Create the SrtpTransportInterface.

webrtc/ortc/ortcrtpreceiveradapter.cc

+/*
+ *  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/ortcrtpreceiveradapter.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/helpers.h"  // For "CreateRandomX".
+#include "webrtc/media/base/mediaconstants.h"
+#include "webrtc/ortc/rtptransportadapter.h"
+
+namespace {
+
+void FillAudioReceiverParameters(webrtc::RtpParameters* parameters) {
+  for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
+    if (!codec.num_channels) {
+      codec.num_channels = rtc::Optional<int>(1);
+    }
+  }
+}
+
+void FillVideoReceiverParameters(webrtc::RtpParameters* parameters) {
+  for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
+    if (!codec.clock_rate) {
+      codec.clock_rate = rtc::Optional<int>(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);
+  return OrtcRtpReceiverProxy::Create(
+      wrapped_receiver->rtp_transport_controller_->signaling_thread(),
+      wrapped_receiver->rtp_transport_controller_->worker_thread(),
+      wrapped_receiver.release());
+}
+
+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;
+  if (!transport_->GetInternal()->TransportReadyToSendAndReceive()) {
+    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
+                         "The underneath transport is not ready to receive.");
+  }
+  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:
+      internal_receiver_ =
+          new AudioRtpReceiver(rtc::CreateRandomUuid(), ssrc,
+                               rtp_transport_controller_->voice_channel());
+      break;
+    case cricket::MEDIA_TYPE_VIDEO:
+      internal_receiver_ = new VideoRtpReceiver(
+          rtc::CreateRandomUuid(), rtp_transport_controller_->worker_thread(),
+          ssrc, rtp_transport_controller_->video_channel());
+      break;
+    case cricket::MEDIA_TYPE_DATA:
+      RTC_NOTREACHED();
+  }
+}
+
+}  // namespace webrtc