Remove VoEVideoSync interface.

webrtc/voice_engine/include/voe_video_sync.h

-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-//  - RTP header modification (time stamp and sequence number fields).
-//  - Playout delay tuning to synchronize the voice with video.
-//  - Playout delay monitoring.
-//
-// Usage example, omitting error checking:
-//
-//  using namespace webrtc;
-//  VoiceEngine* voe = VoiceEngine::Create();
-//  VoEBase* base = VoEBase::GetInterface(voe);
-//  VoEVideoSync* vsync  = VoEVideoSync::GetInterface(voe);
-//  base->Init();
-//  ...
-//  int buffer_ms(0);
-//  vsync->GetPlayoutBufferSize(buffer_ms);
-//  ...
-//  base->Terminate();
-//  base->Release();
-//  vsync->Release();
-//  VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
-#define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class RtpReceiver;
-class RtpRtcp;
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEVideoSync {
- public:
-  // Factory for the VoEVideoSync sub-API. Increases an internal
-  // reference counter if successful. Returns NULL if the API is not
-  // supported or if construction fails.
-  static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine);
-
-  // Releases the VoEVideoSync sub-API and decreases an internal
-  // reference counter. Returns the new reference count. This value should
-  // be zero for all sub-API:s before the VoiceEngine object can be safely
-  // deleted.
-  virtual int Release() = 0;
-
-  // Gets the current sound card buffer size (playout delay).
-  virtual int GetPlayoutBufferSize(int& buffer_ms) = 0;
-
-  // Sets a minimum target delay for the jitter buffer. This delay is
-  // maintained by the jitter buffer, unless channel condition (jitter in
-  // inter-arrival times) dictates a higher required delay. The overall
-  // jitter buffer delay is max of |delay_ms| and the latency that NetEq
-  // computes based on inter-arrival times and its playout mode.
-  virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;
-
-  // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay), and
-  // the |playout_buffer_delay_ms| for a specified |channel|.
-  virtual int GetDelayEstimate(int channel,
-                               int* jitter_buffer_delay_ms,
-                               int* playout_buffer_delay_ms) = 0;
-
-  // Returns the least required jitter buffer delay. This is computed by the
-  // the jitter buffer based on the inter-arrival time of RTP packets and
-  // playout mode. NetEq maintains this latency unless a higher value is
-  // requested by calling SetMinimumPlayoutDelay().
-  virtual int GetLeastRequiredDelayMs(int channel) const = 0;
-
-  // Manual initialization of the RTP timestamp.
-  virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0;
-
-  // Manual initialization of the RTP sequence number.
-  virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0;
-
-  // Get the received RTP timestamp
-  virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0;
-
-  virtual int GetRtpRtcp(int channel,
-                         RtpRtcp** rtpRtcpModule,
-                         RtpReceiver** rtp_receiver) = 0;
-
- protected:
-  VoEVideoSync() {}
-  virtual ~VoEVideoSync() {}
-};
-
-}  // namespace webrtc
-
-#endif  // #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H