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

webrtc/pc/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 2012 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 <memory>
 #include <string>
 #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"
 #include "webrtc/media/engine/fakewebrtccall.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/pc/audiotrack.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/pc/fakemediacontroller.h"
 #include "webrtc/pc/localaudiosource.h"
 #include "webrtc/pc/mediastream.h"
@@ skipping 15 matching lines @@
 static const char kStreamLabel1[] = "local_stream_1";
 static const char kVideoTrackId[] = "video_1";
 static const char kAudioTrackId[] = "audio_1";
 static const uint32_t kVideoSsrc = 98;
 static const uint32_t kVideoSsrc2 = 100;
 static const uint32_t kAudioSsrc = 99;
 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
         // test RtpSenders/RtpReceivers.
         media_engine_(new cricket::FakeMediaEngine()),
         channel_manager_(media_engine_,
                          rtc::Thread::Current(),
                          rtc::Thread::Current()),
         fake_call_(Call::Config(&event_log_)),
         fake_media_controller_(&channel_manager_, &fake_call_),
@@ skipping 35 matching lines @@
     video_media_channel_->AddSendStream(
         cricket::StreamParams::CreateLegacy(kVideoSsrc));
     video_media_channel_->AddRecvStream(
         cricket::StreamParams::CreateLegacy(kVideoSsrc));
     video_media_channel_->AddSendStream(
         cricket::StreamParams::CreateLegacy(kVideoSsrc2));
     video_media_channel_->AddRecvStream(
         cricket::StreamParams::CreateLegacy(kVideoSsrc2));
   }
 
-  void TearDown() override { channel_manager_.Terminate(); }
-
   void AddVideoTrack() { AddVideoTrack(false); }
 
   void AddVideoTrack(bool is_screencast) {
     rtc::scoped_refptr<VideoTrackSourceInterface> source(
         FakeVideoTrackSource::Create(is_screencast));
     video_track_ = VideoTrack::Create(kVideoTrackId, source);
     EXPECT_TRUE(stream_->AddTrack(video_track_));
   }
 
   void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
 
   void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) {
     audio_track_ = AudioTrack::Create(kAudioTrackId, source);
     EXPECT_TRUE(stream_->AddTrack(audio_track_));
     audio_rtp_sender_ =
         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) {
     AddVideoTrack(is_screencast);
     video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
                                            stream_->label(), video_channel_);
     video_rtp_sender_->SetSsrc(kVideoSsrc);
     VerifyVideoChannelInput();
   }
 
@@ skipping 100 matching lines @@
   cricket::VideoChannel* video_channel_;
   cricket::FakeVoiceMediaChannel* voice_media_channel_;
   cricket::FakeVideoMediaChannel* video_media_channel_;
   rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_;
   rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_;
   rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_;
   rtc::scoped_refptr<VideoRtpReceiver> video_rtp_receiver_;
   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
 // and disassociated with an AudioRtpSender.
 TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) {
   CreateAudioRtpSender();
   DestroyAudioRtpSender();
 }
 
 // Test that |video_channel_| is updated when a video track is associated and
@@ skipping 454 matching lines @@
 
   // And removing the hint should go back to false (to verify that false was
   // default correctly).
   video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone);
   EXPECT_EQ(rtc::Optional<bool>(false),
             video_media_channel_->options().is_screencast);
 
   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