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

Issue 1376673004: Add a PacketOptions struct to webrtc::Transport. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comment added Created 5 years, 2 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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371 uint32_t rate) override; 371 uint32_t rate) override;
372 void OnIncomingSSRCChanged(uint32_t ssrc) override; 372 void OnIncomingSSRCChanged(uint32_t ssrc) override;
373 void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override; 373 void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
374 374
375 // From RtpAudioFeedback in the RTP/RTCP module 375 // From RtpAudioFeedback in the RTP/RTCP module
376 void OnPlayTelephoneEvent(uint8_t event, 376 void OnPlayTelephoneEvent(uint8_t event,
377 uint16_t lengthMs, 377 uint16_t lengthMs,
378 uint8_t volume) override; 378 uint8_t volume) override;
379 379
380 // From Transport (called by the RTP/RTCP module) 380 // From Transport (called by the RTP/RTCP module)
381 bool SendRtp(const uint8_t* data, size_t len) override; 381 bool SendRtp(const uint8_t* data,
382 size_t len,
383 const PacketOptions& packet_options) override;
382 bool SendRtcp(const uint8_t* data, size_t len) override; 384 bool SendRtcp(const uint8_t* data, size_t len) override;
383 385
384 // From MixerParticipant 386 // From MixerParticipant
385 int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override; 387 int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override;
386 int32_t NeededFrequency(int32_t id) const override; 388 int32_t NeededFrequency(int32_t id) const override;
387 389
388 // From FileCallback 390 // From FileCallback
389 void PlayNotification(int32_t id, uint32_t durationMs) override; 391 void PlayNotification(int32_t id, uint32_t durationMs) override;
390 void RecordNotification(int32_t id, uint32_t durationMs) override; 392 void RecordNotification(int32_t id, uint32_t durationMs) override;
391 void PlayFileEnded(int32_t id) override; 393 void PlayFileEnded(int32_t id) override;
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581 rtc::scoped_ptr<NetworkPredictor> network_predictor_; 583 rtc::scoped_ptr<NetworkPredictor> network_predictor_;
582 // An associated send channel. 584 // An associated send channel.
583 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; 585 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
584 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); 586 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
585 }; 587 };
586 588
587 } // namespace voe 589 } // namespace voe
588 } // namespace webrtc 590 } // namespace webrtc
589 591
590 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 592 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
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