modules/video_coding/packet_buffer.cc - src - Git at Google

 /*
  *  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 "modules/video_coding/packet_buffer.h"

 #include <string.h>

 #include <algorithm>
 #include <cstdint>
 #include <utility>

 #include "absl/types/variant.h"
 #include "api/video/encoded_frame.h"
 #include "common_video/h264/h264_common.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 #include "modules/video_coding/codecs/h264/include/h264_globals.h"
 #include "modules/video_coding/frame_object.h"
 #include "rtc_base/atomic_ops.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/mod_ops.h"
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/field_trial.h"

 namespace webrtc {
 namespace video_coding {

 rtc::scoped_refptr<PacketBuffer> PacketBuffer::Create(
     Clock* clock,
     size_t start_buffer_size,
     size_t max_buffer_size,
     OnAssembledFrameCallback* assembled_frame_callback) {
   return rtc::scoped_refptr<PacketBuffer>(new PacketBuffer(
       clock, start_buffer_size, max_buffer_size, assembled_frame_callback));
 }

 PacketBuffer::PacketBuffer(Clock* clock,
                            size_t start_buffer_size,
                            size_t max_buffer_size,
                            OnAssembledFrameCallback* assembled_frame_callback)
     : clock_(clock),
       size_(start_buffer_size),
       max_size_(max_buffer_size),
       first_seq_num_(0),
       first_packet_received_(false),
       is_cleared_to_first_seq_num_(false),
       data_buffer_(start_buffer_size),
       sequence_buffer_(start_buffer_size),
       assembled_frame_callback_(assembled_frame_callback),
       unique_frames_seen_(0),
       sps_pps_idr_is_h264_keyframe_(
           field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
   RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
   // 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(VCMPacket* packet) {
   std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
   {
     rtc::CritScope lock(&crit_);

     OnTimestampReceived(packet->timestamp);

     uint16_t seq_num = packet->seqNum;
     size_t index = seq_num % size_;

     if (!first_packet_received_) {
       first_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, just silently ignore it.
       if (is_cleared_to_first_seq_num_) {
         delete[] packet->dataPtr;
         packet->dataPtr = nullptr;
         return true;
       }

       first_seq_num_ = seq_num;
     }

     if (sequence_buffer_[index].used) {
       // Duplicate packet, just delete the payload.
       if (data_buffer_[index].seqNum == packet->seqNum) {
         delete[] packet->dataPtr;
         packet->dataPtr = nullptr;
         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 since we were unable to expand the buffer.
       if (sequence_buffer_[index].used) {
         // Clear the buffer, delete payload, and return false to signal that a
         // new keyframe is needed.
         RTC_LOG(LS_WARNING) << "Clear PacketBuffer and request key frame.";
         Clear();
         delete[] packet->dataPtr;
         packet->dataPtr = nullptr;
         return false;
       }
     }

     sequence_buffer_[index].frame_begin = packet->is_first_packet_in_frame();
     sequence_buffer_[index].frame_end = packet->is_last_packet_in_frame();
     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;
     packet->dataPtr = nullptr;

     UpdateMissingPackets(packet->seqNum);

     int64_t now_ms = clock_->TimeInMilliseconds();
     last_received_packet_ms_ = now_ms;
     if (packet->video_header.frame_type == VideoFrameType::kVideoFrameKey)
       last_received_keyframe_packet_ms_ = now_ms;

     found_frames = FindFrames(seq_num);
   }

   for (std::unique_ptr<RtpFrameObject>& frame : found_frames)
     assembled_frame_callback_->OnAssembledFrame(std::move(frame));

   return true;
 }

 void PacketBuffer::ClearTo(uint16_t seq_num) {
   rtc::CritScope lock(&crit_);
   // We have already cleared past this sequence number, no need to do anything.
   if (is_cleared_to_first_seq_num_ &&
       AheadOf<uint16_t>(first_seq_num_, seq_num)) {
     return;
   }

   // If the packet buffer was cleared between a frame was created and returned.
   if (!first_packet_received_)
     return;

