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Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years 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/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/include/audio_coding_module_typedefs. h" 20 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
21 #include "webrtc/modules/audio_coding/main/test/utility.h" 21 #include "webrtc/modules/audio_coding/test/utility.h"
22 #include "webrtc/system_wrappers/include/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),
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829 printf("%s -> ", send_codec->plname); 829 printf("%s -> ", send_codec->plname);
830 } 830 }
831 CodecInst receive_codec; 831 CodecInst receive_codec;
832 acm_b_->ReceiveCodec(&receive_codec); 832 acm_b_->ReceiveCodec(&receive_codec);
833 if (test_mode_ != 0) { 833 if (test_mode_ != 0) {
834 printf("%s\n", receive_codec.plname); 834 printf("%s\n", receive_codec.plname);
835 } 835 }
836 } 836 }
837 837
838 } // namespace webrtc 838 } // namespace webrtc
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