CodecTSDemux.h File Reference

#include "AudioTools/AudioCodecs/AudioCodecsBase.h"
#include "AudioTools/AudioCodecs/AudioEncoded.h"
#include "stdlib.h"
#include "tsdemux.h"

Go to the source code of this file.

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	AllocSize
struct	AppropriateSumType< T >
	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< T >
	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< T >
	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< T >
	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...
struct	Bin::BinState
struct	Bin::BinStateT< T >
class	BinT< T >
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< T >
	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< T >
	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< T >
	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< T >
	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< T, AvgT >
	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...
struct	ChannelBinDiff::ChannelBinDiffState
struct	ChannelBinDiff::ChannelBinDiffStateT< T >
class	ChannelBinDiffT< T >
	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< T >
	Increasing or decreasing the number of channels. More...
class	ChannelDiff
class	ChannelDiffT< T >
	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< T >
	Increases the channel count. More...
class	ChannelMixer< T, SumT >
	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< T >
	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...
struct	ChannelSplitOutput::ChannelSelectionOutputDef
struct	ChannelsSelectOutput::ChannelSelectionOutputDef
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	Client
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< T >
	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< T >
	Makes sure that the avg of the signal is set to 0. More...
class	ConverterFillLeftAndRight< T >
	Make sure that both channels contain any data. We copy the data from the non-empty channel to the empty one. More...
class	ConverterInvert< T >
	Inverts the signal (multiplies every sample by -1). More...
class	ConverterNChannels< T, FT >
	Converter for n Channels which applies the indicated Filter. More...
class	ConverterScaler< T >
	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< T >
	Both the data of the read or write operations will be converted with the help of the indicated converter. More...
class	ConverterSwitchLeftAndRight< T >
	Switches the left and right channel. More...
class	ConverterToInternalDACFormat< T >
	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< T, Cn, Cx, S >
	Copy channel Cx value of type T shifted by S bits to all Cn channels. More...
class	CsvOutput< T >
	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...
struct	Decimate::DecimateState
struct	Decimate::DecimateStateT< T >
class	DecimateT< T >
	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< T >
	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< T >
	Sine wave which is based on a fast approximation function using floating point math. More...
class	Filter< T >
	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< T, N >
	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< T, TF >
	Stream to which we can apply Filters for each channel. The filter might change the result size! More...
struct	FilterSampleConverter< T, FT >
	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< T >
	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< T >
	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< T >
	A FrameBuffer reads multiple values for array of 2 dimensional frames. More...
class	GeneratedSoundStream< T >
	Source for reading generated tones. Please note. More...
class	GeneratorFixedValue< T >
	Just returns a constant value. More...
class	GeneratorFromArray< T >
	We generate the samples from an array which is provided in the constructor. More...
class	GeneratorFromStream< T >
	An Adapter Class which lets you use any Stream as a Generator. More...
class	GeneratorMixer< T >
	Generator which combines (mixes) multiple sound generators into one output. More...
class	GeneratorQueue< T >
	Queues SoundGenerators and plays them sequentially. Queued SoundGenerators must be limited (eg: setPlayTime()) or no later Generator will ever be played. More...
class	HardwareSerial
class	HexDumpOutput
	Creates a Hex Dump. More...
class	HighPassFilter< T >
