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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/test/gtest.h" | 18 #include "webrtc/test/gtest.h" |
19 #include "webrtc/test/mock_voice_engine.h" | 19 #include "webrtc/test/mock_voice_engine.h" |
20 | 20 |
21 namespace { | 21 namespace { |
22 | 22 |
23 struct CallHelper { | 23 struct CallHelper { |
24 explicit CallHelper( | 24 explicit CallHelper( |
25 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) | 25 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) |
26 : voice_engine_(decoder_factory) { | 26 : voice_engine_(decoder_factory) { |
27 webrtc::AudioState::Config audio_state_config; | 27 webrtc::AudioState::Config audio_state_config; |
28 audio_state_config.voice_engine = &voice_engine_; | 28 audio_state_config.voice_engine = &voice_engine_; |
| 29 EXPECT_CALL(voice_engine_, audio_device_module()); |
| 30 EXPECT_CALL(voice_engine_, audio_processing()); |
| 31 EXPECT_CALL(voice_engine_, audio_transport()); |
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 webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; } | 38 webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; } |
36 | 39 |
37 private: | 40 private: |
38 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; | 41 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; |
(...skipping 240 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
279 stream = call->CreateFlexfecReceiveStream(config); | 282 stream = call->CreateFlexfecReceiveStream(config); |
280 EXPECT_NE(stream, nullptr); | 283 EXPECT_NE(stream, nullptr); |
281 streams.push_back(stream); | 284 streams.push_back(stream); |
282 | 285 |
283 for (auto s : streams) { | 286 for (auto s : streams) { |
284 call->DestroyFlexfecReceiveStream(s); | 287 call->DestroyFlexfecReceiveStream(s); |
285 } | 288 } |
286 } | 289 } |
287 | 290 |
288 } // namespace webrtc | 291 } // namespace webrtc |
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