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

webrtc/ortc/ortcrtpsenderadapter.cc

Index: webrtc/ortc/ortcrtpsenderadapter.cc
diff --git a/webrtc/ortc/ortcrtpsenderadapter.cc b/webrtc/ortc/ortcrtpsenderadapter.cc
new file mode 100644
index 0000000000000000000000000000000000000000..be2b65dac31a2a10dc6d75c7327c94d89c530b33
--- /dev/null
+++ b/webrtc/ortc/ortcrtpsenderadapter.cc
@@ -0,0 +1,178 @@
+/*
+ *  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/ortcrtpsenderadapter.h"
+
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/media/base/mediaconstants.h"
+#include "webrtc/ortc/rtptransportadapter.h"
+
+namespace {
+
+void FillAudioSenderParameters(webrtc::RtpParameters* parameters) {
+  for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
+    if (!codec.num_channels) {
+      codec.num_channels = rtc::Optional<int>(1);
+    }
+  }
+}
+
+void FillVideoSenderParameters(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(OrtcRtpSender)
+PROXY_SIGNALING_THREAD_DESTRUCTOR()
+PROXY_METHOD1(RTCError, SetTrack, MediaStreamTrackInterface*)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, GetTrack)
+PROXY_METHOD1(RTCError, SetTransport, RtpTransportInterface*)
+PROXY_CONSTMETHOD0(RtpTransportInterface*, GetTransport)
+PROXY_METHOD1(RTCError, Send, const RtpParameters&)
+PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
+PROXY_CONSTMETHOD0(cricket::MediaType, GetKind)
+END_PROXY_MAP()
+
+// static
+std::unique_ptr<OrtcRtpSenderInterface> OrtcRtpSenderAdapter::CreateProxy(
+    std::unique_ptr<OrtcRtpSenderAdapter> wrapped_sender) {
+  RTC_DCHECK(wrapped_sender);
+  rtc::Thread* signaling =
+      wrapped_sender->rtp_transport_controller_->signaling_thread();
+  rtc::Thread* worker =
+      wrapped_sender->rtp_transport_controller_->worker_thread();
+  return OrtcRtpSenderProxy::Create(signaling, worker,
+                                    std::move(wrapped_sender));
+}
+
+OrtcRtpSenderAdapter::~OrtcRtpSenderAdapter() {
+  internal_sender_ = nullptr;
+  SignalDestroyed();
+}
+
+RTCError OrtcRtpSenderAdapter::SetTrack(MediaStreamTrackInterface* track) {
+  if (track && cricket::MediaTypeFromString(track->kind()) != kind_) {
+    LOG_AND_RETURN_ERROR(
+        RTCErrorType::INVALID_PARAMETER,
+        "Track kind (audio/video) doesn't match the kind of this sender.");
+  }
+  if (internal_sender_ && !internal_sender_->SetTrack(track)) {
+    // Since we checked the track type above, this should never happen...
+    RTC_NOTREACHED();
+    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
+                         "Failed to set track on RtpSender.");
+  }
+  track_ = track;
+  return RTCError::OK();
+}
+
+rtc::scoped_refptr<MediaStreamTrackInterface> OrtcRtpSenderAdapter::GetTrack()
+    const {
+  return track_;
+}
+
+RTCError OrtcRtpSenderAdapter::SetTransport(RtpTransportInterface* transport) {
+  LOG_AND_RETURN_ERROR(
+      RTCErrorType::UNSUPPORTED_OPERATION,
+      "Changing the transport of an RtpSender is not yet supported.");
+}
+
+RtpTransportInterface* OrtcRtpSenderAdapter::GetTransport() const {
+  return transport_;
+}
+
+RTCError OrtcRtpSenderAdapter::Send(const RtpParameters& parameters) {
+  RtpParameters filled_parameters = parameters;
+  RTCError err;
+  uint32_t ssrc = 0;
+  switch (kind_) {
+    case cricket::MEDIA_TYPE_AUDIO:
+      FillAudioSenderParameters(&filled_parameters);
+      err = rtp_transport_controller_->ValidateAndApplyAudioSenderParameters(
+          filled_parameters, &ssrc);
+      if (!err.ok()) {
+        return err;
+      }
+      break;
+    case cricket::MEDIA_TYPE_VIDEO:
+      FillVideoSenderParameters(&filled_parameters);
+      err = rtp_transport_controller_->ValidateAndApplyVideoSenderParameters(
+          filled_parameters, &ssrc);
+      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 call SetSsrc on the internal sender.
+  // This is analogous to a PeerConnection calling SetSsrc after
+  // SetLocalDescription is successful.
+  //
+  // If there were no encodings, this SSRC may be 0, which is valid.
+  if (!internal_sender_) {
+    CreateInternalSender();
+  }
+  internal_sender_->SetSsrc(ssrc);
+
+  return RTCError::OK();
+}
+
+RtpParameters OrtcRtpSenderAdapter::GetParameters() const {
+  return last_applied_parameters_;
+}
+
+cricket::MediaType OrtcRtpSenderAdapter::GetKind() const {
+  return kind_;
+}
+
+OrtcRtpSenderAdapter::OrtcRtpSenderAdapter(
+    cricket::MediaType kind,
+    RtpTransportInterface* transport,
+    RtpTransportControllerAdapter* rtp_transport_controller)
+    : kind_(kind),
+      transport_(transport),
+      rtp_transport_controller_(rtp_transport_controller) {}
+
+void OrtcRtpSenderAdapter::CreateInternalSender() {
+  switch (kind_) {
+    case cricket::MEDIA_TYPE_AUDIO:
+      internal_sender_ = new AudioRtpSender(
+          rtp_transport_controller_->voice_channel(), nullptr);
+      break;
+    case cricket::MEDIA_TYPE_VIDEO:
+      internal_sender_ =
+          new VideoRtpSender(rtp_transport_controller_->video_channel());
+      break;
+    case cricket::MEDIA_TYPE_DATA:
+      RTC_NOTREACHED();
+  }
+  if (track_) {
+    if (!internal_sender_->SetTrack(track_)) {
+      // Since we checked the track type when it was set, this should never
+      // happen...
+      RTC_NOTREACHED();
+    }
+  }
+}
+
+}  // namespace webrtc