979 lines
31 KiB
C++
979 lines
31 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
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#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
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#include "module.h"
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#include "rtp_rtcp_defines.h"
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namespace webrtc {
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// forward declaration
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class Transport;
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class RtpRtcp : public Module
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{
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public:
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/*
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* create a RTP/RTCP module object
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*
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* id - unique identifier of this RTP/RTCP module object
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* audio - true for a audio version of the RTP/RTCP module object false will create a video version
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*/
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static RtpRtcp* CreateRtpRtcp(const WebRtc_Word32 id,
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const bool audio);
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/*
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* destroy a RTP/RTCP module object
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*
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* module - object to destroy
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*/
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static void DestroyRtpRtcp(RtpRtcp* module);
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/*
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* Returns version of the module and its components
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*
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* version - buffer to which the version will be written
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* remainingBufferInBytes - remaining number of WebRtc_Word8 in the version buffer
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* position - position of the next empty WebRtc_Word8 in the version buffer
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*/
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static WebRtc_Word32 GetVersion(WebRtc_Word8* version,
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WebRtc_UWord32& remainingBufferInBytes,
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WebRtc_UWord32& position);
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/*
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* Change the unique identifier of this object
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*
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* id - new unique identifier of this RTP/RTCP module object
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*/
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virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
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/*
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* De-muxing functionality for conferencing
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*
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* register a module that will act as a default module for this module
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* used for feedback messages back to the encoder when one encoded stream
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* is sent to multiple destinations
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*
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* module - default module
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*/
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virtual WebRtc_Word32 RegisterDefaultModule(RtpRtcp* module) = 0;
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/*
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* unregister the default module
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* will stop the demuxing feedback
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*/
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virtual WebRtc_Word32 DeRegisterDefaultModule() = 0;
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/*
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* returns true if a default module is registered, false otherwise
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*/
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virtual bool DefaultModuleRegistered() = 0;
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/*
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* returns number of registered child modules
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*/
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virtual WebRtc_UWord32 NumberChildModules() = 0;
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/*
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* Lip-sync between voice-video
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*
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* module - audio module
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*
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* Note: only allowed on a video module
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*/
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virtual WebRtc_Word32 RegisterSyncModule(RtpRtcp* module) = 0;
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/*
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* Turn off lip-sync between voice-video
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*/
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virtual WebRtc_Word32 DeRegisterSyncModule() = 0;
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/**************************************************************************
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*
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* Receiver functions
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*
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***************************************************************************/
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/*
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* Initialize receive side
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 InitReceiver() = 0;
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/*
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* Used by the module to deliver the incoming data to the codec module
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*
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* incomingDataCallback - callback object that will receive the incoming data
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback) = 0;
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/*
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* Used by the module to deliver messages to the codec module/appliation
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*
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* incomingMessagesCallback - callback object that will receive the incoming messages
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback) = 0;
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/*
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* configure a RTP packet timeout value
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*
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* RTPtimeoutMS - time in milliseconds after last received RTP packet
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* RTCPtimeoutMS - time in milliseconds after last received RTCP packet
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS,
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const WebRtc_UWord32 RTCPtimeoutMS) = 0;
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/*
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* Set periodic dead or alive notification
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*
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* enable - turn periodic dead or alive notification on/off
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* sampleTimeSeconds - sample interval in seconds for dead or alive notifications
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(const bool enable,
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const WebRtc_UWord8 sampleTimeSeconds) = 0;
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/*
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* Get periodic dead or alive notification status
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*
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* enable - periodic dead or alive notification on/off
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* sampleTimeSeconds - sample interval in seconds for dead or alive notifications
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(bool &enable,
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WebRtc_UWord8 &sampleTimeSeconds) = 0;
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/*
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* set codec name and payload type
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*
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* payloadName - payload name of codec
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* payloadType - payload type of codec
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* frequency - (audio specific) frequency of codec
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* channels - (audio specific) number of channels in codec (1 = mono, 2 = stereo)
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* rate - (audio) rate of codec
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* (video) maxBitrate of codec, bits/sec
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterReceivePayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency = 0,
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const WebRtc_UWord8 channels = 1,
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const WebRtc_UWord32 rate = 0) = 0;
