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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
26 */ | 26 */ |
27 | 27 |
28 #include <string> | 28 #include <string> |
29 | 29 |
30 #include "talk/app/webrtc/audiotrack.h" | 30 #include "talk/app/webrtc/audiotrack.h" |
31 #include "talk/app/webrtc/mediastream.h" | 31 #include "talk/app/webrtc/mediastream.h" |
32 #include "talk/app/webrtc/remoteaudiosource.h" | 32 #include "talk/app/webrtc/remoteaudiosource.h" |
| 33 #include "talk/app/webrtc/remoteaudiotrack.h" |
33 #include "talk/app/webrtc/rtpreceiver.h" | 34 #include "talk/app/webrtc/rtpreceiver.h" |
34 #include "talk/app/webrtc/rtpsender.h" | 35 #include "talk/app/webrtc/rtpsender.h" |
35 #include "talk/app/webrtc/streamcollection.h" | 36 #include "talk/app/webrtc/streamcollection.h" |
36 #include "talk/app/webrtc/videosource.h" | 37 #include "talk/app/webrtc/videosource.h" |
37 #include "talk/app/webrtc/videotrack.h" | 38 #include "talk/app/webrtc/videotrack.h" |
38 #include "talk/media/base/fakevideocapturer.h" | 39 #include "talk/media/base/fakevideocapturer.h" |
39 #include "talk/media/base/mediachannel.h" | 40 #include "talk/media/base/mediachannel.h" |
40 #include "testing/gmock/include/gmock/gmock.h" | 41 #include "testing/gmock/include/gmock/gmock.h" |
41 #include "testing/gtest/include/gtest/gtest.h" | 42 #include "testing/gtest/include/gtest/gtest.h" |
42 #include "webrtc/base/gunit.h" | 43 #include "webrtc/base/gunit.h" |
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60 virtual ~MockAudioProvider() {} | 61 virtual ~MockAudioProvider() {} |
61 MOCK_METHOD2(SetAudioPlayout, | 62 MOCK_METHOD2(SetAudioPlayout, |
62 void(uint32_t ssrc, | 63 void(uint32_t ssrc, |
63 bool enable)); | 64 bool enable)); |
64 MOCK_METHOD4(SetAudioSend, | 65 MOCK_METHOD4(SetAudioSend, |
65 void(uint32_t ssrc, | 66 void(uint32_t ssrc, |
66 bool enable, | 67 bool enable, |
67 const cricket::AudioOptions& options, | 68 const cricket::AudioOptions& options, |
68 cricket::AudioRenderer* renderer)); | 69 cricket::AudioRenderer* renderer)); |
69 MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume)); | 70 MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume)); |
| 71 MOCK_METHOD2(SetRawAudioSink, void(uint32_t, cricket::AudioRenderer::Sink*)); |
70 }; | 72 }; |
71 | 73 |
72 // Helper class to test RtpSender/RtpReceiver. | 74 // Helper class to test RtpSender/RtpReceiver. |
73 class MockVideoProvider : public VideoProviderInterface { | 75 class MockVideoProvider : public VideoProviderInterface { |
74 public: | 76 public: |
75 virtual ~MockVideoProvider() {} | 77 virtual ~MockVideoProvider() {} |
76 MOCK_METHOD2(SetCaptureDevice, | 78 MOCK_METHOD2(SetCaptureDevice, |
77 bool(uint32_t ssrc, cricket::VideoCapturer* camera)); | 79 bool(uint32_t ssrc, cricket::VideoCapturer* camera)); |
78 MOCK_METHOD3(SetVideoPlayout, | 80 MOCK_METHOD3(SetVideoPlayout, |
79 void(uint32_t ssrc, | 81 void(uint32_t ssrc, |
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144 audio_rtp_sender_ = nullptr; | 146 audio_rtp_sender_ = nullptr; |
145 } | 147 } |
146 | 148 |
147 void DestroyVideoRtpSender() { | 149 void DestroyVideoRtpSender() { |
148 EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, NULL)).Times(1); | 150 EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, NULL)).Times(1); |
149 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1); | 151 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1); |
150 video_rtp_sender_ = nullptr; | 152 video_rtp_sender_ = nullptr; |
151 } | 153 } |
152 | 154 |
153 void CreateAudioRtpReceiver() { | 155 void CreateAudioRtpReceiver() { |
154 audio_track_ = | 156 audio_track_ = RemoteAudioTrack::Create( |
155 AudioTrack::Create(kAudioTrackId, RemoteAudioSource::Create().get()); | 157 kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, &audio_provider_)); |
156 EXPECT_TRUE(stream_->AddTrack(audio_track_)); | 158 EXPECT_TRUE(stream_->AddTrack(audio_track_)); |
157 EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true)); | 159 EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true)); |
158 audio_rtp_receiver_ = new AudioRtpReceiver(stream_->GetAudioTracks()[0], | 160 audio_rtp_receiver_ = new AudioRtpReceiver(stream_->GetAudioTracks()[0], |
159 kAudioSsrc, &audio_provider_); | 161 kAudioSsrc, &audio_provider_); |
160 } | 162 } |
161 | 163 |
162 void CreateVideoRtpReceiver() { | 164 void CreateVideoRtpReceiver() { |
163 EXPECT_CALL(video_provider_, | 165 EXPECT_CALL(video_provider_, |
164 SetVideoPlayout(kVideoSsrc, true, | 166 SetVideoPlayout(kVideoSsrc, true, |
165 video_track_->GetSource()->FrameInput())); | 167 video_track_->GetSource()->FrameInput())); |
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481 video_track_->GetSource()->GetVideoCapturer())); | 483 video_track_->GetSource()->GetVideoCapturer())); |
482 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _)); | 484 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _)); |
483 sender->SetSsrc(kVideoSsrc2); | 485 sender->SetSsrc(kVideoSsrc2); |
484 | 486 |
485 // Calls expected from destructor. | 487 // Calls expected from destructor. |
486 EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1); | 488 EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1); |
487 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1); | 489 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1); |
488 } | 490 } |
489 | 491 |
490 } // namespace webrtc | 492 } // namespace webrtc |
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