Don't let time flow backwards in pacer
Bug: webrtc:9716, b:111681259 Change-Id: I1bf8edeaed6c56f3f5a0bdcc1f71108e119e1843 Reviewed-on: https://webrtc-review.googlesource.com/97701 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24561}
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@ -67,6 +67,7 @@ PacedSender::PacedSender(const Clock* clock,
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send_padding_if_silent_(
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field_trial::IsEnabled("WebRTC-Pacer-PadInSilence")),
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video_blocks_audio_(!field_trial::IsDisabled("WebRTC-Pacer-BlockAudio")),
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last_timestamp_ms_(clock_->TimeInMilliseconds()),
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paused_(false),
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media_budget_(absl::make_unique<IntervalBudget>(0)),
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padding_budget_(absl::make_unique<IntervalBudget>(0)),
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@ -94,7 +95,7 @@ PacedSender::~PacedSender() {}
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void PacedSender::CreateProbeCluster(int bitrate_bps) {
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rtc::CritScope cs(&critsect_);
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prober_->CreateProbeCluster(bitrate_bps, clock_->TimeInMilliseconds());
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prober_->CreateProbeCluster(bitrate_bps, TimeMilliseconds());
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}
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void PacedSender::Pause() {
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@ -103,7 +104,7 @@ void PacedSender::Pause() {
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if (!paused_)
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RTC_LOG(LS_INFO) << "PacedSender paused.";
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paused_ = true;
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packets_->SetPauseState(true, clock_->TimeInMilliseconds());
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packets_->SetPauseState(true, TimeMilliseconds());
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}
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rtc::CritScope cs(&process_thread_lock_);
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// Tell the process thread to call our TimeUntilNextProcess() method to get
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@ -118,7 +119,7 @@ void PacedSender::Resume() {
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if (paused_)
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RTC_LOG(LS_INFO) << "PacedSender resumed.";
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paused_ = false;
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packets_->SetPauseState(false, clock_->TimeInMilliseconds());
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packets_->SetPauseState(false, TimeMilliseconds());
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}
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rtc::CritScope cs(&process_thread_lock_);
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// Tell the process thread to call our TimeUntilNextProcess() method to
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@ -143,6 +144,19 @@ bool PacedSender::Congested() const {
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return outstanding_bytes_ >= congestion_window_bytes_;
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}
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int64_t PacedSender::TimeMilliseconds() const {
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int64_t time_ms = clock_->TimeInMilliseconds();
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if (time_ms < last_timestamp_ms_) {
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RTC_LOG(LS_WARNING)
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<< "Non-monotonic clock behavior observed. Previous timestamp: "
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<< last_timestamp_ms_ << ", new timestamp: " << time_ms;
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RTC_DCHECK_GE(time_ms, last_timestamp_ms_);
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time_ms = last_timestamp_ms_;
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}
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last_timestamp_ms_ = time_ms;
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return time_ms;
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}
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void PacedSender::SetProbingEnabled(bool enabled) {
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rtc::CritScope cs(&critsect_);
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RTC_CHECK_EQ(0, packet_counter_);
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@ -192,7 +206,7 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
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RTC_DCHECK(pacing_bitrate_kbps_ > 0)
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<< "SetPacingRate must be called before InsertPacket.";
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int64_t now_ms = clock_->TimeInMilliseconds();
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int64_t now_ms = TimeMilliseconds();
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prober_->OnIncomingPacket(bytes);
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if (capture_time_ms < 0)
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@ -238,7 +252,7 @@ int64_t PacedSender::QueueInMs() const {
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if (oldest_packet == 0)
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return 0;
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return clock_->TimeInMilliseconds() - oldest_packet;
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return TimeMilliseconds() - oldest_packet;
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}
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int64_t PacedSender::TimeUntilNextProcess() {
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@ -252,7 +266,7 @@ int64_t PacedSender::TimeUntilNextProcess() {
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return std::max<int64_t>(kPausedProcessIntervalMs - elapsed_time_ms, 0);
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if (prober_->IsProbing()) {
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int64_t ret = prober_->TimeUntilNextProbe(clock_->TimeInMilliseconds());
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int64_t ret = prober_->TimeUntilNextProbe(TimeMilliseconds());
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if (ret > 0 || (ret == 0 && !probing_send_failure_))
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return ret;
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}
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@ -294,7 +308,7 @@ void PacedSender::Process() {
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// Assuming equal size packets and input/output rate, the average packet
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// has avg_time_left_ms left to get queue_size_bytes out of the queue, if
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// time constraint shall be met. Determine bitrate needed for that.
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packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
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packets_->UpdateQueueTime(TimeMilliseconds());
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if (drain_large_queues_) {
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int64_t avg_time_left_ms = std::max<int64_t>(
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1, queue_time_limit - packets_->AverageQueueTimeMs());
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@ -353,7 +367,7 @@ void PacedSender::Process() {
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if (is_probing) {
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probing_send_failure_ = bytes_sent == 0;
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if (!probing_send_failure_)
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prober_->ProbeSent(clock_->TimeInMilliseconds(), bytes_sent);
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prober_->ProbeSent(TimeMilliseconds(), bytes_sent);
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}
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alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
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}
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@ -384,7 +398,7 @@ bool PacedSender::SendPacket(const PacketQueueInterface::Packet& packet,
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if (success) {
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if (first_sent_packet_ms_ == -1)
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first_sent_packet_ms_ = clock_->TimeInMilliseconds();
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first_sent_packet_ms_ = TimeMilliseconds();
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if (!audio_packet || account_for_audio_) {
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// Update media bytes sent.
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// TODO(eladalon): TimeToSendPacket() can also return |true| in some
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@ -157,6 +157,7 @@ class PacedSender : public Pacer {
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void OnBytesSent(size_t bytes_sent) RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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bool Congested() const RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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int64_t TimeMilliseconds() const RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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const Clock* const clock_;
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PacketSender* const packet_sender_;
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@ -165,7 +166,11 @@ class PacedSender : public Pacer {
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const bool drain_large_queues_;
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const bool send_padding_if_silent_;
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const bool video_blocks_audio_;
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rtc::CriticalSection critsect_;
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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 int64_t last_timestamp_ms_ RTC_GUARDED_BY(critsect_);
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bool paused_ RTC_GUARDED_BY(critsect_);
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// This is the media budget, keeping track of how many bits of media
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// we can pace out during the current interval.
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@ -17,12 +17,12 @@
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namespace webrtc {
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RoundRobinPacketQueue::Stream::Stream() : bytes(0) {}
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RoundRobinPacketQueue::Stream::Stream() : bytes(0), ssrc(0) {}
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RoundRobinPacketQueue::Stream::Stream(const Stream& stream) = default;
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RoundRobinPacketQueue::Stream::~Stream() {}
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RoundRobinPacketQueue::RoundRobinPacketQueue(const Clock* clock)
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: clock_(clock), time_last_updated_(clock_->TimeInMilliseconds()) {}
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: time_last_updated_(clock->TimeInMilliseconds()) {}
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RoundRobinPacketQueue::~RoundRobinPacketQueue() {}
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@ -82,7 +82,6 @@ class RoundRobinPacketQueue : public PacketQueueInterface {
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// Just used to verify correctness.
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bool IsSsrcScheduled(uint32_t ssrc) const;
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const Clock* const clock_;
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int64_t time_last_updated_;
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absl::optional<Packet> pop_packet_;
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absl::optional<Stream*> pop_stream_;
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