RTSPAudioStreamerUsingTask - Task-driven RTP Audio Streaming Engine. More...

#include <RTSPAudioStreamerUsingTask.h>

Inheritance diagram for RTSPAudioStreamerUsingTask< Platform >:

Public Member Functions
	RTSPAudioStreamerUsingTask (bool throttled=true)

	RTSPAudioStreamerUsingTask (IAudioSource &source, bool throttled=true)

bool	checkTimerPeriodChange ()
	Check if timer period has changed and update if necessary.

uint32_t	currentRtpTimestamp () const
	Get current RTP timestamp value that will be used in the next packet.

uint16_t	currentSeq () const
	Get current RTP sequence number that will be used in the next packet.

uint32_t	currentSsrc () const
	Get current SSRC used in RTP header.

IAudioSource *	getAudioSource ()
	Get the configured audio source.

u_short	getRtcpServerPort ()
	Get the RTCP server port number.

u_short	getRtpServerPort ()
	Get the RTP server port number.

uint32_t	getTimerPeriodMs () const
	Get the timer period in milliseconds.

uint32_t	getTimerPeriodUs () const
	Get the timer period in microseconds.

bool	initAudioSource ()
	Initialize audio source configuration.

void	initTcpInterleavedTransport (typename Platform::TcpClientType *tcpSock, int rtpChannel, int rtcpChannel)
	Initialize TCP interleaved transport for RTP over RTSP.

bool	initUdpTransport (IPAddress aClientIP, uint16_t aClientPort)
	Initialize UDP transport for RTP streaming.

bool	isTaskRunning () const

void	releaseUdpTransport (void)
	Release UDP transport resources.

int	sendRtpPacketDirect ()
	Send a single RTP packet with audio data.

virtual void	setAudioSource (IAudioSource *source)
	Configure the audio source for streaming.

void	setFixedDelayMs (uint32_t delayUs)

void	setTaskParameters (uint32_t stackSize, uint8_t priority, int core=-1)

void	setThrottled (bool isThrottled)

void	setThrottleInterval (uint32_t interval)

void	start () override
	Start audio source (base implementation)

void	stop () override
	Stop audio source (base implementation)

Static Public Member Functions
static void	timerCallback (void *audioStreamerObj)
	Static timer callback function for periodic RTP streaming.

Protected Member Functions
void	applyThrottling (unsigned long iterationStartUs)

void	buildRtpHeader ()

uint32_t	computeTimestampIncrement (int bytesSent)
	Compute RTP timestamp increment based on samples sent.

void	sendOut (uint16_t totalLen)

void	streamingTaskLoop ()

void	tryLearnClientFromUdp (bool warnIfNone)

Protected Attributes
const int	HEADER_SIZE = 12

IAudioSource *	m_audioSource = nullptr

IPAddress	m_ClientIP

uint16_t	m_ClientPort

uint16_t	m_fixed_delay_ms = 1

int	m_fragmentSize = 0

unsigned long	m_last_throttle_us = 0

int	m_lastSamplesSent = 0

int	m_payloadType = 96

uint32_t	m_prevMsec

uint16_t	m_RtcpServerPort

Platform::UdpSocketType *	m_RtcpSocket

uint16_t	m_RtpServerPort

Platform::UdpSocketType *	m_RtpSocket

Platform::TcpClientType *	m_RtspTcpSocket

uint32_t	m_send_counter = 0

int	m_SendIdx

u_short	m_SequenceNumber

uint32_t	m_Ssrc = 0x13F97E67

audio_tools::Task	m_streamingTask

int	m_taskCore = -1

uint8_t	m_taskPriority = 5

volatile bool	m_taskRunning

uint32_t	m_taskStackSize = 8192

int	m_TcpRtcpChannel

int	m_TcpRtpChannel

uint32_t	m_throttle_interval = 50

bool	m_throttled = true

int	m_timer_period_us = 20000

volatile bool	m_timer_restart_needed

uint32_t	m_Timestamp

int	m_udpRefCount

bool	m_useTcpInterleaved

audio_tools::Vector< uint8_t >	mMp3Carry

int	mMp3CarryLen = 0

audio_tools::Vector< uint8_t >	mRtpBuf

const int	STREAMING_BUFFER_SIZE = 1024 * 3

Detailed Description

template<typename Platform>
class audio_tools::RTSPAudioStreamerUsingTask< Platform >

RTSPAudioStreamerUsingTask - Task-driven RTP Audio Streaming Engine.

