Move DTMF sender to RtpSender (as opposed to WebRtcSession).

webrtc/pc/rtpsenderreceiver_unittest.cc

Index: webrtc/pc/rtpsenderreceiver_unittest.cc
diff --git a/webrtc/pc/rtpsenderreceiver_unittest.cc b/webrtc/pc/rtpsenderreceiver_unittest.cc
index 86c06121149fc632ac935880795b3cc8cc15002e..b7c5aacb8eaaa649c67d7dcc7d7fb90aea153bc8 100644
--- a/webrtc/pc/rtpsenderreceiver_unittest.cc
+++ b/webrtc/pc/rtpsenderreceiver_unittest.cc
@@ -13,6 +13,7 @@
 #include <utility>
 
 #include "webrtc/base/gunit.h"
+#include "webrtc/base/sigslot.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/base/mediachannel.h"
@@ -48,7 +49,8 @@ static const uint32_t kAudioSsrc2 = 101;
 
 namespace webrtc {
 
-class RtpSenderReceiverTest : public testing::Test {
+class RtpSenderReceiverTest : public testing::Test,
+                              public sigslot::has_slots<> {
  public:
   RtpSenderReceiverTest()
       :  // Create fake media engine/etc. so we can create channels to use to
@@ -104,8 +106,6 @@ class RtpSenderReceiverTest : public testing::Test {
         cricket::StreamParams::CreateLegacy(kVideoSsrc2));
   }
 
-  void TearDown() override { channel_manager_.Terminate(); }
-
   void AddVideoTrack() { AddVideoTrack(false); }
 
   void AddVideoTrack(bool is_screencast) {
@@ -124,9 +124,13 @@ class RtpSenderReceiverTest : public testing::Test {
         new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
                            voice_channel_, nullptr);
     audio_rtp_sender_->SetSsrc(kAudioSsrc);
+    audio_rtp_sender_->GetOnDestroyedSignal()->connect(
+        this, &RtpSenderReceiverTest::OnAudioSenderDestroyed);
     VerifyVoiceChannelInput();
   }
 
+  void OnAudioSenderDestroyed() { audio_sender_destroyed_signal_fired_ = true; }
+
   void CreateVideoRtpSender() { CreateVideoRtpSender(false); }
 
   void CreateVideoRtpSender(bool is_screencast) {
@@ -247,6 +251,7 @@ class RtpSenderReceiverTest : public testing::Test {
   rtc::scoped_refptr<MediaStreamInterface> stream_;
   rtc::scoped_refptr<VideoTrackInterface> video_track_;
   rtc::scoped_refptr<AudioTrackInterface> audio_track_;
+  bool audio_sender_destroyed_signal_fired_ = false;
 };
 
 // Test that |voice_channel_| is updated when an audio track is associated
@@ -721,4 +726,25 @@ TEST_F(RtpSenderReceiverTest,
   DestroyVideoRtpSender();
 }
 
+// There are end-to-end tests for DTMF, but here just ensure the DTMF sender is
+// provided based on sender type.
+TEST_F(RtpSenderReceiverTest, AudioSenderHasDtmfSender) {
+  CreateAudioRtpSender();
+  EXPECT_NE(nullptr, audio_rtp_sender_->GetDtmfSender());
+}
+
+TEST_F(RtpSenderReceiverTest, VideoSenderDoesNotHaveDtmfSender) {
+  CreateVideoRtpSender();
+  EXPECT_EQ(nullptr, video_rtp_sender_->GetDtmfSender());
+}
+
+// Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
+// destroyed, which is needed for the DTMF sender.
+TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
+  CreateAudioRtpSender();
+  EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
+  audio_rtp_sender_ = nullptr;
+  EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
+}
+
 }  // namespace webrtc