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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/test/EncodeDecodeTest.h" | 11 #include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h" |
12 | 12 |
13 #include <memory> | 13 #include <memory> |
14 #include <sstream> | 14 #include <sstream> |
15 #include <stdio.h> | 15 #include <stdio.h> |
16 #include <stdlib.h> | 16 #include <stdlib.h> |
17 | 17 |
18 #include "webrtc/test/gtest.h" | |
19 #include "webrtc/common_types.h" | 18 #include "webrtc/common_types.h" |
| 19 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" |
20 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 20 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
21 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" | |
22 #include "webrtc/modules/audio_coding/test/utility.h" | 21 #include "webrtc/modules/audio_coding/test/utility.h" |
23 #include "webrtc/system_wrappers/include/trace.h" | 22 #include "webrtc/system_wrappers/include/trace.h" |
| 23 #include "webrtc/test/gtest.h" |
24 #include "webrtc/test/testsupport/fileutils.h" | 24 #include "webrtc/test/testsupport/fileutils.h" |
25 | 25 |
26 namespace webrtc { | 26 namespace webrtc { |
27 | 27 |
28 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) | 28 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) |
29 : _rtpStream(rtpStream), | 29 : _rtpStream(rtpStream), |
30 _frequency(frequency), | 30 _frequency(frequency), |
31 _seqNo(0) { | 31 _seqNo(0) { |
32 } | 32 } |
33 | 33 |
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350 if (acm->SendCodec()) { | 350 if (acm->SendCodec()) { |
351 _sender.Run(); | 351 _sender.Run(); |
352 } | 352 } |
353 _sender.Teardown(); | 353 _sender.Teardown(); |
354 rtpFile.Close(); | 354 rtpFile.Close(); |
355 | 355 |
356 return fileName; | 356 return fileName; |
357 } | 357 } |
358 | 358 |
359 } // namespace webrtc | 359 } // namespace webrtc |
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