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/*
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* Remove a registerd payload type from list of accepted payloads
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*
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* payloadType - payload type of codec
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 DeRegisterReceivePayload(const WebRtc_Word8 payloadType) = 0;
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/*
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* get configured payload type
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*
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* payloadName - payload name of codec
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* frequency - frequency of codec, ignored for video
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* payloadType - payload type of codec, ignored for video
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* channels - number of channels in codec (1 = mono, 2 = stereo)
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* rate - (audio) rate of codec (ignored if set to 0)
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 ReceivePayloadType(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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WebRtc_Word8* payloadType,
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const WebRtc_UWord32 rate = 0) const = 0;
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/*
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* get configured payload
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*
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* payloadType - payload type of codec
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* payloadName - payload name of codec
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* frequency - frequency of codec
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* channels - number of channels in codec (1 = mono, 2 = stereo)
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 ReceivePayload(const WebRtc_Word8 payloadType,
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WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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WebRtc_UWord32* frequency,
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WebRtc_UWord8* channels,
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WebRtc_UWord32* rate = NULL) const = 0;
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/*
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* Get last received remote timestamp
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*/
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virtual WebRtc_UWord32 RemoteTimestamp() const = 0;
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/*
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* Get the current estimated remote timestamp
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*
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* timestamp - estimated timestamp
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const = 0;
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/*
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* Get incoming SSRC
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*/
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virtual WebRtc_UWord32 RemoteSSRC() const = 0;
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/*
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* Get remote CSRC
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*
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* arrOfCSRC - array that will receive the CSRCs
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*
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* return -1 on failure else the number of valid entries in the list
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*/
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virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0;
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/*
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* get Current incoming payload
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*
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* payloadName - payload name of codec
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* payloadType - payload type of codec
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* frequency - frequency of codec
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* channels - number of channels in codec (2 = stereo)
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RemotePayload(WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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WebRtc_Word8* payloadType,
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WebRtc_UWord32* frequency,
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WebRtc_UWord8* channels) const = 0;
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/*
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* get the currently configured SSRC filter
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*
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* allowedSSRC - SSRC that will be allowed through
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const = 0;
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/*
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* set a SSRC to be used as a filter for incoming RTP streams
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*
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* allowedSSRC - SSRC that will be allowed through
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC) = 0;
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/*
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* called by the network module when we receive a packet
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*
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* incomingPacket - incoming packet buffer
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* packetLength - length of incoming buffer
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 IncomingPacket( const WebRtc_UWord8* incomingPacket,
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const WebRtc_UWord16 packetLength) = 0;
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/*
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* Option when not using the RegisterSyncModule function
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*
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* Inform the module about the received audion NTP
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 IncomingAudioNTP(const WebRtc_UWord32 audioReceivedNTPsecs,
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const WebRtc_UWord32 audioReceivedNTPfrac,
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const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
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const WebRtc_UWord32 audioRTCPArrivalTimeFrac) = 0;
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/**************************************************************************
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*
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* Sender
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*
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***************************************************************************/
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/*
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* Initialize send side
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 InitSender() = 0;
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/*
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* Used by the module to send RTP and RTCP packet to the network module
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*
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* outgoingTransport - transport object that will be called when packets are ready to be sent out on the network
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport) = 0;
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/*
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* set MTU
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*
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* size - Max transfer unit in bytes, default is 1500
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size) = 0;
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/*
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* set transtport overhead
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* default is IPv4 and UDP with no encryption
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*
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* TCP - true for TCP false UDP
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* IPv6 - true for IP version 6 false for version 4
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* authenticationOverhead - number of bytes to leave for an authentication header
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetTransportOverhead(const bool TCP,
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const bool IPV6,
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const WebRtc_UWord8 authenticationOverhead = 0) = 0;
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/*
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* Get max payload length
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*
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* A combination of the configuration MaxTransferUnit and TransportOverhead.
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* Does not account FEC/ULP/RED overhead if FEC is enabled.
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* Does not account for RTP headers
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*/
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virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
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/*
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* Get max data payload length
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*
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* A combination of the configuration MaxTransferUnit, headers and TransportOverhead.
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* Takes into account FEC/ULP/RED overhead if FEC is enabled.
