guaranteed to return the requested data

◆ readSamples()

template<typename T>

size_t readSamples	(	Stream *	p_stream,
		T *	data,
		int	samples,
		int	retryCount = -1 )

guaranteed to return the requested data

◆ timerCallback()

void timerCallback ( void * obj )

static

◆ timerCallback0()

void timerCallback0 ( )

static

◆ timerCallback1()

void timerCallback1 ( )

static

◆ timerCallback2()

void timerCallback2 ( )

static

◆ timerCallback3()

void timerCallback3 ( )

static

◆ userCallback0()

void userCallback0 ( )

static

◆ userCallback1()

void userCallback1 ( )

static

◆ userCallback2()

void userCallback2 ( )

static

◆ userCallback3()

void userCallback3 ( )

static

◆ writeData()

template<typename T>

size_t writeData	(	Print *	p_out,
		T *	data,
		int	samples,
		int	maxSamples = 512 )

◆ writeDataT()

template<typename T, class P>

size_t writeDataT	(	P *	p_out,
		T *	data,
		int	samples,
		int	maxSamples = 512 )

guaranteed to return the requested data

◆ yield()

void yield ( )

inline

Variable Documentation

◆ DefaultAllocator

TAllocatorExt DefaultAllocator

static

◆ DefaultAllocatorRAM

TAllocatorSTD DefaultAllocatorRAM

static

◆ DefaultESP32AllocatorRAM

AllocatorESP32 DefaultESP32AllocatorRAM

static

◆ FastIntSineGenerator< T >::sine_table

template<class T>

int16_t FastIntSineGenerator< T >::sine_table[]

constexpr

◆ GPIO_NONE

gpio_dt_spec GPIO_NONE = {nullptr, 0, 0}

static

GPIO_NONE is no pin defined.

◆ i2s_formats

const char * i2s_formats

static

Initial value:

= {"I2S_STD_FORMAT",
                                    "I2S_LSB_FORMAT",
                                    "I2S_MSB_FORMAT",
                                    "I2S_PHILIPS_FORMAT",
                                    "I2S_RIGHT_JUSTIFIED_FORMAT",
                                    "I2S_LEFT_JUSTIFIED_FORMAT",
                                    "I2S_PCM"}

◆ i2s_signal_types

const char * i2s_signal_types = {"Digital", "Analog", "PDM", "TDM"}

static

◆ mime_pcm

const char * mime_pcm = "audio/pcm"

static

Mime type for PCM.

◆ RxTxModeNames

const char * RxTxModeNames

static

Initial value:

= {"UNDEFINED_MODE", "TX_MODE", "RX_MODE",

"RXTX_MODE"}

Text string (description) for RxTxMode.

◆ simpleUserCallback

class audio_tools::audio_tools::UserCallback * simpleUserCallback = nullptr

◆ timerAlarmRepeating [1/2]

AudioTimerDriverMBED * timerAlarmRepeating = nullptr

static

◆ timerAlarmRepeating [2/2]

AudioTimerDriverSTM32 * timerAlarmRepeating = nullptr

static

◆ timerAlarmRepeatingRef

AudioTimerDriverAVR * timerAlarmRepeatingRef = nullptr

static

◆ timerCallbackArray

class audio_tools::audio_tools::TimerCallback * timerCallbackArray = nullptr

◆ TimeUnitStr

const char * TimeUnitStr {"MS", "US", "HZ"}

static

Classes

Typedefs

Enumerations

Functions

Variables

Detailed Description

Typedef Documentation

◆ ActivityCallback

◆ AudioTimerDriver

◆ CallbackConverter

◆ digital_pin_t

◆ int24_t

◆ my_repeating_timer_callback_t

◆ PseudoFloat

◆ repeating_timer_callback_t

◆ SineGenerator

◆ SineWaveGenerator

◆ TAllocatorExt

◆ TAllocatorSTD

◆ TimerAlarmRepeating

Enumeration Type Documentation

◆ I2SFormat [1/2]

◆ I2SFormat [2/2]

◆ I2SSignalType [1/2]

◆ I2SSignalType [2/2]

◆ MediaType

◆ RTSPVideoFormat

◆ TimerFunction [1/2]

◆ TimerFunction [2/2]

Function Documentation

◆ calculateBandPassCoeffs()

◆ calculateHighPassCoeffs()

◆ calculateHighShelfCoeffs()

◆ calculateLowPassCoeffs()

◆ calculateLowShelfCoeffs()

◆ calculateNotchCoeffs()

◆ checkCoeffRange()

◆ delay()

◆ delayMicroseconds()

◆ digitalRead()

◆ digitalWrite()

◆ mapT()

◆ micros()

◆ millis()

◆ operator!=() [1/5]

◆ operator!=() [2/5]

◆ operator!=() [3/5]

◆ operator!=() [4/5]

◆ operator!=() [5/5]

◆ operator*() [1/4]

◆ operator*() [2/4]

◆ operator*() [3/4]

◆ operator*() [4/4]

◆ operator+() [1/4]

◆ operator+() [2/4]

◆ operator+() [3/4]

◆ operator+() [4/4]

◆ operator-() [1/4]

◆ operator-() [2/4]

◆ operator-() [3/4]

◆ operator-() [4/4]

◆ operator/() [1/4]

◆ operator/() [2/4]

◆ operator/() [3/4]

◆ operator/() [4/4]

◆ operator<() [1/4]

◆ operator<() [2/4]

◆ operator<() [3/4]

◆ operator<() [4/4]

◆ operator<=() [1/4]

◆ operator<=() [2/4]

◆ operator<=() [3/4]

◆ operator<=() [4/4]

◆ operator==() [1/5]

◆ operator==() [2/5]

◆ operator==() [3/5]

◆ operator==() [4/5]

◆ operator==() [5/5]

◆ operator>() [1/4]

◆ operator>() [2/4]