Packet buffer for the new jitter buffer.

webrtc/modules/video_coding/frame_object.cc

Index: webrtc/modules/video_coding/frame_object.cc
diff --git a/webrtc/modules/video_coding/frame_object.cc b/webrtc/modules/video_coding/frame_object.cc
new file mode 100644
index 0000000000000000000000000000000000000000..d5562dc9d95cc7b1c14bd89cfc0fa164a02262dc
--- /dev/null
+++ b/webrtc/modules/video_coding/frame_object.cc
@@ -0,0 +1,75 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/video_coding/frame_object.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/modules/video_coding/packet_buffer.h"
+
+namespace webrtc {
+namespace video_coding {
+
+RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
+                               uint16_t picture_id,
+                               uint16_t first_packet,
+                               uint16_t last_packet)
+    : packet_buffer_(packet_buffer),
+      first_packet_(first_packet),
+      last_packet_(last_packet) {}
+
+RtpFrameObject::~RtpFrameObject() {
+  rtc::CritScope lock(&packet_buffer_->crit_);
+  int index = first_packet_ % packet_buffer_->size_;
+  int end = ++last_packet_ % packet_buffer_->size_;
+  uint16_t seq_num = first_packet_;
+  while (index != end) {
+    if (packet_buffer_->sequence_buffer_[index].seq_num == seq_num) {
+      packet_buffer_->sequence_buffer_[index].used = false;
+      packet_buffer_->sequence_buffer_[index].continuous = false;
+    }
+    index = (index + 1) % packet_buffer_->size_;
+    ++seq_num;
+  }
+}
+
+uint16_t RtpFrameObject::first_packet() const {
+  return first_packet_;
+}
+
+uint16_t RtpFrameObject::last_packet() const {
+  return last_packet_;
+}
+
+uint16_t RtpFrameObject::picture_id() const {
+  return picture_id_;
+}
+
+bool RtpFrameObject::GetBitstream(uint8_t* destination) const {
+  rtc::CritScope lock(&packet_buffer_->crit_);
+
+  int index = first_packet_ % packet_buffer_->size_;
+  int end = last_packet_ + 1 % packet_buffer_->size_;
+  uint16_t seq_num = first_packet_;
+  while (index != end) {
+    if (!packet_buffer_->sequence_buffer_[index].used ||
+        packet_buffer_->sequence_buffer_[index].seq_num != seq_num)
+      return false;
+
+    const uint8_t* source = packet_buffer_->data_buffer_[index].dataPtr;
+    size_t length = packet_buffer_->data_buffer_[index].sizeBytes;
+    memcpy(destination, source, length);
+    destination += length;
+    index = (index + 1) % packet_buffer_->size_;
+    ++seq_num;
+  }
+  return true;
+}
+
+}  // namespace video_coding
+}  // namespace webrtc