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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 "testing/gtest/include/gtest/gtest.h" | 18 #include "webrtc/test/gtest.h" |
| 19 #include "webrtc/common_types.h" | 19 #include "webrtc/common_types.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" | 21 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" |
| 22 #include "webrtc/modules/audio_coding/test/utility.h" | 22 #include "webrtc/modules/audio_coding/test/utility.h" |
| 23 #include "webrtc/system_wrappers/include/trace.h" | 23 #include "webrtc/system_wrappers/include/trace.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) |
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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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