Reland "Adds trial to use correct overhead calculation in pacer."
This reverts commit 7affd9bcbb7a778408942d8afa4fe3ce29a8fc0b. Reason for revert: The perf issue has been addressed in the reland (https://webrtc-review.googlesource.com/c/src/+/167883). Original change's description: > Revert "Adds trial to use correct overhead calculation in pacer." > > This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7. > > Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict. > > Original change's description: > > Adds trial to use correct overhead calculation in pacer. > > > > Bug: webrtc:9883 > > Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30399} > > TBR=sprang@webrtc.org,srte@webrtc.org > > Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9883 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30409} TBR=mbonadei@webrtc.org,sprang@webrtc.org,srte@webrtc.org Change-Id: Iafdef81d08078000dc368e001f67bee660e2f5bc No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167861 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30414}
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@ -421,6 +421,9 @@ void RtpTransportControllerSend::OnTransportOverheadChanged(
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return;
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}
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pacer()->SetTransportOverhead(
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DataSize::bytes(transport_overhead_bytes_per_packet));
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// TODO(holmer): Call AudioRtpSenders when they have been moved to
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// RtpTransportControllerSend.
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for (auto& rtp_video_sender : video_rtp_senders_) {
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@ -131,6 +131,11 @@ void PacedSender::SetIncludeOverhead() {
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pacing_controller_.SetIncludeOverhead();
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}
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void PacedSender::SetTransportOverhead(DataSize overhead_per_packet) {
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rtc::CritScope cs(&critsect_);
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pacing_controller_.SetTransportOverhead(overhead_per_packet);
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}
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TimeDelta PacedSender::ExpectedQueueTime() const {
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rtc::CritScope cs(&critsect_);
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return pacing_controller_.ExpectedQueueTime();
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@ -98,6 +98,7 @@ class PacedSender : public Module,
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void SetAccountForAudioPackets(bool account_for_audio) override;
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void SetIncludeOverhead() override;
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void SetTransportOverhead(DataSize overhead_per_packet) override;
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// Returns the time since the oldest queued packet was enqueued.
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TimeDelta OldestPacketWaitTime() const override;
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@ -99,7 +99,10 @@ PacingController::PacingController(Clock* clock,
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pace_audio_(IsEnabled(*field_trials_, "WebRTC-Pacer-BlockAudio")),
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small_first_probe_packet_(
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IsEnabled(*field_trials_, "WebRTC-Pacer-SmallFirstProbePacket")),
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ignore_transport_overhead_(
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!IsDisabled(*field_trials_, "WebRTC-Pacer-IgnoreTransportOverhead")),
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min_packet_limit_(kDefaultMinPacketLimit),
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transport_overhead_per_packet_(DataSize::Zero()),
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last_timestamp_(clock_->CurrentTime()),
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paused_(false),
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media_budget_(0),
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@ -230,6 +233,13 @@ void PacingController::SetIncludeOverhead() {
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packet_queue_.SetIncludeOverhead();
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}
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void PacingController::SetTransportOverhead(DataSize overhead_per_packet) {
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if (ignore_transport_overhead_)
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return;
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transport_overhead_per_packet_ = overhead_per_packet;
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packet_queue_.SetTransportOverhead(overhead_per_packet);
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}
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TimeDelta PacingController::ExpectedQueueTime() const {
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RTC_DCHECK_GT(pacing_bitrate_, DataRate::Zero());
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return TimeDelta::ms(
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@ -521,10 +531,13 @@ void PacingController::ProcessPackets() {
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RTC_DCHECK(rtp_packet);
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RTC_DCHECK(rtp_packet->packet_type().has_value());
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const RtpPacketToSend::Type packet_type = *rtp_packet->packet_type();
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const DataSize packet_size =
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DataSize::bytes(include_overhead_ ? rtp_packet->size()
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: rtp_packet->payload_size() +
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rtp_packet->padding_size());
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DataSize packet_size = DataSize::bytes(rtp_packet->payload_size() +
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rtp_packet->padding_size());
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if (include_overhead_) {
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packet_size += DataSize::bytes(rtp_packet->headers_size()) +
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transport_overhead_per_packet_;
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}
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packet_sender_->SendRtpPacket(std::move(rtp_packet), pacing_info);
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data_sent += packet_size;
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@ -109,6 +109,8 @@ class PacingController {
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void SetAccountForAudioPackets(bool account_for_audio);
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void SetIncludeOverhead();
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void SetTransportOverhead(DataSize overhead_per_packet);
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// Returns the time since the oldest queued packet was enqueued.
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TimeDelta OldestPacketWaitTime() const;
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@ -177,9 +179,12 @@ class PacingController {
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const bool send_padding_if_silent_;
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const bool pace_audio_;
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const bool small_first_probe_packet_;
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const bool ignore_transport_overhead_;
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TimeDelta min_packet_limit_;
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DataSize transport_overhead_per_packet_;
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// TODO(webrtc:9716): Remove this when we are certain clocks are monotonic.
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// The last millisecond timestamp returned by |clock_|.
