11beccd712
Feature does not seem to be used and complicates other refactoring of the rtcp module. BUG= R=asapersson@webrtc.org, henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/54569004 Cr-Commit-Position: refs/heads/master@{#9304}
651 lines
19 KiB
C++
651 lines
19 KiB
C++
/*
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* Copyright (c) 2012 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 <set>
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#include <vector>
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#include "webrtc/modules/interface/module.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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namespace webrtc {
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// Forward declarations.
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class PacedSender;
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class ReceiveStatistics;
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class RemoteBitrateEstimator;
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class RtpReceiver;
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class Transport;
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class RtpRtcp : public Module {
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public:
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struct Configuration {
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Configuration();
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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
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* object false will create a video version
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* clock - The clock to use to read time. If NULL object
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* will be using the system clock.
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* incoming_data - Callback object that will receive the incoming
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* data. May not be NULL; default callback will do
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* nothing.
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* incoming_messages - Callback object that will receive the incoming
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* RTP messages. May not be NULL; default callback
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* will do nothing.
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* outgoing_transport - Transport object that will be called when packets
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* are ready to be sent out on the network
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* intra_frame_callback - Called when the receiver request a intra frame.
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* bandwidth_callback - Called when we receive a changed estimate from
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* the receiver of out stream.
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* audio_messages - Telephone events. May not be NULL; default
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* callback will do nothing.
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* remote_bitrate_estimator - Estimates the bandwidth available for a set of
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* streams from the same client.
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* paced_sender - Spread any bursts of packets into smaller
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* bursts to minimize packet loss.
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*/
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int32_t id;
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bool audio;
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bool receiver_only;
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Clock* clock;
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ReceiveStatistics* receive_statistics;
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Transport* outgoing_transport;
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RtcpIntraFrameObserver* intra_frame_callback;
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RtcpBandwidthObserver* bandwidth_callback;
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RtcpRttStats* rtt_stats;
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RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
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RtpAudioFeedback* audio_messages;
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RemoteBitrateEstimator* remote_bitrate_estimator;
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PacedSender* paced_sender;
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BitrateStatisticsObserver* send_bitrate_observer;
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FrameCountObserver* send_frame_count_observer;
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SendSideDelayObserver* send_side_delay_observer;
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};
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/*
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* Create a RTP/RTCP module object using the system clock.
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*
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* configuration - Configuration of the RTP/RTCP module.
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*/
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static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
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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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virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
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size_t incoming_packet_length) = 0;
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virtual void SetRemoteSSRC(uint32_t ssrc) = 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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* 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 int32_t SetMaxTransferUnit(uint16_t 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
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* 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 int32_t SetTransportOverhead(
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bool TCP,
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bool IPV6,
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uint8_t 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
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* 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 uint16_t 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
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* 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 uint16_t MaxDataPayloadLength() const = 0;
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/*
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* set codec name and payload type
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RegisterSendPayload(
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const CodecInst& voiceCodec) = 0;
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/*
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* set codec name and payload type
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RegisterSendPayload(
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const VideoCodec& videoCodec) = 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 int32_t DeRegisterSendPayload(int8_t payloadType) = 0;
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/*
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* (De)register RTP header extension type and id.
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) = 0;
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virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;
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/*
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* get start timestamp
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*/
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virtual uint32_t 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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virtual void SetStartTimestamp(uint32_t timestamp) = 0;
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/*
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* Get SequenceNumber
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*/
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virtual uint16_t 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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virtual void SetSequenceNumber(uint16_t seq) = 0;
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// Returns true if the ssrc matched this module, false otherwise.
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virtual bool SetRtpStateForSsrc(uint32_t ssrc,
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const RtpState& rtp_state) = 0;
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virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
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/*
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* Get SSRC
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*/
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virtual uint32_t 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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virtual void SetSSRC(uint32_t ssrc) = 0;
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/*
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* Set CSRC
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*
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* csrcs - vector of CSRCs
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*/
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virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
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/*
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* Turn on/off sending RTX (RFC 4588). The modes can be set as a combination
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* of values of the enumerator RtxMode.
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*/
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virtual void SetRtxSendStatus(int modes) = 0;
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/*
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* Get status of sending RTX (RFC 4588). The returned value can be
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* a combination of values of the enumerator RtxMode.
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*/
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virtual int RtxSendStatus() const = 0;
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// Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
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// only the SSRC is set.
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virtual void SetRtxSsrc(uint32_t ssrc) = 0;
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// Sets the payload type to use when sending RTX packets. Note that this
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// doesn't enable RTX, only the payload type is set.