	Second-order high-pass filter (BiQuad DF2). Attenuates frequencies below the cutoff frequency. Coefficients are derived from the Audio EQ Cookbook. More...
class	HighPassFilterDF1< T >
	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< T >
	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< T >
	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< T >
	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< T >
	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< T, SumT >
	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< T >::Iterator
class	Vector< T >::iterator
	Iterator for the Vector class. More...
class	List< T >
	Double linked list. More...
class	LowPassFilter< T >
	Second-order low-pass filter (BiQuad DF2). Attenuates frequencies above the cutoff frequency. Coefficients are derived from the Audio EQ Cookbook. More...
class	LowPassFilterDF1< T >
	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< T >
	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< T >
	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...
struct	InputMerge< T >::MergeRecord
class	MimeSource
	Abstract interface for classes that can provide MIME type information. More...
struct	MixGain< T >
	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	MTSDecoderTSDemux
	MPEG-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/codec-tsdemux. More...
class	MultiConverter< T >
	Combines multiple converters. More...
class	MultiOutput
	Replicates the output to multiple destinations. More...
struct	MultiOutput::MultiOutputRecord
	Record describing a single replicated output. More...
class	NBuffer< T >
	A N buffer. If count=2 we create a DoubleBuffer, if count=3 a TripleBuffer etc. More...
class	NBufferExt< T >
	A NBufferExt is a subclass of NBuffer which allows to use a direct access API to the BaseBuffer. More...
class	NBufferFile< File, T >
	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...
struct	List< T >::Node
class	NoFilter< T >
	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< T >
	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< T >
	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	numeric_limits< audio_tools::int24_3bytes_t >
class	numeric_limits< audio_tools::int24_4bytes_t >
struct	RingBufferFile< File, T >::OffsetInfo
class	OutputMixer< T >
class	PinkNoiseGenerator< T >
	Generates pink noise. More...
class	PoppingSoundRemover< T >
	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	Print
class	PriorityQueue< T >
	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...
struct	GeneratorQueue< T >::QNode
class	Queue< T >
	FIFO Queue which is based on a List. More...
class	QueueFromVector< T >
	FIFO Queue which is based on a Vector. More...
class	QueueStream< T >
	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< T >
	Implements a typed Ringbuffer. More...
class	RingBufferFile< File, T >
	An File backed Ring Buffer that we can use to receive streaming audio. We expect an open file as parameter. More...
class	RingBufferSPSC< T >
	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	SawToothGenerator< T >
	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< T >
	Provides a fixed value (e.g. 0) as sound data. This can be used e.g. to test the output functionality which should optimally just output silence and no artifacts. More...
class	SilenceRemovalConverter< T >
	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< T >
	A sine generator based on a table. The table is created using degrees where one full wave is 360 degrees. More...
class	SingleBuffer< T >
	A simple Buffer implementation which just uses a (dynamically sized) array. More...
class	Slice< T >
	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< T >
	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< T, N >
	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< T >
	Base class to define the abstract interface for the sound generating classes. More...
class	SquareWaveGenerator< T >
	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< T >
	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	Stream
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< T >
	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< T >
	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< T >
	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< T >
	Generates a random noise sound with the help of rand() function. More...

