1d0fa5d352
R=mflodman@webrtc.org, stefan@webrtc.org BUG=1667,1788 Review URL: https://webrtc-codereview.appspot.com/37489004 Cr-Commit-Position: refs/heads/master@{#8429} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
419 lines
14 KiB
C++
419 lines
14 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_SOURCE_RTP_RTCP_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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#include <list>
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#include <vector>
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/test/testsupport/gtest_prod_util.h"
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namespace webrtc {
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class ModuleRtpRtcpImpl : public RtpRtcp {
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public:
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explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
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virtual ~ModuleRtpRtcpImpl();
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// Returns the number of milliseconds until the module want a worker thread to
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// call Process.
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virtual int64_t TimeUntilNextProcess() OVERRIDE;
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// Process any pending tasks such as timeouts.
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virtual int32_t Process() OVERRIDE;
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// Receiver part.
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// Called when we receive an RTCP packet.
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virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
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size_t incoming_packet_length) OVERRIDE;
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virtual void SetRemoteSSRC(uint32_t ssrc) OVERRIDE;
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// Sender part.
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virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) OVERRIDE;
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virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) OVERRIDE;
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virtual int32_t DeRegisterSendPayload(int8_t payload_type) OVERRIDE;
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int8_t SendPayloadType() const;
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// Register RTP header extension.
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virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) OVERRIDE;
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virtual int32_t DeregisterSendRtpHeaderExtension(
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RTPExtensionType type) OVERRIDE;
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// Get start timestamp.
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virtual uint32_t StartTimestamp() const OVERRIDE;
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// Configure start timestamp, default is a random number.
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virtual void SetStartTimestamp(uint32_t timestamp) OVERRIDE;
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virtual uint16_t SequenceNumber() const OVERRIDE;
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// Set SequenceNumber, default is a random number.
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virtual void SetSequenceNumber(uint16_t seq) OVERRIDE;
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virtual void SetRtpStateForSsrc(uint32_t ssrc,
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const RtpState& rtp_state) OVERRIDE;
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virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) OVERRIDE;
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virtual uint32_t SSRC() const OVERRIDE;
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// Configure SSRC, default is a random number.
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virtual void SetSSRC(uint32_t ssrc) OVERRIDE;
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virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) OVERRIDE;
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RTCPSender::FeedbackState GetFeedbackState();
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int CurrentSendFrequencyHz() const;
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virtual void SetRtxSendStatus(int mode) OVERRIDE;
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virtual int RtxSendStatus() const OVERRIDE;
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virtual void SetRtxSsrc(uint32_t ssrc) OVERRIDE;
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virtual void SetRtxSendPayloadType(int payload_type) OVERRIDE;
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// Sends kRtcpByeCode when going from true to false.
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virtual int32_t SetSendingStatus(bool sending) OVERRIDE;
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virtual bool Sending() const OVERRIDE;
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// Drops or relays media packets.
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virtual void SetSendingMediaStatus(bool sending) OVERRIDE;
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virtual bool SendingMedia() const OVERRIDE;
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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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virtual int32_t SendOutgoingData(
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FrameType frame_type,
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int8_t payload_type,
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uint32_t time_stamp,
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int64_t capture_time_ms,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation = NULL,
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const RTPVideoHeader* rtp_video_hdr = NULL) OVERRIDE;
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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) OVERRIDE;
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// Returns the number of padding bytes actually sent, which can be more or
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// less than |bytes|.
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virtual size_t TimeToSendPadding(size_t bytes) OVERRIDE;
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virtual bool GetSendSideDelay(int* avg_send_delay_ms,
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int* max_send_delay_ms) const OVERRIDE;
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// RTCP part.
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// Get RTCP status.
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virtual RTCPMethod RTCP() const OVERRIDE;
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// Configure RTCP status i.e on/off.
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virtual void SetRTCPStatus(RTCPMethod method) OVERRIDE;
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// Set RTCP CName.
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virtual int32_t SetCNAME(const char c_name[RTCP_CNAME_SIZE]) OVERRIDE;
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// Get remote CName.
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virtual int32_t RemoteCNAME(uint32_t remote_ssrc,
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char c_name[RTCP_CNAME_SIZE]) const OVERRIDE;
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// Get remote NTP.
