New jitter buffer experiment.

webrtc/modules/video_coding/packet_buffer.cc

 /*
  *  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.
  */
 
@@ skipping 39 matching lines @@
   // 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);
 }
 
 PacketBuffer::~PacketBuffer() {
   Clear();
 }
 
 bool PacketBuffer::InsertPacket(const VCMPacket& packet) {
-  rtc::CritScope lock(&crit_);
-  uint16_t seq_num = packet.seqNum;
-  size_t index = seq_num % size_;
+  std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
+  {
+    rtc::CritScope lock(&crit_);
+    uint16_t seq_num = packet.seqNum;
+    size_t index = seq_num % size_;
 
-  if (!first_packet_received_) {
-    first_seq_num_ = seq_num;
-    last_seq_num_ = seq_num;
-    first_packet_received_ = true;
-  } else if (AheadOf(first_seq_num_, seq_num)) {
-    // If we have explicitly cleared past this packet then it's old,
-    // don't insert it.
-    if (is_cleared_to_first_seq_num_)
-      return false;
+    if (!first_packet_received_) {
+      first_seq_num_ = seq_num;
+      last_seq_num_ = seq_num;
+      first_packet_received_ = true;
+    } else if (AheadOf(first_seq_num_, seq_num)) {
+      // If we have explicitly cleared past this packet then it's old,
+      // don't insert it.
+      if (is_cleared_to_first_seq_num_)
+        return false;
 
-    first_seq_num_ = seq_num;
+      first_seq_num_ = seq_num;
+    }
+
+    if (sequence_buffer_[index].used) {

[nisse-webrtc, 2016/11/15 08:36:29]

I suspect the sequence_buffer_ logic could be simplified a bit (or maybe it's
just using different conventions than what I'm familiar with), but that might
not fit with this cl.

[stefan-webrtc, 2016/11/15 08:55:48]

Could you follow-up on this offline? It would be nice to simplify if possible!

+      // Duplicate packet, do nothing.
+      if (data_buffer_[index].seqNum == packet.seqNum)
+        return true;
+
+      // The packet buffer is full, try to expand the buffer.
+      while (ExpandBufferSize() && sequence_buffer_[seq_num % size_].used) {
+      }
+      index = seq_num % size_;
+
+      // Packet buffer is still full.
+      if (sequence_buffer_[index].used)
+        return false;
+    }
+
+    if (AheadOf(seq_num, last_seq_num_))
+      last_seq_num_ = seq_num;
+
+    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].frame_created = false;
+    sequence_buffer_[index].used = true;
+    data_buffer_[index] = packet;
+
+    found_frames = FindFrames(seq_num);
   }
 
-  if (sequence_buffer_[index].used) {
-    // Duplicate packet, do nothing.
-    if (data_buffer_[index].seqNum == packet.seqNum)
-      return true;
+  for (std::unique_ptr<RtpFrameObject>& frame : found_frames)
+    received_frame_callback_->OnReceivedFrame(std::move(frame));
 
-    // The packet buffer is full, try to expand the buffer.
-    while (ExpandBufferSize() && sequence_buffer_[seq_num % size_].used) {
-    }
-    index = seq_num % size_;
-
-    // Packet buffer is still full.
-    if (sequence_buffer_[index].used)
-      return false;
-  }
-
-  if (AheadOf(seq_num, last_seq_num_))
-    last_seq_num_ = seq_num;
-
-  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].frame_created = false;
-  sequence_buffer_[index].used = true;
-  data_buffer_[index] = packet;
-
-  FindFrames(seq_num);
   return true;
 }
 
 void PacketBuffer::ClearTo(uint16_t seq_num) {
   rtc::CritScope lock(&crit_);
 
   // If the packet buffer was cleared between a frame was created and returned.
   if (!first_packet_received_)
     return;
 
@@ skipping 66 matching lines @@
   if (sequence_buffer_[prev_index].seq_num !=
       sequence_buffer_[index].seq_num - 1) {
     return false;
   }
   if (sequence_buffer_[prev_index].continuous)
     return true;
 
   return false;
 }
 
-void PacketBuffer::FindFrames(uint16_t seq_num) {
+std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames(
+    uint16_t seq_num) {
+  std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
   while (PotentialNewFrame(seq_num)) {
     size_t index = seq_num % size_;
     sequence_buffer_[index].continuous = true;
 
     // If all packets of the frame is continuous, find the first packet of the
     // frame and create an RtpFrameObject.
     if (sequence_buffer_[index].frame_end) {
       size_t frame_size = 0;
       int max_nack_count = -1;
       uint16_t start_seq_num = seq_num;
 
       // Find the start index by searching backward until the packet with
       // the |frame_begin| flag is set.
       int start_index = index;
       while (true) {
         frame_size += data_buffer_[start_index].sizeBytes;
-        max_nack_count = std::max(
-            max_nack_count, data_buffer_[start_index].timesNacked);
+        max_nack_count =
+            std::max(max_nack_count, data_buffer_[start_index].timesNacked);
         sequence_buffer_[start_index].frame_created = true;
 
         if (sequence_buffer_[start_index].frame_begin)
           break;
 
         start_index = start_index > 0 ? start_index - 1 : size_ - 1;
         start_seq_num--;
       }
 
-      std::unique_ptr<RtpFrameObject> frame(
+      found_frames.emplace_back(
           new RtpFrameObject(this, start_seq_num, seq_num, frame_size,
                              max_nack_count, clock_->TimeInMilliseconds()));
-
-      received_frame_callback_->OnReceivedFrame(std::move(frame));
     }
-
     ++seq_num;
   }
+  return found_frames;
 }
 
 void PacketBuffer::ReturnFrame(RtpFrameObject* frame) {
   rtc::CritScope lock(&crit_);
   size_t index = frame->first_seq_num() % size_;
   size_t end = (frame->last_seq_num() + 1) % size_;
   uint16_t seq_num = frame->first_seq_num();
   while (index != end) {
     if (sequence_buffer_[index].seq_num == seq_num) {
       delete[] data_buffer_[index].dataPtr;
@@ skipping 23 matching lines @@
     size_t length = data_buffer_[index].sizeBytes;
     memcpy(destination, source, length);
     destination += length;
     index = (index + 1) % size_;
     ++seq_num;
   }
   return true;
 }
 
 VCMPacket* PacketBuffer::GetPacket(uint16_t seq_num) {
-  rtc::CritScope lock(&crit_);
   size_t index = seq_num % size_;
   if (!sequence_buffer_[index].used ||
       seq_num != sequence_buffer_[index].seq_num) {
     return nullptr;
   }
   return &data_buffer_[index];
 }
 
 int PacketBuffer::AddRef() const {
   return rtc::AtomicOps::Increment(&ref_count_);
 }
 
 int PacketBuffer::Release() const {
   int count = rtc::AtomicOps::Decrement(&ref_count_);
   if (!count) {
     delete this;
   }
   return count;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc