Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(548)

Side by Side Diff: webrtc/modules/audio_device/audio_device_unittest.cc

Issue 3011943002: Move optional.h to webrtc/api/ (Closed)
Patch Set: Created 3 years, 3 months 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
« no previous file with comments | « webrtc/modules/audio_device/BUILD.gn ('k') | webrtc/modules/audio_processing/BUILD.gn » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2017 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 <cstring> 12 #include <cstring>
13 #include <numeric> 13 #include <numeric>
14 14
15 #include "webrtc/api/array_view.h" 15 #include "webrtc/api/array_view.h"
16 #include "webrtc/api/optional.h"
16 #include "webrtc/modules/audio_device/audio_device_impl.h" 17 #include "webrtc/modules/audio_device/audio_device_impl.h"
17 #include "webrtc/modules/audio_device/include/audio_device.h" 18 #include "webrtc/modules/audio_device/include/audio_device.h"
18 #include "webrtc/modules/audio_device/include/mock_audio_transport.h" 19 #include "webrtc/modules/audio_device/include/mock_audio_transport.h"
19 #include "webrtc/rtc_base/buffer.h" 20 #include "webrtc/rtc_base/buffer.h"
20 #include "webrtc/rtc_base/criticalsection.h" 21 #include "webrtc/rtc_base/criticalsection.h"
21 #include "webrtc/rtc_base/event.h" 22 #include "webrtc/rtc_base/event.h"
22 #include "webrtc/rtc_base/logging.h" 23 #include "webrtc/rtc_base/logging.h"
23 #include "webrtc/rtc_base/optional.h"
24 #include "webrtc/rtc_base/race_checker.h" 24 #include "webrtc/rtc_base/race_checker.h"
25 #include "webrtc/rtc_base/safe_conversions.h" 25 #include "webrtc/rtc_base/safe_conversions.h"
26 #include "webrtc/rtc_base/scoped_ref_ptr.h" 26 #include "webrtc/rtc_base/scoped_ref_ptr.h"
27 #include "webrtc/rtc_base/thread_annotations.h" 27 #include "webrtc/rtc_base/thread_annotations.h"
28 #include "webrtc/rtc_base/thread_checker.h" 28 #include "webrtc/rtc_base/thread_checker.h"
29 #include "webrtc/rtc_base/timeutils.h" 29 #include "webrtc/rtc_base/timeutils.h"
30 #include "webrtc/test/gmock.h" 30 #include "webrtc/test/gmock.h"
31 #include "webrtc/test/gtest.h" 31 #include "webrtc/test/gtest.h"
32 32
33 using ::testing::_; 33 using ::testing::_;
(...skipping 656 matching lines...) Expand 10 before | Expand all | Expand 10 after
690 StopPlayout(); 690 StopPlayout();
691 // Verify that the correct number of transmitted impulses are detected. 691 // Verify that the correct number of transmitted impulses are detected.
692 EXPECT_EQ(audio_stream.num_latency_values(), 692 EXPECT_EQ(audio_stream.num_latency_values(),
693 static_cast<size_t>( 693 static_cast<size_t>(
694 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1)); 694 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
695 // Print out min, max and average delay values for debugging purposes. 695 // Print out min, max and average delay values for debugging purposes.
696 audio_stream.PrintResults(); 696 audio_stream.PrintResults();
697 } 697 }
698 698
699 } // namespace webrtc 699 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_device/BUILD.gn ('k') | webrtc/modules/audio_processing/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698