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Side by Side Diff: webrtc/call/call_unittest.cc

Issue 2454373002: Added an empty AudioTransportProxy to AudioState. (Closed)
Patch Set: No heap transport, WillOnce, comparison with constants. Created 4 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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 10
11 #include <list> 11 #include <list>
12 #include <memory> 12 #include <memory>
13 13
14 #include "webrtc/api/call/audio_state.h" 14 #include "webrtc/api/call/audio_state.h"
15 #include "webrtc/call.h" 15 #include "webrtc/call.h"
16 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" 16 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
17 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" 17 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
18 #include "webrtc/modules/audio_device/include/mock_audio_device.h"
18 #include "webrtc/test/gtest.h" 19 #include "webrtc/test/gtest.h"
19 #include "webrtc/test/mock_voice_engine.h" 20 #include "webrtc/test/mock_voice_engine.h"
20 21
21 namespace { 22 namespace {
22 23
23 struct CallHelper { 24 struct CallHelper {
24 explicit CallHelper( 25 explicit CallHelper(
25 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) 26 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
26 : voice_engine_(decoder_factory) { 27 : voice_engine_(decoder_factory) {
27 webrtc::AudioState::Config audio_state_config; 28 webrtc::AudioState::Config audio_state_config;
28 audio_state_config.voice_engine = &voice_engine_; 29 audio_state_config.voice_engine = &voice_engine_;
30 EXPECT_CALL(voice_engine_, audio_device_module())
31 .WillOnce(testing::Return(&mock_audio_device_));
29 webrtc::Call::Config config(&event_log_); 32 webrtc::Call::Config config(&event_log_);
30 config.audio_state = webrtc::AudioState::Create(audio_state_config); 33 config.audio_state = webrtc::AudioState::Create(audio_state_config);
31 call_.reset(webrtc::Call::Create(config)); 34 call_.reset(webrtc::Call::Create(config));
32 } 35 }
33 36
34 webrtc::Call* operator->() { return call_.get(); } 37 webrtc::Call* operator->() { return call_.get(); }
35 38
36 private: 39 private:
37 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; 40 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
41 webrtc::test::MockAudioDeviceModule mock_audio_device_;
38 webrtc::RtcEventLogNullImpl event_log_; 42 webrtc::RtcEventLogNullImpl event_log_;
39 std::unique_ptr<webrtc::Call> call_; 43 std::unique_ptr<webrtc::Call> call_;
40 }; 44 };
41 } // namespace 45 } // namespace
42 46
43 namespace webrtc { 47 namespace webrtc {
44 48
45 TEST(CallTest, ConstructDestruct) { 49 TEST(CallTest, ConstructDestruct) {
46 CallHelper call; 50 CallHelper call;
47 } 51 }
(...skipping 134 matching lines...) Expand 10 before | Expand all | Expand 10 after
182 stream = call->CreateFlexfecReceiveStream(config); 186 stream = call->CreateFlexfecReceiveStream(config);
183 EXPECT_NE(stream, nullptr); 187 EXPECT_NE(stream, nullptr);
184 streams.push_back(stream); 188 streams.push_back(stream);
185 189
186 for (auto s : streams) { 190 for (auto s : streams) {
187 call->DestroyFlexfecReceiveStream(s); 191 call->DestroyFlexfecReceiveStream(s);
188 } 192 }
189 } 193 }
190 194
191 } // namespace webrtc 195 } // namespace webrtc
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