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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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| 273 int GetChannelOutputVolumeScaling(float& scaling) const; | 273 int GetChannelOutputVolumeScaling(float& scaling) const; |
| 274 | 274 |
| 275 // VoENetEqStats | 275 // VoENetEqStats |
| 276 int GetNetworkStatistics(NetworkStatistics& stats); | 276 int GetNetworkStatistics(NetworkStatistics& stats); |
| 277 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; | 277 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; |
| 278 | 278 |
| 279 // VoEVideoSync | 279 // VoEVideoSync |
| 280 bool GetDelayEstimate(int* jitter_buffer_delay_ms, | 280 bool GetDelayEstimate(int* jitter_buffer_delay_ms, |
| 281 int* playout_buffer_delay_ms) const; | 281 int* playout_buffer_delay_ms) const; |
| 282 int LeastRequiredDelayMs() const; | 282 int LeastRequiredDelayMs() const; |
| 283 int SetInitialPlayoutDelay(int delay_ms); | |
| 284 int SetMinimumPlayoutDelay(int delayMs); | 283 int SetMinimumPlayoutDelay(int delayMs); |
| 285 int GetPlayoutTimestamp(unsigned int& timestamp); | 284 int GetPlayoutTimestamp(unsigned int& timestamp); |
| 286 int SetInitTimestamp(unsigned int timestamp); | 285 int SetInitTimestamp(unsigned int timestamp); |
| 287 int SetInitSequenceNumber(short sequenceNumber); | 286 int SetInitSequenceNumber(short sequenceNumber); |
| 288 | 287 |
| 289 // VoEVideoSyncExtended | 288 // VoEVideoSyncExtended |
| 290 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; | 289 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
| 291 | 290 |
| 292 // VoEDtmf | 291 // VoEDtmf |
| 293 int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs, | 292 int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs, |
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| 583 rtc::scoped_ptr<NetworkPredictor> network_predictor_; | 582 rtc::scoped_ptr<NetworkPredictor> network_predictor_; |
| 584 // An associated send channel. | 583 // An associated send channel. |
| 585 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; | 584 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; |
| 586 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); | 585 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); |
| 587 }; | 586 }; |
| 588 | 587 |
| 589 } // namespace voe | 588 } // namespace voe |
| 590 } // namespace webrtc | 589 } // namespace webrtc |
| 591 | 590 |
| 592 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 591 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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