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virtual int32_t RemoteNTP(uint32_t* received_ntp_secs,
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uint32_t* received_ntp_frac,
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uint32_t* rtcp_arrival_time_secs,
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uint32_t* rtcp_arrival_time_frac,
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uint32_t* rtcp_timestamp) const OVERRIDE;
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virtual int32_t AddMixedCNAME(uint32_t ssrc,
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const char c_name[RTCP_CNAME_SIZE]) OVERRIDE;
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virtual int32_t RemoveMixedCNAME(uint32_t ssrc) OVERRIDE;
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// Get RoundTripTime.
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virtual int32_t RTT(uint32_t remote_ssrc,
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int64_t* rtt,
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int64_t* avg_rtt,
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int64_t* min_rtt,
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int64_t* max_rtt) const OVERRIDE;
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// Force a send of an RTCP packet.
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// Normal SR and RR are triggered via the process function.
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virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport) OVERRIDE;
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virtual int32_t ResetSendDataCountersRTP() OVERRIDE;
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// Statistics of the amount of data sent and received.
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virtual int32_t DataCountersRTP(size_t* bytes_sent,
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uint32_t* packets_sent) const OVERRIDE;
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virtual void GetSendStreamDataCounters(
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StreamDataCounters* rtp_counters,
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StreamDataCounters* rtx_counters) const OVERRIDE;
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// Get received RTCP report, sender info.
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virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) OVERRIDE;
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// Get received RTCP report, report block.
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virtual int32_t RemoteRTCPStat(
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std::vector<RTCPReportBlock>* receive_blocks) const OVERRIDE;
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// Set received RTCP report block.
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virtual int32_t AddRTCPReportBlock(
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uint32_t ssrc,
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const RTCPReportBlock* receive_block) OVERRIDE;
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virtual int32_t RemoveRTCPReportBlock(uint32_t ssrc) OVERRIDE;
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// (REMB) Receiver Estimated Max Bitrate.
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virtual bool REMB() const OVERRIDE;
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virtual void SetREMBStatus(bool enable) OVERRIDE;
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virtual void SetREMBData(uint32_t bitrate,
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const std::vector<uint32_t>& ssrcs) OVERRIDE;
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// (IJ) Extended jitter report.
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virtual bool IJ() const OVERRIDE;
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virtual void SetIJStatus(bool enable) OVERRIDE;
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// (TMMBR) Temporary Max Media Bit Rate.
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virtual bool TMMBR() const OVERRIDE;
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virtual void SetTMMBRStatus(bool enable) OVERRIDE;
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int32_t SetTMMBN(const TMMBRSet* bounding_set);
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virtual uint16_t MaxPayloadLength() const OVERRIDE;
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virtual uint16_t MaxDataPayloadLength() const OVERRIDE;
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virtual int32_t SetMaxTransferUnit(uint16_t size) OVERRIDE;
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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 authentication_overhead = 0) OVERRIDE;
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// (NACK) Negative acknowledgment part.
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virtual int SelectiveRetransmissions() const OVERRIDE;
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virtual int SetSelectiveRetransmissions(uint8_t settings) OVERRIDE;
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// Send a Negative acknowledgment packet.
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virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) OVERRIDE;
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// Store the sent packets, needed to answer to a negative acknowledgment
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// requests.
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virtual void SetStorePacketsStatus(bool enable,
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uint16_t number_to_store) OVERRIDE;
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virtual bool StorePackets() const OVERRIDE;
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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) OVERRIDE;
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virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() OVERRIDE;
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// (APP) Application specific data.
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virtual int32_t SetRTCPApplicationSpecificData(uint8_t sub_type,
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uint32_t name,
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const uint8_t* data,
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uint16_t length) OVERRIDE;
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// (XR) VOIP metric.
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virtual int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) OVERRIDE;
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// (XR) Receiver reference time report.
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virtual void SetRtcpXrRrtrStatus(bool enable) OVERRIDE;
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virtual bool RtcpXrRrtrStatus() const OVERRIDE;
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// Audio part.
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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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virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) OVERRIDE;
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// Send a TelephoneEvent tone using RFC 2833 (4733).
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virtual int32_t SendTelephoneEventOutband(uint8_t key,
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uint16_t time_ms,
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uint8_t level) OVERRIDE;
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// Set payload type for Redundant Audio Data RFC 2198.
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virtual int32_t SetSendREDPayloadType(int8_t payload_type) OVERRIDE;
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// Get payload type for Redundant Audio Data RFC 2198.
