Remove VoERTP_RTCP

voice_engine/include/voe_rtp_rtcp.h

Index: voice_engine/include/voe_rtp_rtcp.h
diff --git a/voice_engine/include/voe_rtp_rtcp.h b/voice_engine/include/voe_rtp_rtcp.h
deleted file mode 100644
index e863a28666c05bbf2f8933086cfcc9a73124d7a5..0000000000000000000000000000000000000000
--- a/voice_engine/include/voe_rtp_rtcp.h
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
- *  Copyright (c) 2012 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:
-//
-//  - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
-//  - SSRC handling.
-//  - Transmission of RTCP sender reports.
-//  - Obtaining RTCP data from incoming RTCP sender reports.
-//  - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
-//  - Redundant Coding (RED)
-//  - Writing RTP and RTCP packets to binary files for off-line analysis of
-//    the call quality.
-//
-// Usage example, omitting error checking:
-//
-//  using namespace webrtc;
-//  VoiceEngine* voe = VoiceEngine::Create();
-//  VoEBase* base = VoEBase::GetInterface(voe);
-//  VoERTP_RTCP* rtp_rtcp  = VoERTP_RTCP::GetInterface(voe);
-//  base->Init();
-//  int ch = base->CreateChannel();
-//  ...
-//  rtp_rtcp->SetLocalSSRC(ch, 12345);
-//  ...
-//  base->DeleteChannel(ch);
-//  base->Terminate();
-//  base->Release();
-//  rtp_rtcp->Release();
-//  VoiceEngine::Delete(voe);
-//
-#ifndef VOICE_ENGINE_VOE_RTP_RTCP_H_
-#define VOICE_ENGINE_VOE_RTP_RTCP_H_
-
-#include <vector>
-#include "common_types.h"  // NOLINT(build/include)
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// VoERTPObserver
-class WEBRTC_DLLEXPORT VoERTPObserver {
- public:
-  virtual void OnIncomingCSRCChanged(int channel,
-                                     unsigned int CSRC,
-                                     bool added) = 0;
-
-  virtual void OnIncomingSSRCChanged(int channel, unsigned int SSRC) = 0;
-
- protected:
-  virtual ~VoERTPObserver() {}
-};
-
-// CallStatistics
-struct CallStatistics {
-  unsigned short fractionLost;
-  unsigned int cumulativeLost;
-  unsigned int extendedMax;
-  unsigned int jitterSamples;
-  int64_t rttMs;
-  size_t bytesSent;
-  int packetsSent;
-  size_t bytesReceived;
-  int packetsReceived;
-  // The capture ntp time (in local timebase) of the first played out audio
-  // frame.
-  int64_t capture_start_ntp_time_ms_;
-};
-
-// See section 6.4.1 in http://www.ietf.org/rfc/rfc3550.txt for details.
-struct SenderInfo {
-  uint32_t NTP_timestamp_high;
-  uint32_t NTP_timestamp_low;
-  uint32_t RTP_timestamp;
-  uint32_t sender_packet_count;
-  uint32_t sender_octet_count;
-};
-
-// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
-struct ReportBlock {
-  uint32_t sender_SSRC;  // SSRC of sender
-  uint32_t source_SSRC;
-  uint8_t fraction_lost;
-  uint32_t cumulative_num_packets_lost;
-  uint32_t extended_highest_sequence_number;
-  uint32_t interarrival_jitter;
-  uint32_t last_SR_timestamp;
-  uint32_t delay_since_last_SR;
-};
-
-// VoERTP_RTCP
-class WEBRTC_DLLEXPORT VoERTP_RTCP {
- public:
-  // Factory for the VoERTP_RTCP sub-API. Increases an internal
-  // reference counter if successful. Returns NULL if the API is not
-  // supported or if construction fails.
-  static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine);
-
-  // Releases the VoERTP_RTCP 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;
-
-  // Sets the local RTP synchronization source identifier (SSRC) explicitly.
-  virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
-
-  // Gets the local RTP SSRC of a specified |channel|.
-  virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
-
-  // Gets the SSRC of the incoming RTP packets.
-  virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0;
-
-  // Sets the status of rtp-audio-level-indication on a specific |channel|.
-  virtual int SetSendAudioLevelIndicationStatus(int channel,
-                                                bool enable,
-                                                unsigned char id = 1) = 0;
-
-  // Sets the RTCP status on a specific |channel|.
-  virtual int SetRTCPStatus(int channel, bool enable) = 0;
-
-  // Gets the RTCP status on a specific |channel|.
-  virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
-
-  // Sets the canonical name (CNAME) parameter for RTCP reports on a
-  // specific |channel|.
-  virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
-
-  // Gets the canonical name (CNAME) parameter for incoming RTCP reports
-  // on a specific channel.
-  virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0;
-
-  // Gets RTCP statistics for a specific |channel|.
-  virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
-
- protected:
-  VoERTP_RTCP() {}
-  virtual ~VoERTP_RTCP() {}
-};
-
-}  // namespace webrtc
-
-#endif  // #ifndef VOICE_ENGINE_VOE_RTP_RTCP_H_