Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(352)

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

Issue 1417683006: modules: more interface -> include renames (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix last incorrectly wrapped pragma message Created 5 years, 1 month 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) 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 "testing/gmock/include/gmock/gmock.h" 13 #include "testing/gmock/include/gmock/gmock.h"
14 #include "testing/gtest/include/gtest/gtest.h" 14 #include "testing/gtest/include/gtest/gtest.h"
15 #include "webrtc/config.h" 15 #include "webrtc/config.h"
16 #include "webrtc/modules/audio_processing/test/test_utils.h" 16 #include "webrtc/modules/audio_processing/test/test_utils.h"
17 #include "webrtc/modules/interface/module_common_types.h" 17 #include "webrtc/modules/include/module_common_types.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 23
24 class MockInitialize : public AudioProcessingImpl { 24 class MockInitialize : public AudioProcessingImpl {
25 public: 25 public:
26 explicit MockInitialize(const Config& config) : AudioProcessingImpl(config) { 26 explicit MockInitialize(const Config& config) : AudioProcessingImpl(config) {
27 } 27 }
(...skipping 39 matching lines...) Expand 10 before | Expand all | Expand 10 after
67 67
68 // A new sample rate passed to AnalyzeReverseStream should be an error and 68 // A new sample rate passed to AnalyzeReverseStream should be an error and
69 // not cause an init. 69 // not cause an init.
70 SetFrameSampleRate(&frame, 16000); 70 SetFrameSampleRate(&frame, 16000);
71 EXPECT_CALL(mock, InitializeLocked()) 71 EXPECT_CALL(mock, InitializeLocked())
72 .Times(0); 72 .Times(0);
73 EXPECT_EQ(mock.kBadSampleRateError, mock.AnalyzeReverseStream(&frame)); 73 EXPECT_EQ(mock.kBadSampleRateError, mock.AnalyzeReverseStream(&frame));
74 } 74 }
75 75
76 } // namespace webrtc 76 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698