Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/rtp_rtcp/include/rtp_receiver.h

@@ -0,0 +1,100 @@
+/*
+ *  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_
+
+#include <vector>
+
+#include "webrtc/api/rtpreceiverinterface.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct CodecInst;
+class RTPPayloadRegistry;
+class VideoCodec;
+
+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,
+      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 CodecInst& audio_codec) = 0;
+  // Registers a receive payload in the payload registry.
+  virtual int32_t RegisterReceivePayload(const VideoCodec& video_codec) = 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;
+
+  // 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;
+
+  virtual std::vector<RtpSource> GetSources() const = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_