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virtual void SetRtxSendPayloadType(int payload_type,
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int associated_payload_type) = 0;
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// Gets the payload type pair of (RTX, associated) to use when sending RTX
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// packets.
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virtual std::pair<int, int> RtxSendPayloadType() const = 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 int32_t SetSendingStatus(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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virtual void SetSendingMediaStatus(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 void BitrateSent(uint32_t* totalRate,
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uint32_t* videoRate,
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uint32_t* fecRate,
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uint32_t* nackRate) const = 0;
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/*
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* Used by the codec module to deliver a video or audio frame for
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* 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
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* 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 int32_t SendOutgoingData(
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FrameType frameType,
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int8_t payloadType,
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uint32_t timeStamp,
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int64_t capture_time_ms,
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const uint8_t* payloadData,
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size_t payloadSize,
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const RTPFragmentationHeader* fragmentation = NULL,
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const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
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virtual bool TimeToSendPacket(uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_time_ms,
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bool retransmission) = 0;
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virtual size_t TimeToSendPadding(size_t bytes) = 0;
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virtual bool GetSendSideDelay(int* avg_send_delay_ms,
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int* max_send_delay_ms) const = 0;
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// Called on generation of new statistics after an RTP send.
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virtual void RegisterSendChannelRtpStatisticsCallback(
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StreamDataCountersCallback* callback) = 0;
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virtual StreamDataCountersCallback*
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GetSendChannelRtpStatisticsCallback() const = 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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* Get RTCP status
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*/
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virtual RTCPMethod RTCP() const = 0;
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/*
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* configure RTCP status i.e on(compound or non- compound)/off
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*
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* method - RTCP method to use
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*/
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virtual void SetRTCPStatus(RTCPMethod method) = 0;
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/*
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* Set RTCP CName (i.e unique identifier)
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*
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* return -1 on failure else 0
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*/
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virtual int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]) = 0;
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/*
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* Get remote CName
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RemoteCNAME(uint32_t remoteSSRC,
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char cName[RTCP_CNAME_SIZE]) const = 0;
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/*
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* Get remote NTP
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RemoteNTP(
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uint32_t *ReceivedNTPsecs,
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uint32_t *ReceivedNTPfrac,
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uint32_t *RTCPArrivalTimeSecs,
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uint32_t *RTCPArrivalTimeFrac,
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uint32_t *rtcp_timestamp) const = 0;
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/*
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* AddMixedCNAME
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*
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* return -1 on failure else 0
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*/
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virtual int32_t AddMixedCNAME(uint32_t SSRC,
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const char cName[RTCP_CNAME_SIZE]) = 0;
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/*
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* RemoveMixedCNAME
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0;
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/*
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* Get RoundTripTime
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RTT(uint32_t remoteSSRC,
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int64_t* RTT,
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int64_t* avgRTT,
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int64_t* minRTT,
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int64_t* maxRTT) const = 0;
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/*
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* Force a send of a RTCP packet
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* periodic SR and RR are triggered via the process function
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*
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* return -1 on failure else 0
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*/
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virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0;
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/*
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* Force a send of a RTCP packet with more than one packet type.
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* periodic SR and RR are triggered via the process function
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*
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* return -1 on failure else 0
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*/
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virtual int32_t SendCompoundRTCP(
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const std::set<RTCPPacketType>& rtcpPacketTypes) = 0;
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/*
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* Good state of RTP receiver inform sender
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*/
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virtual int32_t SendRTCPReferencePictureSelection(
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const uint64_t pictureID) = 0;
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/*
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* Send a RTCP Slice Loss Indication (SLI)
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* 6 least significant bits of pictureID
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*/
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virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
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/*
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* Reset RTP data counters for the sending side
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*
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* return -1 on failure else 0
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*/
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virtual int32_t ResetSendDataCountersRTP() = 0;
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/*
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* Statistics of the amount of data sent
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*
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* return -1 on failure else 0
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*/
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virtual int32_t DataCountersRTP(
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size_t* bytesSent,
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uint32_t* packetsSent) const = 0;
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/*
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* Get send statistics for the RTP and RTX stream.