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virtual int32_t SendREDPayloadType(int8_t& payload_type) const OVERRIDE;
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// Store the audio level in d_bov for header-extension-for-audio-level-
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// indication.
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virtual int32_t SetAudioLevel(uint8_t level_d_bov) OVERRIDE;
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// Video part.
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virtual int32_t SendRTCPSliceLossIndication(uint8_t picture_id) OVERRIDE;
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// Set method for requesting a new key frame.
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virtual int32_t SetKeyFrameRequestMethod(
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KeyFrameRequestMethod method) OVERRIDE;
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// Send a request for a keyframe.
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virtual int32_t RequestKeyFrame() OVERRIDE;
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virtual void SetTargetSendBitrate(
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const std::vector<uint32_t>& stream_bitrates) OVERRIDE;
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virtual int32_t SetGenericFECStatus(bool enable,
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uint8_t payload_type_red,
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uint8_t payload_type_fec) OVERRIDE;
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virtual int32_t GenericFECStatus(
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bool& enable,
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uint8_t& payload_type_red,
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uint8_t& payload_type_fec) OVERRIDE;
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virtual int32_t SetFecParameters(
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const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params) OVERRIDE;
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bool LastReceivedNTP(uint32_t* NTPsecs,
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uint32_t* NTPfrac,
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uint32_t* remote_sr) const;
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bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
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virtual int32_t BoundingSet(bool& tmmbr_owner, TMMBRSet*& bounding_set_rec);
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virtual void BitrateSent(uint32_t* total_rate,
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uint32_t* video_rate,
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uint32_t* fec_rate,
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uint32_t* nackRate) const OVERRIDE;
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int64_t SendTimeOfSendReport(uint32_t send_report);
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bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
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// Good state of RTP receiver inform sender.
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virtual int32_t SendRTCPReferencePictureSelection(
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uint64_t picture_id) OVERRIDE;
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virtual void RegisterSendChannelRtpStatisticsCallback(
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StreamDataCountersCallback* callback) OVERRIDE;
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virtual StreamDataCountersCallback*
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GetSendChannelRtpStatisticsCallback() const OVERRIDE;
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void OnReceivedTMMBR();
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// Bad state of RTP receiver request a keyframe.
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void OnRequestIntraFrame();
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// Received a request for a new SLI.
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void OnReceivedSliceLossIndication(uint8_t picture_id);
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// Received a new reference frame.
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void OnReceivedReferencePictureSelectionIndication(uint64_t picture_id);
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void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
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void OnRequestSendReport();
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protected:
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void RegisterChildModule(RtpRtcp* module);
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void DeRegisterChildModule(RtpRtcp* module);
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bool UpdateRTCPReceiveInformationTimers();
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uint32_t BitrateReceivedNow() const;
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// Get remote SequenceNumber.
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uint16_t RemoteSequenceNumber() const;
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// Only for internal testing.
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uint32_t LastSendReport(int64_t& last_rtcptime);
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RTPSender rtp_sender_;
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RTCPSender rtcp_sender_;
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RTCPReceiver rtcp_receiver_;
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Clock* clock_;
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private:
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FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
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FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
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int64_t RtcpReportInterval();
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void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
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void set_rtt_ms(int64_t rtt_ms);
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int64_t rtt_ms() const;
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bool TimeToSendFullNackList(int64_t now) const;
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bool IsDefaultModule() const;
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int32_t id_;
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const bool audio_;
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bool collision_detected_;
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int64_t last_process_time_;
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int64_t last_bitrate_process_time_;
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int64_t last_rtt_process_time_;
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uint16_t packet_overhead_;
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scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_;
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scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_feedback_;
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ModuleRtpRtcpImpl* default_module_;
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std::vector<ModuleRtpRtcpImpl*> child_modules_;
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size_t padding_index_;
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// Send side
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NACKMethod nack_method_;
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int64_t nack_last_time_sent_full_;
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uint32_t nack_last_time_sent_full_prev_;
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uint16_t nack_last_seq_number_sent_;
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bool simulcast_;
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VideoCodec send_video_codec_;
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KeyFrameRequestMethod key_frame_req_method_;
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RemoteBitrateEstimator* remote_bitrate_;
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RtcpRttStats* rtt_stats_;
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// The processed RTT from RtcpRttStats.
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scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
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int64_t rtt_ms_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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