   // Avoid iterating over the buffer more than once by capping the number of
   // iterations to the |size_| of the buffer.
   ++seq_num;
   size_t diff = ForwardDiff<uint16_t>(first_seq_num_, seq_num);
   size_t iterations = std::min(diff, size_);
   for (size_t i = 0; i < iterations; ++i) {
     size_t index = first_seq_num_ % size_;
     RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num);
     if (AheadOf<uint16_t>(seq_num, sequence_buffer_[index].seq_num)) {
       delete[] data_buffer_[index].dataPtr;
       data_buffer_[index].dataPtr = nullptr;
       sequence_buffer_[index].used = false;
     }
     ++first_seq_num_;
   }

   // If |diff| is larger than |iterations| it means that we don't increment
   // |first_seq_num_| until we reach |seq_num|, so we set it here.
   first_seq_num_ = seq_num;

   is_cleared_to_first_seq_num_ = true;
   auto clear_to_it = missing_packets_.upper_bound(seq_num);
   if (clear_to_it != missing_packets_.begin()) {
     --clear_to_it;
     missing_packets_.erase(missing_packets_.begin(), clear_to_it);
   }
 }

 void PacketBuffer::Clear() {
   rtc::CritScope lock(&crit_);
   for (size_t i = 0; i < size_; ++i) {
     delete[] data_buffer_[i].dataPtr;
     data_buffer_[i].dataPtr = nullptr;
     sequence_buffer_[i].used = false;
   }

   first_packet_received_ = false;
   is_cleared_to_first_seq_num_ = false;
   last_received_packet_ms_.reset();
   last_received_keyframe_packet_ms_.reset();
   newest_inserted_seq_num_.reset();
   missing_packets_.clear();
 }

 void PacketBuffer::PaddingReceived(uint16_t seq_num) {
   std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
   {
     rtc::CritScope lock(&crit_);
     UpdateMissingPackets(seq_num);
     found_frames = FindFrames(static_cast<uint16_t>(seq_num + 1));
   }

   for (std::unique_ptr<RtpFrameObject>& frame : found_frames)
     assembled_frame_callback_->OnAssembledFrame(std::move(frame));
 }

 absl::optional<int64_t> PacketBuffer::LastReceivedPacketMs() const {
   rtc::CritScope lock(&crit_);
   return last_received_packet_ms_;
 }

 absl::optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
   rtc::CritScope lock(&crit_);
   return last_received_keyframe_packet_ms_;
 }

 int PacketBuffer::GetUniqueFramesSeen() const {
   rtc::CritScope lock(&crit_);
   return unique_frames_seen_;
 }

 bool PacketBuffer::ExpandBufferSize() {
   if (size_ == max_size_) {
     RTC_LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_
                         << "), failed to increase size.";
     return false;
   }

   size_t 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 (size_t i = 0; i < size_; ++i) {
     if (sequence_buffer_[i].used) {
       size_t 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);
   RTC_LOG(LS_INFO) << "PacketBuffer size expanded to " << new_size;
   return true;
 }

 bool PacketBuffer::PotentialNewFrame(uint16_t seq_num) const {
   size_t index = seq_num % size_;
   int prev_index = index > 0 ? index - 1 : size_ - 1;

   if (!sequence_buffer_[index].used)
     return false;
   if (sequence_buffer_[index].seq_num != seq_num)
     return false;
   if (sequence_buffer_[index].frame_created)
     return false;
   if (sequence_buffer_[index].frame_begin)
     return true;
   if (!sequence_buffer_[prev_index].used)
     return false;
   if (sequence_buffer_[prev_index].frame_created)
     return false;
   if (sequence_buffer_[prev_index].seq_num !=
       static_cast<uint16_t>(sequence_buffer_[index].seq_num - 1)) {
     return false;
   }
   if (data_buffer_[prev_index].timestamp != data_buffer_[index].timestamp)
     return false;
   if (sequence_buffer_[prev_index].continuous)
     return true;

   return false;
 }

 std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames(
     uint16_t seq_num) {
   std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
   for (size_t i = 0; i < size_ && PotentialNewFrame(seq_num); ++i) {
     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;
       int64_t min_recv_time = data_buffer_[index].packet_info.receive_time_ms();
       int64_t max_recv_time = data_buffer_[index].packet_info.receive_time_ms();
       RtpPacketInfos::vector_type packet_infos;

       // Find the start index by searching backward until the packet with
       // the |frame_begin| flag is set.
       int start_index = index;
       size_t tested_packets = 0;
       int64_t frame_timestamp = data_buffer_[start_index].timestamp;

