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/video_coding/frame_object.h

@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2016 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_VIDEO_CODING_FRAME_OBJECT_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_
+
+#include "webrtc/api/optional.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class FrameObject : public webrtc::VCMEncodedFrame {
+ public:
+  static const uint8_t kMaxFrameReferences = 5;
+
+  FrameObject();
+  virtual ~FrameObject() {}
+
+  virtual bool GetBitstream(uint8_t* destination) const = 0;
+
+  // The capture timestamp of this frame.
+  virtual uint32_t Timestamp() const = 0;
+
+  // When this frame was received.
+  virtual int64_t ReceivedTime() const = 0;
+
+  // When this frame should be rendered.
+  virtual int64_t RenderTime() const = 0;
+
+  // This information is currently needed by the timing calculation class.
+  // TODO(philipel): Remove this function when a new timing class has
+  //                 been implemented.
+  virtual bool delayed_by_retransmission() const { return 0; }
+
+  size_t size() const { return _length; }
+
+  bool is_keyframe() const { return num_references == 0; }
+
+  // The tuple (|picture_id|, |spatial_layer|) uniquely identifies a frame
+  // object. For codec types that don't necessarily have picture ids they
+  // have to be constructed from the header data relevant to that codec.
+  int64_t picture_id;
+  uint8_t spatial_layer;
+  uint32_t timestamp;
+
+  // TODO(philipel): Add simple modify/access functions to prevent adding too
+  // many |references|.
+  size_t num_references;
+  int64_t references[kMaxFrameReferences];
+  bool inter_layer_predicted;
+};
+
+class PacketBuffer;
+
+class RtpFrameObject : public FrameObject {
+ public:
+  RtpFrameObject(PacketBuffer* packet_buffer,
+                 uint16_t first_seq_num,
+                 uint16_t last_seq_num,
+                 size_t frame_size,
+                 int times_nacked,
+                 int64_t received_time);
+
+  ~RtpFrameObject();
+  uint16_t first_seq_num() const;
+  uint16_t last_seq_num() const;
+  int times_nacked() const;
+  enum FrameType frame_type() const;
+  VideoCodecType codec_type() const;
+  bool GetBitstream(uint8_t* destination) const override;
+  uint32_t Timestamp() const override;
+  int64_t ReceivedTime() const override;
+  int64_t RenderTime() const override;
+  bool delayed_by_retransmission() const override;
+  rtc::Optional<RTPVideoTypeHeader> GetCodecHeader() const;
+
+ private:
+  rtc::scoped_refptr<PacketBuffer> packet_buffer_;
+  enum FrameType frame_type_;
+  VideoCodecType codec_type_;
+  uint16_t first_seq_num_;
+  uint16_t last_seq_num_;
+  uint32_t timestamp_;
+  int64_t received_time_;
+
+  // Equal to times nacked of the packet with the highet times nacked
+  // belonging to this frame.
+  int times_nacked_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_