RTSPMediaStreamerUsingTask - Task-driven RTP Media Streaming Engine. More...
#include <RTSPMediaStreamerUsingTask.h>
Public Member Functions | |
| RTSPMediaStreamerUsingTask (bool throttled=true) | |
| RTSPMediaStreamerUsingTask (IMediaSource &source, bool throttled=true) | |
| virtual | ~RTSPMediaStreamerUsingTask () |
| 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. | |
| IMediaSource * | getMediaSource () |
| 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 | initMediaSource () |
| 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 | sendPacketizedFrame (int firstPacketSize) |
| Send every already-fragmented RTP payload belonging to the next queued video frame (packetized sources, e.g. JPEGRtpEncoder). | |
| int | sendRtpPacketDirect () |
| Send a single RTP packet with audio data. | |
| void | setFixedDelayMs (uint32_t delayUs) |
| virtual void | setMediaSource (IMediaSource *source) |
| Configure the audio source for streaming. | |
| 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 (bool markerBit=false) |
| uint32_t | computeTimestampIncrement (int bytesSent) |
| Compute RTP timestamp increment based on samples sent. | |
| void | sendOut (uint16_t totalLen) |
| void | streamingTaskLoop () |
| void | tryLearnClientFromUdp (bool warnIfNone) |
Detailed Description
class audio_tools::RTSPMediaStreamerUsingTask< Platform >
RTSPMediaStreamerUsingTask - Task-driven RTP Media Streaming Engine.
The RTSPMediaStreamerUsingTask class extends RTSPMediaStreamerBase 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 (media 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 media 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.
Constructor & Destructor Documentation
◆ RTSPMediaStreamerUsingTask() [1/2]
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inline |
◆ RTSPMediaStreamerUsingTask() [2/2]
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inline |
◆ ~RTSPMediaStreamerUsingTask()
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inlinevirtual |
Member Function Documentation
◆ applyThrottling()
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inlineprotected |
◆ buildRtpHeader()
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inlineprotectedinherited |
◆ checkTimerPeriodChange()
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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()
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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()
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inlineinherited |
Get current RTP timestamp value that will be used in the next packet.
- Returns
- Current RTP timestamp
◆ currentSeq()
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inlineinherited |
Get current RTP sequence number that will be used in the next packet.
- Returns
- Current RTP sequence number
◆ currentSsrc()
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inlineinherited |
Get current SSRC used in RTP header.
◆ getMediaSource()
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inlineinherited |
Get the configured audio source.
- Returns
- Pointer to the current audio source, nullptr if not set
- See also
- setAudioSource()
◆ getRtcpServerPort()
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inlineinherited |
Get the RTCP server port number.
- Returns
- Port number used for RTCP communication, 0 if not initialized
- See also
- initUdpTransport()
◆ getRtpServerPort()
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inlineinherited |
Get the RTP server port number.
- Returns
- Port number used for RTP packet transmission, 0 if not initialized
- See also
- initUdpTransport()
◆ getTimerPeriodMs()
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inlineinherited |
Get the timer period in milliseconds.
- Returns
- Timer period in milliseconds for convenience
- See also
- getTimerPeriodUs()
◆ getTimerPeriodUs()
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inlineinherited |
Get the timer period in microseconds.
- Returns
- Timer period configured from audio source format
- See also
- initAudioSource()
◆ initMediaSource()
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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()
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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()
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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
◆ isTaskRunning()
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inline |
◆ releaseUdpTransport()
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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()
◆ sendOut()
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inlineprotectedinherited |
◆ sendPacketizedFrame()
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inlineinherited |
Send every already-fragmented RTP payload belonging to the next queued video frame (packetized sources, e.g. JPEGRtpEncoder).
Each fragment already carries its own payload-format header (e.g. RFC 2435); this method only wraps it in the generic 12-byte RTP header. The marker bit is set on a fragment when packetSize() reports nothing left queued right after sending it - i.e. the source is expected to keep at most one frame's worth of fragments queued at a time (see JPEGRtpEncoder::m_maxQueuedFrames), so "nothing left" means "that was the last fragment of this frame". All fragments of the frame share a single timestamp (advanced once by the caller after this method returns, matching one call = one frame).
- Parameters
-
firstPacketSize Result of the packetSize() call the caller already made to detect that this source is packetized; reused here instead of querying it again for the first fragment.
- Returns
- Total payload bytes sent across all fragments of the frame, 0 if nothing was queued, negative on error
◆ sendRtpPacketDirect()
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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()
◆ setFixedDelayMs()
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inline |
◆ setMediaSource()
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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
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source Pointer to an IMediaSource implementation. Must not be nullptr.
- Note
- Calling this method will reinitialize audio source parameters
- See also
- IMediaSource, initAudioSource()
◆ setTaskParameters()
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◆ setThrottled()
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◆ setThrottleInterval()
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◆ start()
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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 RTSPMediaStreamerBase< Platform >.
◆ stop()
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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 RTSPMediaStreamerBase< Platform >.
◆ streamingTaskLoop()
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inlineprotected |
◆ timerCallback()
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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 RTSPMediaStreamerBase 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()
◆ tryLearnClientFromUdp()
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inlineprotectedinherited |
Member Data Documentation
◆ HEADER_SIZE
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protectedinherited |
◆ m_audioSource
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protectedinherited |
◆ m_ClientIP
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protectedinherited |
◆ m_ClientPort
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protectedinherited |
◆ m_fixed_delay_ms
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◆ m_fragmentSize
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protectedinherited |
◆ m_last_throttle_us
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protected |
◆ m_lastSamplesSent
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protectedinherited |
◆ m_payloadType
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protectedinherited |
◆ m_prevMsec
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protectedinherited |
◆ m_RtcpServerPort
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protectedinherited |
◆ m_RtcpSocket
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protectedinherited |
◆ m_RtpServerPort
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protectedinherited |
◆ m_RtpSocket
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protectedinherited |
◆ m_RtspTcpSocket
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protectedinherited |
◆ m_send_counter
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protected |
◆ m_SendIdx
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protectedinherited |
◆ m_SequenceNumber
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protectedinherited |
◆ m_Ssrc
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protectedinherited |
◆ m_streamingTask
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◆ m_taskCore
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◆ m_taskPriority
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◆ m_taskRunning
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◆ m_taskStackSize
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◆ m_TcpRtcpChannel
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protectedinherited |
◆ m_TcpRtpChannel
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protectedinherited |
◆ m_throttle_interval
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◆ m_throttled
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protected |
◆ m_timer_period_us
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protectedinherited |
◆ m_timer_restart_needed
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protectedinherited |
◆ m_Timestamp
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protectedinherited |
◆ m_udpRefCount
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protectedinherited |
◆ m_useTcpInterleaved
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protectedinherited |
◆ mMp3Carry
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protectedinherited |
◆ mMp3CarryLen
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protectedinherited |
◆ mRtpBuf
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protectedinherited |
◆ STREAMING_BUFFER_SIZE
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protectedinherited |
The documentation for this class was generated from the following file:
- src/AudioTools/Communication/RTSP/RTSPMediaStreamerUsingTask.h
