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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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