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}
169 lines
5.4 KiB
C++
169 lines
5.4 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 "webrtc/ortc/ortcrtpreceiveradapter.h"
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#include <utility>
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#include "webrtc/base/checks.h"
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#include "webrtc/base/helpers.h" // For "CreateRandomX".
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#include "webrtc/media/base/mediaconstants.h"
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#include "webrtc/ortc/rtptransportadapter.h"
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namespace {
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void FillAudioReceiverParameters(webrtc::RtpParameters* parameters) {
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for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
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if (!codec.num_channels) {
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codec.num_channels = rtc::Optional<int>(1);
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}
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}
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}
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void FillVideoReceiverParameters(webrtc::RtpParameters* parameters) {
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for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
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if (!codec.clock_rate) {
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codec.clock_rate = rtc::Optional<int>(cricket::kVideoCodecClockrate);
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}
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}
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}
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} // namespace
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namespace webrtc {
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BEGIN_OWNED_PROXY_MAP(OrtcRtpReceiver)
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PROXY_SIGNALING_THREAD_DESTRUCTOR()
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, GetTrack)
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PROXY_METHOD1(RTCError, SetTransport, RtpTransportInterface*)
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PROXY_CONSTMETHOD0(RtpTransportInterface*, GetTransport)
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PROXY_METHOD1(RTCError, Receive, const RtpParameters&)
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PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
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PROXY_CONSTMETHOD0(cricket::MediaType, GetKind)
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END_PROXY_MAP()
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// static
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std::unique_ptr<OrtcRtpReceiverInterface> OrtcRtpReceiverAdapter::CreateProxy(
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std::unique_ptr<OrtcRtpReceiverAdapter> wrapped_receiver) {
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RTC_DCHECK(wrapped_receiver);
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rtc::Thread* signaling =
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wrapped_receiver->rtp_transport_controller_->signaling_thread();
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rtc::Thread* worker =
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wrapped_receiver->rtp_transport_controller_->worker_thread();
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return OrtcRtpReceiverProxy::Create(signaling, worker,
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std::move(wrapped_receiver));
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}
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OrtcRtpReceiverAdapter::~OrtcRtpReceiverAdapter() {
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internal_receiver_ = nullptr;
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SignalDestroyed();
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}
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rtc::scoped_refptr<MediaStreamTrackInterface> OrtcRtpReceiverAdapter::GetTrack()
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const {
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return internal_receiver_ ? internal_receiver_->track() : nullptr;
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}
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RTCError OrtcRtpReceiverAdapter::SetTransport(
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RtpTransportInterface* transport) {
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LOG_AND_RETURN_ERROR(
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RTCErrorType::UNSUPPORTED_OPERATION,
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"Changing the transport of an RtpReceiver is not yet supported.");
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}
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RtpTransportInterface* OrtcRtpReceiverAdapter::GetTransport() const {
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return transport_;
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}
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RTCError OrtcRtpReceiverAdapter::Receive(const RtpParameters& parameters) {
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RtpParameters filled_parameters = parameters;
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RTCError err;
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switch (kind_) {
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case cricket::MEDIA_TYPE_AUDIO:
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FillAudioReceiverParameters(&filled_parameters);
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err = rtp_transport_controller_->ValidateAndApplyAudioReceiverParameters(
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filled_parameters);
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if (!err.ok()) {
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return err;
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}
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break;
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case cricket::MEDIA_TYPE_VIDEO:
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FillVideoReceiverParameters(&filled_parameters);
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err = rtp_transport_controller_->ValidateAndApplyVideoReceiverParameters(
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filled_parameters);
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if (!err.ok()) {
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return err;
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}
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break;
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case cricket::MEDIA_TYPE_DATA:
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RTC_NOTREACHED();
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return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
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}
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last_applied_parameters_ = filled_parameters;
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// Now that parameters were applied, can create (or recreate) the internal
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// receiver.
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//
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// This is analogous to a PeerConnection creating a receiver after
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// SetRemoteDescription is successful.
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MaybeRecreateInternalReceiver();
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return RTCError::OK();
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}
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RtpParameters OrtcRtpReceiverAdapter::GetParameters() const {
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return last_applied_parameters_;
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}
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cricket::MediaType OrtcRtpReceiverAdapter::GetKind() const {
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return kind_;
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}
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OrtcRtpReceiverAdapter::OrtcRtpReceiverAdapter(
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cricket::MediaType kind,
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RtpTransportInterface* transport,
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RtpTransportControllerAdapter* rtp_transport_controller)
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: kind_(kind),
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transport_(transport),
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rtp_transport_controller_(rtp_transport_controller) {}
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void OrtcRtpReceiverAdapter::MaybeRecreateInternalReceiver() {
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if (last_applied_parameters_.encodings.empty()) {
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internal_receiver_ = nullptr;
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return;
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}
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// An SSRC of 0 is valid; this is used to identify "the default SSRC" (which
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// is the first one seen by the underlying media engine).
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uint32_t ssrc = 0;
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if (last_applied_parameters_.encodings[0].ssrc) {
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ssrc = *last_applied_parameters_.encodings[0].ssrc;
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}
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if (internal_receiver_ && ssrc == internal_receiver_->ssrc()) {
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// SSRC not changing; nothing to do.
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return;
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}
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internal_receiver_ = nullptr;
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switch (kind_) {
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case cricket::MEDIA_TYPE_AUDIO:
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internal_receiver_ =
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new AudioRtpReceiver(rtc::CreateRandomUuid(), ssrc,
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rtp_transport_controller_->voice_channel());
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break;
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case cricket::MEDIA_TYPE_VIDEO:
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internal_receiver_ = new VideoRtpReceiver(
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rtc::CreateRandomUuid(), rtp_transport_controller_->worker_thread(),
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ssrc, rtp_transport_controller_->video_channel());
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break;
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case cricket::MEDIA_TYPE_DATA:
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RTC_NOTREACHED();
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}
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}
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} // namespace webrtc
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