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| 1 /* | 1 /* |
| 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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/neteq/tools/neteq_performance_test.h" | 11 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h" |
| 12 | 12 |
| 13 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" | 13 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" |
| 14 #include "webrtc/modules/audio_coding/neteq/interface/neteq.h" | 14 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" |
| 15 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" | 15 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" |
| 16 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" | 16 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" |
| 17 #include "webrtc/system_wrappers/include/clock.h" | 17 #include "webrtc/system_wrappers/include/clock.h" |
| 18 #include "webrtc/test/testsupport/fileutils.h" | 18 #include "webrtc/test/testsupport/fileutils.h" |
| 19 #include "webrtc/typedefs.h" | 19 #include "webrtc/typedefs.h" |
| 20 | 20 |
| 21 using webrtc::NetEq; | 21 using webrtc::NetEq; |
| 22 using webrtc::test::AudioLoop; | 22 using webrtc::test::AudioLoop; |
| 23 using webrtc::test::RtpGenerator; | 23 using webrtc::test::RtpGenerator; |
| 24 using webrtc::WebRtcRTPHeader; | 24 using webrtc::WebRtcRTPHeader; |
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| 122 drift_flipped = true; | 122 drift_flipped = true; |
| 123 } | 123 } |
| 124 } | 124 } |
| 125 int64_t end_time_ms = clock->TimeInMilliseconds(); | 125 int64_t end_time_ms = clock->TimeInMilliseconds(); |
| 126 delete neteq; | 126 delete neteq; |
| 127 return end_time_ms - start_time_ms; | 127 return end_time_ms - start_time_ms; |
| 128 } | 128 } |
| 129 | 129 |
| 130 } // namespace test | 130 } // namespace test |
| 131 } // namespace webrtc | 131 } // namespace webrtc |
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