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* Takes into account RTP headers
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*/
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virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
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/*
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* set RTPKeepaliveStatus
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*
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* enable - on/off
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* unknownPayloadType - payload type to use for RTP keepalive
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* deltaTransmitTimeMS - delta time between RTP keepalive packets
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetRTPKeepaliveStatus(const bool enable,
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const WebRtc_Word8 unknownPayloadType,
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const WebRtc_UWord16 deltaTransmitTimeMS) = 0;
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/*
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* Get RTPKeepaliveStatus
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*
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* enable - on/off
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* unknownPayloadType - payload type in use for RTP keepalive
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* deltaTransmitTimeMS - delta time between RTP keepalive packets
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
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WebRtc_Word8* unknownPayloadType,
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WebRtc_UWord16* deltaTransmitTimeMS) const = 0;
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/*
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* check if RTPKeepaliveStatus is enabled
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*/
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virtual bool RTPKeepalive() const = 0;
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/*
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* set codec name and payload type
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*
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* payloadName - payload name of codec
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* payloadType - payload type of codec
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* frequency - frequency of codec
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* channels - number of channels in codec (1 = mono, 2 = stereo)
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* rate - (audio) rate of codec
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* (video) maxBitrate of codec, bits/sec
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 RegisterSendPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency = 0,
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const WebRtc_UWord8 channels = 1,
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const WebRtc_UWord32 rate = 0) = 0;
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/*
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* Unregister a send payload
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*
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* payloadType - payload type of codec
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType) = 0;
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/*
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* get start timestamp
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*/
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virtual WebRtc_UWord32 StartTimestamp() const = 0;
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/*
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* configure start timestamp, default is a random number
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*
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* timestamp - start timestamp
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp) = 0;
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/*
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* Get SequenceNumber
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*/
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virtual WebRtc_UWord16 SequenceNumber() const = 0;
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/*
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* Set SequenceNumber, default is a random number
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq) = 0;
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/*
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* Get SSRC
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*/
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virtual WebRtc_UWord32 SSRC() const = 0;
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/*
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* configure SSRC, default is a random number
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc) = 0;
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/*
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* Get CSRC
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*
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* arrOfCSRC - array of CSRCs
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*
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* return -1 on failure else number of valid entries in the array
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*/
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virtual WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0;
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/*
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* Set CSRC
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*
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* arrOfCSRC - array of CSRCs
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* arrLength - number of valid entries in the array
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetCSRCs( const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
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const WebRtc_UWord8 arrLength) = 0;
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/*
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* includes CSRCs in RTP header if enabled
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*
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* include CSRC - on/off
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*
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* default:on
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetCSRCStatus(const bool include) = 0;
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/*
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* sends kRtcpByeCode when going from true to false
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*
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* sending - on/off
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSendingStatus(const bool sending) = 0;
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/*
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* get send status
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*/
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virtual bool Sending() const = 0;
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/*
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* Starts/Stops media packets, on by default
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*
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* sending - on/off
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending) = 0;
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/*
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* get send status
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*/
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virtual bool SendingMedia() const = 0;
