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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_VOICE_ENGINE_CHANNEL_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/audio_sink.h" | 16 #include "webrtc/api/call/audio_sink.h" |
17 #include "webrtc/base/criticalsection.h" | 17 #include "webrtc/base/criticalsection.h" |
18 #include "webrtc/base/optional.h" | 18 #include "webrtc/base/optional.h" |
19 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 19 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
20 #include "webrtc/common_types.h" | 20 #include "webrtc/common_types.h" |
21 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" | 21 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
22 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 22 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
24 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" | 24 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" |
25 #include "webrtc/modules/audio_processing/rms_level.h" | 25 #include "webrtc/modules/audio_processing/rms_level.h" |
26 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 26 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
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588 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 588 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
589 | 589 |
590 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 590 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
591 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 591 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
592 }; | 592 }; |
593 | 593 |
594 } // namespace voe | 594 } // namespace voe |
595 } // namespace webrtc | 595 } // namespace webrtc |
596 | 596 |
597 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 597 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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