Remove interface directories kept to avoid breaking downstream.

webrtc/modules/rtp_rtcp/interface/rtp_receiver.h

-/*
- *  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.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class RTPPayloadRegistry;
-
-class TelephoneEventHandler {
- public:
-  virtual ~TelephoneEventHandler() {}
-
-  // The following three methods implement the TelephoneEventHandler interface.
-  // Forward DTMFs to decoder for playout.
-  virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
-
-  // Is forwarding of outband telephone events turned on/off?
-  virtual bool TelephoneEventForwardToDecoder() const = 0;
-
-  // Is TelephoneEvent configured with payload type payload_type
-  virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
-};
-
-class RtpReceiver {
- public:
-  // Creates a video-enabled RTP receiver.
-  static RtpReceiver* CreateVideoReceiver(
-      Clock* clock,
-      RtpData* incoming_payload_callback,
-      RtpFeedback* incoming_messages_callback,
-      RTPPayloadRegistry* rtp_payload_registry);
-
-  // Creates an audio-enabled RTP receiver.
-  static RtpReceiver* CreateAudioReceiver(
-      Clock* clock,
-      RtpAudioFeedback* incoming_audio_feedback,
-      RtpData* incoming_payload_callback,
-      RtpFeedback* incoming_messages_callback,
-      RTPPayloadRegistry* rtp_payload_registry);
-
-  virtual ~RtpReceiver() {}
-
-  // Returns a TelephoneEventHandler if available.
-  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
-
-  // Registers a receive payload in the payload registry and notifies the media
-  // receiver strategy.
-  virtual int32_t RegisterReceivePayload(
-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-      const int8_t payload_type,
-      const uint32_t frequency,
-      const uint8_t channels,
-      const uint32_t rate) = 0;
-
-  // De-registers |payload_type| from the payload registry.
-  virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
-
-  // Parses the media specific parts of an RTP packet and updates the receiver
-  // state. This for instance means that any changes in SSRC and payload type is
-  // detected and acted upon.
-  virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
-                                 const uint8_t* payload,
-                                 size_t payload_length,
-                                 PayloadUnion payload_specific,
-                                 bool in_order) = 0;
-
-  // Returns the currently configured NACK method.
-  virtual NACKMethod NACK() const = 0;
-
-  // Turn negative acknowledgement (NACK) requests on/off.
-  virtual void SetNACKStatus(const NACKMethod method) = 0;
-
-  // Gets the last received timestamp. Returns true if a packet has been
-  // received, false otherwise.
-  virtual bool Timestamp(uint32_t* timestamp) const = 0;
-  // Gets the time in milliseconds when the last timestamp was received.
-  // Returns true if a packet has been received, false otherwise.
-  virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0;
-
-  // Returns the remote SSRC of the currently received RTP stream.
-  virtual uint32_t SSRC() const = 0;
-
-  // Returns the current remote CSRCs.
-  virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
-
-  // Returns the current energy of the RTP stream received.
-  virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_