| Index: media/cast/net/rtcp/sender_rtcp_session.h
|
| diff --git a/media/cast/net/rtcp/rtcp.h b/media/cast/net/rtcp/sender_rtcp_session.h
|
| similarity index 36%
|
| rename from media/cast/net/rtcp/rtcp.h
|
| rename to media/cast/net/rtcp/sender_rtcp_session.h
|
| index 96f5147e5123b3f0d8042ed585cddc8b404a2c7b..f22dd589c681d707d74c4569b90a650295b032b5 100644
|
| --- a/media/cast/net/rtcp/rtcp.h
|
| +++ b/media/cast/net/rtcp/sender_rtcp_session.h
|
| @@ -1,179 +1,114 @@
|
| -// Copyright 2014 The Chromium Authors. All rights reserved.
|
| +// 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.
|
|
|
| -// This class maintains a bi-directional RTCP connection with a remote
|
| -// peer.
|
| -
|
| -#ifndef MEDIA_CAST_RTCP_RTCP_H_
|
| -#define MEDIA_CAST_RTCP_RTCP_H_
|
| -
|
| -#include <stddef.h>
|
| -#include <stdint.h>
|
| +#ifndef MEDIA_CAST_NET_RTCP_SENDER_RTCP_SESSION_H_
|
| +#define MEDIA_CAST_NET_RTCP_SENDER_RTCP_SESSION_H_
|
|
|
| #include <map>
|
| #include <queue>
|
| -#include <string>
|
| +#include <utility>
|
|
|
| -#include "base/macros.h"
|
| -#include "base/memory/scoped_ptr.h"
|
| -#include "base/memory/weak_ptr.h"
|
| -#include "base/time/tick_clock.h"
|
| +#include "base/containers/hash_tables.h"
|
| #include "base/time/time.h"
|
| -#include "media/cast/common/clock_drift_smoother.h"
|
| -#include "media/cast/net/cast_transport_defines.h"
|
| -#include "media/cast/net/cast_transport_sender.h"
|
| -#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
|
| -#include "media/cast/net/rtcp/rtcp_builder.h"
|
| +#include "media/cast/net/pacing/paced_sender.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 LocalRtcpReceiverFeedback;
|
| -class PacedPacketSender;
|
| -class RtcpReceiver;
|
| -class RtcpBuilder;
|
| -
|
| typedef std::pair<uint32_t, base::TimeTicks> RtcpSendTimePair;
|
| typedef std::map<uint32_t, base::TimeTicks> RtcpSendTimeMap;
|
| typedef std::queue<RtcpSendTimePair> RtcpSendTimeQueue;
|
|
|
| -// TODO(hclam): This should be renamed to RtcpSession.
|
| -class Rtcp {
|
| +// This class represents a RTCP session on a RTP sender. It provides an
|
| +// interface to send RTCP sender report (SR). RTCP SR packets allow
|
| +// receiver to maintain clock offsets and synchronize between audio and video.
|
| +//
|
| +// RTCP session on sender handles the following incoming RTCP reports
|
| +// from receiver:
|
| +// - Receiver reference time report: Helps with tracking largest timestamp
|
| +// seen and as a result rejecting old RTCP reports.
|
| +// - Receiver logs: The sender receives log events from the receiver and
|
| +// invokes a callback passed.
|
| +// - cast message: Receives feedback from receiver on missing packets/frames
|
| +// and last frame id received and invokes a callback passed.
|
| +// - Last report: The receiver provides feedback on delay since last report
|
| +// received which helps it compute round trip time and invoke a callback.
|
| +class SenderRtcpSession : public RtcpSession {
|
| public:
|
| - Rtcp(const RtcpCastMessageCallback& cast_callback,
|
| - const RtcpRttCallback& rtt_callback,
|
| - const RtcpLogMessageCallback& log_callback,
|
| - base::TickClock* clock, // Not owned.
|
| - PacedPacketSender* packet_sender, // Not owned.
|
| - uint32_t local_ssrc,
|
| - uint32_t remote_ssrc);
|
| -
|
| - virtual ~Rtcp();
|
| -
|
| - // Send a RTCP sender report.
