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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 #include <functional> | 10 #include <functional> |
11 #include <list> | 11 #include <list> |
12 #include <string> | 12 #include <string> |
13 | 13 |
14 #include "testing/gtest/include/gtest/gtest.h" | 14 #include "testing/gtest/include/gtest/gtest.h" |
15 | 15 |
| 16 #include "webrtc/audio_state.h" |
16 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
17 #include "webrtc/base/scoped_ptr.h" | 18 #include "webrtc/base/scoped_ptr.h" |
18 #include "webrtc/base/thread_annotations.h" | 19 #include "webrtc/base/thread_annotations.h" |
19 #include "webrtc/call.h" | 20 #include "webrtc/call.h" |
20 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 21 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
21 #include "webrtc/system_wrappers/include/event_wrapper.h" | 22 #include "webrtc/system_wrappers/include/event_wrapper.h" |
22 #include "webrtc/system_wrappers/include/trace.h" | 23 #include "webrtc/system_wrappers/include/trace.h" |
23 #include "webrtc/test/call_test.h" | 24 #include "webrtc/test/call_test.h" |
24 #include "webrtc/test/direct_transport.h" | 25 #include "webrtc/test/direct_transport.h" |
25 #include "webrtc/test/encoder_settings.h" | 26 #include "webrtc/test/encoder_settings.h" |
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110 static const int kTOFExtensionId = 4; | 111 static const int kTOFExtensionId = 4; |
111 static const int kASTExtensionId = 5; | 112 static const int kASTExtensionId = 5; |
112 | 113 |
113 class BitrateEstimatorTest : public test::CallTest { | 114 class BitrateEstimatorTest : public test::CallTest { |
114 public: | 115 public: |
115 BitrateEstimatorTest() : receive_config_(nullptr) {} | 116 BitrateEstimatorTest() : receive_config_(nullptr) {} |
116 | 117 |
117 virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); } | 118 virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); } |
118 | 119 |
119 virtual void SetUp() { | 120 virtual void SetUp() { |
| 121 EXPECT_CALL(mock_voice_engine_, |
| 122 RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0)); |
| 123 EXPECT_CALL(mock_voice_engine_, |
| 124 DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0)); |
120 EXPECT_CALL(mock_voice_engine_, GetEventLog()) | 125 EXPECT_CALL(mock_voice_engine_, GetEventLog()) |
121 .WillRepeatedly(testing::Return(nullptr)); | 126 .WillRepeatedly(testing::Return(nullptr)); |
122 | 127 |
| 128 AudioState::Config audio_state_config; |
| 129 audio_state_config.voice_engine = &mock_voice_engine_; |
123 Call::Config config; | 130 Call::Config config; |
124 config.voice_engine = &mock_voice_engine_; | 131 config.audio_state = AudioState::Create(audio_state_config); |
125 receiver_call_.reset(Call::Create(config)); | 132 receiver_call_.reset(Call::Create(config)); |
126 sender_call_.reset(Call::Create(config)); | 133 sender_call_.reset(Call::Create(config)); |
127 | 134 |
128 send_transport_.reset(new test::DirectTransport(sender_call_.get())); | 135 send_transport_.reset(new test::DirectTransport(sender_call_.get())); |
129 send_transport_->SetReceiver(receiver_call_->Receiver()); | 136 send_transport_->SetReceiver(receiver_call_->Receiver()); |
130 receive_transport_.reset(new test::DirectTransport(receiver_call_.get())); | 137 receive_transport_.reset(new test::DirectTransport(receiver_call_.get())); |
131 receive_transport_->SetReceiver(sender_call_->Receiver()); | 138 receive_transport_->SetReceiver(sender_call_->Receiver()); |
132 | 139 |
133 send_config_ = VideoSendStream::Config(send_transport_.get()); | 140 send_config_ = VideoSendStream::Config(send_transport_.get()); |
134 send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]); | 141 send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]); |
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155 | 162 |
156 send_transport_->StopSending(); | 163 send_transport_->StopSending(); |
157 receive_transport_->StopSending(); | 164 receive_transport_->StopSending(); |
158 | 165 |
159 while (!streams_.empty()) { | 166 while (!streams_.empty()) { |
160 delete streams_.back(); | 167 delete streams_.back(); |
161 streams_.pop_back(); | 168 streams_.pop_back(); |
162 } | 169 } |
163 | 170 |
164 receiver_call_.reset(); | 171 receiver_call_.reset(); |
| 172 sender_call_.reset(); |
165 } | 173 } |
166 | 174 |
167 protected: | 175 protected: |
168 friend class Stream; | 176 friend class Stream; |
169 | 177 |
170 class Stream { | 178 class Stream { |
171 public: | 179 public: |
172 Stream(BitrateEstimatorTest* test, bool receive_audio) | 180 Stream(BitrateEstimatorTest* test, bool receive_audio) |
173 : test_(test), | 181 : test_(test), |
174 is_sending_receiving_(false), | 182 is_sending_receiving_(false), |
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352 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId); | 360 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId); |
353 receiver_trace_.PushExpectedLogLine( | 361 receiver_trace_.PushExpectedLogLine( |
354 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); | 362 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); |
355 receiver_trace_.PushExpectedLogLine(kSingleStreamLog); | 363 receiver_trace_.PushExpectedLogLine(kSingleStreamLog); |
356 streams_.push_back(new Stream(this, false)); | 364 streams_.push_back(new Stream(this, false)); |
357 streams_[0]->StopSending(); | 365 streams_[0]->StopSending(); |
358 streams_[1]->StopSending(); | 366 streams_[1]->StopSending(); |
359 EXPECT_EQ(kEventSignaled, receiver_trace_.Wait()); | 367 EXPECT_EQ(kEventSignaled, receiver_trace_.Wait()); |
360 } | 368 } |
361 } // namespace webrtc | 369 } // namespace webrtc |
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