       // Identify H.264 keyframes by means of SPS, PPS, and IDR.
       bool is_h264 = data_buffer_[start_index].codec() == kVideoCodecH264;
       bool has_h264_sps = false;
       bool has_h264_pps = false;
       bool has_h264_idr = false;
       bool is_h264_keyframe = false;

       while (true) {
         ++tested_packets;
         frame_size += data_buffer_[start_index].sizeBytes;
         max_nack_count =
             std::max(max_nack_count, data_buffer_[start_index].timesNacked);
         sequence_buffer_[start_index].frame_created = true;

         min_recv_time =
             std::min(min_recv_time,
                      data_buffer_[start_index].packet_info.receive_time_ms());
         max_recv_time =
             std::max(max_recv_time,
                      data_buffer_[start_index].packet_info.receive_time_ms());

         // Should use |push_front()| since the loop traverses backwards. But
         // it's too inefficient to do so on a vector so we'll instead fix the
         // order afterwards.
         packet_infos.push_back(data_buffer_[start_index].packet_info);

         if (!is_h264 && sequence_buffer_[start_index].frame_begin)
           break;

         if (is_h264 && !is_h264_keyframe) {
           const auto* h264_header = absl::get_if<RTPVideoHeaderH264>(
               &data_buffer_[start_index].video_header.video_type_header);
           if (!h264_header || h264_header->nalus_length >= kMaxNalusPerPacket)
             return found_frames;

           for (size_t j = 0; j < h264_header->nalus_length; ++j) {
             if (h264_header->nalus[j].type == H264::NaluType::kSps) {
               has_h264_sps = true;
             } else if (h264_header->nalus[j].type == H264::NaluType::kPps) {
               has_h264_pps = true;
             } else if (h264_header->nalus[j].type == H264::NaluType::kIdr) {
               has_h264_idr = true;
             }
           }
           if ((sps_pps_idr_is_h264_keyframe_ && has_h264_idr && has_h264_sps &&
                has_h264_pps) ||
               (!sps_pps_idr_is_h264_keyframe_ && has_h264_idr)) {
             is_h264_keyframe = true;
           }
         }

         if (tested_packets == size_)
           break;

         start_index = start_index > 0 ? start_index - 1 : size_ - 1;

         // In the case of H264 we don't have a frame_begin bit (yes,
         // |frame_begin| might be set to true but that is a lie). So instead
         // we traverese backwards as long as we have a previous packet and
         // the timestamp of that packet is the same as this one. This may cause
         // the PacketBuffer to hand out incomplete frames.
         // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7106
         if (is_h264 &&
             (!sequence_buffer_[start_index].used ||
              data_buffer_[start_index].timestamp != frame_timestamp)) {
           break;
         }

         --start_seq_num;
       }

       // Fix the order since the packet-finding loop traverses backwards.
       std::reverse(packet_infos.begin(), packet_infos.end());

       if (is_h264) {
         // Warn if this is an unsafe frame.
         if (has_h264_idr && (!has_h264_sps || !has_h264_pps)) {
           RTC_LOG(LS_WARNING)
               << "Received H.264-IDR frame "
               << "(SPS: " << has_h264_sps << ", PPS: " << has_h264_pps
               << "). Treating as "
               << (sps_pps_idr_is_h264_keyframe_ ? "delta" : "key")
               << " frame since WebRTC-SpsPpsIdrIsH264Keyframe is "
               << (sps_pps_idr_is_h264_keyframe_ ? "enabled." : "disabled");
         }

         // Now that we have decided whether to treat this frame as a key frame
         // or delta frame in the frame buffer, we update the field that
         // determines if the RtpFrameObject is a key frame or delta frame.
         const size_t first_packet_index = start_seq_num % size_;
         RTC_CHECK_LT(first_packet_index, size_);
         if (is_h264_keyframe) {
           data_buffer_[first_packet_index].video_header.frame_type =
               VideoFrameType::kVideoFrameKey;
         } else {
           data_buffer_[first_packet_index].video_header.frame_type =
               VideoFrameType::kVideoFrameDelta;
         }

