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