blob: 2ee27e02fb673725ec3c66ab9260d723d6083dda [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 "webrtc/pc/rtptransport.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/copyonwritebuffer.h"
#include "webrtc/p2p/base/packettransportinterface.h"
namespace webrtc {
void RtpTransport::SetRtcpMuxEnabled(bool enable) {
rtcp_mux_enabled_ = enable;
MaybeSignalReadyToSend();
}
void RtpTransport::SetRtpPacketTransport(
rtc::PacketTransportInternal* new_packet_transport) {
if (new_packet_transport == rtp_packet_transport_) {
return;
}
if (rtp_packet_transport_) {
rtp_packet_transport_->SignalReadyToSend.disconnect(this);
}
if (new_packet_transport) {
new_packet_transport->SignalReadyToSend.connect(
this, &RtpTransport::OnReadyToSend);
}
rtp_packet_transport_ = new_packet_transport;
// Assumes the transport is ready to send if it is writable. If we are wrong,
// ready to send will be updated the next time we try to send.
SetReadyToSend(false,
rtp_packet_transport_ && rtp_packet_transport_->writable());
}
void RtpTransport::SetRtcpPacketTransport(
rtc::PacketTransportInternal* new_packet_transport) {
if (new_packet_transport == rtcp_packet_transport_) {
return;
}
if (rtcp_packet_transport_) {
rtcp_packet_transport_->SignalReadyToSend.disconnect(this);
}
if (new_packet_transport) {
new_packet_transport->SignalReadyToSend.connect(
this, &RtpTransport::OnReadyToSend);
}
rtcp_packet_transport_ = new_packet_transport;
// Assumes the transport is ready to send if it is writable. If we are wrong,
// ready to send will be updated the next time we try to send.
SetReadyToSend(true,
rtcp_packet_transport_ && rtcp_packet_transport_->writable());
}
bool RtpTransport::IsWritable(bool rtcp) const {
rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
? rtcp_packet_transport_
: rtp_packet_transport_;
return transport && transport->writable();
}
bool RtpTransport::SendPacket(bool rtcp,
const rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
? rtcp_packet_transport_
: rtp_packet_transport_;
int ret = transport->SendPacket(packet->data<char>(), packet->size(), options,
flags);
if (ret != static_cast<int>(packet->size())) {
if (transport->GetError() == ENOTCONN) {
LOG(LS_WARNING) << "Got ENOTCONN from transport.";
SetReadyToSend(rtcp, false);
}
return false;
}
return true;
}
PacketTransportInterface* RtpTransport::GetRtpPacketTransport() const {
return rtp_packet_transport_;
}
PacketTransportInterface* RtpTransport::GetRtcpPacketTransport() const {
return rtcp_packet_transport_;
}
RTCError RtpTransport::SetRtcpParameters(const RtcpParameters& parameters) {
if (rtcp_parameters_.mux && !parameters.mux) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"Disabling RTCP muxing is not allowed.");
}
RtcpParameters new_parameters = parameters;
if (new_parameters.cname.empty()) {
new_parameters.cname = rtcp_parameters_.cname;
}
rtcp_parameters_ = new_parameters;
return RTCError::OK();
}
RtcpParameters RtpTransport::GetRtcpParameters() const {
return rtcp_parameters_;
}
RtpTransportAdapter* RtpTransport::GetInternal() {
return nullptr;
}
void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
SetReadyToSend(transport == rtcp_packet_transport_, true);
}
void RtpTransport::SetReadyToSend(bool rtcp, bool ready) {
if (rtcp) {
rtcp_ready_to_send_ = ready;
} else {
rtp_ready_to_send_ = ready;
}
MaybeSignalReadyToSend();
}
void RtpTransport::MaybeSignalReadyToSend() {
bool ready_to_send =
rtp_ready_to_send_ && (rtcp_ready_to_send_ || rtcp_mux_enabled_);
if (ready_to_send != ready_to_send_) {
ready_to_send_ = ready_to_send;
SignalReadyToSend(ready_to_send);
}
}
} // namespace webrtc