This makes the receiver know about its associated set of streams, the equivalent of the [[AssociatedRemoteMediaStreams]] slot in the spec, https://w3c.github.io/webrtc-pc/#dfn-x%5B%5Bassociatedremotemediastreams%5D%5D This does not change layers below peerconnection.cc. The streams are set upon the receiver's construction and is not modified for the duration of its lifetime. When we support modifying the associated set of streams of a receiver the receiver needs to know about it. The receiver's streams() should be used in all places where a receiver's streams need to be known. Bug: webrtc:8473 Change-Id: I31202973aed98e61fa9b6a78b52e815227b6c17d Reviewed-on: https://webrtc-review.googlesource.com/22922 Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20825}
170 lines
5.4 KiB
C++
170 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 "ortc/ortcrtpreceiveradapter.h"
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#include <utility>
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#include "media/base/mediaconstants.h"
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#include "ortc/rtptransportadapter.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/helpers.h" // For "CreateRandomX".
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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_ =
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new VideoRtpReceiver(rtc::CreateRandomUuid(), {},
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rtp_transport_controller_->worker_thread(), ssrc,
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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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