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Side by Side Diff: webrtc/voice_engine/channel.h

Issue 1213603002: Remove ResetStatistics from RTP feedback. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: comment + rebase Created 5 years, 5 months ago
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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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361 361
362 // From RtpFeedback in the RTP/RTCP module 362 // From RtpFeedback in the RTP/RTCP module
363 int32_t OnInitializeDecoder(int32_t id, 363 int32_t OnInitializeDecoder(int32_t id,
364 int8_t payloadType, 364 int8_t payloadType,
365 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 365 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
366 int frequency, 366 int frequency,
367 uint8_t channels, 367 uint8_t channels,
368 uint32_t rate) override; 368 uint32_t rate) override;
369 void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override; 369 void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override;
370 void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override; 370 void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override;
371 void ResetStatistics(uint32_t ssrc) override;
372 371
373 // From RtpAudioFeedback in the RTP/RTCP module 372 // From RtpAudioFeedback in the RTP/RTCP module
374 void OnPlayTelephoneEvent(int32_t id, 373 void OnPlayTelephoneEvent(int32_t id,
375 uint8_t event, 374 uint8_t event,
376 uint16_t lengthMs, 375 uint16_t lengthMs,
377 uint8_t volume) override; 376 uint8_t volume) override;
378 377
379 // From Transport (called by the RTP/RTCP module) 378 // From Transport (called by the RTP/RTCP module)
380 int SendPacket(int /*channel*/, const void* data, size_t len) override; 379 int SendPacket(int /*channel*/, const void* data, size_t len) override;
381 int SendRTCPPacket(int /*channel*/, const void* data, size_t len) override; 380 int SendRTCPPacket(int /*channel*/, const void* data, size_t len) override;
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579 rtc::scoped_ptr<NetworkPredictor> network_predictor_; 578 rtc::scoped_ptr<NetworkPredictor> network_predictor_;
580 // An associated send channel. 579 // An associated send channel.
581 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; 580 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
582 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); 581 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
583 }; 582 };
584 583
585 } // namespace voe 584 } // namespace voe
586 } // namespace webrtc 585 } // namespace webrtc
587 586
588 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 587 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
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