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Side by Side Diff: webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h

Issue 1417173004: audio_coding: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Restored incorrectly renamed header guards and fixed an old error 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 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
13 13
14 #include <stdio.h> 14 #include <stdio.h>
15 #include <string.h> 15 #include <string.h>
16 16
17 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" 17 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
18 #include "webrtc/modules/audio_coding/main/test/ACMTest.h" 18 #include "webrtc/modules/audio_coding/main/test/ACMTest.h"
19 #include "webrtc/modules/audio_coding/main/test/PCMFile.h" 19 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
20 #include "webrtc/modules/audio_coding/main/test/RTPFile.h" 20 #include "webrtc/modules/audio_coding/main/test/RTPFile.h"
21 #include "webrtc/typedefs.h" 21 #include "webrtc/typedefs.h"
22 22
23 namespace webrtc { 23 namespace webrtc {
24 24
25 #define MAX_INCOMING_PAYLOAD 8096 25 #define MAX_INCOMING_PAYLOAD 8096
26 26
27 // TestPacketization callback which writes the encoded payloads to file 27 // TestPacketization callback which writes the encoded payloads to file
(...skipping 86 matching lines...) Expand 10 before | Expand all | Expand 10 after
114 int testMode); 114 int testMode);
115 115
116 protected: 116 protected:
117 Sender _sender; 117 Sender _sender;
118 Receiver _receiver; 118 Receiver _receiver;
119 }; 119 };
120 120
121 } // namespace webrtc 121 } // namespace webrtc
122 122
123 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ 123 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
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