Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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442
modules/video_coding/packet_buffer.cc
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442
modules/video_coding/packet_buffer.cc
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/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/video_coding/packet_buffer.h"
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#include <algorithm>
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#include <limits>
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#include <utility>
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#include "webrtc/common_video/h264/h264_common.h"
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#include "webrtc/modules/video_coding/frame_object.h"
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#include "webrtc/rtc_base/atomicops.h"
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/logging.h"
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#include "webrtc/system_wrappers/include/clock.h"
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namespace webrtc {
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namespace video_coding {
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rtc::scoped_refptr<PacketBuffer> PacketBuffer::Create(
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Clock* clock,
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size_t start_buffer_size,
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size_t max_buffer_size,
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OnReceivedFrameCallback* received_frame_callback) {
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return rtc::scoped_refptr<PacketBuffer>(new PacketBuffer(
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clock, start_buffer_size, max_buffer_size, received_frame_callback));
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}
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PacketBuffer::PacketBuffer(Clock* clock,
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size_t start_buffer_size,
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size_t max_buffer_size,
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OnReceivedFrameCallback* received_frame_callback)
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: clock_(clock),
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size_(start_buffer_size),
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max_size_(max_buffer_size),
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first_seq_num_(0),
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first_packet_received_(false),
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is_cleared_to_first_seq_num_(false),
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data_buffer_(start_buffer_size),
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sequence_buffer_(start_buffer_size),
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received_frame_callback_(received_frame_callback) {
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RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
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// Buffer size must always be a power of 2.
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RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0);
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RTC_DCHECK((max_buffer_size & (max_buffer_size - 1)) == 0);
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}
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PacketBuffer::~PacketBuffer() {
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Clear();
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}
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bool PacketBuffer::InsertPacket(VCMPacket* packet) {
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std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
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{
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rtc::CritScope lock(&crit_);
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uint16_t seq_num = packet->seqNum;
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size_t index = seq_num % size_;
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if (!first_packet_received_) {
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first_seq_num_ = seq_num;
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first_packet_received_ = true;
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} else if (AheadOf(first_seq_num_, seq_num)) {
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// If we have explicitly cleared past this packet then it's old,
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// don't insert it.
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if (is_cleared_to_first_seq_num_) {
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delete[] packet->dataPtr;
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packet->dataPtr = nullptr;
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return false;
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}
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first_seq_num_ = seq_num;
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}
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if (sequence_buffer_[index].used) {
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// Duplicate packet, just delete the payload.
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if (data_buffer_[index].seqNum == packet->seqNum) {
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delete[] packet->dataPtr;
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packet->dataPtr = nullptr;
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return true;
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}
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// The packet buffer is full, try to expand the buffer.
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while (ExpandBufferSize() && sequence_buffer_[seq_num % size_].used) {
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}
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index = seq_num % size_;
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// Packet buffer is still full.
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if (sequence_buffer_[index].used) {
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delete[] packet->dataPtr;
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packet->dataPtr = nullptr;
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return false;
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}
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}
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sequence_buffer_[index].frame_begin = packet->is_first_packet_in_frame;
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sequence_buffer_[index].frame_end = packet->markerBit;
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sequence_buffer_[index].seq_num = packet->seqNum;
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sequence_buffer_[index].continuous = false;
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sequence_buffer_[index].frame_created = false;
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sequence_buffer_[index].used = true;
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data_buffer_[index] = *packet;
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packet->dataPtr = nullptr;
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UpdateMissingPackets(packet->seqNum);
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int64_t now_ms = clock_->TimeInMilliseconds();
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last_received_packet_ms_ = rtc::Optional<int64_t>(now_ms);
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if (packet->frameType == kVideoFrameKey)
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last_received_keyframe_packet_ms_ = rtc::Optional<int64_t>(now_ms);
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found_frames = FindFrames(seq_num);
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}
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for (std::unique_ptr<RtpFrameObject>& frame : found_frames)
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received_frame_callback_->OnReceivedFrame(std::move(frame));
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return true;
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}
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void PacketBuffer::ClearTo(uint16_t seq_num) {
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rtc::CritScope lock(&crit_);
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// We have already cleared past this sequence number, no need to do anything.
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if (is_cleared_to_first_seq_num_ &&
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AheadOf<uint16_t>(first_seq_num_, seq_num)) {
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return;
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}
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// If the packet buffer was cleared between a frame was created and returned.
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if (!first_packet_received_)
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return;
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// Avoid iterating over the buffer more than once by capping the number of
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// iterations to the |size_| of the buffer.
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++seq_num;
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size_t diff = ForwardDiff<uint16_t>(first_seq_num_, seq_num);
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size_t iterations = std::min(diff, size_);
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for (size_t i = 0; i < iterations; ++i) {
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size_t index = first_seq_num_ % size_;
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RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num);
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if (AheadOf<uint16_t>(seq_num, sequence_buffer_[index].seq_num)) {
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delete[] data_buffer_[index].dataPtr;
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data_buffer_[index].dataPtr = nullptr;
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sequence_buffer_[index].used = false;
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}
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++first_seq_num_;
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}
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// If |diff| is larger than |iterations| it means that we don't increment
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// |first_seq_num_| until we reach |seq_num|, so we set it here.
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first_seq_num_ = seq_num;
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is_cleared_to_first_seq_num_ = true;
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missing_packets_.erase(missing_packets_.begin(),
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missing_packets_.upper_bound(seq_num));
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}
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void PacketBuffer::Clear() {
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rtc::CritScope lock(&crit_);
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for (size_t i = 0; i < size_; ++i) {
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delete[] data_buffer_[i].dataPtr;
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data_buffer_[i].dataPtr = nullptr;
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sequence_buffer_[i].used = false;
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}
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first_packet_received_ = false;
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is_cleared_to_first_seq_num_ = false;
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last_received_packet_ms_.reset();
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last_received_keyframe_packet_ms_.reset();
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newest_inserted_seq_num_.reset();
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missing_packets_.clear();
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}
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void PacketBuffer::PaddingReceived(uint16_t seq_num) {
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std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
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{
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rtc::CritScope lock(&crit_);
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UpdateMissingPackets(seq_num);
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found_frames = FindFrames(static_cast<uint16_t>(seq_num + 1));
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}
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for (std::unique_ptr<RtpFrameObject>& frame : found_frames)
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received_frame_callback_->OnReceivedFrame(std::move(frame));
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}
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rtc::Optional<int64_t> PacketBuffer::LastReceivedPacketMs() const {
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rtc::CritScope lock(&crit_);
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return last_received_packet_ms_;
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}
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rtc::Optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
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rtc::CritScope lock(&crit_);
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return last_received_keyframe_packet_ms_;
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}
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bool PacketBuffer::ExpandBufferSize() {
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if (size_ == max_size_) {
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LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_
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<< "), failed to increase size. Clearing PacketBuffer.";
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Clear();
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return false;
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}
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size_t new_size = std::min(max_size_, 2 * size_);
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std::vector<VCMPacket> new_data_buffer(new_size);
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std::vector<ContinuityInfo> new_sequence_buffer(new_size);
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for (size_t i = 0; i < size_; ++i) {
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if (sequence_buffer_[i].used) {
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size_t index = sequence_buffer_[i].seq_num % new_size;
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new_sequence_buffer[index] = sequence_buffer_[i];
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new_data_buffer[index] = data_buffer_[i];
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}
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}
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size_ = new_size;
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sequence_buffer_ = std::move(new_sequence_buffer);
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data_buffer_ = std::move(new_data_buffer);
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LOG(LS_INFO) << "PacketBuffer size expanded to " << new_size;
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return true;
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}
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bool PacketBuffer::PotentialNewFrame(uint16_t seq_num) const {
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size_t index = seq_num % size_;
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int prev_index = index > 0 ? index - 1 : size_ - 1;
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if (!sequence_buffer_[index].used)
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return false;
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if (sequence_buffer_[index].seq_num != seq_num)
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return false;
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if (sequence_buffer_[index].frame_created)
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return false;
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if (sequence_buffer_[index].frame_begin)
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return true;
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if (!sequence_buffer_[prev_index].used)
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return false;
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if (sequence_buffer_[prev_index].frame_created)
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return false;
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if (sequence_buffer_[prev_index].seq_num !=
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static_cast<uint16_t>(sequence_buffer_[index].seq_num - 1)) {
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return false;
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}
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if (sequence_buffer_[prev_index].continuous)
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return true;
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return false;
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}
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std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames(
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uint16_t seq_num) {
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std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
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for (size_t i = 0; i < size_ && PotentialNewFrame(seq_num); ++i) {
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size_t index = seq_num % size_;
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sequence_buffer_[index].continuous = true;
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// If all packets of the frame is continuous, find the first packet of the
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// frame and create an RtpFrameObject.
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if (sequence_buffer_[index].frame_end) {
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size_t frame_size = 0;
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int max_nack_count = -1;
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uint16_t start_seq_num = seq_num;
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// Find the start index by searching backward until the packet with
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// the |frame_begin| flag is set.
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int start_index = index;
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size_t tested_packets = 0;
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bool is_h264 = data_buffer_[start_index].codec == kVideoCodecH264;
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bool is_h264_keyframe = false;
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int64_t frame_timestamp = data_buffer_[start_index].timestamp;
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while (true) {
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++tested_packets;
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frame_size += data_buffer_[start_index].sizeBytes;
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max_nack_count =
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std::max(max_nack_count, data_buffer_[start_index].timesNacked);
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sequence_buffer_[start_index].frame_created = true;
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if (!is_h264 && sequence_buffer_[start_index].frame_begin)
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break;
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if (is_h264 && !is_h264_keyframe) {
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const RTPVideoHeaderH264& header =
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data_buffer_[start_index].video_header.codecHeader.H264;
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for (size_t i = 0; i < header.nalus_length; ++i) {
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if (header.nalus[i].type == H264::NaluType::kIdr) {
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is_h264_keyframe = true;
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break;
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}
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}
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}
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if (tested_packets == size_)
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break;
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start_index = start_index > 0 ? start_index - 1 : size_ - 1;
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// In the case of H264 we don't have a frame_begin bit (yes,
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// |frame_begin| might be set to true but that is a lie). So instead
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// we traverese backwards as long as we have a previous packet and
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// the timestamp of that packet is the same as this one. This may cause
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// the PacketBuffer to hand out incomplete frames.
