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// Copyright 2014 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 transport for audio and video in a Cast Streaming
// session.
// Audio, video frames and RTCP messages are submitted to this object
// and then packetized and paced to the underlying UDP socket.
// The hierarchy of send transport in a Cast Streaming session:
// CastTransport RTP RTCP
// ------------------------------------------------------------------
// TransportEncryptionHandler (A/V)
// RtpSender (A/V) Rtcp (A/V)
// PacedSender (Shared)
// UdpTransport (Shared)
// There are objects of TransportEncryptionHandler, RtpSender and Rtcp
// for each audio and video stream.
// PacedSender and UdpTransport are shared between all RTP and RTCP
// streams.
#include <stdint.h>
#include <memory>
#include <set>
#include <vector>
#include "base/callback.h"
#include "base/gtest_prod_util.h"
#include "base/macros.h"
#include "base/memory/ref_counted.h"
#include "base/memory/weak_ptr.h"
#include "base/time/tick_clock.h"
#include "base/time/time.h"
#include "media/cast/common/transport_encryption_handler.h"
#include "media/cast/logging/logging_defines.h"
#include "media/cast/net/cast_transport.h"
#include "media/cast/net/cast_transport_config.h"
#include "media/cast/net/pacing/paced_sender.h"
#include "media/cast/net/rtcp/rtcp_builder.h"
#include "media/cast/net/rtcp/sender_rtcp_session.h"
#include "media/cast/net/rtp/rtp_parser.h"
#include "media/cast/net/rtp/rtp_sender.h"
#include "net/base/network_interfaces.h"
namespace media {
namespace cast {
class CastTransportImpl final : public CastTransport {
const base::TickClock* clock, // Owned by the caller.
base::TimeDelta logging_flush_interval,
std::unique_ptr<Client> client,
std::unique_ptr<PacketTransport> transport,
const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner);
~CastTransportImpl() final;
// CastTransport implementation for sending.
void InitializeStream(const CastTransportRtpConfig& config,
std::unique_ptr<RtcpObserver> rtcp_observer) final;
void InsertFrame(uint32_t ssrc, const EncodedFrame& frame) final;
void SendSenderReport(uint32_t ssrc,
base::TimeTicks current_time,
RtpTimeTicks current_time_as_rtp_timestamp) final;
void CancelSendingFrames(uint32_t ssrc,
const std::vector<FrameId>& frame_ids) final;
void ResendFrameForKickstart(uint32_t ssrc, FrameId frame_id) final;
PacketReceiverCallback PacketReceiverForTesting() final;
// Possible keys of |options| handled here are:
// "pacer_target_burst_size": int
// - Specifies how many packets to send per 10 ms ideally.
// "pacer_max_burst_size": int
// - Specifies how many pakcets to send per 10 ms, maximum.
// "send_buffer_min_size": int
// - Specifies the minimum socket send buffer size.
// "disable_wifi_scan" (value ignored)
// - Disable wifi scans while streaming.
// "media_streaming_mode" (value ignored)
// - Turn media streaming mode on.
// Note, these options may be ignored on some platforms.
void SetOptions(const base::DictionaryValue& options) final;
// CastTransport implementation for receiving.
void AddValidRtpReceiver(uint32_t rtp_sender_ssrc,
uint32_t rtp_receiver_ssrc) final;
// Building and sending RTCP packet from RTP receiver implementation.
void InitializeRtpReceiverRtcpBuilder(uint32_t rtp_receiver_ssrc,
const RtcpTimeData& time_data) final;
void AddCastFeedback(const RtcpCastMessage& cast_message,
base::TimeDelta target_delay) final;
void AddPli(const RtcpPliMessage& pli_message) final;
void AddRtcpEvents(
const ReceiverRtcpEventSubscriber::RtcpEvents& rtcp_events) final;
void AddRtpReceiverReport(
const RtcpReportBlock& rtp_receiver_report_block) final;
void SendRtcpFromRtpReceiver() final;
// Handle received RTCP messages on RTP sender.
class RtcpClient;
struct RtpStreamSession;
FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, NacksCancelRetransmits);
FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, CancelRetransmits);
FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, Kickstart);
FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, DedupRetransmissionWithAudio);
// Resend packets for the stream identified by |ssrc|.
// If |cancel_rtx_if_not_in_list| is true then transmission of packets for the
// frames but not in the list will be dropped.
// See PacedSender::ResendPackets() to see how |dedup_info| works.
void ResendPackets(uint32_t ssrc,
const MissingFramesAndPacketsMap& missing_packets,
bool cancel_rtx_if_not_in_list,
const DedupInfo& dedup_info);
// If |logging_flush_interval| is set, this is called at approximate periodic
// intervals.
void SendRawEvents();
// Called when a packet is received.
bool OnReceivedPacket(std::unique_ptr<Packet> packet);
// Called when a log message is received.
void OnReceivedLogMessage(EventMediaType media_type,
const RtcpReceiverLogMessage& log);
// Called when a RTCP Cast message is received.
void OnReceivedCastMessage(uint32_t ssrc,
const RtcpCastMessage& cast_message);
const base::TickClock* const clock_; // Not owned by this class.
const base::TimeDelta logging_flush_interval_;
const std::unique_ptr<Client> transport_client_;
const std::unique_ptr<PacketTransport> transport_;
const scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
// FrameEvents and PacketEvents pending delivery via raw events callback.
// Do not add elements to these when |logging_flush_interval| is
// |base::TimeDelta()|.
std::vector<FrameEvent> recent_frame_events_;
std::vector<PacketEvent> recent_packet_events_;
// Packet sender that performs pacing.
PacedSender pacer_;
// Right after a frame is sent we record the number of bytes sent to the
// socket. We record the corresponding bytes sent for the most recent ACKed
// audio packet.
int64_t last_byte_acked_for_audio_;
// Packets that don't match these sender ssrcs are ignored.
std::set<uint32_t> valid_sender_ssrcs_;
// While non-null, global WiFi behavior modifications are in effect. This is
// used, for example, to turn off WiFi scanning that tends to interfere with
// the reliability of UDP packet transmission.
std::unique_ptr<net::ScopedWifiOptions> wifi_options_autoreset_;
// Do not initialize the |rtcp_builder_at_rtp_receiver_| if the RTP receiver
// SSRC does not match these ssrcs. Only RTP receiver needs to register its
// SSRC in this set.
std::set<uint32_t> valid_rtp_receiver_ssrcs_;
std::unique_ptr<RtcpBuilder> rtcp_builder_at_rtp_receiver_;
// Records the initialized stream sessions on RTP sender. The sender SSRC is
// used as key since it is unique for each RTP stream.
using SessionMap = std::map<uint32_t, std::unique_ptr<RtpStreamSession>>;
SessionMap sessions_;
base::WeakPtrFactory<CastTransportImpl> weak_factory_{this};
} // namespace cast
} // namespace media