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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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326 const char* data, | 326 const char* data, |
327 unsigned short dataLengthInBytes); | 327 unsigned short dataLengthInBytes); |
328 int GetRTPStatistics(unsigned int& averageJitterMs, | 328 int GetRTPStatistics(unsigned int& averageJitterMs, |
329 unsigned int& maxJitterMs, | 329 unsigned int& maxJitterMs, |
330 unsigned int& discardedPackets); | 330 unsigned int& discardedPackets); |
331 int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks); | 331 int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks); |
332 int GetRTPStatistics(CallStatistics& stats); | 332 int GetRTPStatistics(CallStatistics& stats); |
333 int SetCodecFECStatus(bool enable); | 333 int SetCodecFECStatus(bool enable); |
334 bool GetCodecFECStatus(); | 334 bool GetCodecFECStatus(); |
335 void SetNACKStatus(bool enable, int maxNumberOfPackets); | 335 void SetNACKStatus(bool enable, int maxNumberOfPackets); |
| 336 void AdaptCodec(); |
336 | 337 |
337 // From AudioPacketizationCallback in the ACM | 338 // From AudioPacketizationCallback in the ACM |
338 int32_t SendData(FrameType frameType, | 339 int32_t SendData(FrameType frameType, |
339 uint8_t payloadType, | 340 uint8_t payloadType, |
340 uint32_t timeStamp, | 341 uint32_t timeStamp, |
341 const uint8_t* payloadData, | 342 const uint8_t* payloadData, |
342 size_t payloadSize, | 343 size_t payloadSize, |
343 const RTPFragmentationHeader* fragmentation) override; | 344 const RTPFragmentationHeader* fragmentation) override; |
344 | 345 |
345 // From ACMVADCallback in the ACM | 346 // From ACMVADCallback in the ACM |
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541 bool pacing_enabled_; | 542 bool pacing_enabled_; |
542 PacketRouter* packet_router_ = nullptr; | 543 PacketRouter* packet_router_ = nullptr; |
543 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 544 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
544 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 545 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
545 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 546 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
546 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 547 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
547 | 548 |
548 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 549 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
549 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 550 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
550 | 551 |
551 SmoothingFilterImpl bitrate_smoother_; | 552 rtc::CriticalSection bitrate_smoother_lock_; |
| 553 SmoothingFilterImpl bitrate_smoother_ GUARDED_BY(bitrate_smoother_lock_); |
552 }; | 554 }; |
553 | 555 |
554 } // namespace voe | 556 } // namespace voe |
555 } // namespace webrtc | 557 } // namespace webrtc |
556 | 558 |
557 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 559 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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