Namespaces
namespace	audio_tools
	Generic Implementation of sound input and output for desktop environments using portaudio.
namespace	audio_tools::audio_tools
	AutioTools Configuration.
namespace	audio_tools::std

Macros
#define	A2DP_BUFFER_COUNT   30
#define	A2DP_BUFFER_SIZE   512
#define	ALLOC_MEM_INIT   0
#define	ANALOG_BUFFER_COUNT   6
#define	ANALOG_BUFFER_SIZE   512
#define	ANALOG_BUFFER_SIZE   512
#define	ANALOG_BUFFERS   50
#define	ANALOG_BUFFERS   5
#define	ANALOG_MAX_OUT_CHANNELS   1
#define	ANALOG_MAX_SAMPLE_RATE   16000
#define	ARDUINO
#define	assert(T)
#define	AUDIO_INCLUDE_CORE   true
#define	AUDIOTOOLS_MAJOR_VERSION   1
#define	AUDIOTOOLS_METADATA_ICY_ASCII_ONLY   true
#define	AUDIOTOOLS_METADATA_ICY_LIMIT   400
#define	AUDIOTOOLS_MINOR_VERSION   2
#define	AUDIOTOOLS_VERSION   "1.2.4"
#define	CHANNEL_SELECT_BUFFER_SIZE   256
#define	CHECK_MEMORY()
#define	CODEC_DELAY_MS   10
#define	COPY_DELAY_ON_NODATA   10
#define	COPY_RETRY_LIMIT   20
#define	DEFAULT_BITS_PER_SAMPLE   16
#define	DEFAULT_BUFFER_SIZE   125
#define	DEFAULT_BUFFER_SIZE   1024
#define	DEFAULT_CHANNELS   2
#define	DEFAULT_SAMPLE_RATE   44100
#define	DESKTOP_MILLIS_DEFINED
#define	df2str(s)
#define	df2xstr(s)
#define	ESP_ARDUINO_VERSION_STR   df2str(ESP_ARDUINO_VERSION_MAJOR) "." df2str(ESP_ARDUINO_VERSION_MINOR) "." df2str(ESP_ARDUINO_VERSION_PATCH)
#define	ESP_IDF_VERSION_VAL(a, b, c)
#define	EXIT_ON_STOP
#define	FIX_SYNC_SYNCHRONIZE
#define	GPIO_NONE   -1
#define	GPIO_TO_INT(pin)
#define	GPIO_TO_INT(pin)
#define	GPIO_TO_STR(pin)
#define	GPIO_TO_STR(pin)
#define	HIGH   0x1
#define	HTTP_CHUNKED_SIZE_MAX_LEN   80
#define	HTTP_MAX_LEN   1024
#define	I2S_BUFFER_COUNT   6
#define	I2S_BUFFER_SIZE   512
#define	I2S_DEFAULT_PORT   0
#define	I2S_USE_APLL   false
#define	I2S_USE_APLL   false
#define	INPUT   0x0
#define	INPUT_PULLUP   0x2
#define	INT24_MAX   0x7FFFFF
#define	INT24_MAX   0x7FFFFF
#define	IRAM_ATTR
#define	IS_GPIO(pin)
#define	IS_GPIO(pin)
#define	IS_NOARDUINO
#define	IS_RENESAS   1
#define	LOG_LEVEL   AudioLogger::Warning
#define	LOG_METHOD   __PRETTY_FUNCTION__
#define	LOG_PRINTF_BUFFER_SIZE   60
#define	LOG_PRINTF_BUFFER_SIZE   303
#define	LOG_STREAM   Serial
#define	LOGD(...)
#define	LOGE(...)
#define	LOGI(...)
#define	LOGW(...)
#define	LOW   0x0
#define	MAX_SINGLE_CHARS   8
#define	MAX_ZERO_READ_COUNT   3
#define	MTS_PRINT_PIDS_LEN   (16)
#define	MTS_UNDERFLOW_LIMIT   188
#define	MTS_WRITE_BUFFER_SIZE   2000
#define	NO_INPLACE_INIT_SUPPORT
#define	NO_INPLACE_INIT_SUPPORT
#define	NO_TRACE
#define	NO_TRACED
#define	NO_TRACEE
#define	NO_TRACEI
#define	OUTPUT   0x1
#define	PI   3.14159265359f
#define	PIN_ANALOG_START   26
#define	PIN_ANALOG_START   PA0
#define	PIN_ANALOG_START   A0
#define	PIN_CS   SS
#define	PIN_CS   SS
#define	PIN_CS   1
#define	PIN_CS   4
#define	PIN_CS   -1
#define	PIN_CS   9
#define	PIN_I2S_BCK   13
#define	PIN_I2S_BCK   -1
#define	PIN_I2S_BCK   26
#define	PIN_I2S_BCK   1
#define	PIN_I2S_BCK   -1
#define	PIN_I2S_DATA_IN   12
#define	PIN_I2S_DATA_IN   -1
