Chromium Code Reviews| OLD | NEW |
|---|---|
| 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 |
| (...skipping 16 matching lines...) Expand all Loading... | |
| 27 #include "webrtc/modules/audio_processing/rms_level.h" | 27 #include "webrtc/modules/audio_processing/rms_level.h" |
| 28 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 28 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
| 29 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 29 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| 30 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 30 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| 31 #include "webrtc/voice_engine/audio_level.h" | 31 #include "webrtc/voice_engine/audio_level.h" |
| 32 #include "webrtc/voice_engine/file_player.h" | 32 #include "webrtc/voice_engine/file_player.h" |
| 33 #include "webrtc/voice_engine/file_recorder.h" | 33 #include "webrtc/voice_engine/file_recorder.h" |
| 34 #include "webrtc/voice_engine/include/voe_base.h" | 34 #include "webrtc/voice_engine/include/voe_base.h" |
| 35 #include "webrtc/voice_engine/include/voe_network.h" | 35 #include "webrtc/voice_engine/include/voe_network.h" |
| 36 #include "webrtc/voice_engine/shared_data.h" | 36 #include "webrtc/voice_engine/shared_data.h" |
| 37 #include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h" | |
|
the sun
2017/03/22 12:06:42
No need to include this here.
elad.alon_webrtc.org
2017/03/22 14:34:47
Done. (FYI, relic from an older revision where Cha
| |
| 37 #include "webrtc/voice_engine/voice_engine_defines.h" | 38 #include "webrtc/voice_engine/voice_engine_defines.h" |
| 38 | 39 |
| 39 namespace rtc { | 40 namespace rtc { |
| 40 class TimestampWrapAroundHandler; | 41 class TimestampWrapAroundHandler; |
| 41 } | 42 } |
| 42 | 43 |
| 43 namespace webrtc { | 44 namespace webrtc { |
| 44 | 45 |
| 45 class AudioDeviceModule; | 46 class AudioDeviceModule; |
| 46 class FileWrapper; | 47 class FileWrapper; |
| (...skipping 324 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 371 | 372 |
| 372 // Set a RtcEventLog logging object. | 373 // Set a RtcEventLog logging object. |
| 373 void SetRtcEventLog(RtcEventLog* event_log); | 374 void SetRtcEventLog(RtcEventLog* event_log); |
| 374 | 375 |
| 375 void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats); | 376 void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats); |
| 376 void SetTransportOverhead(size_t transport_overhead_per_packet); | 377 void SetTransportOverhead(size_t transport_overhead_per_packet); |
| 377 | 378 |
| 378 // From OverheadObserver in the RTP/RTCP module | 379 // From OverheadObserver in the RTP/RTCP module |
| 379 void OnOverheadChanged(size_t overhead_bytes_per_packet) override; | 380 void OnOverheadChanged(size_t overhead_bytes_per_packet) override; |
| 380 | 381 |
| 382 // Note: The existence of this function alongside OnUplinkPacketLossRate is | |
|
the sun
2017/03/22 12:06:42
"Note:" is superfluous.
elad.alon_webrtc.org
2017/03/22 14:34:47
I've removed, but IMHO, it made it immediately cle
| |
| 383 // a compromise. We want the encoder to be agnostic of the PLR source, but | |
| 384 // we also don't want it to receive conflicting information from TWCC and | |
| 385 // from RTCP-XR. | |
| 386 void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate); | |
| 387 | |
| 381 protected: | 388 protected: |
|
the sun
2017/03/22 12:06:42
Nobody inherits from this class so you should move
elad.alon_webrtc.org
2017/03/22 14:34:47
Done.
| |
| 382 void OnIncomingFractionLoss(int fraction_lost); | 389 void OnUplinkPacketLossRate(float packet_loss_rate); |
| 383 | 390 |
| 384 private: | 391 private: |
| 385 bool InputMute() const; | 392 bool InputMute() const; |
| 386 bool OnRtpPacketWithHeader(const uint8_t* received_packet, | 393 bool OnRtpPacketWithHeader(const uint8_t* received_packet, |
| 387 size_t length, | 394 size_t length, |
| 388 RTPHeader *header); | 395 RTPHeader *header); |
| 389 bool ReceivePacket(const uint8_t* packet, | 396 bool ReceivePacket(const uint8_t* packet, |
| 390 size_t packet_length, | 397 size_t packet_length, |
| 391 const RTPHeader& header, | 398 const RTPHeader& header, |
| 392 bool in_order); | 399 bool in_order); |
| (...skipping 108 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 501 PacketRouter* packet_router_ = nullptr; | 508 PacketRouter* packet_router_ = nullptr; |
| 502 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 509 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
| 503 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 510 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
| 504 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 511 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
| 505 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 512 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
| 506 | 513 |
| 507 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 514 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
| 508 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 515 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
| 509 | 516 |
| 510 rtc::ThreadChecker construction_thread_; | 517 rtc::ThreadChecker construction_thread_; |
| 518 | |
| 519 const bool use_twcc_plr_for_ana_; | |
| 511 }; | 520 }; |
| 512 | 521 |
| 513 } // namespace voe | 522 } // namespace voe |
| 514 } // namespace webrtc | 523 } // namespace webrtc |
| 515 | 524 |
| 516 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 525 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
| OLD | NEW |