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

Issue 1421013006: Delete a chain of methods in ViE, VoE and ACM (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month 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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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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