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mutable Timestamp last_timestamp_;
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@ -73,12 +73,6 @@ uint64_t RoundRobinPacketQueue::QueuedPacket::EnqueueOrder() const {
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return enqueue_order_;
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}
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DataSize RoundRobinPacketQueue::QueuedPacket::Size(bool count_overhead) const {
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return DataSize::bytes(count_overhead ? owned_packet_->size()
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: owned_packet_->payload_size() +
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owned_packet_->padding_size());
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}
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RtpPacketToSend* RoundRobinPacketQueue::QueuedPacket::RtpPacket() const {
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return owned_packet_;
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}
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@ -117,7 +111,8 @@ bool IsEnabled(const WebRtcKeyValueConfig* field_trials, const char* name) {
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RoundRobinPacketQueue::RoundRobinPacketQueue(
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Timestamp start_time,
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const WebRtcKeyValueConfig* field_trials)
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: time_last_updated_(start_time),
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: transport_overhead_per_packet_(DataSize::Zero()),
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time_last_updated_(start_time),
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paused_(false),
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size_packets_(0),
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size_(DataSize::Zero()),
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@ -167,7 +162,13 @@ std::unique_ptr<RtpPacketToSend> RoundRobinPacketQueue::Pop() {
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// case a "budget" will be built up for the stream sending at the lower
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// rate. To avoid building a too large budget we limit |bytes| to be within
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// kMaxLeading bytes of the stream that has sent the most amount of bytes.
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DataSize packet_size = queued_packet.Size(include_overhead_);
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DataSize packet_size =
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DataSize::bytes(queued_packet.RtpPacket()->payload_size() +
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queued_packet.RtpPacket()->padding_size());
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if (include_overhead_) {
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packet_size += DataSize::bytes(queued_packet.RtpPacket()->headers_size()) +
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transport_overhead_per_packet_;
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}
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stream->size =
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std::max(stream->size + packet_size, max_size_ - kMaxLeadingSize);
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max_size_ = std::max(max_size_, stream->size);
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@ -250,14 +251,18 @@ void RoundRobinPacketQueue::SetPauseState(bool paused, Timestamp now) {
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void RoundRobinPacketQueue::SetIncludeOverhead() {
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include_overhead_ = true;
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// We need to update the size to reflect overhead for existing packets.
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size_ = DataSize::Zero();
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for (const auto& stream : streams_) {
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for (const QueuedPacket& packet : stream.second.packet_queue) {
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size_ += packet.Size(include_overhead_);
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size_ += DataSize::bytes(packet.RtpPacket()->headers_size()) +
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transport_overhead_per_packet_;
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}
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}
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}
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void RoundRobinPacketQueue::SetTransportOverhead(DataSize overhead_per_packet) {
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transport_overhead_per_packet_ = overhead_per_packet;
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}
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TimeDelta RoundRobinPacketQueue::AverageQueueTime() const {
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if (Empty())
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return TimeDelta::Zero();
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@ -299,7 +304,12 @@ void RoundRobinPacketQueue::Push(QueuedPacket packet) {
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packet.SubtractPauseTime(pause_time_sum_);
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size_packets_ += 1;
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size_ += packet.Size(include_overhead_);
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size_ += DataSize::bytes(packet.RtpPacket()->payload_size() +
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packet.RtpPacket()->padding_size());
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if (include_overhead_) {
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size_ += DataSize::bytes(packet.RtpPacket()->headers_size()) +
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transport_overhead_per_packet_;
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}
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stream->packet_queue.push(packet);
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}
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@ -53,6 +53,7 @@ class RoundRobinPacketQueue {
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void UpdateQueueTime(Timestamp now);
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void SetPauseState(bool paused, Timestamp now);
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void SetIncludeOverhead();
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void SetTransportOverhead(DataSize overhead_per_packet);
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private:
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struct QueuedPacket {
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@ -73,7 +74,6 @@ class RoundRobinPacketQueue {
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Timestamp EnqueueTime() const;
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bool IsRetransmission() const;
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uint64_t EnqueueOrder() const;
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DataSize Size(bool count_overhead) const;
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RtpPacketToSend* RtpPacket() const;
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std::multiset<Timestamp>::iterator EnqueueTimeIterator() const;
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@ -137,6 +137,8 @@ class RoundRobinPacketQueue {
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// Just used to verify correctness.
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bool IsSsrcScheduled(uint32_t ssrc) const;
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DataSize transport_overhead_per_packet_;
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Timestamp time_last_updated_;
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bool paused_;
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@ -65,6 +65,7 @@ class RtpPacketPacer {
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// at high priority.
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virtual void SetAccountForAudioPackets(bool account_for_audio) = 0;
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virtual void SetIncludeOverhead() = 0;
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virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0;
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};
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} // namespace webrtc
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@ -143,6 +143,13 @@ void TaskQueuePacedSender::SetIncludeOverhead() {
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});
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}
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void TaskQueuePacedSender::SetTransportOverhead(DataSize overhead_per_packet) {
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task_queue_.PostTask([this, overhead_per_packet]() {
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RTC_DCHECK_RUN_ON(&task_queue_);
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pacing_controller_.SetTransportOverhead(overhead_per_packet);
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});
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}
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void TaskQueuePacedSender::SetQueueTimeLimit(TimeDelta limit) {
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task_queue_.PostTask([this, limit]() {
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RTC_DCHECK_RUN_ON(&task_queue_);
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@ -80,6 +80,8 @@ class TaskQueuePacedSender : public RtpPacketPacer,
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void SetAccountForAudioPackets(bool account_for_audio) override;
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void SetIncludeOverhead() override;
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void SetTransportOverhead(DataSize overhead_per_packet) override;
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// Returns the time since the oldest queued packet was enqueued.
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TimeDelta OldestPacketWaitTime() const override;
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