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

webrtc/pc/rtpsendershim.cc

Index: webrtc/pc/rtpsendershim.cc
diff --git a/webrtc/pc/rtpsendershim.cc b/webrtc/pc/rtpsendershim.cc
new file mode 100644
index 0000000000000000000000000000000000000000..1514df631be578dc5a200fd0650e056ed4f30d6b
--- /dev/null
+++ b/webrtc/pc/rtpsendershim.cc
@@ -0,0 +1,197 @@
+/*
+ *  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/rtpsendershim.h"
+
+#include "webrtc/base/checks.h"
+
+namespace {
+
+static const int kVideoClockrate = 90000;
+
+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>(kVideoClockrate);
+    }
+  }
+}
+
+}  // 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, SetParameters, const RtpParameters&)
+PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
+PROXY_CONSTMETHOD0(cricket::MediaType, GetKind)
+END_PROXY_MAP()
+
+// static
+RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>
+RtpSenderShim::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 sender of the same type
+  // isn't attached to the same transport.
+  RTCError err;
+  switch (kind) {
+    case cricket::MEDIA_TYPE_AUDIO:
+      err = rtp_transport_controller->AttachAudioSender(transport);
+      break;
+    case cricket::MEDIA_TYPE_VIDEO:
+      err = rtp_transport_controller->AttachVideoSender(transport);
+      break;
+    case cricket::MEDIA_TYPE_DATA:
+      RTC_NOTREACHED();
+  }
+  if (!err.ok()) {
+    return err;
+  }
+
+  // Attempt to set parameters.
+  std::unique_ptr<RtpSenderShim> sender_shim(
+      new RtpSenderShim(kind, transport, rtp_transport_controller));
+  err = sender_shim->SetParameters(parameters);
+  if (!err.ok()) {
+    // Note: Destructor will automatically call "Detach" method.
+    return err;
+  }
+  return OrtcRtpSenderProxy::Create(
+      rtp_transport_controller->signaling_thread(),
+      rtp_transport_controller->worker_thread(), sender_shim.release());
+}
+
+RtpSenderShim::~RtpSenderShim() {
+  internal_sender_ = nullptr;
+  // Need to detach from transport (was attached in Create method).
+  switch (kind_) {
+    case cricket::MEDIA_TYPE_AUDIO:
+      rtp_transport_controller_->DetachAudioSender();
+      break;
+    case cricket::MEDIA_TYPE_VIDEO:
+      rtp_transport_controller_->DetachVideoSender();
+      break;
+    case cricket::MEDIA_TYPE_DATA:
+      RTC_NOTREACHED();
+  }
+}
+
+RTCError RtpSenderShim::SetTrack(MediaStreamTrackInterface* track) {
+  RTCError ret;
+  if (!internal_sender_->SetTrack(track)) {
+    // Should only happen if track type is invalid.
+    ret.set_type(RTCErrorType::INVALID_PARAMETER);
+  }
+  return ret;
+}
+
+rtc::scoped_refptr<MediaStreamTrackInterface> RtpSenderShim::GetTrack() const {
+  return internal_sender_->track();
+}
+
+RTCError RtpSenderShim::SetTransport(RtpTransportInterface* transport) {
+  LOG(LS_ERROR) << "Changing the transport of an RtpSender is not yet "
+                << "supported.";
+  return RTCError(RTCErrorType::UNSUPPORTED_PARAMETER);
+}
+
+RtpTransportInterface* RtpSenderShim::GetTransport() const {
+  return transport_;
+}
+
+RTCError RtpSenderShim::SetParameters(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();
+}
+
+RtpParameters RtpSenderShim::GetParameters() const {
+  return last_applied_parameters_;
+}
+
+cricket::MediaType RtpSenderShim::GetKind() const {
+  return internal_sender_->media_type();
+}
+
+RtpSenderShim::RtpSenderShim(
+    cricket::MediaType kind,
+    RtpTransportShim* transport,
+    RtpTransportControllerShim* rtp_transport_controller)
+    : kind_(kind),
+      transport_(transport),
+      rtp_transport_controller_(rtp_transport_controller) {}
+
+void RtpSenderShim::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();
+  }
+}
+
+}  // namespace webrtc