cast: Split Rtcp into two for sender and receiver

media/cast/net/rtcp/receiver_rtcp_session.h

+// Copyright 2015 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef MEDIA_CAST_NET_RTCP_RECEIVER_RTCP_SESSION_H_
+#define MEDIA_CAST_NET_RTCP_RECEIVER_RTCP_SESSION_H_
+
+#include "base/time/tick_clock.h"
+#include "media/cast/common/clock_drift_smoother.h"
+#include "media/cast/net/pacing/paced_sender.h"
+#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
+#include "media/cast/net/rtcp/receiver_rtcp_session.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
+#include "media/cast/net/rtcp/rtcp_session.h"
+#include "media/cast/net/rtcp/rtcp_utility.h"
+
+namespace media {
+namespace cast {
+
+class ReceiverRtcpSession : public RtcpSession {

[xjz, 2015/12/11 18:02:47]

Where is RtcpSession defined? Did you forget "rtcp_session.h"?

[Irfan, 2015/12/11 18:42:56]

Thanks for catching. Added now.

+ public:
+  ReceiverRtcpSession(base::TickClock* clock,            // Not owned.
+                      PacedPacketSender* packet_sender,  // Not owned.
+                      uint32_t local_ssrc,
+                      uint32_t remote_ssrc);
+
+  ~ReceiverRtcpSession() override;
+
+  uint32_t local_ssrc() const { return local_ssrc_; }
+  uint32_t remote_ssrc() const { return remote_ssrc_; }
+
+  // |cast_message|, |rtcp_events| and |rtp_receiver_statistics| are optional;
+  // if |cast_message| is provided the RTCP receiver report will append a Cast
+  // message containing Acks and Nacks; |target_delay| is sent together with
+  // |cast_message|. If |rtcp_events| is provided the RTCP receiver report will
+  // append the log messages.
+  void SendRtcpReport(
+      RtcpTimeData time_data,
+      const RtcpCastMessage* cast_message,
+      base::TimeDelta target_delay,
+      const ReceiverRtcpEventSubscriber::RtcpEvents* rtcp_events,
+      const RtpReceiverStatistics* rtp_receiver_statistics) const;
+
+  // Handle incoming RTCP packet.
+  // Returns false if it is not a RTCP packet or it is not directed to
+  // this session, e.g. SSRC doesn't match.
+  bool IncomingRtcpPacket(const uint8* data, size_t length) override;
+
+  // If available, returns true and sets the output arguments to the latest
+  // lip-sync timestamps gleaned from the sender reports.  While the sender
+  // provides reference NTP times relative to its own wall clock, the
+  // |reference_time| returned here has been translated to the local
+  // CastEnvironment clock.
+  bool GetLatestLipSyncTimes(RtpTimeTicks* rtp_timestamp,
+                             base::TimeTicks* reference_time) const;
+
+ private:
+  void OnReceivedNtp(uint32_t ntp_seconds, uint32_t ntp_fraction);
+
+  void OnReceivedLipSyncInfo(RtpTimeTicks rtp_timestamp,
+                             uint32_t ntp_seconds,
+                             uint32_t ntp_fraction);
+
+  base::TickClock* const clock_;      // Not owned.
+  PacedPacketSender* packet_sender_;  // Not owned.
+  const uint32_t local_ssrc_;
+  const uint32_t remote_ssrc_;
+
+  // The truncated (i.e., 64-->32-bit) NTP timestamp provided in the last report
+  // from the remote peer, along with the local time at which the report was
+  // received.  These values are used for ping-pong'ing NTP timestamps between
+  // the peers so that they can estimate the network's round-trip time.
+  uint32_t last_report_truncated_ntp_;
+  base::TimeTicks time_last_report_received_;
+
+  // Maintains a smoothed offset between the local clock and the remote clock.
+  // Calling this member's Current() method is only valid if
+  // |time_last_report_received_| has a valid value.
+  ClockDriftSmoother local_clock_ahead_by_;
+
+  // Latest "lip sync" info from the sender.  The sender provides the RTP
+  // timestamp of some frame of its choosing and also a corresponding reference
+  // NTP timestamp sampled from a clock common to all media streams.  It is
+  // expected that the sender will update this data regularly and in a timely
+  // manner (e.g., about once per second).
+  RtpTimeTicks lip_sync_rtp_timestamp_;
+  uint64_t lip_sync_ntp_timestamp_;
+
+  RtcpParser parser_;
+
+  DISALLOW_COPY_AND_ASSIGN(ReceiverRtcpSession);
+};
+
+}  // namespace cast
+}  // namespace media
+
+#endif  // MEDIA_CAST_NET_RTCP_RECEIVER_RTCP_SESSION_H_