|
| - // |current_time| is the current time reported by a tick clock.
|
| - // |current_time_as_rtp_timestamp| is the corresponding RTP timestamp.
|
| - // |send_packet_count| is the number of packets sent.
|
| - // |send_octet_count| is the number of octets sent.
|
| - void SendRtcpFromRtpSender(base::TimeTicks current_time,
|
| - RtpTimeTicks current_time_as_rtp_timestamp,
|
| - uint32_t send_packet_count,
|
| - size_t send_octet_count);
|
| -
|
| - // This function is meant to be used in conjunction with
|
| - // SendRtcpFromRtpReceiver.
|
| - // |now| is converted to NTP and saved internally for
|
| - // future round-trip/lip-sync calculations.
|
| - // This is done in a separate method so that SendRtcpFromRtpReceiver can
|
| - // be done on a separate (temporary) RTCP object.
|
| - RtcpTimeData ConvertToNTPAndSave(base::TimeTicks now);
|
| -
|
| - // |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 SendRtcpFromRtpReceiver(
|
| - RtcpTimeData time_data,
|
| - const RtcpCastMessage* cast_message,
|
| - base::TimeDelta target_delay,
|
| - const ReceiverRtcpEventSubscriber::RtcpEvents* rtcp_events,
|
| - const RtpReceiverStatistics* rtp_receiver_statistics) const;
|
| -
|
| - // Submit a received packet to this object. The packet will be parsed
|
| - // and used to maintain a RTCP session.
|
| - // Returns false if this is not a RTCP packet or it is not directed to
|
| - // this session, e.g. SSRC doesn't match.
|
| - bool IncomingRtcpPacket(const uint8_t* data, size_t length);
|
| + SenderRtcpSession(const RtcpCastMessageCallback& cast_callback,
|
| + const RtcpRttCallback& rtt_callback,
|
| + const RtcpLogMessageCallback& log_callback,
|
| + base::TickClock* clock, // Not owned.
|
| + PacedPacketSender* packet_sender, // Not owned.
|
| + uint32_t local_ssrc,
|
| + uint32_t remote_ssrc);
|
|
|
| - // 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;
|
| -
|
| - void OnReceivedReceiverLog(const RtcpReceiverLogMessage& receiver_log);
|
| + ~SenderRtcpSession() override;
|
|
|
| // If greater than zero, this is the last measured network round trip time.
|
| base::TimeDelta current_round_trip_time() const {
|
| return current_round_trip_time_;
|
| }
|
|
|
| - static bool IsRtcpPacket(const uint8_t* packet, size_t length);
|
| - static uint32_t GetSsrcOfSender(const uint8_t* rtcp_buffer, size_t length);
|
| -
|
| - uint32_t GetLocalSsrc() const { return local_ssrc_; }
|
| - uint32_t GetRemoteSsrc() const { return remote_ssrc_; }
|
| + // Send a RTCP sender report.
|
| + // |current_time| is the current time reported by a tick clock.
|
| + // |current_time_as_rtp_timestamp| is the corresponding RTP timestamp.
|
| + // |send_packet_count| is the number of packets sent.
|
| + // |send_octet_count| is the number of octets sent.
|
| + void SendRtcpReport(base::TimeTicks current_time,
|
| + RtpTimeTicks current_time_as_rtp_timestamp,
|
| + uint32_t send_packet_count,
|
| + size_t send_octet_count);
|
|
|
| - protected:
|
| - void OnReceivedNtp(uint32_t ntp_seconds, uint32_t ntp_fraction);
|
| - void OnReceivedLipSyncInfo(RtpTimeTicks rtp_timestamp,
|
| - uint32_t ntp_seconds,
|
| - uint32_t ntp_fraction);
|
| + // 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_t* data, size_t length) override;
|
|
|
| private:
|
| + // Received last report information from RTP receiver which helps compute
|
| + // round trip time.
|
| void OnReceivedDelaySinceLastReport(uint32_t last_report,
|
| uint32_t delay_since_last_report);
|
|
|
| + // Received logs from RTP receiver.
