OLD | NEW |
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 <algorithm> | 11 #include <algorithm> |
12 #include <limits> | 12 #include <limits> |
13 #include <list> | 13 #include <list> |
14 #include <memory> | 14 #include <memory> |
15 #include <numeric> | 15 #include <numeric> |
16 #include <string> | 16 #include <string> |
17 #include <vector> | 17 #include <vector> |
18 | 18 |
19 #include "webrtc/test/gmock.h" | |
20 #include "webrtc/test/gtest.h" | |
21 #include "webrtc/base/arraysize.h" | 19 #include "webrtc/base/arraysize.h" |
22 #include "webrtc/base/criticalsection.h" | 20 #include "webrtc/base/criticalsection.h" |
23 #include "webrtc/base/format_macros.h" | 21 #include "webrtc/base/format_macros.h" |
24 #include "webrtc/base/scoped_ref_ptr.h" | 22 #include "webrtc/base/scoped_ref_ptr.h" |
25 #include "webrtc/modules/audio_device/android/audio_common.h" | 23 #include "webrtc/modules/audio_device/android/audio_common.h" |
26 #include "webrtc/modules/audio_device/android/audio_manager.h" | 24 #include "webrtc/modules/audio_device/android/audio_manager.h" |
27 #include "webrtc/modules/audio_device/android/build_info.h" | 25 #include "webrtc/modules/audio_device/android/build_info.h" |
28 #include "webrtc/modules/audio_device/android/ensure_initialized.h" | 26 #include "webrtc/modules/audio_device/android/ensure_initialized.h" |
29 #include "webrtc/modules/audio_device/audio_device_impl.h" | 27 #include "webrtc/modules/audio_device/audio_device_impl.h" |
30 #include "webrtc/modules/audio_device/include/audio_device.h" | 28 #include "webrtc/modules/audio_device/include/audio_device.h" |
31 #include "webrtc/system_wrappers/include/clock.h" | 29 #include "webrtc/system_wrappers/include/clock.h" |
32 #include "webrtc/system_wrappers/include/event_wrapper.h" | 30 #include "webrtc/system_wrappers/include/event_wrapper.h" |
33 #include "webrtc/system_wrappers/include/sleep.h" | 31 #include "webrtc/system_wrappers/include/sleep.h" |
| 32 #include "webrtc/test/gmock.h" |
| 33 #include "webrtc/test/gtest.h" |
34 #include "webrtc/test/testsupport/fileutils.h" | 34 #include "webrtc/test/testsupport/fileutils.h" |
35 | 35 |
36 using std::cout; | 36 using std::cout; |
37 using std::endl; | 37 using std::endl; |
38 using ::testing::_; | 38 using ::testing::_; |
39 using ::testing::AtLeast; | 39 using ::testing::AtLeast; |
40 using ::testing::Gt; | 40 using ::testing::Gt; |
41 using ::testing::Invoke; | 41 using ::testing::Invoke; |
42 using ::testing::NiceMock; | 42 using ::testing::NiceMock; |
43 using ::testing::NotNull; | 43 using ::testing::NotNull; |
(...skipping 1008 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
1052 StopPlayout(); | 1052 StopPlayout(); |
1053 StopRecording(); | 1053 StopRecording(); |
1054 // Verify that the correct number of transmitted impulses are detected. | 1054 // Verify that the correct number of transmitted impulses are detected. |
1055 EXPECT_EQ(latency_audio_stream->num_latency_values(), | 1055 EXPECT_EQ(latency_audio_stream->num_latency_values(), |
1056 static_cast<size_t>( | 1056 static_cast<size_t>( |
1057 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1)); | 1057 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1)); |
1058 latency_audio_stream->PrintResults(); | 1058 latency_audio_stream->PrintResults(); |
1059 } | 1059 } |
1060 | 1060 |
1061 } // namespace webrtc | 1061 } // namespace webrtc |
OLD | NEW |