#define	PIN_I2S_DATA_IN   28
#define	PIN_I2S_DATA_IN   3
#define	PIN_I2S_DATA_IN   -1
#define	PIN_I2S_DATA_OUT   11
#define	PIN_I2S_DATA_OUT   -1
#define	PIN_I2S_DATA_OUT   28
#define	PIN_I2S_DATA_OUT   3
#define	PIN_I2S_DATA_OUT   -1
#define	PIN_I2S_MUTE   -1
#define	PIN_I2S_MUTE   23
#define	PIN_I2S_MUTE   -1
#define	PIN_I2S_MUTE   -1
#define	PIN_I2S_MUTE   -1
#define	PIN_I2S_WS   10
#define	PIN_I2S_WS   -1
#define	PIN_I2S_WS   PIN_I2S_BCK+1
#define	PIN_I2S_WS   PIN_I2S_BCK+1
#define	PIN_I2S_WS   -1
#define	PIN_PWM_COUNT   12
#define	PIN_PWM_START   6
#define	PIN_PWM_START   12
#define	PIN_PWM_START   6
#define	PIN_PWM_START   PA0
#define	PIN_PWM_START   D2
#define	PREFER_FIXEDPOINT   false
#define	PRINT_FLUSH_OVERRIDE
#define	PSTR(fmt)
#define	PWM_AUDIO_FREQUENCY   30000
#define	PWM_BUFFER_COUNT   4
#define	PWM_BUFFER_SIZE   125
#define	PWM_BUFFER_SIZE   1024
#define	PWM_DEFAULT_TIMER   TIM2
#define	PWM_FREQ_TIMER_NO   3
#define	PWM_MAX_SAMPLE_RATE   8000
#define	RP2040_MBED
#define	SOFT_MUTE_VALUE   0
#define	SOFT_MUTE_VALUE   0
#define	SOFT_MUTE_VALUE   0
#define	SOFT_MUTE_VALUE   0
#define	STREAM_READ_OVERRIDE
#define	STREAM_READCHAR_OVERRIDE
#define	STREAM_WRITE_OVERRIDE
#define	TRACED()
#define	TRACEE()
#define	TRACEI()
#define	TRACEW()
#define	URL_CLIENT_TIMEOUT   60000
#define	URL_CLIENT_TIMEOUT   60000;
#define	URL_HANDSHAKE_TIMEOUT   120000
#define	USE_ALLOCATOR   false
#define	USE_ANALOG
#define	USE_ANALOG
#define	USE_ANALOG_ARDUINO
#define	USE_ANALOG_ARDUINO
#define	USE_ANALOG_ARDUINO
#define	USE_AUDIO_LOGGING   true
#define	USE_AUDIO_SERVER
#define	USE_AUDIO_SERVER
#define	USE_AUDIOTOOLS_NS   true
#define	USE_CHECK_MEMORY   false
#define	USE_CONCURRENCY
#define	USE_CPP_TASK
#define	USE_ESP32_LOGGER   false
#define	USE_ETHERNET
#define	USE_ETHERNET
#define	USE_I2S
#define	USE_I2S
#define	USE_I2S
#define	USE_INT24_FROM_INT
#define	USE_INT24_FROM_INT
#define	USE_INT24_FROM_INT
#define	USE_INT24_FROM_INT
#define	USE_LEGACY_I2S   (ESP_IDF_VERSION < ESP_IDF_VERSION_VAL(5, 0, 0))
#define	USE_OBSOLETE   false
#define	USE_OPENCV
#define	USE_PRINT_FLUSH   true
#define	USE_PSRAM
#define	USE_PWM
#define	USE_PWM
#define	USE_PWM
#define	USE_PWM
#define	USE_RESAMPLE_BUFFER   true
#define	USE_SD_NO_NS
#define	USE_SD_NO_NS
#define	USE_SD_NO_NS
#define	USE_SD_NO_NS
#define	USE_SD_NO_NS
#define	USE_SD_SUPPORTS_SPI
#define	USE_SD_SUPPORTS_SPI
#define	USE_SERVER_ACCEPT   true
#define	USE_SERVER_ACCEPT   1
#define	USE_STD_CONCURRENCY
#define	USE_STRTOD
#define	USE_TASK   false
#define	USE_TIMER
#define	USE_TIMER
#define	USE_TIMER
#define	USE_TIMER
#define	USE_TIMER
#define	USE_TIMER
#define	USE_TYPETRAITS
#define	USE_TYPETRAITS
#define	USE_TYPETRAITS
#define	USE_URL_ARDUINO
#define	USE_URL_ARDUINO
#define	USE_URL_ARDUINO
#define	USE_URL_ARDUINO
#define	USE_WIFI
#define	USE_WIFI
#define	VS1053_CS   6
#define	VS1053_CS_SD   9
#define	VS1053_DCS   7
#define	VS1053_DEFAULT_VOLUME   0.7
#define	VS1053_DEFINED
#define	VS1053_DREQ   2
#define	VS1053_EXT   1
#define	VS1053_RESET   8
#define	ZLOGE(...)