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/*
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* get sent bitrate in Kbit/s
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*/
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virtual WebRtc_UWord32 BitrateSent() const = 0;
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/*
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* Used by the codec module to deliver a video or audio frame for packetization
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*
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* frameType - type of frame to send
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* payloadType - payload type of frame to send
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* timestamp - timestamp of frame to send
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* payloadData - payload buffer of frame to send
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* payloadSize - size of payload buffer to send
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* fragmentation - fragmentation offset data for fragmented frames such as layers or RED
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*
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* return -1 on failure else 0
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*/
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virtual WebRtc_Word32 SendOutgoingData(const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 timeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 payloadSize,
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const RTPFragmentationHeader* fragmentation = NULL) = 0;
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/**************************************************************************
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*
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* RTCP
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*
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***************************************************************************/
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/*
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* RegisterIncomingRTCPCallback
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*
|
|
* incomingMessagesCallback - callback object that will receive messages from RTCP
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback) = 0;
|
|
|
|
/*
|
|
* Get RTCP status
|
|
*/
|
|
virtual RTCPMethod RTCP() const = 0;
|
|
|
|
/*
|
|
* configure RTCP status i.e on(compound or non- compound)/off
|
|
*
|
|
* method - RTCP method to use
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method) = 0;
|
|
|
|
/*
|
|
* Set RTCP CName (i.e unique identifier)
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetCNAME(const WebRtc_Word8 cName[RTCP_CNAME_SIZE]) = 0;
|
|
|
|
/*
|
|
* Get RTCP CName (i.e unique identifier)
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 CNAME(WebRtc_Word8 cName[RTCP_CNAME_SIZE]) = 0;
|
|
|
|
/*
|
|
* Get remote CName
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC,
|
|
WebRtc_Word8 cName[RTCP_CNAME_SIZE]) const = 0;
|
|
|
|
/*
|
|
* Get remote NTP
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32 *ReceivedNTPsecs,
|
|
WebRtc_UWord32 *ReceivedNTPfrac,
|
|
WebRtc_UWord32 *RTCPArrivalTimeSecs,
|
|
WebRtc_UWord32 *RTCPArrivalTimeFrac) const = 0;
|
|
|
|
/*
|
|
* AddMixedCNAME
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
|
|
const WebRtc_Word8 cName[RTCP_CNAME_SIZE]) = 0;
|
|
|
|
/*
|
|
* RemoveMixedCNAME
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC) = 0;
|
|
|
|
/*
|
|
* Get RoundTripTime
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
|
|
WebRtc_UWord16* RTT,
|
|
WebRtc_UWord16* avgRTT,
|
|
WebRtc_UWord16* minRTT,
|
|
WebRtc_UWord16* maxRTT) const = 0 ;
|
|
|
|
/*
|
|
* Reset RoundTripTime statistics
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC)= 0 ;
|
|
|
|
/*
|
|
* Force a send of a RTCP packet
|
|
* normal SR and RR are triggered via the process function
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport) = 0;
|
|
|
|
/*
|
|
* Good state of RTP receiver inform sender
|
|
*/
|
|
virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID) = 0;
|
|
|
|
/*
|
|
* Send a RTCP Slice Loss Indication (SLI)
|
|
* 6 least significant bits of pictureID
|
|
*/
|
|
virtual WebRtc_Word32 SendRTCPSliceLossIndication(const WebRtc_UWord8 pictureID) = 0;
|
|
|
|
/*
|
|
* Reset RTP statistics
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 ResetStatisticsRTP() = 0;
|
|
|
|
/*
|
|
* statistics of our localy created statistics of the received RTP stream
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8 *fraction_lost, // scale 0 to 255
|
|
WebRtc_UWord32 *cum_lost, // number of lost packets
|
|
WebRtc_UWord32 *ext_max, // highest sequence number received
|
|
WebRtc_UWord32 *jitter,
|
|
WebRtc_UWord32 *max_jitter = NULL) const = 0;
|
|
|
|
/*
|
|
* Reset RTP data counters for the receiving side
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 ResetReceiveDataCountersRTP() = 0;
|
|
|
|
/*
|
|
* Reset RTP data counters for the sending side
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 ResetSendDataCountersRTP() = 0;
|
|
|
|
/*
|
|
* statistics of the amount of data sent and received
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent,
|
|
WebRtc_UWord32 *packetsSent,
|
|
WebRtc_UWord32 *bytesReceived,
|
|
WebRtc_UWord32 *packetsReceived) const = 0;
|
|
/*
|
|
* Get received RTCP sender info
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoteRTCPStat( RTCPSenderInfo* senderInfo) = 0;
|
|
|
|
/*
|
|
* Get received RTCP report block
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoteRTCPStat( const WebRtc_UWord32 remoteSSRC,
|
|
RTCPReportBlock* receiveBlock) = 0;
|
|
/*
|
|
* Set received RTCP report block
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 AddRTCPReportBlock(const WebRtc_UWord32 SSRC,
|
|
const RTCPReportBlock* receiveBlock) = 0;
|
|
|
|
/*
|
|
* RemoveRTCPReportBlock
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC) = 0;
|
|
|
|
/*
|
|
* (APP) Application specific data
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetRTCPApplicationSpecificData(const WebRtc_UWord8 subType,
|
|
const WebRtc_UWord32 name,
|
|
const WebRtc_UWord8* data,
|
|
const WebRtc_UWord16 length) = 0;
|
|
/*
|
|
* (XR) VOIP metric
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) = 0;
|
|
|
|
/*
|
|
* (TMMBR) Temporary Max Media Bit Rate
|
|
*/
|
|
virtual bool TMMBR() const = 0;
|
|
|
|
/*
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetTMMBRStatus(const bool enable) = 0;
|
|
|
|
/*
|
|
* local bw estimation changed
|
|
*
|
|
* for video called by internal estimator
|
|
* for audio (iSAC) called by engine, geting the data from the decoder
|
|
*/
|
|
virtual void OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit) = 0;
|
|
|
|
/*
|
|
* (NACK)
|
|
*/
|
|
virtual NACKMethod NACK() const = 0;
|
|
|
|
/*
|
|
* Turn negative acknowledgement requests on/off
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method) = 0;
|
|
|
|
/*
|
|
* Send a Negative acknowledgement packet
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
|
|
const WebRtc_UWord16 size) = 0;
|
|
|
|
/*
|
|
* Store the sent packets, needed to answer to a Negative acknowledgement requests
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200) = 0;
|
|
|
|
/**************************************************************************
|
|
*
|
|
* Audio
|
|
*
|
|
***************************************************************************/
|
|
|
|
/*
|
|
* RegisterAudioCallback
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback) = 0;
|
|
|
|
/*
|
|
* set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples) = 0;
|
|
|
|
/*
|
|
* Outband TelephoneEvent(DTMF) detection
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
|
|
const bool forwardToDecoder,
|
|
const bool detectEndOfTone = false) = 0;
|
|
|
|
/*
|
|
* Is outband TelephoneEvent(DTMF) turned on/off?
