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Side by Side Diff: webrtc/modules/audio_processing/audio_processing_impl_unittest.cc

Issue 2961723004: Allow an external audio processing module to be used in WebRTC (Closed)
Patch Set: Allow an external audio processing module to be used in WebRTC Created 3 years, 5 months ago
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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 #if 0
the sun 2017/06/29 10:41:52 Uh-oh!
peah-webrtc 2017/06/29 11:46:32 Done.
12
11 #include "webrtc/modules/audio_processing/audio_processing_impl.h" 13 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
12 14
13 #include "webrtc/config.h" 15 #include "webrtc/config.h"
14 #include "webrtc/modules/audio_processing/test/test_utils.h" 16 #include "webrtc/modules/audio_processing/test/test_utils.h"
15 #include "webrtc/modules/include/module_common_types.h" 17 #include "webrtc/modules/include/module_common_types.h"
16 #include "webrtc/test/gmock.h" 18 #include "webrtc/test/gmock.h"
17 #include "webrtc/test/gtest.h" 19 #include "webrtc/test/gtest.h"
18 20
19 using ::testing::Invoke; 21 using ::testing::Invoke;
20 using ::testing::Return; 22 using ::testing::Return;
21 23
22 namespace webrtc { 24 namespace webrtc {
25 namespace {
23 26
24 class MockInitialize : public AudioProcessingImpl { 27 class MockInitialize : public AudioProcessingImpl {
25 public: 28 public:
26 explicit MockInitialize(const webrtc::Config& config) 29 explicit MockInitialize(const webrtc::Config& config)
27 : AudioProcessingImpl(config) {} 30 : AudioProcessingImpl(config) {}
28 31
29 MOCK_METHOD0(InitializeLocked, int()); 32 MOCK_METHOD0(InitializeLocked, int());
30 int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS { 33 int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS {
31 return AudioProcessingImpl::InitializeLocked(); 34 return AudioProcessingImpl::InitializeLocked();
32 } 35 }
36
37 MOCK_CONST_METHOD0(AddRef, int());
38 MOCK_CONST_METHOD0(Release, int());
33 }; 39 };
34 40
41 } // namespace
42
35 TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { 43 TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
36 webrtc::Config config; 44 webrtc::Config config;
37 MockInitialize mock(config); 45 MockInitialize mock(config);
38 ON_CALL(mock, InitializeLocked()) 46 ON_CALL(mock, InitializeLocked())
39 .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked)); 47 .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
40 48
41 EXPECT_CALL(mock, InitializeLocked()).Times(1); 49 EXPECT_CALL(mock, InitializeLocked()).Times(1);
42 mock.Initialize(); 50 mock.Initialize();
43 51
44 AudioFrame frame; 52 AudioFrame frame;
(...skipping 20 matching lines...) Expand all
65 frame.num_channels_ = 2; 73 frame.num_channels_ = 2;
66 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); 74 EXPECT_NOERR(mock.ProcessReverseStream(&frame));
67 75
68 // A new sample rate passed to ProcessReverseStream should cause an init. 76 // A new sample rate passed to ProcessReverseStream should cause an init.
69 SetFrameSampleRate(&frame, 16000); 77 SetFrameSampleRate(&frame, 16000);
70 EXPECT_CALL(mock, InitializeLocked()).Times(1); 78 EXPECT_CALL(mock, InitializeLocked()).Times(1);
71 EXPECT_NOERR(mock.ProcessReverseStream(&frame)); 79 EXPECT_NOERR(mock.ProcessReverseStream(&frame));
72 } 80 }
73 81
74 } // namespace webrtc 82 } // namespace webrtc
83
84 #endif
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