Typedefs
using	EncodedAudioPrint = EncodedAudioOutput
	Legacy alias for EncodedAudioOutput.

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	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	I2SSignalType {   Digital , Analog , PDM , TDM ,   Digital , Analog , PDM , TDM }
	I2S Signal Types: Digital, Analog, PDM. More...
enum	MemoryType {   RAM , PS_RAM , FLASH_RAM , RAM ,   PS_RAM , FLASH_RAM }
	Memory types. More...
enum	PrintCharFmt { DEC = 10 , HEX = 16 , DEC = 10 , HEX = 16 }
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	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).
long	map (long x, long in_min, long in_max, long out_min, long out_max)
	Maps input to output values.
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 ()

Macro Definition Documentation

A2DP_BUFFER_COUNT

#define A2DP_BUFFER_COUNT   30

A2DP_BUFFER_SIZE

#define A2DP_BUFFER_SIZE   512

ALLOC_MEM_INIT

#define ALLOC_MEM_INIT   0

ANALOG_BUFFER_COUNT

#define ANALOG_BUFFER_COUNT   6

ANALOG_BUFFER_SIZE [1/2]

#define ANALOG_BUFFER_SIZE   512

ANALOG_BUFFER_SIZE [2/2]

#define ANALOG_BUFFER_SIZE   512

ANALOG_BUFFERS [1/2]

#define ANALOG_BUFFERS   50

ANALOG_BUFFERS [2/2]

#define ANALOG_BUFFERS   5

ANALOG_MAX_OUT_CHANNELS

#define ANALOG_MAX_OUT_CHANNELS   1

ANALOG_MAX_SAMPLE_RATE

#define ANALOG_MAX_SAMPLE_RATE   16000

ARDUINO

#define ARDUINO

assert

#define assert

(

T

)

AUDIO_INCLUDE_CORE

#define AUDIO_INCLUDE_CORE   true

AUDIOTOOLS_MAJOR_VERSION

#define AUDIOTOOLS_MAJOR_VERSION   1

AUDIOTOOLS_METADATA_ICY_ASCII_ONLY

#define AUDIOTOOLS_METADATA_ICY_ASCII_ONLY   true

AUDIOTOOLS_METADATA_ICY_LIMIT

#define AUDIOTOOLS_METADATA_ICY_LIMIT   400

AUDIOTOOLS_MINOR_VERSION

#define AUDIOTOOLS_MINOR_VERSION   2

AUDIOTOOLS_VERSION

#define AUDIOTOOLS_VERSION   "1.2.4"

CHANNEL_SELECT_BUFFER_SIZE

#define CHANNEL_SELECT_BUFFER_SIZE   256

CHECK_MEMORY

#define CHECK_MEMORY

(

)

CODEC_DELAY_MS

#define CODEC_DELAY_MS   10

COPY_DELAY_ON_NODATA

#define COPY_DELAY_ON_NODATA   10

COPY_RETRY_LIMIT

#define COPY_RETRY_LIMIT   20

DEFAULT_BITS_PER_SAMPLE

#define DEFAULT_BITS_PER_SAMPLE   16

DEFAULT_BUFFER_SIZE [1/2]

#define DEFAULT_BUFFER_SIZE   125

DEFAULT_BUFFER_SIZE [2/2]

#define DEFAULT_BUFFER_SIZE   1024

DEFAULT_CHANNELS

#define DEFAULT_CHANNELS   2

DEFAULT_SAMPLE_RATE

#define DEFAULT_SAMPLE_RATE   44100

DESKTOP_MILLIS_DEFINED

#define DESKTOP_MILLIS_DEFINED

df2str

#define df2str

(

s

)

Value:

df2xstr(s)

df2xstr

#define df2xstr

(

s

)

Value:

#s

ESP_ARDUINO_VERSION_STR

#define ESP_ARDUINO_VERSION_STR   df2str(ESP_ARDUINO_VERSION_MAJOR) "." df2str(ESP_ARDUINO_VERSION_MINOR) "." df2str(ESP_ARDUINO_VERSION_PATCH)

ESP_IDF_VERSION_VAL

#define ESP_IDF_VERSION_VAL	(		a,
			b,
			c )

Value:

0

EXIT_ON_STOP

#define EXIT_ON_STOP

FIX_SYNC_SYNCHRONIZE

#define FIX_SYNC_SYNCHRONIZE

GPIO_NONE

#define GPIO_NONE   -1

GPIO_TO_INT [1/2]

#define GPIO_TO_INT

(

pin

)

Value:

pin.pin

GPIO_TO_INT [2/2]

#define GPIO_TO_INT

(

pin

)

Value:

pin

GPIO_TO_STR [1/2]

#define GPIO_TO_STR

(

pin

)

Value:

std::to_string(GPIO_TO_INT(pin)).c_str()

GPIO_TO_STR [2/2]

#define GPIO_TO_STR

(

pin

)

Value:

std::to_string(pin).c_str()

HIGH

#define HIGH   0x1

HTTP_CHUNKED_SIZE_MAX_LEN

#define HTTP_CHUNKED_SIZE_MAX_LEN   80

HTTP_MAX_LEN

#define HTTP_MAX_LEN   1024

I2S_BUFFER_COUNT

#define I2S_BUFFER_COUNT   6

I2S_BUFFER_SIZE

#define I2S_BUFFER_SIZE   512

I2S_DEFAULT_PORT

#define I2S_DEFAULT_PORT   0

I2S_USE_APLL [1/2]

