/* * Copyright (c) 2017 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/pacing/packet_queue2.h" #include #include "rtc_base/checks.h" #include "system_wrappers/include/clock.h" namespace webrtc { PacketQueue2::Stream::Stream() : bytes(0) {} PacketQueue2::Stream::~Stream() {} PacketQueue2::Packet::Packet(RtpPacketSender::Priority priority, uint32_t ssrc, uint16_t seq_number, int64_t capture_time_ms, int64_t enqueue_time_ms, size_t length_in_bytes, bool retransmission, uint64_t enqueue_order) : priority(priority), ssrc(ssrc), sequence_number(seq_number), capture_time_ms(capture_time_ms), enqueue_time_ms(enqueue_time_ms), bytes(length_in_bytes), retransmission(retransmission), enqueue_order(enqueue_order) {} PacketQueue2::Packet::Packet(const Packet& other) = default; PacketQueue2::Packet::~Packet() {} PacketQueue2::PacketQueue2(const Clock* clock) : clock_(clock), time_last_updated_(clock_->TimeInMilliseconds()) {} PacketQueue2::~PacketQueue2() {} void PacketQueue2::Push(const Packet& packet_to_insert) { Packet packet(packet_to_insert); auto stream_info_it = streams_.find(packet.ssrc); if (stream_info_it == streams_.end()) { stream_info_it = streams_.emplace(packet.ssrc, Stream()).first; stream_info_it->second.priority_it = stream_priorities_.end(); stream_info_it->second.ssrc = packet.ssrc; } Stream* streams_ = &stream_info_it->second; if (streams_->priority_it == stream_priorities_.end()) { // If the SSRC is not currently scheduled, add it to |stream_priorities_|. RTC_CHECK(!IsSsrcScheduled(streams_->ssrc)); streams_->priority_it = stream_priorities_.emplace( StreamPrioKey(packet.priority, streams_->bytes), packet.ssrc); } else if (packet.priority < streams_->priority_it->first.priority) { // If the priority of this SSRC increased, remove the outdated StreamPrioKey // and insert a new one with the new priority. Note that // RtpPacketSender::Priority uses lower ordinal for higher priority. stream_priorities_.erase(streams_->priority_it); streams_->priority_it = stream_priorities_.emplace( StreamPrioKey(packet.priority, streams_->bytes), packet.ssrc); } RTC_CHECK(streams_->priority_it != stream_priorities_.end()); packet.enqueue_time_it = enqueue_times_.insert(packet.enqueue_time_ms); // In order to figure out how much time a packet has spent in the queue while // not in a paused state, we subtract the total amount of time the queue has // been paused so far, and when the packet is poped we subtract the total // amount of time the queue has been paused at that moment. This way we // subtract the total amount of time the packet has spent in the queue while // in a paused state. UpdateQueueTime(packet.enqueue_time_ms); packet.enqueue_time_ms -= pause_time_sum_ms_; streams_->packet_queue.push(packet); size_packets_ += 1; size_bytes_ += packet.bytes; } const PacketQueue2::Packet& PacketQueue2::Top() { return GetHighestPriorityStream()->packet_queue.top(); } void PacketQueue2::Pop() { RTC_CHECK(!paused_); if (!Empty()) { Stream* streams_ = GetHighestPriorityStream(); stream_priorities_.erase(streams_->priority_it); const Packet& packet = streams_->packet_queue.top(); // Calculate the total amount of time spent by this packet in the queue // while in a non-paused state. Note that the |pause_time_sum_ms_| was // subtracted from |packet.enqueue_time_ms| when the packet was pushed, and // by subtracting it now we effectively remove the time spent in in the // queue while in a paused state. int64_t time_in_non_paused_state_ms = time_last_updated_ - packet.enqueue_time_ms - pause_time_sum_ms_; queue_time_sum_ms_ -= time_in_non_paused_state_ms; RTC_CHECK(packet.enqueue_time_it != enqueue_times_.end()); enqueue_times_.erase(packet.enqueue_time_it); // Update |bytes| of this stream. The general idea is that the stream that // has sent the least amount of bytes should have the highest priority. // The problem with that is if streams send with different rates, in which // case a "budget" will be built up for the stream sending at the lower // rate. To avoid building a too large budget we limit |bytes| to be within // kMaxLeading bytes of the stream that has sent the most amount of bytes. streams_->bytes = std::max(streams_->bytes + packet.bytes, max_bytes_ - kMaxLeadingBytes); max_bytes_ = std::max(max_bytes_, streams_->bytes); size_bytes_ -= packet.bytes; size_packets_ -= 1; RTC_CHECK(size_packets_ > 0 || queue_time_sum_ms_ == 0); streams_->packet_queue.pop(); // If there are packets left to be sent, schedule the stream again. RTC_CHECK(!IsSsrcScheduled(streams_->ssrc)); if (streams_->packet_queue.empty()) { streams_->priority_it = stream_priorities_.end(); } else { RtpPacketSender::Priority priority = streams_->packet_queue.top().priority; streams_->priority_it = stream_priorities_.emplace( StreamPrioKey(priority, streams_->bytes), streams_->ssrc); } } } bool PacketQueue2::Empty() const { RTC_CHECK((!stream_priorities_.empty() && size_packets_ > 0) || (stream_priorities_.empty() && size_packets_ == 0)); return stream_priorities_.empty(); } size_t PacketQueue2::SizeInPackets() const { return size_packets_; } uint64_t PacketQueue2::SizeInBytes() const { return size_bytes_; } int64_t PacketQueue2::OldestEnqueueTimeMs() const { if (Empty()) return 0; RTC_CHECK(!enqueue_times_.empty()); return *enqueue_times_.begin(); } void PacketQueue2::UpdateQueueTime(int64_t timestamp_ms) { RTC_CHECK_GE(timestamp_ms, time_last_updated_); if (timestamp_ms == time_last_updated_) return; int64_t delta_ms = timestamp_ms - time_last_updated_; if (paused_) { pause_time_sum_ms_ += delta_ms; } else { queue_time_sum_ms_ += delta_ms * size_packets_; } time_last_updated_ = timestamp_ms; } void PacketQueue2::SetPauseState(bool paused, int64_t timestamp_ms) { if (paused_ == paused) return; UpdateQueueTime(timestamp_ms); paused_ = paused; } int64_t PacketQueue2::AverageQueueTimeMs() const { if (Empty()) return 0; return queue_time_sum_ms_ / size_packets_; } PacketQueue2::Stream* PacketQueue2::GetHighestPriorityStream() { RTC_CHECK(!stream_priorities_.empty()); uint32_t ssrc = stream_priorities_.begin()->second; auto stream_info_it = streams_.find(ssrc); RTC_CHECK(stream_info_it != streams_.end()); RTC_CHECK(stream_info_it->second.priority_it == stream_priorities_.begin()); RTC_CHECK(!stream_info_it->second.packet_queue.empty()); return &stream_info_it->second; } bool PacketQueue2::IsSsrcScheduled(uint32_t ssrc) const { for (const auto& scheduled_stream : stream_priorities_) { if (scheduled_stream.second == ssrc) return true; } return false; } } // namespace webrtc