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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 "webrtc/config.h" | 13 #include "webrtc/config.h" |
| 14 #include "webrtc/modules/audio_processing/test/test_utils.h" | 14 #include "webrtc/modules/audio_processing/test/test_utils.h" |
| 15 #include "webrtc/modules/include/module_common_types.h" | 15 #include "webrtc/modules/include/module_common_types.h" |
| 16 #include "webrtc/test/gmock.h" | 16 #include "webrtc/test/gmock.h" |
| 17 #include "webrtc/test/gtest.h" | 17 #include "webrtc/test/gtest.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 namespace { |
| 23 | 24 |
| 24 class MockInitialize : public AudioProcessingImpl { | 25 class MockInitialize : public AudioProcessingImpl { |
| 25 public: | 26 public: |
| 26 explicit MockInitialize(const webrtc::Config& config) | 27 explicit MockInitialize(const webrtc::Config& config) |
| 27 : AudioProcessingImpl(config) {} | 28 : AudioProcessingImpl(config) {} |
| 28 | 29 |
| 29 MOCK_METHOD0(InitializeLocked, int()); | 30 MOCK_METHOD0(InitializeLocked, int()); |
| 30 int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS { | 31 int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS { |
| 31 return AudioProcessingImpl::InitializeLocked(); | 32 return AudioProcessingImpl::InitializeLocked(); |
| 32 } | 33 } |
| 34 |
| 35 MOCK_CONST_METHOD0(AddRef, int()); |
| 36 MOCK_CONST_METHOD0(Release, int()); |
| 33 }; | 37 }; |
| 34 | 38 |
| 39 } // namespace |
| 40 |
| 35 TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { | 41 TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { |
| 36 webrtc::Config config; | 42 webrtc::Config config; |
| 37 MockInitialize mock(config); | 43 MockInitialize mock(config); |
| 38 ON_CALL(mock, InitializeLocked()) | 44 ON_CALL(mock, InitializeLocked()) |
| 39 .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked)); | 45 .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked)); |
| 40 | 46 |
| 41 EXPECT_CALL(mock, InitializeLocked()).Times(1); | 47 EXPECT_CALL(mock, InitializeLocked()).Times(1); |
| 42 mock.Initialize(); | 48 mock.Initialize(); |
| 43 | 49 |
| 44 AudioFrame frame; | 50 AudioFrame frame; |
| (...skipping 20 matching lines...) Expand all Loading... |
| 65 frame.num_channels_ = 2; | 71 frame.num_channels_ = 2; |
| 66 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); | 72 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); |
| 67 | 73 |
| 68 // A new sample rate passed to ProcessReverseStream should cause an init. | 74 // A new sample rate passed to ProcessReverseStream should cause an init. |
| 69 SetFrameSampleRate(&frame, 16000); | 75 SetFrameSampleRate(&frame, 16000); |
| 70 EXPECT_CALL(mock, InitializeLocked()).Times(1); | 76 EXPECT_CALL(mock, InitializeLocked()).Times(1); |
| 71 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); | 77 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); |
| 72 } | 78 } |
| 73 | 79 |
| 74 } // namespace webrtc | 80 } // namespace webrtc |
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