#define I2S_USE_APLL   false

I2S_USE_APLL [2/2]

#define I2S_USE_APLL   false

INPUT

#define INPUT   0x0

INPUT_PULLUP

#define INPUT_PULLUP   0x2

INT24_MAX [1/2]

#define INT24_MAX   0x7FFFFF

INT24_MAX [2/2]

#define INT24_MAX   0x7FFFFF

IRAM_ATTR

#define IRAM_ATTR

IS_GPIO [1/2]

#define IS_GPIO

(

pin

)

Value:

(pin.port != nullptr)

IS_GPIO [2/2]

#define IS_GPIO

(

pin

)

Value:

(pin >= 0)

IS_NOARDUINO

#define IS_NOARDUINO

IS_RENESAS

#define IS_RENESAS   1

LOG_LEVEL

#define LOG_LEVEL   AudioLogger::Warning

LOG_METHOD

#define LOG_METHOD   __PRETTY_FUNCTION__

LOG_PRINTF_BUFFER_SIZE [1/2]

#define LOG_PRINTF_BUFFER_SIZE   60

LOG_PRINTF_BUFFER_SIZE [2/2]

#define LOG_PRINTF_BUFFER_SIZE   303

LOG_STREAM

#define LOG_STREAM   Serial

LOGD

#define LOGD

(

...

)

LOGE

#define LOGE

(

...

)

LOGI

#define LOGI

(

...

)

LOGW

#define LOGW

(

...

)

LOW

#define LOW   0x0

MAX_SINGLE_CHARS

#define MAX_SINGLE_CHARS   8

MAX_ZERO_READ_COUNT

#define MAX_ZERO_READ_COUNT   3

MTS_PRINT_PIDS_LEN

#define MTS_PRINT_PIDS_LEN   (16)

MTS_UNDERFLOW_LIMIT

#define MTS_UNDERFLOW_LIMIT   188

MTS_WRITE_BUFFER_SIZE

#define MTS_WRITE_BUFFER_SIZE   2000

NO_INPLACE_INIT_SUPPORT [1/2]

#define NO_INPLACE_INIT_SUPPORT

NO_INPLACE_INIT_SUPPORT [2/2]

#define NO_INPLACE_INIT_SUPPORT

NO_TRACE

#define NO_TRACE

NO_TRACED

#define NO_TRACED

NO_TRACEE

#define NO_TRACEE

NO_TRACEI

#define NO_TRACEI

OUTPUT

#define OUTPUT   0x1

PI

#define PI   3.14159265359f

PIN_ANALOG_START [1/3]

#define PIN_ANALOG_START   26

PIN_ANALOG_START [2/3]

#define PIN_ANALOG_START   PA0

PIN_ANALOG_START [3/3]

#define PIN_ANALOG_START   A0

PIN_CS [1/6]

#define PIN_CS   SS

PIN_CS [2/6]

#define PIN_CS   SS

PIN_CS [3/6]

#define PIN_CS   1

PIN_CS [4/6]

#define PIN_CS   4

PIN_CS [5/6]

#define PIN_CS   -1

PIN_CS [6/6]

#define PIN_CS   9

PIN_I2S_BCK [1/5]

#define PIN_I2S_BCK   13

PIN_I2S_BCK [2/5]

#define PIN_I2S_BCK   -1

PIN_I2S_BCK [3/5]

#define PIN_I2S_BCK   26

PIN_I2S_BCK [4/5]

#define PIN_I2S_BCK   1

PIN_I2S_BCK [5/5]

#define PIN_I2S_BCK   -1

PIN_I2S_DATA_IN [1/5]

#define PIN_I2S_DATA_IN   12

PIN_I2S_DATA_IN [2/5]

#define PIN_I2S_DATA_IN   -1

PIN_I2S_DATA_IN [3/5]

#define PIN_I2S_DATA_IN   28

PIN_I2S_DATA_IN [4/5]

#define PIN_I2S_DATA_IN   3

PIN_I2S_DATA_IN [5/5]

#define PIN_I2S_DATA_IN   -1

PIN_I2S_DATA_OUT [1/5]

#define PIN_I2S_DATA_OUT   11

PIN_I2S_DATA_OUT [2/5]

#define PIN_I2S_DATA_OUT   -1

PIN_I2S_DATA_OUT [3/5]

#define PIN_I2S_DATA_OUT   28

PIN_I2S_DATA_OUT [4/5]