         // With IPPP, if this is not a keyframe, make sure there are no gaps
         // in the packet sequence numbers up until this point.
         const uint8_t h264tid =
             data_buffer_[start_index].video_header.frame_marking.temporal_id;
         if (h264tid == kNoTemporalIdx && !is_h264_keyframe &&
             missing_packets_.upper_bound(start_seq_num) !=
                 missing_packets_.begin()) {
           uint16_t stop_index = (index + 1) % size_;
           while (start_index != stop_index) {
             sequence_buffer_[start_index].frame_created = false;
             start_index = (start_index + 1) % size_;
           }

           return found_frames;
         }
       }

       missing_packets_.erase(missing_packets_.begin(),
                              missing_packets_.upper_bound(seq_num));

       found_frames.emplace_back(
           new RtpFrameObject(this, start_seq_num, seq_num, frame_size,
                              max_nack_count, min_recv_time, max_recv_time,
                              RtpPacketInfos(std::move(packet_infos))));
     }
     ++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();
   uint32_t timestamp = frame->Timestamp();
   while (index != end) {
     // Check both seq_num and timestamp to handle the case when seq_num wraps
     // around too quickly for high packet rates.
     if (sequence_buffer_[index].seq_num == seq_num &&
         data_buffer_[index].timestamp == timestamp) {
       delete[] data_buffer_[index].dataPtr;
       data_buffer_[index].dataPtr = nullptr;
       sequence_buffer_[index].used = false;
     }

     index = (index + 1) % size_;
     ++seq_num;
   }
 }

 bool PacketBuffer::GetBitstream(const RtpFrameObject& frame,
                                 uint8_t* destination) {
   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();
   uint32_t timestamp = frame.Timestamp();
   uint8_t* destination_end = destination + frame.size();

   do {
     // Check both seq_num and timestamp to handle the case when seq_num wraps
     // around too quickly for high packet rates.
     if (!sequence_buffer_[index].used ||
         sequence_buffer_[index].seq_num != seq_num ||
         data_buffer_[index].timestamp != timestamp) {
       return false;
     }

     RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num);
     size_t length = data_buffer_[index].sizeBytes;
     if (destination + length > destination_end) {
       RTC_LOG(LS_WARNING) << "Frame (" << frame.id.picture_id << ":"
                           << static_cast<int>(frame.id.spatial_layer) << ")"
                           << " bitstream buffer is not large enough.";
       return false;
     }

     const uint8_t* source = data_buffer_[index].dataPtr;
     memcpy(destination, source, length);
     destination += length;
     index = (index + 1) % size_;
     ++seq_num;
   } while (index != end);

   return true;
 }

 VCMPacket* PacketBuffer::GetPacket(uint16_t seq_num) {
   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;
 }

 void PacketBuffer::UpdateMissingPackets(uint16_t seq_num) {
   if (!newest_inserted_seq_num_)
     newest_inserted_seq_num_ = seq_num;

   const int kMaxPaddingAge = 1000;
   if (AheadOf(seq_num, *newest_inserted_seq_num_)) {
     uint16_t old_seq_num = seq_num - kMaxPaddingAge;
     auto erase_to = missing_packets_.lower_bound(old_seq_num);
     missing_packets_.erase(missing_packets_.begin(), erase_to);

     // Guard against inserting a large amount of missing packets if there is a
     // jump in the sequence number.
     if (AheadOf(old_seq_num, *newest_inserted_seq_num_))
       *newest_inserted_seq_num_ = old_seq_num;

     ++*newest_inserted_seq_num_;
     while (AheadOf(seq_num, *newest_inserted_seq_num_)) {
       missing_packets_.insert(*newest_inserted_seq_num_);
       ++*newest_inserted_seq_num_;
     }
   } else {
     missing_packets_.erase(seq_num);
   }
 }

 void PacketBuffer::OnTimestampReceived(uint32_t rtp_timestamp) {
   const size_t kMaxTimestampsHistory = 1000;
   if (rtp_timestamps_history_set_.insert(rtp_timestamp).second) {
     rtp_timestamps_history_queue_.push(rtp_timestamp);
     ++unique_frames_seen_;
     if (rtp_timestamps_history_set_.size() > kMaxTimestampsHistory) {
       uint32_t discarded_timestamp = rtp_timestamps_history_queue_.front();
       rtp_timestamps_history_set_.erase(discarded_timestamp);
       rtp_timestamps_history_queue_.pop();
     }
   }
 }

 }  // namespace video_coding
 }  // namespace webrtc
	/*
	* 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 "modules/video_coding/packet_buffer.h"