|
| + void OnReceivedReceiverLog(const RtcpReceiverLogMessage& receiver_log);
|
| +
|
| + // Received cast message containing details of missing packets/frames and
|
| + // last frame id received.
|
| void OnReceivedCastFeedback(const RtcpCastMessage& cast_message);
|
|
|
| + // Remove duplicate events in |receiver_log|.
|
| + // Returns true if any events remain.
|
| + bool DedupeReceiverLog(RtcpReceiverLogMessage* receiver_log);
|
| +
|
| + // Save last sent NTP time on RTPC SR. This helps map the sent time when a
|
| + // last report is received from RTP receiver to compute RTT.
|
| void SaveLastSentNtpTime(const base::TimeTicks& now,
|
| uint32_t last_ntp_seconds,
|
| uint32_t last_ntp_fraction);
|
|
|
| - // Remove duplicate events in |receiver_log|.
|
| - // Returns true if any events remain.
|
| - bool DedupeReceiverLog(RtcpReceiverLogMessage* receiver_log);
|
| + base::TickClock* const clock_; // Not owned.
|
| + PacedPacketSender* packet_sender_; // Not owned.
|
| + const uint32_t local_ssrc_;
|
| + const uint32_t remote_ssrc_;
|
|
|
| const RtcpCastMessageCallback cast_callback_;
|
| const RtcpRttCallback rtt_callback_;
|
| const RtcpLogMessageCallback log_callback_;
|
| - base::TickClock* const clock_; // Not owned by this class.
|
| - RtcpBuilder rtcp_builder_;
|
| - PacedPacketSender* packet_sender_; // Not owned.
|
| - const uint32_t local_ssrc_;
|
| - const uint32_t remote_ssrc_;
|
| +
|
| + // Computed from RTCP RRTR report.
|
| + base::TimeTicks largest_seen_timestamp_;
|
|
|
| // The RTCP packet parser is re-used when parsing each RTCP packet. It
|
| // remembers state about prior RTP timestamps and other sequence values to
|
| // re-construct "expanded" values.
|
| RtcpParser parser_;
|
|
|
| - RtcpSendTimeMap last_reports_sent_map_;
|
| - RtcpSendTimeQueue last_reports_sent_queue_;
|
| -
|
| - // 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_| is not "null."
|
| - 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_;
|
| -
|
| - // The last measured network round trip time. This is updated with each
|
| - // sender report --> receiver report round trip. If this is zero, then the
|
| - // round trip time has not been measured yet.
|
| - base::TimeDelta current_round_trip_time_;
|
| -
|
| - base::TimeTicks largest_seen_timestamp_;
|
| -
|
| // For extending received ACK frame IDs from 8-bit to 32-bit.
|
| FrameIdWrapHelper ack_frame_id_wrap_helper_;
|
|
|
| @@ -182,10 +117,20 @@ class Rtcp {
|
| base::hash_set<ReceiverEventKey> receiver_event_key_set_;
|
| std::queue<ReceiverEventKey> receiver_event_key_queue_;
|
|
|
| - DISALLOW_COPY_AND_ASSIGN(Rtcp);
|
| + // The last measured network round trip time. This is updated with each
|
| + // sender report --> receiver report round trip. If this is zero, then the
|
| + // round trip time has not been measured yet.
|
| + base::TimeDelta current_round_trip_time_;
|
| +
|
| + // Map of NTP timestamp to local time that helps with RTT calculation
|
| + // when last report is received from RTP receiver.
|
| + RtcpSendTimeMap last_reports_sent_map_;
|
| + RtcpSendTimeQueue last_reports_sent_queue_;
|
| +
|
| + DISALLOW_COPY_AND_ASSIGN(SenderRtcpSession);
|
| };
|
|
|
| } // namespace cast
|
| } // namespace media
|
|
|
| -#endif // MEDIA_CAST_RTCP_RTCP_H_
|
| +#endif // MEDIA_CAST_NET_RTCP_SENDER_RTCP_SESSION_H_
|
|
|