Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(638)

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

Issue 1457023002: Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 "webrtc/modules/audio_processing/audio_processing_impl.h" 11 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <vector> 14 #include <vector>
15 15
16 #include "testing/gtest/include/gtest/gtest.h" 16 #include "testing/gtest/include/gtest/gtest.h"
17 #include "webrtc/base/array_view.h" 17 #include "webrtc/base/array_view.h"
18 #include "webrtc/base/criticalsection.h" 18 #include "webrtc/base/criticalsection.h"
19 #include "webrtc/base/event.h" 19 #include "webrtc/base/event.h"
20 #include "webrtc/base/platform_thread.h" 20 #include "webrtc/base/platform_thread.h"
21 #include "webrtc/base/random.h"
21 #include "webrtc/config.h" 22 #include "webrtc/config.h"
22 #include "webrtc/modules/audio_processing/test/test_utils.h" 23 #include "webrtc/modules/audio_processing/test/test_utils.h"
23 #include "webrtc/modules/include/module_common_types.h" 24 #include "webrtc/modules/include/module_common_types.h"
24 #include "webrtc/system_wrappers/include/sleep.h" 25 #include "webrtc/system_wrappers/include/sleep.h"
25 #include "webrtc/test/random.h"
26 26
27 namespace webrtc { 27 namespace webrtc {
28 28
29 namespace { 29 namespace {
30 30
31 class AudioProcessingImplLockTest; 31 class AudioProcessingImplLockTest;
32 32
33 // Type of the render thread APM API call to use in the test. 33 // Type of the render thread APM API call to use in the test.
34 enum class RenderApiImpl { 34 enum class RenderApiImpl {
35 ProcessReverseStreamImpl1, 35 ProcessReverseStreamImpl1,
(...skipping 41 matching lines...) Expand 10 before | Expand all | Expand 10 after
77 return rand_gen_.Rand(max); 77 return rand_gen_.Rand(max);
78 } 78 }
79 79
80 float RandFloat() { 80 float RandFloat() {
81 rtc::CritScope cs(&crit_); 81 rtc::CritScope cs(&crit_);
82 return rand_gen_.Rand<float>(); 82 return rand_gen_.Rand<float>();
83 } 83 }
84 84
85 private: 85 private:
86 rtc::CriticalSection crit_; 86 rtc::CriticalSection crit_;
87 test::Random rand_gen_ GUARDED_BY(crit_); 87 Random rand_gen_ GUARDED_BY(crit_);
88 }; 88 };
89 89
90 // Variables related to the audio data and formats. 90 // Variables related to the audio data and formats.
91 struct AudioFrameData { 91 struct AudioFrameData {
92 explicit AudioFrameData(int max_frame_size) { 92 explicit AudioFrameData(int max_frame_size) {
93 // Set up the two-dimensional arrays needed for the APM API calls. 93 // Set up the two-dimensional arrays needed for the APM API calls.
94 input_framechannels.resize(2 * max_frame_size); 94 input_framechannels.resize(2 * max_frame_size);
95 input_frame.resize(2); 95 input_frame.resize(2);
96 input_frame[0] = &input_framechannels[0]; 96 input_frame[0] = &input_framechannels[0];
97 input_frame[1] = &input_framechannels[max_frame_size]; 97 input_frame[1] = &input_framechannels[max_frame_size];
(...skipping 1026 matching lines...) Expand 10 before | Expand all | Expand 10 after
1124 DISABLED_AudioProcessingImplLockExtensive, 1124 DISABLED_AudioProcessingImplLockExtensive,
1125 AudioProcessingImplLockTest, 1125 AudioProcessingImplLockTest,
1126 ::testing::ValuesIn(TestConfig::GenerateExtensiveTestConfigs())); 1126 ::testing::ValuesIn(TestConfig::GenerateExtensiveTestConfigs()));
1127 1127
1128 INSTANTIATE_TEST_CASE_P( 1128 INSTANTIATE_TEST_CASE_P(
1129 AudioProcessingImplLockBrief, 1129 AudioProcessingImplLockBrief,
1130 AudioProcessingImplLockTest, 1130 AudioProcessingImplLockTest,
1131 ::testing::ValuesIn(TestConfig::GenerateBriefTestConfigs())); 1131 ::testing::ValuesIn(TestConfig::GenerateBriefTestConfigs()));
1132 1132
1133 } // namespace webrtc 1133 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/call/rtc_event_log_unittest.cc ('k') | webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698