1 // Copyright 2015 The Chromium Authors. All rights reserved. 2 // Use of this source code is governed by a BSD-style license that can be 3 // found in the LICENSE file. 4 5 #ifndef MEDIA_CAST_NET_RTCP_RECEIVER_RTCP_SESSION_H_ 6 #define MEDIA_CAST_NET_RTCP_RECEIVER_RTCP_SESSION_H_ 7 8 #include "base/time/tick_clock.h" 9 #include "media/cast/common/clock_drift_smoother.h" 10 #include "media/cast/net/pacing/paced_sender.h" 11 #include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h" 12 #include "media/cast/net/rtcp/receiver_rtcp_session.h" 13 #include "media/cast/net/rtcp/rtcp_defines.h" 14 #include "media/cast/net/rtcp/rtcp_session.h" 15 #include "media/cast/net/rtcp/rtcp_utility.h" 16 17 namespace media { 18 namespace cast { 19 20 // The RTCP session on a receiver handles incoming RTCP SR packets and maintains 21 // the offset of local clock from the remote clock and lip sync info (RTP 22 // & NTP timestamps). 23 class ReceiverRtcpSession : public RtcpSession { 24 public: 25 ReceiverRtcpSession(const base::TickClock* clock, // Not owned. 26 uint32_t local_ssrc, 27 uint32_t remote_ssrc); 28 29 ~ReceiverRtcpSession() override; 30 local_ssrc()31 uint32_t local_ssrc() const { return local_ssrc_; } remote_ssrc()32 uint32_t remote_ssrc() const { return remote_ssrc_; } 33 34 // Handle incoming RTCP packet. 35 // Returns false if it is not a RTCP packet or it is not directed to 36 // this session, e.g. SSRC doesn't match. 37 bool IncomingRtcpPacket(const uint8_t* data, size_t length) override; 38 39 // If available, returns true and sets the output arguments to the latest 40 // lip-sync timestamps gleaned from the sender reports. While the sender 41 // provides reference NTP times relative to its own wall clock, the 42 // |reference_time| returned here has been translated to the local 43 // CastEnvironment clock. 44 bool GetLatestLipSyncTimes(RtpTimeTicks* rtp_timestamp, 45 base::TimeTicks* reference_time) const; 46 last_report_truncated_ntp()47 uint32_t last_report_truncated_ntp() const { 48 return last_report_truncated_ntp_; 49 } 50 time_last_report_received()51 base::TimeTicks time_last_report_received() const { 52 return time_last_report_received_; 53 } 54 55 private: 56 // Received NTP timestamps from RTCP SR packets. 57 void OnReceivedNtp(uint32_t ntp_seconds, uint32_t ntp_fraction); 58 59 // Received RTP and NTP timestamps from RTCP SR packets. 60 void OnReceivedLipSyncInfo(RtpTimeTicks rtp_timestamp, 61 uint32_t ntp_seconds, 62 uint32_t ntp_fraction); 63 64 const base::TickClock* const clock_; // Not owned. 65 const uint32_t local_ssrc_; 66 const uint32_t remote_ssrc_; 67 68 // The truncated (i.e., 64-->32-bit) NTP timestamp provided in the last report 69 // from the remote peer, along with the local time at which the report was 70 // received. These values are used for ping-pong'ing NTP timestamps between 71 // the peers so that they can estimate the network's round-trip time. 72 uint32_t last_report_truncated_ntp_; 73 base::TimeTicks time_last_report_received_; 74 75 // Maintains a smoothed offset between the local clock and the remote clock. 76 // Calling this member's Current() method is only valid if 77 // |time_last_report_received_| has a valid value. 78 ClockDriftSmoother local_clock_ahead_by_; 79 80 // Latest "lip sync" info from the sender. The sender provides the RTP 81 // timestamp of some frame of its choosing and also a corresponding reference 82 // NTP timestamp sampled from a clock common to all media streams. It is 83 // expected that the sender will update this data regularly and in a timely 84 // manner (e.g., about once per second). 85 RtpTimeTicks lip_sync_rtp_timestamp_; 86 uint64_t lip_sync_ntp_timestamp_; 87 88 // The RTCP packet parser is re-used when parsing each RTCP packet. It 89 // remembers state about prior RTP timestamps and other sequence values to 90 // re-construct "expanded" values. 91 RtcpParser parser_; 92 93 DISALLOW_COPY_AND_ASSIGN(ReceiverRtcpSession); 94 }; 95 96 } // namespace cast 97 } // namespace media 98 99 #endif // MEDIA_CAST_NET_RTCP_RECEIVER_RTCP_SESSION_H_ 100