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Side by Side Diff: webrtc/modules/audio_coding/main/test/TestStereo.cc

Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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_coding/main/test/TestStereo.h" 11 #include "webrtc/modules/audio_coding/main/test/TestStereo.h"
12 12
13 #include <assert.h> 13 #include <assert.h>
14 14
15 #include <string> 15 #include <string>
16 16
17 #include "testing/gtest/include/gtest/gtest.h" 17 #include "testing/gtest/include/gtest/gtest.h"
18 #include "webrtc/common_types.h" 18 #include "webrtc/common_types.h"
19 #include "webrtc/engine_configurations.h" 19 #include "webrtc/engine_configurations.h"
20 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedef s.h" 20 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedef s.h"
21 #include "webrtc/modules/audio_coding/main/test/utility.h" 21 #include "webrtc/modules/audio_coding/main/test/utility.h"
22 #include "webrtc/system_wrappers/interface/trace.h" 22 #include "webrtc/system_wrappers/include/trace.h"
23 #include "webrtc/test/testsupport/fileutils.h" 23 #include "webrtc/test/testsupport/fileutils.h"
24 24
25 namespace webrtc { 25 namespace webrtc {
26 26
27 // Class for simulating packet handling 27 // Class for simulating packet handling
28 TestPackStereo::TestPackStereo() 28 TestPackStereo::TestPackStereo()
29 : receiver_acm_(NULL), 29 : receiver_acm_(NULL),
30 seq_no_(0), 30 seq_no_(0),
31 timestamp_diff_(0), 31 timestamp_diff_(0),
32 last_in_timestamp_(0), 32 last_in_timestamp_(0),
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828 if (test_mode_ != 0) { 828 if (test_mode_ != 0) {
829 printf("%s -> ", my_codec_param.plname); 829 printf("%s -> ", my_codec_param.plname);
830 } 830 }
831 acm_b_->ReceiveCodec(&my_codec_param); 831 acm_b_->ReceiveCodec(&my_codec_param);
832 if (test_mode_ != 0) { 832 if (test_mode_ != 0) {
833 printf("%s\n", my_codec_param.plname); 833 printf("%s\n", my_codec_param.plname);
834 } 834 }
835 } 835 }
836 836
837 } // namespace webrtc 837 } // namespace webrtc
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