	#include <string.h>

	#include <algorithm>
	#include <cstdint>
	#include <utility>

	#include "absl/types/variant.h"
	#include "api/video/encoded_frame.h"
	#include "common_video/h264/h264_common.h"
	#include "modules/rtp_rtcp/source/rtp_video_header.h"
	#include "modules/video_coding/codecs/h264/include/h264_globals.h"
	#include "modules/video_coding/frame_object.h"
	#include "rtc_base/atomic_ops.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/logging.h"
	#include "rtc_base/numerics/mod_ops.h"
	#include "system_wrappers/include/clock.h"
	#include "system_wrappers/include/field_trial.h"

	namespace webrtc {
	namespace video_coding {

	rtc::scoped_refptr<PacketBuffer> PacketBuffer::Create(
	Clock* clock,
	size_t start_buffer_size,
	size_t max_buffer_size,
	OnAssembledFrameCallback* assembled_frame_callback) {
	return rtc::scoped_refptr<PacketBuffer>(new PacketBuffer(
	clock, start_buffer_size, max_buffer_size, assembled_frame_callback));
	}

	PacketBuffer::PacketBuffer(Clock* clock,
	size_t start_buffer_size,
	size_t max_buffer_size,
	OnAssembledFrameCallback* assembled_frame_callback)
	: clock_(clock),
	size_(start_buffer_size),
	max_size_(max_buffer_size),
	first_seq_num_(0),
	first_packet_received_(false),
	is_cleared_to_first_seq_num_(false),
	data_buffer_(start_buffer_size),
	sequence_buffer_(start_buffer_size),
	assembled_frame_callback_(assembled_frame_callback),
	unique_frames_seen_(0),
	sps_pps_idr_is_h264_keyframe_(
	field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
	RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
	// 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(VCMPacket* packet) {
	std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
	{
	rtc::CritScope lock(&crit_);

	OnTimestampReceived(packet->timestamp);

	uint16_t seq_num = packet->seqNum;
	size_t index = seq_num % size_;

	if (!first_packet_received_) {
	first_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, just silently ignore it.
	if (is_cleared_to_first_seq_num_) {
	delete[] packet->dataPtr;
	packet->dataPtr = nullptr;
	return true;
	}

	first_seq_num_ = seq_num;
	}

	if (sequence_buffer_[index].used) {
	// Duplicate packet, just delete the payload.
	if (data_buffer_[index].seqNum == packet->seqNum) {
	delete[] packet->dataPtr;
	packet->dataPtr = nullptr;
	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 since we were unable to expand the buffer.
	if (sequence_buffer_[index].used) {
	// Clear the buffer, delete payload, and return false to signal that a
	// new keyframe is needed.
	RTC_LOG(LS_WARNING) << "Clear PacketBuffer and request key frame.";
	Clear();
	delete[] packet->dataPtr;
	packet->dataPtr = nullptr;
	return false;
	}
	}

	sequence_buffer_[index].frame_begin = packet->is_first_packet_in_frame();
	sequence_buffer_[index].frame_end = packet->is_last_packet_in_frame();
	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;
	packet->dataPtr = nullptr;

	UpdateMissingPackets(packet->seqNum);

	int64_t now_ms = clock_->TimeInMilliseconds();
	last_received_packet_ms_ = now_ms;
	if (packet->video_header.frame_type == VideoFrameType::kVideoFrameKey)
	last_received_keyframe_packet_ms_ = now_ms;

	found_frames = FindFrames(seq_num);
	}

	for (std::unique_ptr<RtpFrameObject>& frame : found_frames)
	assembled_frame_callback_->OnAssembledFrame(std::move(frame));

	return true;
	}

	void PacketBuffer::ClearTo(uint16_t seq_num) {
	rtc::CritScope lock(&crit_);
	// We have already cleared past this sequence number, no need to do anything.
	if (is_cleared_to_first_seq_num_ &&
	AheadOf<uint16_t>(first_seq_num_, seq_num)) {
	return;
	}

	// If the packet buffer was cleared between a frame was created and returned.
	if (!first_packet_received_)
	return;

