Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

webrtc/pc/rtpreceivershim.cc

Index: webrtc/pc/rtpreceivershim.cc
diff --git a/webrtc/pc/rtpreceivershim.cc b/webrtc/pc/rtpreceivershim.cc
new file mode 100644
index 0000000000000000000000000000000000000000..5edbf62d71dda93ec3164f27388eff2c7df20d58
--- /dev/null
+++ b/webrtc/pc/rtpreceivershim.cc
@@ -0,0 +1,201 @@
+/*
+ *  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/pc/rtpreceivershim.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/helpers.h"  // For "CreateRandomX".
+
+namespace {
+
+static const int kDefaultVideoClockrate = 90000;
+
+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>(kDefaultVideoClockrate);
+    }
+  }
+}
+
+}  // 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, SetParameters, const RtpParameters&)
+PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
+PROXY_CONSTMETHOD0(cricket::MediaType, GetKind)
+END_PROXY_MAP()
+
+// static
+RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
+RtpReceiverShim::CreateProxied(cricket::MediaType kind,
+                               const RtpParameters& parameters,
+                               RtpTransportShim* transport) {
+  RTC_DCHECK(transport);
+  RtpTransportControllerShim* rtp_transport_controller =
+      transport->rtp_transport_controller();
+  // Call "attach" method to ensure more than one receiver of the same type
+  // isn't attached to the same transport.
+  RTCError err;
+  switch (kind) {
+    case cricket::MEDIA_TYPE_AUDIO:
+      err = rtp_transport_controller->AttachAudioReceiver(transport);
+      break;
+    case cricket::MEDIA_TYPE_VIDEO:
+      err = rtp_transport_controller->AttachVideoReceiver(transport);
+      break;
+    case cricket::MEDIA_TYPE_DATA:
+      RTC_NOTREACHED();
+  }
+  if (!err.ok()) {
+    return err;
+  }
+
+  // Attempt to set parameters.
+  std::unique_ptr<RtpReceiverShim> receiver_shim(
+      new RtpReceiverShim(kind, transport, rtp_transport_controller));
+  err = receiver_shim->SetParameters(parameters);
+  if (!err.ok()) {
+    // Note: Destructor will automatically call "Detach" method.
+    return err;
+  }
+  return OrtcRtpReceiverProxy::Create(
+      rtp_transport_controller->signaling_thread(),
+      rtp_transport_controller->worker_thread(), receiver_shim.release());
+}
+
+RtpReceiverShim::~RtpReceiverShim() {
+  internal_receiver_ = nullptr;
+  // Need to detach from transport (was attached in Create method).
+  switch (kind_) {
+    case cricket::MEDIA_TYPE_AUDIO:
+      rtp_transport_controller_->DetachAudioReceiver();
+      break;
+    case cricket::MEDIA_TYPE_VIDEO:
+      rtp_transport_controller_->DetachVideoReceiver();
+      break;
+    case cricket::MEDIA_TYPE_DATA:
+      RTC_NOTREACHED();
+  }
+}
+
+rtc::scoped_refptr<MediaStreamTrackInterface> RtpReceiverShim::GetTrack()
+    const {
+  return internal_receiver_->track();
+}
+
+RTCError RtpReceiverShim::SetTransport(RtpTransportInterface* transport) {
+  LOG(LS_ERROR) << "Changing the transport of an RtpReceiver is not yet "
+                << "supported.";
+  return RTCError(RTCErrorType::UNSUPPORTED_PARAMETER);
+}
+
+RtpTransportInterface* RtpReceiverShim::GetTransport() const {
+  return transport_;
+}
+
+RTCError RtpReceiverShim::SetParameters(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();
+}
+
+RtpParameters RtpReceiverShim::GetParameters() const {
+  return last_applied_parameters_;
+}
+
+cricket::MediaType RtpReceiverShim::GetKind() const {
+  return cricket::MediaTypeFromString(GetTrack()->kind());
+}
+
+RtpReceiverShim::RtpReceiverShim(
+    cricket::MediaType kind,
+    RtpTransportShim* transport,
+    RtpTransportControllerShim* rtp_transport_controller)
+    : kind_(kind),
+      transport_(transport),
+      rtp_transport_controller_(rtp_transport_controller) {}
+
+void RtpReceiverShim::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