mirror of
https://github.com/DrKLO/Telegram.git
synced 2024-12-23 06:50:36 +01:00
144 lines
4.6 KiB
C++
144 lines
4.6 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef PC_RTP_TRANSPORT_H_
|
|
#define PC_RTP_TRANSPORT_H_
|
|
|
|
#include <stddef.h>
|
|
#include <stdint.h>
|
|
|
|
#include <string>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "call/rtp_demuxer.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"
|
|
#include "rtc_base/third_party/sigslot/sigslot.h"
|
|
|
|
namespace rtc {
|
|
|
|
class CopyOnWriteBuffer;
|
|
struct PacketOptions;
|
|
class PacketTransportInternal;
|
|
|
|
} // namespace rtc
|
|
|
|
namespace webrtc {
|
|
|
|
class RtpTransport : public RtpTransportInternal {
|
|
public:
|
|
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;
|
|
|
|
protected:
|
|
// 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);
|
|
|
|
private:
|
|
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
|
|
|
|
#endif // PC_RTP_TRANSPORT_H_
|