Files
platform-external-webrtc/webrtc/ortc/rtptransportadapter.cc
deadbeef e814a0dee0 Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver

They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:

* You can only have one of each type of sender and receiver (audio/video) on top
  of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.

Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:

ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine

And later we hope to have simply:

PeerConnection -> "Real" ORTC objects -> Media engine

See the linked bug for more context.

BUG=webrtc:7013
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-26 02:15:09 +00:00

130 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.
*/
#include "webrtc/ortc/rtptransportadapter.h"
#include <algorithm> // For std::find.
#include <set>
#include <sstream>
#include <utility> // For std::move.
#include "webrtc/api/proxy.h"
#include "webrtc/base/logging.h"
namespace webrtc {
BEGIN_OWNED_PROXY_MAP(RtpTransport)
PROXY_SIGNALING_THREAD_DESTRUCTOR()
PROXY_CONSTMETHOD0(PacketTransportInterface*, GetRtpPacketTransport)
PROXY_CONSTMETHOD0(PacketTransportInterface*, GetRtcpPacketTransport)
PROXY_METHOD1(RTCError, SetRtcpParameters, const RtcpParameters&)
PROXY_CONSTMETHOD0(RtcpParameters, GetRtcpParameters)
protected:
RtpTransportAdapter* GetInternal() override {
return internal();
}
END_PROXY_MAP()
// static
RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
RtpTransportAdapter::CreateProxied(
const RtcpParameters& rtcp_parameters,
PacketTransportInterface* rtp,
PacketTransportInterface* rtcp,
RtpTransportControllerAdapter* rtp_transport_controller) {
if (!rtp) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Must provide an RTP packet transport.");
}
if (!rtcp_parameters.mux && !rtcp) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Must provide an RTCP packet transport when RTCP muxing is not used.");
}
if (rtcp_parameters.mux && rtcp) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Creating an RtpTransport with RTCP muxing enabled, "
"with a separate RTCP packet transport?");
}
if (!rtp_transport_controller) {
// Since OrtcFactory::CreateRtpTransport creates an RtpTransportController
// automatically when one isn't passed in, this should never be reached.
RTC_NOTREACHED();
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Must provide an RTP transport controller.");
}
return RtpTransportProxyWithInternal<RtpTransportAdapter>::Create(
rtp_transport_controller->signaling_thread(),
rtp_transport_controller->worker_thread(),
std::unique_ptr<RtpTransportAdapter>(new RtpTransportAdapter(
rtcp_parameters, rtp, rtcp, rtp_transport_controller)));
}
void RtpTransportAdapter::TakeOwnershipOfRtpTransportController(
std::unique_ptr<RtpTransportControllerInterface> controller) {
RTC_DCHECK_EQ(rtp_transport_controller_, controller->GetInternal());
RTC_DCHECK(owned_rtp_transport_controller_.get() == nullptr);
owned_rtp_transport_controller_ = std::move(controller);
}
RtpTransportAdapter::RtpTransportAdapter(
const RtcpParameters& rtcp_parameters,
PacketTransportInterface* rtp,
PacketTransportInterface* rtcp,
RtpTransportControllerAdapter* rtp_transport_controller)
: rtp_packet_transport_(rtp),
rtcp_packet_transport_(rtcp),
rtp_transport_controller_(rtp_transport_controller),
rtcp_parameters_(rtcp_parameters) {
RTC_DCHECK(rtp_transport_controller);
// CNAME should have been filled by OrtcFactory if empty.
RTC_DCHECK(!rtcp_parameters_.cname.empty());
}
RtpTransportAdapter::~RtpTransportAdapter() {
SignalDestroyed(this);
}
PacketTransportInterface* RtpTransportAdapter::GetRtpPacketTransport() const {
return rtp_packet_transport_;
}
PacketTransportInterface* RtpTransportAdapter::GetRtcpPacketTransport() const {
return rtcp_packet_transport_;
}
RTCError RtpTransportAdapter::SetRtcpParameters(
const RtcpParameters& parameters) {
if (!parameters.mux && rtcp_parameters_.mux) {
LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::INVALID_STATE,
"Can't disable RTCP muxing after enabling.");
}
if (!parameters.cname.empty() && parameters.cname != rtcp_parameters_.cname) {
LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
"Changing the RTCP CNAME is currently unsupported.");
}
// If the CNAME is empty, use the existing one.
RtcpParameters copy = parameters;
if (copy.cname.empty()) {
copy.cname = rtcp_parameters_.cname;
}
RTCError err = rtp_transport_controller_->SetRtcpParameters(copy, this);
if (!err.ok()) {
return err;
}
rtcp_parameters_ = copy;
if (rtcp_parameters_.mux) {
rtcp_packet_transport_ = nullptr;
}
return RTCError::OK();
}
} // namespace webrtc