|
|
*/
|
|
virtual bool TelephoneEvent() const = 0;
|
|
|
|
/*
|
|
* Returns true if received DTMF events are forwarded to the decoder using
|
|
* the OnPlayTelephoneEvent callback.
|
|
*/
|
|
virtual bool TelephoneEventForwardToDecoder() const = 0;
|
|
|
|
/*
|
|
* SendTelephoneEventActive
|
|
*
|
|
* return true if we currently send a telephone event and 100 ms after an event is sent
|
|
* used to prevent teh telephone event tone to be recorded by the microphone and send inband
|
|
* just after the tone has ended
|
|
*/
|
|
virtual bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const = 0;
|
|
|
|
/*
|
|
* Send a TelephoneEvent tone using RFC 2833 (4733)
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key,
|
|
const WebRtc_UWord16 time_ms,
|
|
const WebRtc_UWord8 level) = 0;
|
|
|
|
/*
|
|
* Set payload type for Redundant Audio Data RFC 2198
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payloadType) = 0;
|
|
|
|
/*
|
|
* Get payload type for Redundant Audio Data RFC 2198
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payloadType) const = 0;
|
|
|
|
/*
|
|
* Set status and ID for header-extension-for-audio-level-indication.
|
|
* See https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
|
|
* for more details.
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(const bool enable,
|
|
const WebRtc_UWord8 ID) = 0;
|
|
|
|
/*
|
|
* Get status and ID for header-extension-for-audio-level-indication.
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(bool& enable,
|
|
WebRtc_UWord8& ID) const = 0;
|
|
|
|
/*
|
|
* Store the audio level in dBov for header-extension-for-audio-level-indication.
|
|
* This API shall be called before transmision of an RTP packet to ensure
|
|
* that the |level| part of the extended RTP header is updated.
|
|
*
|
|
* return -1 on failure else 0.
|
|
*/
|
|
virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov) = 0;
|
|
|
|
/**************************************************************************
|
|
*
|
|
* Video
|
|
*
|
|
***************************************************************************/
|
|
|
|
/*
|
|
* Register a callback object that will receive callbacks for video related events
|
|
* such as an incoming key frame request.
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback) = 0;
|
|
|
|
/*
|
|
* Set the estimated camera delay in MS
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS) = 0;
|
|
|
|
/*
|
|
* Set the start and max send bitrate
|
|
* used by the bandwidth management
|
|
*
|
|
* Not calling this or setting startBitrateKbit to 0 disables the bandwidth management
|
|
*
|
|
* minBitrateKbit = 0 equals no min bitrate
|
|
* maxBitrateKbit = 0 equals no max bitrate
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetSendBitrate(const WebRtc_UWord32 startBitrate,
|
|
const WebRtc_UWord16 minBitrateKbit,
|
|
const WebRtc_UWord16 maxBitrateKbit) = 0;
|
|
|
|
/*
|
|
* Turn on/off generic FEC
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetGenericFECStatus(const bool enable,
|
|
const WebRtc_UWord8 payloadTypeRED,
|
|
const WebRtc_UWord8 payloadTypeFEC) = 0;
|
|
|
|
/*
|
|
* Get generic FEC setting
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 GenericFECStatus(bool& enable,
|
|
WebRtc_UWord8& payloadTypeRED,
|
|
WebRtc_UWord8& payloadTypeFEC) = 0;
|
|
|
|
/*
|
|
* Set FEC code rate of key and delta frames
|
|
* codeRate on a scale of 0 to 255 where 255 is 100% added packets, hence protect up to 50% packet loss
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
|
|
const WebRtc_UWord8 deltaFrameCodeRate) = 0;
|
|
|
|
/*
|
|
* Set method for requestion a new key frame
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method) = 0;
|
|
|
|
/*
|
|
* send a request for a keyframe
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual WebRtc_Word32 RequestKeyFrame(const FrameType frameType = kVideoFrameKey) = 0;
|
|
|
|
/*
|
|
* Only for H.263 to interop with bad endpoints
|
|
*/
|
|
virtual WebRtc_Word32 SetH263InverseLogic(const bool enable) = 0;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
|