Add RtpTransceiverInterface and implementing class

Introduces the public API interface corresponding to the
standardized RtpTransceiver object in the WebRTC spec.
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver

The RtpTransceiver will be the internal representation for both
Plan B and Unified Plan SDP, but the public API interface will
only support Unified Plan (existing users should continue to use
GetSenders/GetReceivers, which will still be supported).

Bug: webrtc:7600
Change-Id: I417ffda683209ba9a9b4cbd274f91ca8295779a7
Reviewed-on: https://webrtc-review.googlesource.com/21460
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20659}
diff --git a/pc/rtptransceiver.cc b/pc/rtptransceiver.cc
new file mode 100644
index 0000000..7f6ed6d
--- /dev/null
+++ b/pc/rtptransceiver.cc
@@ -0,0 +1,154 @@
+/*
+ *  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 "pc/rtptransceiver.h"
+
+#include <string>
+
+namespace webrtc {
+
+RtpTransceiver::RtpTransceiver(cricket::MediaType media_type)
+    : unified_plan_(false), media_type_(media_type) {
+  RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
+             media_type == cricket::MEDIA_TYPE_VIDEO);
+}
+
+RtpTransceiver::~RtpTransceiver() {
+  Stop();
+}
+
+void RtpTransceiver::SetChannel(cricket::BaseChannel* channel) {
+  if (channel) {
+    RTC_DCHECK_EQ(media_type(), channel->media_type());
+  }
+  channel_ = channel;
+  for (auto sender : senders_) {
+    if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
+      static_cast<AudioRtpSender*>(sender->internal())
+          ->SetChannel(static_cast<cricket::VoiceChannel*>(channel));
+    } else {
+      static_cast<VideoRtpSender*>(sender->internal())
+          ->SetChannel(static_cast<cricket::VideoChannel*>(channel));
+    }
+  }
+  for (auto receiver : receivers_) {
+    if (!channel) {
+      receiver->internal()->Stop();
+    }
+    if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
+      static_cast<AudioRtpReceiver*>(receiver->internal())
+          ->SetChannel(static_cast<cricket::VoiceChannel*>(channel));
+    } else {
+      static_cast<VideoRtpReceiver*>(receiver->internal())
+          ->SetChannel(static_cast<cricket::VideoChannel*>(channel));
+    }
+  }
+}
+
+void RtpTransceiver::AddSender(
+    rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender) {
+  RTC_DCHECK(!unified_plan_);
+  RTC_DCHECK(sender);
+  RTC_DCHECK_EQ(media_type(), sender->internal()->media_type());
+  RTC_DCHECK(std::find(senders_.begin(), senders_.end(), sender) ==
+             senders_.end());
+  senders_.push_back(sender);
+}
+
+bool RtpTransceiver::RemoveSender(RtpSenderInterface* sender) {
+  RTC_DCHECK(!unified_plan_);
+  if (sender) {
+    RTC_DCHECK_EQ(media_type(), sender->media_type());
+  }
+  auto it = std::find(senders_.begin(), senders_.end(), sender);
+  if (it == senders_.end()) {
+    return false;
+  }
+  (*it)->internal()->Stop();
+  senders_.erase(it);
+  return true;
+}
+
+void RtpTransceiver::AddReceiver(
+    rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
+        receiver) {
+  RTC_DCHECK(!unified_plan_);
+  RTC_DCHECK(receiver);
+  RTC_DCHECK_EQ(media_type(), receiver->internal()->media_type());
+  RTC_DCHECK(std::find(receivers_.begin(), receivers_.end(), receiver) ==
+             receivers_.end());
+  receivers_.push_back(receiver);
+}
+
+bool RtpTransceiver::RemoveReceiver(RtpReceiverInterface* receiver) {
+  RTC_DCHECK(!unified_plan_);
+  if (receiver) {
+    RTC_DCHECK_EQ(media_type(), receiver->media_type());
+  }
+  auto it = std::find(receivers_.begin(), receivers_.end(), receiver);
+  if (it == receivers_.end()) {
+    return false;
+  }
+  (*it)->internal()->Stop();
+  receivers_.erase(it);
+  return true;
+}
+
+rtc::Optional<std::string> RtpTransceiver::mid() const {
+  return mid_;
+}
+
+rtc::scoped_refptr<RtpSenderInterface> RtpTransceiver::sender() const {
+  RTC_DCHECK(unified_plan_);
+  RTC_CHECK_EQ(1u, senders_.size());
+  return senders_[0];
+}
+
+rtc::scoped_refptr<RtpReceiverInterface> RtpTransceiver::receiver() const {
+  RTC_DCHECK(unified_plan_);
+  RTC_CHECK_EQ(1u, receivers_.size());
+  return receivers_[0];
+}
+
+bool RtpTransceiver::stopped() const {
+  return stopped_;
+}
+
+RtpTransceiverDirection RtpTransceiver::direction() const {
+  return direction_;
+}
+
+void RtpTransceiver::SetDirection(RtpTransceiverDirection new_direction) {
+  // TODO(steveanton): This should fire OnNegotiationNeeded.
+  direction_ = new_direction;
+}
+
+rtc::Optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
+    const {
+  return current_direction_;
+}
+
+void RtpTransceiver::Stop() {
+  for (auto sender : senders_) {
+    sender->internal()->Stop();
+  }
+  for (auto receiver : receivers_) {
+    receiver->internal()->Stop();
+  }
+  stopped_ = true;
+}
+
+void RtpTransceiver::SetCodecPreferences(
+    rtc::ArrayView<RtpCodecCapability> codecs) {
+  // TODO(steveanton): Implement this.
+  RTC_NOTREACHED() << "Not implemented";
+}
+
+}  // namespace webrtc