#define PIN_I2S_DATA_OUT   3

PIN_I2S_DATA_OUT [5/5]

#define PIN_I2S_DATA_OUT   -1

PIN_I2S_MUTE [1/5]

#define PIN_I2S_MUTE   -1

PIN_I2S_MUTE [2/5]

#define PIN_I2S_MUTE   23

PIN_I2S_MUTE [3/5]

#define PIN_I2S_MUTE   -1

PIN_I2S_MUTE [4/5]

#define PIN_I2S_MUTE   -1

PIN_I2S_MUTE [5/5]

#define PIN_I2S_MUTE   -1

PIN_I2S_WS [1/5]

#define PIN_I2S_WS   10

PIN_I2S_WS [2/5]

#define PIN_I2S_WS   -1

PIN_I2S_WS [3/5]

#define PIN_I2S_WS   PIN_I2S_BCK+1

PIN_I2S_WS [4/5]

#define PIN_I2S_WS   PIN_I2S_BCK+1

PIN_I2S_WS [5/5]

#define PIN_I2S_WS   -1

PIN_PWM_COUNT

#define PIN_PWM_COUNT   12

PIN_PWM_START [1/5]

#define PIN_PWM_START   6

PIN_PWM_START [2/5]

#define PIN_PWM_START   12

PIN_PWM_START [3/5]

#define PIN_PWM_START   6

PIN_PWM_START [4/5]

#define PIN_PWM_START   PA0

PIN_PWM_START [5/5]

#define PIN_PWM_START   D2

PREFER_FIXEDPOINT

#define PREFER_FIXEDPOINT   false

PRINT_FLUSH_OVERRIDE

#define PRINT_FLUSH_OVERRIDE

PSTR

#define PSTR

(

fmt

)

Value:

fmt

PWM_AUDIO_FREQUENCY

#define PWM_AUDIO_FREQUENCY   30000

PWM_BUFFER_COUNT

#define PWM_BUFFER_COUNT   4

PWM_BUFFER_SIZE [1/2]

#define PWM_BUFFER_SIZE   125

PWM_BUFFER_SIZE [2/2]

#define PWM_BUFFER_SIZE   1024

PWM_DEFAULT_TIMER

#define PWM_DEFAULT_TIMER   TIM2

PWM_FREQ_TIMER_NO

#define PWM_FREQ_TIMER_NO   3

PWM_MAX_SAMPLE_RATE

#define PWM_MAX_SAMPLE_RATE   8000

RP2040_MBED

#define RP2040_MBED

SOFT_MUTE_VALUE [1/4]

#define SOFT_MUTE_VALUE   0

SOFT_MUTE_VALUE [2/4]

#define SOFT_MUTE_VALUE   0

SOFT_MUTE_VALUE [3/4]

#define SOFT_MUTE_VALUE   0

SOFT_MUTE_VALUE [4/4]

#define SOFT_MUTE_VALUE   0

STREAM_READ_OVERRIDE

#define STREAM_READ_OVERRIDE

STREAM_READCHAR_OVERRIDE

#define STREAM_READCHAR_OVERRIDE

STREAM_WRITE_OVERRIDE

#define STREAM_WRITE_OVERRIDE

TRACED

#define TRACED

(

)

TRACEE

#define TRACEE

(

)

TRACEI

#define TRACEI

(

)

TRACEW

#define TRACEW

(

)

URL_CLIENT_TIMEOUT [1/2]

#define URL_CLIENT_TIMEOUT   60000

URL_CLIENT_TIMEOUT [2/2]

#define URL_CLIENT_TIMEOUT   60000;

URL_HANDSHAKE_TIMEOUT

#define URL_HANDSHAKE_TIMEOUT   120000

USE_ALLOCATOR

#define USE_ALLOCATOR   false

USE_ANALOG [1/2]

#define USE_ANALOG

USE_ANALOG [2/2]

#define USE_ANALOG

USE_ANALOG_ARDUINO [1/3]

#define USE_ANALOG_ARDUINO

USE_ANALOG_ARDUINO [2/3]

#define USE_ANALOG_ARDUINO

USE_ANALOG_ARDUINO [3/3]

#define USE_ANALOG_ARDUINO

USE_AUDIO_LOGGING

#define USE_AUDIO_LOGGING   true

USE_AUDIO_SERVER [1/2]

#define USE_AUDIO_SERVER

USE_AUDIO_SERVER [2/2]

