Packet buffer for the new jitter buffer.

webrtc/modules/video_coding/packet_buffer.cc

Index: webrtc/modules/video_coding/packet_buffer.cc
diff --git a/webrtc/modules/video_coding/packet_buffer.cc b/webrtc/modules/video_coding/packet_buffer.cc
new file mode 100644
index 0000000000000000000000000000000000000000..128f6b32fdd3edc89b6c2b845411731129bca6a9
--- /dev/null
+++ b/webrtc/modules/video_coding/packet_buffer.cc
@@ -0,0 +1,150 @@
+/*
+ *  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/packet_buffer.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "webrtc/base/mod_ops.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_coding/frame_object.h"
+
+namespace webrtc {
+namespace video_coding {
+
+PacketBuffer::PacketBuffer(int start_buffer_size,
+                           int max_buffer_size,
+                           OnCompleteFrameCallback* frame_callback) :
+    size_(start_buffer_size),
+    max_size_(max_buffer_size),
+    clear_up_to_(0),
+    initialized_(false),
+    data_buffer_(start_buffer_size),
+    sequence_buffer_(start_buffer_size),
+    frame_callback_(frame_callback) {
+      RTC_DCHECK_LE(start_buffer_size, max_buffer_size);

[stefan-webrtc, 2016/03/17 12:06:31]

Indentation, will probably be fixed with git cl format.

[philipel, 2016/03/17 15:00:35]

Formated, lets hope for the best :)

+      // Buffer size must always be a power of 2.
+      RTC_DCHECK((start_buffer_size & start_buffer_size - 1) == 0);
+      RTC_DCHECK((max_buffer_size & max_buffer_size - 1) == 0);
+}
+
+bool PacketBuffer::InsertPacket(const VCMPacket& packet) {
+  rtc::CritScope lock(&crit_);
+  uint16_t seq_num = packet.seqNum;
+  int index = seq_num % size_;
+
+  if (!initialized_) {
+    clear_up_to_ = seq_num;
+    initialized_ = true;
+  }
+
+  if (sequence_buffer_[index].used) {
+    // Duplicate packet, do nothing.
+    if (data_buffer_[index].seqNum == packet.seqNum)
+      return false;
+
+    // The Packet Buffer is full, try to expand the buffer.

[stefan-webrtc, 2016/03/17 12:06:31]

PacketBuffer or "packet buffer"

[philipel, 2016/03/17 15:00:35]

Done.

+    while (ExpandBufferSize() && sequence_buffer_[seq_num % size_].used) {}
+    index = seq_num % size_;
+    // If still full, test if the old packet can be discarded (overwritten
+    // later in the code), if not, return false.
+    if (sequence_buffer_[index].used &&
+        AheadOrAt(sequence_buffer_[index].seq_num, clear_up_to_))
+      return false;
+  }
+
+  sequence_buffer_[index].frame_begin = packet.isFirstPacket;
+  sequence_buffer_[index].frame_end = packet.markerBit;
+  sequence_buffer_[index].seq_num = packet.seqNum;
+  sequence_buffer_[index].continuous = false;
+  sequence_buffer_[index].used = true;
+  data_buffer_[index] = packet;
+
+  FindCompleteFrames(seq_num);
+  return true;
+}
+
+void PacketBuffer::ClearUpTo(uint16_t seq_num) {
+  rtc::CritScope lock(&crit_);
+  clear_up_to_ = seq_num;
+}
+
+bool PacketBuffer::ExpandBufferSize() {
+  if (size_ == max_size_) {
+    return false;
+  }
+
+  int new_size = std::min(max_size_, 2*size_);
+  std::vector<VCMPacket> new_data_buffer(new_size);
+  std::vector<ContinuityInfo> new_sequence_buffer(new_size);
+  for (int i = 0; i < size_; ++i) {
+    if (sequence_buffer_[i].used) {
+      int index = sequence_buffer_[i].seq_num % new_size;
+      new_sequence_buffer[index] = sequence_buffer_[i];
+      new_data_buffer[index] = data_buffer_[i];
+    }
+  }
+  size_ = new_size;
+  sequence_buffer_ = std::move(new_sequence_buffer);
+  data_buffer_ = std::move(new_data_buffer);
+  return true;
+}
+
+bool PacketBuffer::IsContinous(uint16_t seq_num) const {
+  int index = seq_num % size_;
+  int prev_index = index > 0 ? index - 1 : size_- 1;
+  if (!sequence_buffer_[index].used)
+    return false;
+  if (sequence_buffer_[index].frame_begin)
+    return true;
+  if (!sequence_buffer_[prev_index].used)
+    return false;
+  if (sequence_buffer_[prev_index].continuous)
+    return true;
+
+  return false;
+}
+
+void PacketBuffer::FindCompleteFrames(uint16_t seq_num) {
+  int index = seq_num % size_;
+  while (IsContinous(seq_num)) {

[stefan-webrtc, 2016/03/17 12:06:31]

Do we always have to deliver continuous frames (based on sequence numbers) to
the frame buffer? Doesn't that mean we can't skip incomplete frames in, e.g.,
TL1 as long as we have a complete TL0 sequence?

[philipel, 2016/03/17 15:00:35]

This function only test the continuity between packets within a frame, so frames
are still assembled out of order.

In short, as soon as a frame is complete a frame object is created and the
callback is called. This logic will have to be changed slightly in the case of
not having a picture id available.

+    sequence_buffer_[index].continuous = true;
+
+    // If the frame is complete, find the first packet of the frame and
+    // create a FrameObject.
+    if (sequence_buffer_[index].frame_end) {
+      int rindex = index;
+      uint16_t start_seq_num = seq_num;
+      while (!sequence_buffer_[rindex].frame_begin) {
+        rindex = rindex > 0 ? rindex - 1 : size_ - 1;
+        start_seq_num--;
+      }
+
+      std::unique_ptr<FrameObject> frame(
+        new RtpFrameObject(this, start_seq_num, seq_num));
+      frame_callback_->OnCompleteFrame(std::move(frame));
+    }
+
+  index = (index + 1) % size_;
+  ++seq_num;
+  }
+}
+
+void PacketBuffer::Flush() {
+  rtc::CritScope lock(&crit_);
+  for (int i = 0; i < size_; ++i) {
+    sequence_buffer_[i].used = false;
+    sequence_buffer_[i].continuous = false;
+  }
+}
+
+}  // namespace video_coding
+}  // namespace webrtc