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/source/rtp_receiver_strategy.h

@@ -0,0 +1,99 @@
+/*
+ *  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_SOURCE_RTP_RECEIVER_STRATEGY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct CodecInst;
+
+class TelephoneEventHandler;
+
+// This strategy deals with media-specific RTP packet processing.
+// This class is not thread-safe and must be protected by its caller.
+class RTPReceiverStrategy {
+ public:
+  static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
+  static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback);
+
+  virtual ~RTPReceiverStrategy() {}
+
+  // Parses the RTP packet and calls the data callback with the payload data.
+  // Implementations are encouraged to use the provided packet buffer and RTP
+  // header as arguments to the callback; implementations are also allowed to
+  // make changes in the data as necessary. The specific_payload argument
+  // provides audio or video-specific data. The is_first_packet argument is true
+  // if this packet is either the first packet ever or the first in its frame.
+  virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
+                                 const PayloadUnion& specific_payload,
+                                 bool is_red,
+                                 const uint8_t* payload,
+                                 size_t payload_length,
+                                 int64_t timestamp_ms,
+                                 bool is_first_packet) = 0;
+
+  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
+
+  // Computes the current dead-or-alive state.
+  virtual RTPAliveType ProcessDeadOrAlive(
+      uint16_t last_payload_length) const = 0;
+
+  // Returns true if we should report CSRC changes for this payload type.
+  // TODO(phoglund): should move out of here along with other payload stuff.
+  virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
+
+  // Notifies the strategy that we have created a new non-RED audio payload type
+  // in the payload registry.
+  virtual int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) = 0;
+
+  // Invokes the OnInitializeDecoder callback in a media-specific way.
+  virtual int32_t InvokeOnInitializeDecoder(
+      RtpFeedback* callback,
+      int8_t payload_type,
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const PayloadUnion& specific_payload) const = 0;
+
+  // Checks if the payload type has changed, and returns whether we should
+  // reset statistics and/or discard this packet.
+  virtual void CheckPayloadChanged(int8_t payload_type,
+                                   PayloadUnion* specific_payload,
+                                   bool* should_discard_changes);
+
+  virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
+
+  // Stores / retrieves the last media specific payload for later reference.
+  void GetLastMediaSpecificPayload(PayloadUnion* payload) const;
+  void SetLastMediaSpecificPayload(const PayloadUnion& payload);
+
+ protected:
+  // The data callback is where we should send received payload data.
+  // See ParseRtpPacket. This class does not claim ownership of the callback.
+  // Implementations must NOT hold any critical sections while calling the
+  // callback.
+  //
+  // Note: Implementations may call the callback for other reasons than calls
+  // to ParseRtpPacket, for instance if the implementation somehow recovers a
+  // packet.
+  explicit RTPReceiverStrategy(RtpData* data_callback);
+
+  rtc::CriticalSection crit_sect_;
+  PayloadUnion last_payload_;
+  RtpData* data_callback_;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_