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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/api/audio/audio_mixer.h" | 16 #include "webrtc/api/audio/audio_mixer.h" |
17 #include "webrtc/api/call/audio_sink.h" | 17 #include "webrtc/api/call/audio_sink.h" |
18 #include "webrtc/base/criticalsection.h" | 18 #include "webrtc/base/criticalsection.h" |
19 #include "webrtc/base/optional.h" | 19 #include "webrtc/base/optional.h" |
20 #include "webrtc/base/smoothing_filter.h" | |
20 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 21 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
21 #include "webrtc/common_types.h" | 22 #include "webrtc/common_types.h" |
22 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" | 23 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
23 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 24 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
24 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 25 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
25 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h" | 26 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h" |
26 #include "webrtc/modules/audio_processing/rms_level.h" | 27 #include "webrtc/modules/audio_processing/rms_level.h" |
27 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 28 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
28 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 29 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
29 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 30 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
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539 | 540 |
540 bool pacing_enabled_; | 541 bool pacing_enabled_; |
541 PacketRouter* packet_router_ = nullptr; | 542 PacketRouter* packet_router_ = nullptr; |
542 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 543 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
543 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 544 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
544 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 545 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
545 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 546 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
546 | 547 |
547 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 548 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
548 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 549 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
550 | |
551 SmoothingFilterImpl bitrate_bps_smoothed_; | |
minyue-webrtc
2016/11/17 12:04:02
better bitrate_bps_smoothed_ -> bitrate_smoother_
michaelt
2016/11/21 08:57:32
Done.
| |
549 }; | 552 }; |
550 | 553 |
551 } // namespace voe | 554 } // namespace voe |
552 } // namespace webrtc | 555 } // namespace webrtc |
553 | 556 |
554 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 557 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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