Packet buffer for the new jitter buffer.

webrtc/modules/video_coding/packet_buffer.h

+/*
+ *  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_PACKET_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
+
+#include <vector>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/video_coding/packet.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class FrameObject;
+class RtpFrameObject;
+
+class OnCompleteFrameCallback {
+ public:
+  virtual ~OnCompleteFrameCallback() {}
+  virtual void OnCompleteFrame(std::unique_ptr<FrameObject> frame) = 0;
+};
+
+class PacketBuffer {
+ public:
+  // Both |start_buffer_size| and |max_buffer_size| must be a power of 2.
+  PacketBuffer(size_t start_buffer_size,
+               size_t max_buffer_size,
+               OnCompleteFrameCallback* frame_callback);
+
+  bool InsertPacket(const VCMPacket& packet);
+  void ClearTo(uint16_t seq_num);
+  void Flush();
+
+ private:
+  friend RtpFrameObject;
+  // Since we want the packet buffer to be as packet type agnostic
+  // as possible we extract only the information needed in order
+  // to determin whether a sequence of packets is continuous or not.
+  struct ContinuityInfo {
+    uint16_t seq_num = 0;
+    bool frame_begin = false;
+    bool frame_end = false;
+    bool used = false;
+    bool continuous = false;
+  };
+
+  bool ExpandBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  bool IsContinuous(uint16_t seq_num) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  void FindCompleteFrames(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination);
+  void ReturnFrame(RtpFrameObject* frame);
+
+  rtc::CriticalSection crit_;
+
+  // Buffer size_ and max_size_ must always be a power of two.
+  size_t size_ GUARDED_BY(crit_);
+  const size_t max_size_;
+
+  uint16_t last_seq_num_ GUARDED_BY(crit_);
+  uint16_t first_seq_num_ GUARDED_BY(crit_);
+  bool initialized_ GUARDED_BY(crit_);
+  std::vector<VCMPacket> data_buffer_ GUARDED_BY(crit_);
+  std::vector<ContinuityInfo> sequence_buffer_ GUARDED_BY(crit_);
+
+  OnCompleteFrameCallback* const frame_callback_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_