The RTSPAudioStreamerUsingTask class extends RTSPAudioStreamerBase with AudioTools Task-driven streaming functionality. Instead of using hardware timers, this class creates a dedicated task that continuously calls the timer callback at the appropriate intervals. This approach provides:

All base class functionality (audio source, UDP transport, RTP packets)
Task-based periodic streaming using AudioTools Task class
More predictable scheduling compared to hardware timers
Better resource control and priority management
Reduced timer resource usage
Optional throttling for timing control

The throttling feature is particularly important when working with audio sources that can provide data faster than the defined sampling rate, such as file readers, buffer sources, or fast generators. Without throttling, these sources would flood the network with packets faster than real-time playback.

Author: Phil Schatzmann

Member Function Documentation

◆ checkTimerPeriodChange()

template<typename Platform >

bool checkTimerPeriodChange ( )

inlineinherited

Check if timer period has changed and update if necessary.

Checks the audio source format for timer period changes and updates the cached value if different. This enables dynamic timer period adjustments during playback when the audio format changes.

Returns: true if timer period changed, false if unchanged or no audio source

Note: Call this periodically to detect format changes during streaming

See also: updateTimerPeriod()

◆ computeTimestampIncrement()

template<typename Platform >

uint32_t computeTimestampIncrement ( int bytesSent )

inlineprotectedinherited

Compute RTP timestamp increment based on samples sent.

Calculates the appropriate RTP timestamp increment based on the number of audio samples sent in the last packet. This ensures proper timing and synchronization in the RTP stream.

Parameters

bytesSent Number of bytes sent in the last RTP packet

Returns: Calculated timestamp increment value

Note: Uses exact samples sent if available, otherwise falls back to audio source format or a crude estimate

◆ currentRtpTimestamp()

template<typename Platform >

uint32_t currentRtpTimestamp ( ) const

inlineinherited

Get current RTP timestamp value that will be used in the next packet.

Returns: Current RTP timestamp

◆ currentSeq()

template<typename Platform >

uint16_t currentSeq ( ) const

inlineinherited

Get current RTP sequence number that will be used in the next packet.

Returns: Current RTP sequence number

◆ getAudioSource()

template<typename Platform >

IAudioSource * getAudioSource ( )

inlineinherited

Get the configured audio source.

Returns: Pointer to the current audio source, nullptr if not set

See also: setAudioSource()

◆ getRtcpServerPort()

template<typename Platform >

u_short getRtcpServerPort ( )

inlineinherited

Get the RTCP server port number.

Returns: Port number used for RTCP communication, 0 if not initialized

See also: initUdpTransport()

◆ getRtpServerPort()

template<typename Platform >

u_short getRtpServerPort ( )

inlineinherited

Get the RTP server port number.

Returns: Port number used for RTP packet transmission, 0 if not initialized

See also: initUdpTransport()

◆ getTimerPeriodMs()

template<typename Platform >

uint32_t getTimerPeriodMs ( ) const

inlineinherited

Get the timer period in milliseconds.

Returns: Timer period in milliseconds for convenience

See also: getTimerPeriodUs()

◆ getTimerPeriodUs()

template<typename Platform >

uint32_t getTimerPeriodUs ( ) const

inlineinherited

Get the timer period in microseconds.

Returns: Timer period configured from audio source format

See also: initAudioSource()

◆ initAudioSource()

template<typename Platform >

bool initAudioSource ( )

inlineinherited

Initialize audio source configuration.

Extracts and caches configuration parameters from the audio source including fragment size (bytes per packet) and timer period (microseconds between packets). This method must be called before streaming can begin.

Returns: true if initialization succeeded, false if audio source or format is invalid

Note: Fragment size determines how much audio data is sent per RTP packet; Timer period controls the streaming rate and must match the audio sample rate

◆ initTcpInterleavedTransport()

template<typename Platform >

void initTcpInterleavedTransport	(	typename Platform::TcpClientType *	tcpSock,
		int	rtpChannel,
		int	rtcpChannel
	)

inlineinherited

Initialize TCP interleaved transport for RTP over RTSP.

Configures the streamer to send RTP packets using RTSP TCP interleaving. RTP and RTCP packets are framed with '$', channel byte, and 2-byte length per RFC 2326 section 10.12 and sent over the RTSP control socket.

Parameters

tcpSock	Pointer to the RTSP TCP client socket
rtpChannel	Interleaved channel number for RTP packets (e.g., 0)
rtcpChannel	Interleaved channel number for RTCP packets (e.g., 1)

◆ initUdpTransport()

template<typename Platform >

bool initUdpTransport	(	IPAddress	aClientIP,
		uint16_t	aClientPort
	)

inlineinherited

Initialize UDP transport for RTP streaming.

Sets up UDP sockets for RTP and RTCP communication with a specific client. Automatically allocates consecutive port pairs starting from 6970. Supports multiple clients through reference counting - subsequent calls with different clients will reuse existing sockets.

