298 lines
11 KiB
C++
298 lines
11 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/rtp_transport.h"
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#include <errno.h>
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#include <string>
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#include <utility>
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#include "api/rtp_headers.h"
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#include "api/rtp_parameters.h"
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#include "media/base/rtp_utils.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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void RtpTransport::SetRtcpMuxEnabled(bool enable) {
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rtcp_mux_enabled_ = enable;
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MaybeSignalReadyToSend();
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}
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const std::string& RtpTransport::transport_name() const {
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return rtp_packet_transport_->transport_name();
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}
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int RtpTransport::SetRtpOption(rtc::Socket::Option opt, int value) {
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return rtp_packet_transport_->SetOption(opt, value);
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}
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int RtpTransport::SetRtcpOption(rtc::Socket::Option opt, int value) {
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if (rtcp_packet_transport_) {
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return rtcp_packet_transport_->SetOption(opt, value);
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}
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return -1;
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}
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void RtpTransport::SetRtpPacketTransport(
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rtc::PacketTransportInternal* new_packet_transport) {
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if (new_packet_transport == rtp_packet_transport_) {
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return;
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}
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if (rtp_packet_transport_) {
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rtp_packet_transport_->SignalReadyToSend.disconnect(this);
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rtp_packet_transport_->SignalReadPacket.disconnect(this);
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rtp_packet_transport_->SignalNetworkRouteChanged.disconnect(this);
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rtp_packet_transport_->SignalWritableState.disconnect(this);
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rtp_packet_transport_->SignalSentPacket.disconnect(this);
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// Reset the network route of the old transport.
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SignalNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
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}
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if (new_packet_transport) {
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new_packet_transport->SignalReadyToSend.connect(
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this, &RtpTransport::OnReadyToSend);
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new_packet_transport->SignalReadPacket.connect(this,
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&RtpTransport::OnReadPacket);
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new_packet_transport->SignalNetworkRouteChanged.connect(
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this, &RtpTransport::OnNetworkRouteChanged);
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new_packet_transport->SignalWritableState.connect(
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this, &RtpTransport::OnWritableState);
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new_packet_transport->SignalSentPacket.connect(this,
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&RtpTransport::OnSentPacket);
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// Set the network route for the new transport.
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SignalNetworkRouteChanged(new_packet_transport->network_route());
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}
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rtp_packet_transport_ = new_packet_transport;
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// Assumes the transport is ready to send if it is writable. If we are wrong,
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// ready to send will be updated the next time we try to send.
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SetReadyToSend(false,
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rtp_packet_transport_ && rtp_packet_transport_->writable());
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}
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void RtpTransport::SetRtcpPacketTransport(
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rtc::PacketTransportInternal* new_packet_transport) {
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if (new_packet_transport == rtcp_packet_transport_) {
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return;
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}
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if (rtcp_packet_transport_) {
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rtcp_packet_transport_->SignalReadyToSend.disconnect(this);
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rtcp_packet_transport_->SignalReadPacket.disconnect(this);
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rtcp_packet_transport_->SignalNetworkRouteChanged.disconnect(this);
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rtcp_packet_transport_->SignalWritableState.disconnect(this);
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rtcp_packet_transport_->SignalSentPacket.disconnect(this);
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// Reset the network route of the old transport.
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SignalNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
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}
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if (new_packet_transport) {
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new_packet_transport->SignalReadyToSend.connect(
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this, &RtpTransport::OnReadyToSend);
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new_packet_transport->SignalReadPacket.connect(this,
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&RtpTransport::OnReadPacket);
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new_packet_transport->SignalNetworkRouteChanged.connect(
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this, &RtpTransport::OnNetworkRouteChanged);
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new_packet_transport->SignalWritableState.connect(
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this, &RtpTransport::OnWritableState);
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new_packet_transport->SignalSentPacket.connect(this,
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&RtpTransport::OnSentPacket);
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// Set the network route for the new transport.
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SignalNetworkRouteChanged(new_packet_transport->network_route());
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}
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rtcp_packet_transport_ = new_packet_transport;
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// Assumes the transport is ready to send if it is writable. If we are wrong,
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// ready to send will be updated the next time we try to send.