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*/
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virtual void GetSendStreamDataCounters(
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StreamDataCounters* rtp_counters,
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StreamDataCounters* rtx_counters) const = 0;
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/*
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* Get received RTCP sender info
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
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/*
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* Get received RTCP report block
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*
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* return -1 on failure else 0
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*/
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virtual int32_t RemoteRTCPStat(
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std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
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/*
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* (APP) Application specific data
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*
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* return -1 on failure else 0
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*/
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virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType,
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uint32_t name,
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const uint8_t* data,
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uint16_t length) = 0;
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/*
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* (XR) VOIP metric
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*
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* return -1 on failure else 0
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*/
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virtual int32_t SetRTCPVoIPMetrics(
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const RTCPVoIPMetric* VoIPMetric) = 0;
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/*
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* (XR) Receiver Reference Time Report
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*/
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virtual void SetRtcpXrRrtrStatus(bool enable) = 0;
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virtual bool RtcpXrRrtrStatus() const = 0;
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/*
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* (REMB) Receiver Estimated Max Bitrate
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*/
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virtual bool REMB() const = 0;
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virtual void SetREMBStatus(bool enable) = 0;
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virtual void SetREMBData(uint32_t bitrate,
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const std::vector<uint32_t>& ssrcs) = 0;
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/*
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* (IJ) Extended jitter report.
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*/
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virtual bool IJ() const = 0;
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virtual void SetIJStatus(bool enable) = 0;
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/*
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* (TMMBR) Temporary Max Media Bit Rate
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*/
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virtual bool TMMBR() const = 0;
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virtual void SetTMMBRStatus(bool enable) = 0;
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/*
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* (NACK)
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*/
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/*
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* TODO(holmer): Propagate this API to VideoEngine.
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* Returns the currently configured selective retransmission settings.
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*/
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virtual int SelectiveRetransmissions() const = 0;
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/*
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* TODO(holmer): Propagate this API to VideoEngine.
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* Sets the selective retransmission settings, which will decide which
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* packets will be retransmitted if NACKed. Settings are constructed by
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* combining the constants in enum RetransmissionMode with bitwise OR.
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* All packets are retransmitted if kRetransmitAllPackets is set, while no
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* packets are retransmitted if kRetransmitOff is set.
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* By default all packets except FEC packets are retransmitted. For VP8
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* with temporal scalability only base layer packets are retransmitted.
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*
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* Returns -1 on failure, otherwise 0.
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*/
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virtual int SetSelectiveRetransmissions(uint8_t settings) = 0;
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/*
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* Send a Negative acknowledgement packet
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*
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* return -1 on failure else 0
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*/
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virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0;
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/*
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* Store the sent packets, needed to answer to a Negative acknowledgement
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* requests
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*/
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virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
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// Returns true if the module is configured to store packets.
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virtual bool StorePackets() const = 0;
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// Called on receipt of RTCP report block from remote side.
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virtual void RegisterRtcpStatisticsCallback(
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RtcpStatisticsCallback* callback) = 0;
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virtual RtcpStatisticsCallback*
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GetRtcpStatisticsCallback() = 0;
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/**************************************************************************
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*
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* Audio
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*
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***************************************************************************/
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/*
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* set audio packet size, used to determine when it's time to send a DTMF
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* packet in silence (CNG)
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*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0;
|
|
|
|
/*
|
|
* Send a TelephoneEvent tone using RFC 2833 (4733)
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual int32_t SendTelephoneEventOutband(uint8_t key,
|
|
uint16_t time_ms,
|
|
uint8_t level) = 0;
|
|
|
|
/*
|
|
* Set payload type for Redundant Audio Data RFC 2198
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0;
|
|
|
|
/*
|
|
* Get payload type for Redundant Audio Data RFC 2198
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual int32_t SendREDPayloadType(
|
|
int8_t& payloadType) 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 int32_t SetAudioLevel(uint8_t level_dBov) = 0;
|
|
|
|
/**************************************************************************
|
|
*
|
|
* Video
|
|
*
|
|
***************************************************************************/
|
|
|
|
/*
|
|
* Set the target send bitrate
|
|
*/
|
|
virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0;
|
|
|
|
/*
|
|
* Turn on/off generic FEC
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual int32_t SetGenericFECStatus(bool enable,
|
|
uint8_t payloadTypeRED,
|
|
uint8_t payloadTypeFEC) = 0;
|
|
|
|
/*
|
|
* Get generic FEC setting
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual int32_t GenericFECStatus(bool& enable,
|
|
uint8_t& payloadTypeRED,
|
|
uint8_t& payloadTypeFEC) = 0;
|
|
|
|
|
|
virtual int32_t SetFecParameters(
|
|
const FecProtectionParams* delta_params,
|
|
const FecProtectionParams* key_params) = 0;
|
|
|
|
/*
|
|
* Set method for requestion a new key frame
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
|
|
|
|
/*
|
|
* send a request for a keyframe
|
|
*
|
|
* return -1 on failure else 0
|
|
*/
|
|
virtual int32_t RequestKeyFrame() = 0;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
|