	// Avoid iterating over the buffer more than once by capping the number of
	// iterations to the \|size_\| of the buffer.
	++seq_num;
	size_t diff = ForwardDiff<uint16_t>(first_seq_num_, seq_num);
	size_t iterations = std::min(diff, size_);
	for (size_t i = 0; i < iterations; ++i) {
	size_t index = first_seq_num_ % size_;
	RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num);
	if (AheadOf<uint16_t>(seq_num, sequence_buffer_[index].seq_num)) {
	delete[] data_buffer_[index].dataPtr;
	data_buffer_[index].dataPtr = nullptr;
	sequence_buffer_[index].used = false;
	}
	++first_seq_num_;
	}

	// If \|diff\| is larger than \|iterations\| it means that we don't increment
	// \|first_seq_num_\| until we reach \|seq_num\|, so we set it here.
	first_seq_num_ = seq_num;

	is_cleared_to_first_seq_num_ = true;
	auto clear_to_it = missing_packets_.upper_bound(seq_num);
	if (clear_to_it != missing_packets_.begin()) {
	--clear_to_it;
	missing_packets_.erase(missing_packets_.begin(), clear_to_it);
	}
	}

	void PacketBuffer::Clear() {
	rtc::CritScope lock(&crit_);
	for (size_t i = 0; i < size_; ++i) {
	delete[] data_buffer_[i].dataPtr;
	data_buffer_[i].dataPtr = nullptr;
	sequence_buffer_[i].used = false;
	}

	first_packet_received_ = false;
	is_cleared_to_first_seq_num_ = false;
	last_received_packet_ms_.reset();
	last_received_keyframe_packet_ms_.reset();
	newest_inserted_seq_num_.reset();
	missing_packets_.clear();
	}

	void PacketBuffer::PaddingReceived(uint16_t seq_num) {
	std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
	{
	rtc::CritScope lock(&crit_);
	UpdateMissingPackets(seq_num);
	found_frames = FindFrames(static_cast<uint16_t>(seq_num + 1));
	}

	for (std::unique_ptr<RtpFrameObject>& frame : found_frames)
	assembled_frame_callback_->OnAssembledFrame(std::move(frame));
	}

	absl::optional<int64_t> PacketBuffer::LastReceivedPacketMs() const {
	rtc::CritScope lock(&crit_);
	return last_received_packet_ms_;
	}

	absl::optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
	rtc::CritScope lock(&crit_);
	return last_received_keyframe_packet_ms_;
	}

	int PacketBuffer::GetUniqueFramesSeen() const {
	rtc::CritScope lock(&crit_);
	return unique_frames_seen_;
	}

	bool PacketBuffer::ExpandBufferSize() {
	if (size_ == max_size_) {
	RTC_LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_
	<< "), failed to increase size.";
	return false;
	}

	size_t 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 (size_t i = 0; i < size_; ++i) {
	if (sequence_buffer_[i].used) {
	size_t 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);
	RTC_LOG(LS_INFO) << "PacketBuffer size expanded to " << new_size;
	return true;
	}

	bool PacketBuffer::PotentialNewFrame(uint16_t seq_num) const {
	size_t index = seq_num % size_;
	int prev_index = index > 0 ? index - 1 : size_ - 1;

	if (!sequence_buffer_[index].used)
	return false;
	if (sequence_buffer_[index].seq_num != seq_num)
	return false;
	if (sequence_buffer_[index].frame_created)
	return false;
	if (sequence_buffer_[index].frame_begin)
	return true;
	if (!sequence_buffer_[prev_index].used)
	return false;
	if (sequence_buffer_[prev_index].frame_created)
	return false;
	if (sequence_buffer_[prev_index].seq_num !=
	static_cast<uint16_t>(sequence_buffer_[index].seq_num - 1)) {
	return false;
	}
	if (data_buffer_[prev_index].timestamp != data_buffer_[index].timestamp)
	return false;
	if (sequence_buffer_[prev_index].continuous)
	return true;

	return false;
	}

	std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames(
	uint16_t seq_num) {
	std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
	for (size_t i = 0; i < size_ && PotentialNewFrame(seq_num); ++i) {
	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;
	int64_t min_recv_time = data_buffer_[index].packet_info.receive_time_ms();
	int64_t max_recv_time = data_buffer_[index].packet_info.receive_time_ms();
	RtpPacketInfos::vector_type packet_infos;

	// Find the start index by searching backward until the packet with
	// the \|frame_begin\| flag is set.
	int start_index = index;
	size_t tested_packets = 0;
	int64_t frame_timestamp = data_buffer_[start_index].timestamp;

