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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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