Revert of Merge webrtc/video_engine/ into webrtc/video/

webrtc/video/payload_router.h

Index: webrtc/video/payload_router.h
diff --git a/webrtc/video/payload_router.h b/webrtc/video/payload_router.h
deleted file mode 100644
index 881145976d13bd28639f1474c0e56613899bdb5c..0000000000000000000000000000000000000000
--- a/webrtc/video/payload_router.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- *  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_