	// Identify H.264 keyframes by means of SPS, PPS, and IDR.
	bool is_h264 = data_buffer_[start_index].codec() == kVideoCodecH264;
	bool has_h264_sps = false;
	bool has_h264_pps = false;
	bool has_h264_idr = false;
	bool is_h264_keyframe = false;

	while (true) {
	++tested_packets;
	frame_size += data_buffer_[start_index].sizeBytes;
	max_nack_count =
	std::max(max_nack_count, data_buffer_[start_index].timesNacked);
	sequence_buffer_[start_index].frame_created = true;

	min_recv_time =
	std::min(min_recv_time,
	data_buffer_[start_index].packet_info.receive_time_ms());
	max_recv_time =
	std::max(max_recv_time,
	data_buffer_[start_index].packet_info.receive_time_ms());

	// Should use \|push_front()\| since the loop traverses backwards. But
	// it's too inefficient to do so on a vector so we'll instead fix the
	// order afterwards.
	packet_infos.push_back(data_buffer_[start_index].packet_info);

	if (!is_h264 && sequence_buffer_[start_index].frame_begin)
	break;

	if (is_h264 && !is_h264_keyframe) {
	const auto* h264_header = absl::get_if<RTPVideoHeaderH264>(
	&data_buffer_[start_index].video_header.video_type_header);
	if (!h264_header \|\| h264_header->nalus_length >= kMaxNalusPerPacket)
	return found_frames;

	for (size_t j = 0; j < h264_header->nalus_length; ++j) {
	if (h264_header->nalus[j].type == H264::NaluType::kSps) {
	has_h264_sps = true;
	} else if (h264_header->nalus[j].type == H264::NaluType::kPps) {
	has_h264_pps = true;
	} else if (h264_header->nalus[j].type == H264::NaluType::kIdr) {
	has_h264_idr = true;
	}
	}
	if ((sps_pps_idr_is_h264_keyframe_ && has_h264_idr && has_h264_sps &&
	has_h264_pps) \|\|
	(!sps_pps_idr_is_h264_keyframe_ && has_h264_idr)) {
	is_h264_keyframe = true;
	}
	}

	if (tested_packets == size_)
	break;

	start_index = start_index > 0 ? start_index - 1 : size_ - 1;

	// In the case of H264 we don't have a frame_begin bit (yes,
	// \|frame_begin\| might be set to true but that is a lie). So instead
	// we traverese backwards as long as we have a previous packet and
	// the timestamp of that packet is the same as this one. This may cause
	// the PacketBuffer to hand out incomplete frames.
	// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7106
	if (is_h264 &&
	(!sequence_buffer_[start_index].used \|\|
	data_buffer_[start_index].timestamp != frame_timestamp)) {
	break;
	}

	--start_seq_num;
	}

	// Fix the order since the packet-finding loop traverses backwards.
	std::reverse(packet_infos.begin(), packet_infos.end());

	if (is_h264) {
	// Warn if this is an unsafe frame.
	if (has_h264_idr && (!has_h264_sps \|\| !has_h264_pps)) {
	RTC_LOG(LS_WARNING)
	<< "Received H.264-IDR frame "
	<< "(SPS: " << has_h264_sps << ", PPS: " << has_h264_pps
	<< "). Treating as "
	<< (sps_pps_idr_is_h264_keyframe_ ? "delta" : "key")
	<< " frame since WebRTC-SpsPpsIdrIsH264Keyframe is "
	<< (sps_pps_idr_is_h264_keyframe_ ? "enabled." : "disabled");
	}

	// Now that we have decided whether to treat this frame as a key frame
	// or delta frame in the frame buffer, we update the field that
	// determines if the RtpFrameObject is a key frame or delta frame.
	const size_t first_packet_index = start_seq_num % size_;
	RTC_CHECK_LT(first_packet_index, size_);
	if (is_h264_keyframe) {
	data_buffer_[first_packet_index].video_header.frame_type =
	VideoFrameType::kVideoFrameKey;
	} else {
	data_buffer_[first_packet_index].video_header.frame_type =
	VideoFrameType::kVideoFrameDelta;
	}

