Index: media/cast/net/rtcp/receiver_rtcp_session.h |
diff --git a/media/cast/net/rtcp/receiver_rtcp_session.h b/media/cast/net/rtcp/receiver_rtcp_session.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..e52a9cf5fe1ae1ecb54fb8b2c9d51f95600da520 |
--- /dev/null |
+++ b/media/cast/net/rtcp/receiver_rtcp_session.h |
@@ -0,0 +1,97 @@ |
+// 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
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_ |