Move ready to send logic from BaseChannel to RtpTransport.

webrtc/pc/rtptransport_unittest.cc

Index: webrtc/pc/rtptransport_unittest.cc
diff --git a/webrtc/pc/rtptransport_unittest.cc b/webrtc/pc/rtptransport_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..ba492517b511846e56912901e601b2bc35a81073
--- /dev/null
+++ b/webrtc/pc/rtptransport_unittest.cc
@@ -0,0 +1,150 @@
+/*
+ *  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/base/gunit.h"
+#include "webrtc/p2p/base/fakepackettransport.h"
+#include "webrtc/pc/rtptransport.h"
+
+namespace webrtc {
+
+class RtpTransportTest : public testing::Test {};
+
+constexpr bool kMuxDisabled = false;
+constexpr bool kMuxEnabled = true;
+
+TEST_F(RtpTransportTest, SetRtcpParametersCantDisableRtcpMux) {
+  RtpTransport transport(kMuxDisabled);
+  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(kMuxDisabled);
+  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:
+  explicit SignalObserver(RtpTransport* transport) {
+    transport->SignalReadyToSend.connect(this, &SignalObserver::OnReadyToSend);
+  }
+  void OnReadyToSend(bool ready) { ready_ = ready; }
+  bool ready_ = false;
+};
+
+TEST_F(RtpTransportTest, SettingRtcpAndRtpSignalsReady) {
+  RtpTransport transport(kMuxDisabled);
+  SignalObserver observer(&transport);
+  rtc::FakePacketTransport fake_rtcp("fake_rtcp");
+  fake_rtcp.SetWritable(true);
+  rtc::FakePacketTransport fake_rtp("fake_rtp");
+  fake_rtp.SetWritable(true);
+
+  transport.SetRtcpPacketTransport(&fake_rtcp);  // rtcp ready
+  EXPECT_FALSE(observer.ready_);
+  transport.SetRtpPacketTransport(&fake_rtp);  // rtp ready
+  EXPECT_TRUE(observer.ready_);
+}
+
+TEST_F(RtpTransportTest, SettingRtpAndRtcpSignalsReady) {
+  RtpTransport transport(kMuxDisabled);
+  SignalObserver observer(&transport);
+  rtc::FakePacketTransport fake_rtcp("fake_rtcp");
+  fake_rtcp.SetWritable(true);
+  rtc::FakePacketTransport fake_rtp("fake_rtp");
+  fake_rtp.SetWritable(true);
+
+  transport.SetRtpPacketTransport(&fake_rtp);  // rtp ready
+  EXPECT_FALSE(observer.ready_);
+  transport.SetRtcpPacketTransport(&fake_rtcp);  // rtcp ready
+  EXPECT_TRUE(observer.ready_);
+}
+
+TEST_F(RtpTransportTest, SettingRtpWithRtcpMuxEnabledSignalsReady) {
+  RtpTransport transport(kMuxEnabled);
+  SignalObserver observer(&transport);
+  rtc::FakePacketTransport fake_rtp("fake_rtp");
+  fake_rtp.SetWritable(true);
+
+  transport.SetRtpPacketTransport(&fake_rtp);  // rtp ready
+  EXPECT_TRUE(observer.ready_);
+}
+
+TEST_F(RtpTransportTest, DisablingRtcpMuxSignalsNotReady) {
+  RtpTransport transport(kMuxEnabled);
+  SignalObserver observer(&transport);
+  rtc::FakePacketTransport fake_rtp("fake_rtp");
+  fake_rtp.SetWritable(true);
+
+  transport.SetRtpPacketTransport(&fake_rtp);  // rtp ready
+  EXPECT_TRUE(observer.ready_);
+
+  transport.SetRtcpMuxEnabled(false);
+  EXPECT_FALSE(observer.ready_);
+}
+
+TEST_F(RtpTransportTest, EnablingRtcpMuxSignalsReady) {
+  RtpTransport transport(kMuxDisabled);
+  SignalObserver observer(&transport);
+  rtc::FakePacketTransport fake_rtp("fake_rtp");
+  fake_rtp.SetWritable(true);
+
+  transport.SetRtpPacketTransport(&fake_rtp);  // rtp ready
+  EXPECT_FALSE(observer.ready_);
+
+  transport.SetRtcpMuxEnabled(true);
+  EXPECT_TRUE(observer.ready_);
+}
+
+class SignalCounter : public sigslot::has_slots<> {
+ public:
+  explicit SignalCounter(RtpTransport* transport) {
+    transport->SignalReadyToSend.connect(this, &SignalCounter::OnReadyToSend);
+  }
+  void OnReadyToSend(bool ready) { ++count_; }
+  int count_ = 0;
+};
+
+TEST_F(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
+  RtpTransport transport(kMuxEnabled);
+  SignalCounter observer(&transport);
+  rtc::FakePacketTransport fake_rtp("fake_rtp");
+  fake_rtp.SetWritable(true);
+
+  // State changes, so we should signal.
+  transport.SetRtpPacketTransport(&fake_rtp);
+  EXPECT_EQ(observer.count_, 1);
+
+  // State does not change, so we should not signal.
+  transport.SetRtpPacketTransport(&fake_rtp);
+  EXPECT_EQ(observer.count_, 1);
+
+  // State does not change, so we should not signal.
+  transport.SetRtcpMuxEnabled(true);
+  EXPECT_EQ(observer.count_, 1);
+
+  // State changes, so we should signal.
+  transport.SetRtcpMuxEnabled(false);
+  EXPECT_EQ(observer.count_, 2);
+}
+
+}  // namespace webrtc