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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 "webrtc/modules/audio_processing/audio_processing_impl.h" | 11 #include "webrtc/modules/audio_processing/audio_processing_impl.h" |
12 | 12 |
13 #include "testing/gmock/include/gmock/gmock.h" | 13 #include "testing/gmock/include/gmock/gmock.h" |
14 #include "testing/gtest/include/gtest/gtest.h" | 14 #include "testing/gtest/include/gtest/gtest.h" |
15 #include "webrtc/config.h" | 15 #include "webrtc/config.h" |
16 #include "webrtc/modules/audio_processing/test/test_utils.h" | 16 #include "webrtc/modules/audio_processing/test/test_utils.h" |
17 #include "webrtc/modules/include/module_common_types.h" | 17 #include "webrtc/modules/include/module_common_types.h" |
18 | 18 |
19 using ::testing::Invoke; | 19 using ::testing::Invoke; |
20 using ::testing::Return; | 20 using ::testing::Return; |
21 | 21 |
22 namespace webrtc { | 22 namespace webrtc { |
23 | 23 |
24 class MockInitialize : public AudioProcessingImpl { | 24 class MockInitialize : public AudioProcessingImpl { |
25 public: | 25 public: |
26 explicit MockInitialize(const webrtc::Config& config) | 26 explicit MockInitialize(const Config& config) : AudioProcessingImpl(config) { |
27 : AudioProcessingImpl(config) {} | 27 } |
28 | 28 |
29 MOCK_METHOD0(InitializeLocked, int()); | 29 MOCK_METHOD0(InitializeLocked, int()); |
30 int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS { | 30 int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS { |
31 return AudioProcessingImpl::InitializeLocked(); | 31 return AudioProcessingImpl::InitializeLocked(); |
32 } | 32 } |
33 }; | 33 }; |
34 | 34 |
35 TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { | 35 TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { |
36 webrtc::Config config; | 36 Config config; |
37 MockInitialize mock(config); | 37 MockInitialize mock(config); |
38 ON_CALL(mock, InitializeLocked()) | 38 ON_CALL(mock, InitializeLocked()) |
39 .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked)); | 39 .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked)); |
40 | 40 |
41 EXPECT_CALL(mock, InitializeLocked()).Times(1); | 41 EXPECT_CALL(mock, InitializeLocked()).Times(1); |
42 mock.Initialize(); | 42 mock.Initialize(); |
43 | 43 |
44 AudioFrame frame; | 44 AudioFrame frame; |
45 // Call with the default parameters; there should be no init. | 45 // Call with the default parameters; there should be no init. |
46 frame.num_channels_ = 1; | 46 frame.num_channels_ = 1; |
(...skipping 18 matching lines...) Expand all Loading... |
65 frame.num_channels_ = 2; | 65 frame.num_channels_ = 2; |
66 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); | 66 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); |
67 | 67 |
68 // A new sample rate passed to ProcessReverseStream should cause an init. | 68 // A new sample rate passed to ProcessReverseStream should cause an init. |
69 SetFrameSampleRate(&frame, 16000); | 69 SetFrameSampleRate(&frame, 16000); |
70 EXPECT_CALL(mock, InitializeLocked()).Times(1); | 70 EXPECT_CALL(mock, InitializeLocked()).Times(1); |
71 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); | 71 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); |
72 } | 72 } |
73 | 73 |
74 } // namespace webrtc | 74 } // namespace webrtc |
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