Move ready to send logic from BaseChannel to RtpTransport.

webrtc/pc/rtptransport_unittest.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 <string>
+
+#include "webrtc/pc/rtptransport.h"
+#include "webrtc/base/gunit.h"
+
+namespace webrtc {
+
+class RtpTransportTest : public testing::Test {};
+
+TEST_F(RtpTransportTest, SetRtcpParametersCantDisableRtcpMux) {
+  RtpTransport transport(false);
+  RtcpParameters params;
+  transport.SetRtcpParameters(params);
+  params.mux = false;
+  EXPECT_FALSE(transport.SetRtcpParameters(params).ok());
+}
+
+TEST_F(RtpTransportTest, SetRtcpParametersEmptyCnameUsesExisting) {
+  static const char kName[] = "name";
+  RtpTransport transport(false);
+  RtcpParameters params_with_name;
+  params_with_name.cname = kName;
+  transport.SetRtcpParameters(params_with_name);
+  EXPECT_EQ(transport.GetRtcpParameters().cname, kName);
+
+  RtcpParameters params_without_name;
+  transport.SetRtcpParameters(params_without_name);
+  EXPECT_EQ(transport.GetRtcpParameters().cname, kName);
+}
+
+class SignalObserver : public sigslot::has_slots<> {
+ public:
+  void OnReadyToSend(bool ready) { ready_ = ready; }
+  bool ready_ = false;
+};
+
+TEST_F(RtpTransportTest, SetReadyToSend) {
+  RtpTransport transport(false);
+  SignalObserver observer;
+  transport.SignalReadyToSend.connect(&observer,
+                                      &SignalObserver::OnReadyToSend);
+
+  transport.SetReadyToSend(true, true);  // rtcp ready
+  EXPECT_FALSE(observer.ready_);
+  transport.SetReadyToSend(false, true);  // rtp ready
+  EXPECT_TRUE(observer.ready_);
+}
+
+}  // namespace webrtc