blob: 1afcb5ee3d1a6088c0008fa2b822494e839b4399 [file] [log] [blame]
* Copyright 2017 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <stddef.h>
#include <stdint.h>
#include <string>
#include "absl/types/optional.h"
#include "call/rtp_demuxer.h"
#include "call/video_receive_stream.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "p2p/base/packet_transport_internal.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"
namespace rtc {
class CopyOnWriteBuffer;
struct PacketOptions;
class PacketTransportInternal;
} // namespace rtc
namespace webrtc {
class RtpTransport : public RtpTransportInternal {
RtpTransport(const RtpTransport&) = delete;
RtpTransport& operator=(const RtpTransport&) = delete;
explicit RtpTransport(bool rtcp_mux_enabled)
: rtcp_mux_enabled_(rtcp_mux_enabled) {}
bool rtcp_mux_enabled() const override { return rtcp_mux_enabled_; }
void SetRtcpMuxEnabled(bool enable) override;
const std::string& transport_name() const override;
int SetRtpOption(rtc::Socket::Option opt, int value) override;
int SetRtcpOption(rtc::Socket::Option opt, int value) override;
rtc::PacketTransportInternal* rtp_packet_transport() const {
return rtp_packet_transport_;
void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp);
rtc::PacketTransportInternal* rtcp_packet_transport() const {
return rtcp_packet_transport_;
void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp);
bool IsReadyToSend() const override { return ready_to_send_; }
bool IsWritable(bool rtcp) const override;
bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) override;
bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) override;
bool IsSrtpActive() const override { return false; }
void UpdateRtpHeaderExtensionMap(
const cricket::RtpHeaderExtensions& header_extensions) override;
bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
RtpPacketSinkInterface* sink) override;
bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override;
// These methods will be used in the subclasses.
void DemuxPacket(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us);
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags);
// Overridden by SrtpTransport.
virtual void OnNetworkRouteChanged(
absl::optional<rtc::NetworkRoute> network_route);
virtual void OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us);
virtual void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us);
// Overridden by SrtpTransport and DtlsSrtpTransport.
virtual void OnWritableState(rtc::PacketTransportInternal* packet_transport);
void OnReadyToSend(rtc::PacketTransportInternal* transport);
void OnSentPacket(rtc::PacketTransportInternal* packet_transport,
const rtc::SentPacket& sent_packet);
void OnReadPacket(rtc::PacketTransportInternal* transport,
const char* data,
size_t len,
const int64_t& packet_time_us,
int flags);
// Updates "ready to send" for an individual channel and fires
// SignalReadyToSend.
void SetReadyToSend(bool rtcp, bool ready);
void MaybeSignalReadyToSend();
bool IsTransportWritable();
bool rtcp_mux_enabled_;
rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr;
bool ready_to_send_ = false;
bool rtp_ready_to_send_ = false;
bool rtcp_ready_to_send_ = false;
RtpDemuxer rtp_demuxer_;
// Used for identifying the MID for RtpDemuxer.
RtpHeaderExtensionMap header_extension_map_;
} // namespace webrtc