rtp_transport.cc | Explore in Territory

/*
 *  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 "pc/rtp_transport.h"

#include <errno.h>

#include <cstdint>
#include <utility>

#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "media/base/rtp_utils.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"

namespace webrtc {

void RtpTransport::SetRtcpMuxEnabled(bool enable) { … }

const std::string& RtpTransport::transport_name() const { … }

int RtpTransport::SetRtpOption(rtc::Socket::Option opt, int value) { … }

int RtpTransport::SetRtcpOption(rtc::Socket::Option opt, int value) { … }

void RtpTransport::SetRtpPacketTransport(
    rtc::PacketTransportInternal* new_packet_transport) { … }

void RtpTransport::SetRtcpPacketTransport(
    rtc::PacketTransportInternal* new_packet_transport) { … }

bool RtpTransport::IsWritable(bool rtcp) const { … }

bool RtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
                                 const rtc::PacketOptions& options,
                                 int flags) { … }

bool RtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
                                  const rtc::PacketOptions& options,
                                  int flags) { … }

bool RtpTransport::SendPacket(bool rtcp,
                              rtc::CopyOnWriteBuffer* packet,
                              const rtc::PacketOptions& options,
                              int flags) { … }

void RtpTransport::UpdateRtpHeaderExtensionMap(
    const cricket::RtpHeaderExtensions& header_extensions) { … }

bool RtpTransport::RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
                                          RtpPacketSinkInterface* sink) { … }

bool RtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) { … }

flat_set<uint32_t> RtpTransport::GetSsrcsForSink(RtpPacketSinkInterface* sink) { … }

void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet,
                               webrtc::Timestamp arrival_time,
                               rtc::EcnMarking ecn) { … }

bool RtpTransport::IsTransportWritable() { … }

void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) { … }

void RtpTransport::OnNetworkRouteChanged(
    absl::optional<rtc::NetworkRoute> network_route) { … }

void RtpTransport::OnWritableState(
    rtc::PacketTransportInternal* packet_transport) { … }

void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
                                const rtc::SentPacket& sent_packet) { … }

void RtpTransport::OnRtpPacketReceived(
    const rtc::ReceivedPacket& received_packet) { … }

void RtpTransport::OnRtcpPacketReceived(
    const rtc::ReceivedPacket& received_packet) { … }

void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
                                const rtc::ReceivedPacket& received_packet) { … }

void RtpTransport::SetReadyToSend(bool rtcp, bool ready) { … }

void RtpTransport::MaybeSignalReadyToSend() { … }

}  // namespace webrtc
chromium/third_party/webrtc/pc/rtp_transport.cc