Parameters

aClientIP	IP address of the RTP client that will receive the stream
aClientPort	Port number on the client for receiving RTP packets

Returns: true if transport initialization succeeded, false on socket creation failure

Note: Port allocation: RTP uses even ports (6970, 6972, ...), RTCP uses odd ports (6971, 6973, ...); Call releaseUdpTransport() when streaming to this client is complete

See also: releaseUdpTransport(), getRtpServerPort(), getRtcpServerPort()

◆ releaseUdpTransport()

template<typename Platform >

void releaseUdpTransport ( void )

inlineinherited

Release UDP transport resources.

Decrements the reference count for UDP socket usage and closes sockets when no more clients are using them. This implements proper resource management for multiple concurrent streaming sessions.

Note: Only closes sockets when reference count reaches zero; Always call this method when streaming to a client is complete

See also: initUdpTransport()

◆ sendRtpPacketDirect()

template<typename Platform >

int sendRtpPacketDirect ( )

inlineinherited

Send a single RTP packet with audio data.

Constructs and sends one RTP packet containing audio data from the configured source. This method handles the complete RTP packet creation process including:

RTP header construction with sequence numbers and timestamps
Audio data retrieval from the source
Network byte order conversion
UDP transmission to the configured client

Returns: Number of bytes sent in the packet payload, or negative value on error

Return values

-1	RTP buffer not allocated or audio source not configured

Note: This method is typically called periodically by the timer-driven streaming; Automatically increments sequence numbers and updates timestamps

See also: start(), timerCallback()

◆ setAudioSource()

template<typename Platform >

virtual void setAudioSource ( IAudioSource * source )

inlinevirtualinherited

Configure the audio source for streaming.

Sets or changes the audio source that provides data for the RTP stream. This method automatically initializes the audio source configuration including fragment size and timer period based on the source's format specifications.

Parameters

source Pointer to an IAudioSource implementation. Must not be nullptr.

Note: Calling this method will reinitialize audio source parameters

See also: IAudioSource, initAudioSource()

◆ start()

template<typename Platform >

void start ( )

inlineoverridevirtual

Start audio source (base implementation)

Begins the audio source and initializes the RTP buffer if not already done. This base implementation does not start any timer - derived classes should override this method to add timer functionality.

Note: UDP transport must be initialized before calling this method; Audio source must be configured before calling this method

Warning: Will log errors if audio source is not properly configured

See also: stop(), initUdpTransport(), setAudioSource()

Reimplemented from RTSPAudioStreamerBase< Platform >.

◆ stop()

template<typename Platform >

void stop ( )

inlineoverridevirtual

Stop audio source (base implementation)

Stops the audio source. This base implementation does not manage any timer - derived classes should override this method to add timer management.

Note: Does not release UDP transport or buffers - use releaseUdpTransport() separately; Can be restarted by calling start() again

See also: start(), releaseUdpTransport()

Reimplemented from RTSPAudioStreamerBase< Platform >.

◆ timerCallback()

template<typename Platform >

static void timerCallback ( void * audioStreamerObj )

inlinestaticinherited

Static timer callback function for periodic RTP streaming.

This static method is called periodically by timers to send RTP packets. It handles timing measurements and calls sendRtpPacketDirect() to transmit audio data. Performance monitoring ensures the streaming keeps up with real-time requirements.

This method can be used by any derived class that implements timer-driven streaming functionality.

Parameters

audioStreamerObj Void pointer to the RTSPAudioStreamerBase instance (passed as callback parameter)

Note: This is a static method suitable for use as a timer callback; Logs warnings if packet transmission takes too long

See also: sendRtpPacketDirect()

Member Data Documentation

◆ m_timer_restart_needed

template<typename Platform >

volatile bool m_timer_restart_needed

protectedinherited

Initial value:

=

false

The documentation for this class was generated from the following file:

src/AudioTools/Communication/RTSP/RTSPAudioStreamerUsingTask.h

Public Member Functions

Static Public Member Functions

Protected Member Functions

Protected Attributes

Detailed Description

Member Function Documentation

◆ checkTimerPeriodChange()

◆ computeTimestampIncrement()

◆ currentRtpTimestamp()

◆ currentSeq()

◆ getAudioSource()

◆ getRtcpServerPort()

◆ getRtpServerPort()

◆ getTimerPeriodMs()

◆ getTimerPeriodUs()

◆ initAudioSource()

◆ initTcpInterleavedTransport()

◆ initUdpTransport()

◆ releaseUdpTransport()

◆ sendRtpPacketDirect()

◆ setAudioSource()

◆ start()

◆ stop()

◆ timerCallback()

Member Data Documentation

◆ m_timer_restart_needed