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// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7106
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if (is_h264 &&
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(!sequence_buffer_[start_index].used ||
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data_buffer_[start_index].timestamp != frame_timestamp)) {
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break;
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}
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--start_seq_num;
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}
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// If this is H264 but not a keyframe, make sure there are no gaps in the
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// packet sequence numbers up until this point.
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if (is_h264 && !is_h264_keyframe &&
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missing_packets_.upper_bound(start_seq_num) !=
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missing_packets_.begin()) {
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uint16_t stop_index = (index + 1) % size_;
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while (start_index != stop_index) {
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sequence_buffer_[start_index].frame_created = false;
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start_index = (start_index + 1) % size_;
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}
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return found_frames;
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}
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missing_packets_.erase(missing_packets_.begin(),
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missing_packets_.upper_bound(seq_num));
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found_frames.emplace_back(
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new RtpFrameObject(this, start_seq_num, seq_num, frame_size,
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max_nack_count, clock_->TimeInMilliseconds()));
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}
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++seq_num;
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}
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return found_frames;
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}
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void PacketBuffer::ReturnFrame(RtpFrameObject* frame) {
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rtc::CritScope lock(&crit_);
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size_t index = frame->first_seq_num() % size_;
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size_t end = (frame->last_seq_num() + 1) % size_;
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uint16_t seq_num = frame->first_seq_num();
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while (index != end) {
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if (sequence_buffer_[index].seq_num == seq_num) {
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delete[] data_buffer_[index].dataPtr;
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data_buffer_[index].dataPtr = nullptr;
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sequence_buffer_[index].used = false;
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}
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index = (index + 1) % size_;
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++seq_num;
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}
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}
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bool PacketBuffer::GetBitstream(const RtpFrameObject& frame,
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uint8_t* destination) {
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rtc::CritScope lock(&crit_);
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size_t index = frame.first_seq_num() % size_;
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size_t end = (frame.last_seq_num() + 1) % size_;
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uint16_t seq_num = frame.first_seq_num();
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uint8_t* destination_end = destination + frame.size();
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do {
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if (!sequence_buffer_[index].used ||
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sequence_buffer_[index].seq_num != seq_num) {
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return false;
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}
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RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num);
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size_t length = data_buffer_[index].sizeBytes;
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if (destination + length > destination_end) {
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LOG(LS_WARNING) << "Frame (" << frame.picture_id << ":"
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<< static_cast<int>(frame.spatial_layer) << ")"
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<< " bitstream buffer is not large enough.";
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return false;
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}
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const uint8_t* source = data_buffer_[index].dataPtr;
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memcpy(destination, source, length);
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destination += length;
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index = (index + 1) % size_;
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++seq_num;
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} while (index != end);
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return true;
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}
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VCMPacket* PacketBuffer::GetPacket(uint16_t seq_num) {
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size_t index = seq_num % size_;
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if (!sequence_buffer_[index].used ||
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seq_num != sequence_buffer_[index].seq_num) {
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return nullptr;
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}
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return &data_buffer_[index];
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}
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int PacketBuffer::AddRef() const {
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return rtc::AtomicOps::Increment(&ref_count_);
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}
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int PacketBuffer::Release() const {
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int count = rtc::AtomicOps::Decrement(&ref_count_);
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if (!count) {
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delete this;
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}
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return count;
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}
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void PacketBuffer::UpdateMissingPackets(uint16_t seq_num) {
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if (!newest_inserted_seq_num_)
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newest_inserted_seq_num_ = rtc::Optional<uint16_t>(seq_num);
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const int kMaxPaddingAge = 1000;
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if (AheadOf(seq_num, *newest_inserted_seq_num_)) {
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uint16_t old_seq_num = seq_num - kMaxPaddingAge;
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auto erase_to = missing_packets_.lower_bound(old_seq_num);
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missing_packets_.erase(missing_packets_.begin(), erase_to);
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// Guard against inserting a large amount of missing packets if there is a
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// jump in the sequence number.
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if (AheadOf(old_seq_num, *newest_inserted_seq_num_))
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*newest_inserted_seq_num_ = old_seq_num;
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++*newest_inserted_seq_num_;
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while (AheadOf(seq_num, *newest_inserted_seq_num_)) {
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missing_packets_.insert(*newest_inserted_seq_num_);
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++*newest_inserted_seq_num_;
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}
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} else {
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missing_packets_.erase(seq_num);
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}
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}
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} // namespace video_coding
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} // namespace webrtc
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