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diff --git a/webrtc/video/payload_router.h b/webrtc/video/payload_router.h
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+/*
+ *  Copyright (c) 2015 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_VIDEO_PAYLOAD_ROUTER_H_
+#define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
+
+#include <list>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_types.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class RTPFragmentationHeader;
+class RtpRtcp;
+struct RTPVideoHeader;
+
+// PayloadRouter routes outgoing data to the correct sending RTP module, based
+// on the simulcast layer in RTPVideoHeader.
+class PayloadRouter {
+ public:
+  PayloadRouter();
+  ~PayloadRouter();
+
+  static size_t DefaultMaxPayloadLength();
+
+  // Rtp modules are assumed to be sorted in simulcast index order.
+  void SetSendingRtpModules(const std::list<RtpRtcp*>& rtp_modules);
+
+  // PayloadRouter will only route packets if being active, all packets will be
+  // dropped otherwise.
+  void set_active(bool active);
+  bool active();
+
+  // Input parameters according to the signature of RtpRtcp::SendOutgoingData.
+  // Returns true if the packet was routed / sent, false otherwise.
+  bool RoutePayload(FrameType frame_type,
+                    int8_t payload_type,
+                    uint32_t time_stamp,
+                    int64_t capture_time_ms,
+                    const uint8_t* payload_data,
+                    size_t payload_size,
+                    const RTPFragmentationHeader* fragmentation,
+                    const RTPVideoHeader* rtp_video_hdr);
+
+  // Configures current target bitrate per module. 'stream_bitrates' is assumed
+  // to be in the same order as 'SetSendingRtpModules'.
+  void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates);
+
+  // Returns the maximum allowed data payload length, given the configured MTU
+  // and RTP headers.
+  size_t MaxPayloadLength() const;
+
+  void AddRef() { ++ref_count_; }
+  void Release() { if (--ref_count_ == 0) { delete this; } }
+
+ private:
+  // TODO(mflodman): When the new video API has launched, remove crit_ and
+  // assume rtp_modules_ will never change during a call.
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+
+  // Active sending RTP modules, in layer order.
+  std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
+  bool active_ GUARDED_BY(crit_.get());
+
+  Atomic32 ref_count_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_