	// With IPPP, if this is not a keyframe, make sure there are no gaps
	// in the packet sequence numbers up until this point.
	const uint8_t h264tid =
	data_buffer_[start_index].video_header.frame_marking.temporal_id;
	if (h264tid == kNoTemporalIdx && !is_h264_keyframe &&
	missing_packets_.upper_bound(start_seq_num) !=
	missing_packets_.begin()) {
	uint16_t stop_index = (index + 1) % size_;
	while (start_index != stop_index) {
	sequence_buffer_[start_index].frame_created = false;
	start_index = (start_index + 1) % size_;
	}

	return found_frames;
	}
	}

	missing_packets_.erase(missing_packets_.begin(),
	missing_packets_.upper_bound(seq_num));

	found_frames.emplace_back(
	new RtpFrameObject(this, start_seq_num, seq_num, frame_size,
	max_nack_count, min_recv_time, max_recv_time,
	RtpPacketInfos(std::move(packet_infos))));
	}
	++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();
	uint32_t timestamp = frame->Timestamp();
	while (index != end) {
	// Check both seq_num and timestamp to handle the case when seq_num wraps
	// around too quickly for high packet rates.
	if (sequence_buffer_[index].seq_num == seq_num &&
	data_buffer_[index].timestamp == timestamp) {
	delete[] data_buffer_[index].dataPtr;
	data_buffer_[index].dataPtr = nullptr;
	sequence_buffer_[index].used = false;
	}

	index = (index + 1) % size_;
	++seq_num;
	}
	}

	bool PacketBuffer::GetBitstream(const RtpFrameObject& frame,
	uint8_t* destination) {
	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();
	uint32_t timestamp = frame.Timestamp();
	uint8_t* destination_end = destination + frame.size();

	do {
	// Check both seq_num and timestamp to handle the case when seq_num wraps
	// around too quickly for high packet rates.
	if (!sequence_buffer_[index].used \|\|
	sequence_buffer_[index].seq_num != seq_num \|\|
	data_buffer_[index].timestamp != timestamp) {
	return false;
	}

	RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num);
	size_t length = data_buffer_[index].sizeBytes;
	if (destination + length > destination_end) {
	RTC_LOG(LS_WARNING) << "Frame (" << frame.id.picture_id << ":"
	<< static_cast<int>(frame.id.spatial_layer) << ")"
	<< " bitstream buffer is not large enough.";
	return false;
	}

	const uint8_t* source = data_buffer_[index].dataPtr;
	memcpy(destination, source, length);
	destination += length;
	index = (index + 1) % size_;
	++seq_num;
	} while (index != end);

	return true;
	}

	VCMPacket* PacketBuffer::GetPacket(uint16_t seq_num) {
	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;
	}

	void PacketBuffer::UpdateMissingPackets(uint16_t seq_num) {
	if (!newest_inserted_seq_num_)
	newest_inserted_seq_num_ = seq_num;

	const int kMaxPaddingAge = 1000;
	if (AheadOf(seq_num, *newest_inserted_seq_num_)) {
	uint16_t old_seq_num = seq_num - kMaxPaddingAge;
	auto erase_to = missing_packets_.lower_bound(old_seq_num);
	missing_packets_.erase(missing_packets_.begin(), erase_to);

	// Guard against inserting a large amount of missing packets if there is a
	// jump in the sequence number.
	if (AheadOf(old_seq_num, *newest_inserted_seq_num_))
	*newest_inserted_seq_num_ = old_seq_num;

	++*newest_inserted_seq_num_;
	while (AheadOf(seq_num, *newest_inserted_seq_num_)) {
	missing_packets_.insert(*newest_inserted_seq_num_);
	++*newest_inserted_seq_num_;
	}
	} else {
	missing_packets_.erase(seq_num);
	}
	}

	void PacketBuffer::OnTimestampReceived(uint32_t rtp_timestamp) {
	const size_t kMaxTimestampsHistory = 1000;
	if (rtp_timestamps_history_set_.insert(rtp_timestamp).second) {
	rtp_timestamps_history_queue_.push(rtp_timestamp);
	++unique_frames_seen_;
	if (rtp_timestamps_history_set_.size() > kMaxTimestampsHistory) {
	uint32_t discarded_timestamp = rtp_timestamps_history_queue_.front();
	rtp_timestamps_history_set_.erase(discarded_timestamp);
	rtp_timestamps_history_queue_.pop();
	}
	}
	}

	} // namespace video_coding
	} // namespace webrtc