Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(11)

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

Issue 2784873002: Delete unneeded includes of deprecated system_wrappers include files. (Closed)
Patch Set: Delete unneeded includes of system_wrappers/include/critical_section_wrapper.h Created 3 years, 8 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
OLDNEW
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/test/TestStereo.h" 11 #include "webrtc/modules/audio_coding/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 "webrtc/common_types.h" 17 #include "webrtc/common_types.h"
18 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" 18 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
19 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" 19 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
20 #include "webrtc/modules/audio_coding/test/utility.h" 20 #include "webrtc/modules/audio_coding/test/utility.h"
21 #include "webrtc/system_wrappers/include/trace.h"
22 #include "webrtc/test/gtest.h" 21 #include "webrtc/test/gtest.h"
23 #include "webrtc/test/testsupport/fileutils.h" 22 #include "webrtc/test/testsupport/fileutils.h"
24 #include "webrtc/typedefs.h" 23 #include "webrtc/typedefs.h"
25 24
26 namespace webrtc { 25 namespace webrtc {
27 26
28 // Class for simulating packet handling 27 // Class for simulating packet handling
29 TestPackStereo::TestPackStereo() 28 TestPackStereo::TestPackStereo()
30 : receiver_acm_(NULL), 29 : receiver_acm_(NULL),
31 seq_no_(0), 30 seq_no_(0),
(...skipping 814 matching lines...) Expand 10 before | Expand all | Expand 10 after
846 printf("%s -> ", send_codec->plname); 845 printf("%s -> ", send_codec->plname);
847 } 846 }
848 CodecInst receive_codec; 847 CodecInst receive_codec;
849 acm_b_->ReceiveCodec(&receive_codec); 848 acm_b_->ReceiveCodec(&receive_codec);
850 if (test_mode_ != 0) { 849 if (test_mode_ != 0) {
851 printf("%s\n", receive_codec.plname); 850 printf("%s\n", receive_codec.plname);
852 } 851 }
853 } 852 }
854 853
855 } // namespace webrtc 854 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/test/TestRedFec.cc ('k') | webrtc/modules/audio_coding/test/TwoWayCommunication.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698