#define USE_AUDIO_SERVER

USE_AUDIOTOOLS_NS

#define USE_AUDIOTOOLS_NS   true

USE_CHECK_MEMORY

#define USE_CHECK_MEMORY   false

USE_CONCURRENCY

#define USE_CONCURRENCY

USE_CPP_TASK

#define USE_CPP_TASK

USE_ESP32_LOGGER

#define USE_ESP32_LOGGER   false

USE_ETHERNET [1/2]

#define USE_ETHERNET

USE_ETHERNET [2/2]

#define USE_ETHERNET

USE_I2S [1/3]

#define USE_I2S

USE_I2S [2/3]

#define USE_I2S

USE_I2S [3/3]

#define USE_I2S

USE_INT24_FROM_INT [1/4]

#define USE_INT24_FROM_INT

USE_INT24_FROM_INT [2/4]

#define USE_INT24_FROM_INT

USE_INT24_FROM_INT [3/4]

#define USE_INT24_FROM_INT

USE_INT24_FROM_INT [4/4]

#define USE_INT24_FROM_INT

USE_LEGACY_I2S

#define USE_LEGACY_I2S   (ESP_IDF_VERSION < ESP_IDF_VERSION_VAL(5, 0, 0))

USE_OBSOLETE

#define USE_OBSOLETE   false

USE_OPENCV

#define USE_OPENCV

USE_PRINT_FLUSH

#define USE_PRINT_FLUSH   true

USE_PSRAM

#define USE_PSRAM

USE_PWM [1/4]

#define USE_PWM

USE_PWM [2/4]

#define USE_PWM

USE_PWM [3/4]

#define USE_PWM

USE_PWM [4/4]

#define USE_PWM

USE_RESAMPLE_BUFFER

#define USE_RESAMPLE_BUFFER   true

USE_SD_NO_NS [1/5]

#define USE_SD_NO_NS

USE_SD_NO_NS [2/5]

#define USE_SD_NO_NS

USE_SD_NO_NS [3/5]

#define USE_SD_NO_NS

USE_SD_NO_NS [4/5]

#define USE_SD_NO_NS

USE_SD_NO_NS [5/5]

#define USE_SD_NO_NS

USE_SD_SUPPORTS_SPI [1/2]

#define USE_SD_SUPPORTS_SPI

USE_SD_SUPPORTS_SPI [2/2]

#define USE_SD_SUPPORTS_SPI

USE_SERVER_ACCEPT [1/2]

#define USE_SERVER_ACCEPT   true

USE_SERVER_ACCEPT [2/2]

#define USE_SERVER_ACCEPT   1

USE_STD_CONCURRENCY

#define USE_STD_CONCURRENCY

USE_STRTOD

#define USE_STRTOD

USE_TASK

#define USE_TASK   false

USE_TIMER [1/6]

#define USE_TIMER

USE_TIMER [2/6]

#define USE_TIMER

USE_TIMER [3/6]

#define USE_TIMER

USE_TIMER [4/6]

#define USE_TIMER

USE_TIMER [5/6]

#define USE_TIMER

USE_TIMER [6/6]

#define USE_TIMER

USE_TYPETRAITS [1/3]

#define USE_TYPETRAITS

USE_TYPETRAITS [2/3]

#define USE_TYPETRAITS

USE_TYPETRAITS [3/3]

#define USE_TYPETRAITS

USE_URL_ARDUINO [1/4]

#define USE_URL_ARDUINO

USE_URL_ARDUINO [2/4]

#define USE_URL_ARDUINO

USE_URL_ARDUINO [3/4]

#define USE_URL_ARDUINO

USE_URL_ARDUINO [4/4]

#define USE_URL_ARDUINO

USE_WIFI [1/2]

#define USE_WIFI

USE_WIFI [2/2]

#define USE_WIFI

VS1053_CS

#define VS1053_CS   6

VS1053_CS_SD

#define VS1053_CS_SD   9

VS1053_DCS

#define VS1053_DCS   7

VS1053_DEFAULT_VOLUME

#define VS1053_DEFAULT_VOLUME   0.7

VS1053_DEFINED

#define VS1053_DEFINED

VS1053_DREQ

#define VS1053_DREQ   2

VS1053_EXT

#define VS1053_EXT   1

VS1053_RESET

#define VS1053_RESET   8

ZLOGE

#define ZLOGE

(

...

)

Value:

printk(__VA_ARGS__)