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SetReadyToSend(true,
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rtcp_packet_transport_ && rtcp_packet_transport_->writable());
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}
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bool RtpTransport::IsWritable(bool rtcp) const {
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rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
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? rtcp_packet_transport_
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: rtp_packet_transport_;
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return transport && transport->writable();
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}
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bool RtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) {
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return SendPacket(false, packet, options, flags);
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}
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bool RtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) {
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return SendPacket(true, packet, options, flags);
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}
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bool RtpTransport::SendPacket(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) {
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rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
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? rtcp_packet_transport_
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: rtp_packet_transport_;
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int ret = transport->SendPacket(packet->cdata<char>(), packet->size(),
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options, flags);
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if (ret != static_cast<int>(packet->size())) {
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if (transport->GetError() == ENOTCONN) {
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RTC_LOG(LS_WARNING) << "Got ENOTCONN from transport.";
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SetReadyToSend(rtcp, false);
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}
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return false;
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}
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return true;
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}
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void RtpTransport::UpdateRtpHeaderExtensionMap(
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const cricket::RtpHeaderExtensions& header_extensions) {
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header_extension_map_ = RtpHeaderExtensionMap(header_extensions);
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}
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bool RtpTransport::RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
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RtpPacketSinkInterface* sink) {
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rtp_demuxer_.RemoveSink(sink);
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if (!rtp_demuxer_.AddSink(criteria, sink)) {
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RTC_LOG(LS_ERROR) << "Failed to register the sink for RTP demuxer.";
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return false;
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}
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return true;
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}
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bool RtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) {
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if (!rtp_demuxer_.RemoveSink(sink)) {
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RTC_LOG(LS_ERROR) << "Failed to unregister the sink for RTP demuxer.";
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return false;
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}
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return true;
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}
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void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) {
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webrtc::RtpPacketReceived parsed_packet(&header_extension_map_);
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if (!parsed_packet.Parse(std::move(packet))) {
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RTC_LOG(LS_ERROR)
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<< "Failed to parse the incoming RTP packet before demuxing. Drop it.";
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return;
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}
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if (packet_time_us != -1) {
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parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
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}
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if (!rtp_demuxer_.OnRtpPacket(parsed_packet)) {
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RTC_LOG(LS_WARNING) << "Failed to demux RTP packet: "
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<< RtpDemuxer::DescribePacket(parsed_packet);
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uint32_t ssrc = parsed_packet.Ssrc();
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OnErrorDemuxingPacket(ssrc);
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}
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}
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void RtpTransport::OnErrorDemuxingPacket(uint32_t ssrc) {
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}
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bool RtpTransport::IsTransportWritable() {
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auto rtcp_packet_transport =
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rtcp_mux_enabled_ ? nullptr : rtcp_packet_transport_;
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return rtp_packet_transport_ && rtp_packet_transport_->writable() &&
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(!rtcp_packet_transport || rtcp_packet_transport->writable());
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}
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void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
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SetReadyToSend(transport == rtcp_packet_transport_, true);
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}
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void RtpTransport::OnNetworkRouteChanged(
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absl::optional<rtc::NetworkRoute> network_route) {
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SignalNetworkRouteChanged(network_route);
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}
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void RtpTransport::OnWritableState(
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rtc::PacketTransportInternal* packet_transport) {
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RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
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packet_transport == rtcp_packet_transport_);
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SignalWritableState(IsTransportWritable());
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}
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void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
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const rtc::SentPacket& sent_packet) {
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RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
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packet_transport == rtcp_packet_transport_);
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SignalSentPacket(sent_packet);
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}
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void RtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) {
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DemuxPacket(packet, packet_time_us);
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}
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void RtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) {
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SignalRtcpPacketReceived(&packet, packet_time_us);
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}
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void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
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const char* data,
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size_t len,
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const int64_t& packet_time_us,
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int flags) {
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TRACE_EVENT0("webrtc", "RtpTransport::OnReadPacket");
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// When using RTCP multiplexing we might get RTCP packets on the RTP
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// transport. We check the RTP payload type to determine if it is RTCP.
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auto array_view = rtc::MakeArrayView(data, len);
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cricket::RtpPacketType packet_type = cricket::InferRtpPacketType(array_view);
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// Filter out the packet that is neither RTP nor RTCP.
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if (packet_type == cricket::RtpPacketType::kUnknown) {
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return;
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}
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// Protect ourselves against crazy data.
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if (!cricket::IsValidRtpPacketSize(packet_type, len)) {
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RTC_LOG(LS_ERROR) << "Dropping incoming "
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<< cricket::RtpPacketTypeToString(packet_type)
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<< " packet: wrong size=" << len;
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return;
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}
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rtc::CopyOnWriteBuffer packet(data, len);
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if (packet_type == cricket::RtpPacketType::kRtcp) {
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OnRtcpPacketReceived(std::move(packet), packet_time_us);
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} else {
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OnRtpPacketReceived(std::move(packet), packet_time_us);
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}
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}
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void RtpTransport::SetReadyToSend(bool rtcp, bool ready) {
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if (rtcp) {
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rtcp_ready_to_send_ = ready;
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} else {
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rtp_ready_to_send_ = ready;
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}
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MaybeSignalReadyToSend();
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}
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void RtpTransport::MaybeSignalReadyToSend() {
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bool ready_to_send =
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rtp_ready_to_send_ && (rtcp_ready_to_send_ || rtcp_mux_enabled_);
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if (ready_to_send != ready_to_send_) {
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ready_to_send_ = ready_to_send;
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SignalReadyToSend(ready_to_send);
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}
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}
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} // namespace webrtc
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