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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 |
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371 | 371 |
372 // Set a RtcEventLog logging object. | 372 // Set a RtcEventLog logging object. |
373 void SetRtcEventLog(RtcEventLog* event_log); | 373 void SetRtcEventLog(RtcEventLog* event_log); |
374 | 374 |
375 void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats); | 375 void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats); |
376 void SetTransportOverhead(size_t transport_overhead_per_packet); | 376 void SetTransportOverhead(size_t transport_overhead_per_packet); |
377 | 377 |
378 // From OverheadObserver in the RTP/RTCP module | 378 // From OverheadObserver in the RTP/RTCP module |
379 void OnOverheadChanged(size_t overhead_bytes_per_packet) override; | 379 void OnOverheadChanged(size_t overhead_bytes_per_packet) override; |
380 | 380 |
381 protected: | 381 // The existence of this function alongside OnUplinkPacketLossRate is |
382 void OnIncomingFractionLoss(int fraction_lost); | 382 // a compromise. We want the encoder to be agnostic of the PLR source, but |
| 383 // we also don't want it to receive conflicting information from TWCC and |
| 384 // from RTCP-XR. |
| 385 void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate); |
383 | 386 |
384 private: | 387 private: |
| 388 void OnUplinkPacketLossRate(float packet_loss_rate); |
| 389 |
385 bool InputMute() const; | 390 bool InputMute() const; |
386 bool OnRtpPacketWithHeader(const uint8_t* received_packet, | 391 bool OnRtpPacketWithHeader(const uint8_t* received_packet, |
387 size_t length, | 392 size_t length, |
388 RTPHeader *header); | 393 RTPHeader *header); |
389 bool ReceivePacket(const uint8_t* packet, | 394 bool ReceivePacket(const uint8_t* packet, |
390 size_t packet_length, | 395 size_t packet_length, |
391 const RTPHeader& header, | 396 const RTPHeader& header, |
392 bool in_order); | 397 bool in_order); |
393 bool HandleRtxPacket(const uint8_t* packet, | 398 bool HandleRtxPacket(const uint8_t* packet, |
394 size_t packet_length, | 399 size_t packet_length, |
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501 PacketRouter* packet_router_ = nullptr; | 506 PacketRouter* packet_router_ = nullptr; |
502 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 507 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
503 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 508 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
504 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 509 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
505 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 510 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
506 | 511 |
507 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 512 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
508 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 513 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
509 | 514 |
510 rtc::ThreadChecker construction_thread_; | 515 rtc::ThreadChecker construction_thread_; |
| 516 |
| 517 const bool use_twcc_plr_for_ana_; |
511 }; | 518 }; |
512 | 519 |
513 } // namespace voe | 520 } // namespace voe |
514 } // namespace webrtc | 521 } // namespace webrtc |
515 | 522 |
516 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 523 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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