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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc')
-rw-r--r-- | third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc | 189 |
1 files changed, 189 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc new file mode 100644 index 0000000000..1605008299 --- /dev/null +++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc @@ -0,0 +1,189 @@ +/* + * Copyright (c) 2012 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/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h" + +#include <cstdint> +#include <utility> + +#include "absl/types/optional.h" +#include "modules/remote_bitrate_estimator/aimd_rate_control.h" +#include "modules/remote_bitrate_estimator/include/bwe_defines.h" +#include "modules/remote_bitrate_estimator/inter_arrival.h" +#include "modules/remote_bitrate_estimator/overuse_detector.h" +#include "modules/remote_bitrate_estimator/overuse_estimator.h" +#include "modules/rtp_rtcp/source/rtp_header_extensions.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "system_wrappers/include/clock.h" +#include "system_wrappers/include/metrics.h" + +namespace webrtc { +namespace { + +constexpr int kTimestampGroupLengthMs = 5; +constexpr double kTimestampToMs = 1.0 / 90.0; + +} // namespace + +RemoteBitrateEstimatorSingleStream::Detector::Detector() + : last_packet_time(Timestamp::Zero()), + inter_arrival(90 * kTimestampGroupLengthMs, kTimestampToMs) {} + +RemoteBitrateEstimatorSingleStream::RemoteBitrateEstimatorSingleStream( + RemoteBitrateObserver* observer, + Clock* clock) + : clock_(clock), + incoming_bitrate_(kBitrateWindow), + last_valid_incoming_bitrate_(DataRate::Zero()), + remote_rate_(field_trials_), + observer_(observer), + process_interval_(kProcessInterval), + uma_recorded_(false) { + RTC_LOG(LS_INFO) << "RemoteBitrateEstimatorSingleStream: Instantiating."; +} + +RemoteBitrateEstimatorSingleStream::~RemoteBitrateEstimatorSingleStream() = + default; + +void RemoteBitrateEstimatorSingleStream::IncomingPacket( + const RtpPacketReceived& rtp_packet) { + absl::optional<int32_t> transmission_time_offset = + rtp_packet.GetExtension<TransmissionOffset>(); + if (!uma_recorded_) { + BweNames type = transmission_time_offset.has_value() + ? BweNames::kReceiverTOffset + : BweNames::kReceiverNoExtension; + RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram, type, BweNames::kBweNamesMax); + uma_recorded_ = true; + } + uint32_t ssrc = rtp_packet.Ssrc(); + uint32_t rtp_timestamp = + rtp_packet.Timestamp() + transmission_time_offset.value_or(0); + Timestamp now = clock_->CurrentTime(); + Detector& estimator = overuse_detectors_[ssrc]; + estimator.last_packet_time = now; + + // Check if incoming bitrate estimate is valid, and if it needs to be reset. + absl::optional<DataRate> incoming_bitrate = incoming_bitrate_.Rate(now); + if (incoming_bitrate) { + last_valid_incoming_bitrate_ = *incoming_bitrate; + } else if (last_valid_incoming_bitrate_ > DataRate::Zero()) { + // Incoming bitrate had a previous valid value, but now not enough data + // point are left within the current window. Reset incoming bitrate + // estimator so that the window size will only contain new data points. + incoming_bitrate_.Reset(); + last_valid_incoming_bitrate_ = DataRate::Zero(); + } + size_t payload_size = rtp_packet.payload_size() + rtp_packet.padding_size(); + incoming_bitrate_.Update(payload_size, now); + + const BandwidthUsage prior_state = estimator.detector.State(); + uint32_t timestamp_delta = 0; + int64_t time_delta = 0; + int size_delta = 0; + int64_t now_ms = now.ms(); + if (estimator.inter_arrival.ComputeDeltas( + rtp_timestamp, rtp_packet.arrival_time().ms(), now_ms, payload_size, + ×tamp_delta, &time_delta, &size_delta)) { + double timestamp_delta_ms = timestamp_delta * kTimestampToMs; + estimator.estimator.Update(time_delta, timestamp_delta_ms, size_delta, + estimator.detector.State(), now_ms); + estimator.detector.Detect(estimator.estimator.offset(), timestamp_delta_ms, + estimator.estimator.num_of_deltas(), now_ms); + } + if (estimator.detector.State() == BandwidthUsage::kBwOverusing) { + absl::optional<DataRate> incoming_bitrate = incoming_bitrate_.Rate(now); + if (incoming_bitrate.has_value() && + (prior_state != BandwidthUsage::kBwOverusing || + remote_rate_.TimeToReduceFurther(now, *incoming_bitrate))) { + // The first overuse should immediately trigger a new estimate. + // We also have to update the estimate immediately if we are overusing + // and the target bitrate is too high compared to what we are receiving. + UpdateEstimate(now); + } + } +} + +TimeDelta RemoteBitrateEstimatorSingleStream::Process() { + Timestamp now = clock_->CurrentTime(); + Timestamp next_process_time = last_process_time_.has_value() + ? *last_process_time_ + process_interval_ + : now; + // TODO(bugs.webrtc.org/13756): Removing rounding to milliseconds after + // investigating why tests fails without that rounding. + if (now.ms() >= next_process_time.ms()) { + UpdateEstimate(now); + last_process_time_ = now; + return process_interval_; + } + + return next_process_time - now; +} + +void RemoteBitrateEstimatorSingleStream::UpdateEstimate(Timestamp now) { + BandwidthUsage bw_state = BandwidthUsage::kBwNormal; + auto it = overuse_detectors_.begin(); + while (it != overuse_detectors_.end()) { + if (now - it->second.last_packet_time > kStreamTimeOut) { + // This over-use detector hasn't received packets for `kStreamTimeOut` + // and is considered stale. + overuse_detectors_.erase(it++); + } else { + // Make sure that we trigger an over-use if any of the over-use detectors + // is detecting over-use. + if (it->second.detector.State() > bw_state) { + bw_state = it->second.detector.State(); + } + ++it; + } + } + // We can't update the estimate if we don't have any active streams. + if (overuse_detectors_.empty()) { + return; + } + + const RateControlInput input(bw_state, incoming_bitrate_.Rate(now)); + uint32_t target_bitrate = remote_rate_.Update(input, now).bps<uint32_t>(); + if (remote_rate_.ValidEstimate()) { + process_interval_ = remote_rate_.GetFeedbackInterval(); + RTC_DCHECK_GT(process_interval_, TimeDelta::Zero()); + if (observer_) + observer_->OnReceiveBitrateChanged(GetSsrcs(), target_bitrate); + } +} + +void RemoteBitrateEstimatorSingleStream::OnRttUpdate(int64_t avg_rtt_ms, + int64_t max_rtt_ms) { + remote_rate_.SetRtt(TimeDelta::Millis(avg_rtt_ms)); +} + +void RemoteBitrateEstimatorSingleStream::RemoveStream(uint32_t ssrc) { + overuse_detectors_.erase(ssrc); +} + +DataRate RemoteBitrateEstimatorSingleStream::LatestEstimate() const { + if (!remote_rate_.ValidEstimate() || overuse_detectors_.empty()) { + return DataRate::Zero(); + } + return remote_rate_.LatestEstimate(); +} + +std::vector<uint32_t> RemoteBitrateEstimatorSingleStream::GetSsrcs() const { + std::vector<uint32_t> ssrcs; + ssrcs.reserve(overuse_detectors_.size()); + for (const auto& [ssrc, unused] : overuse_detectors_) { + ssrcs.push_back(ssrc); + } + return ssrcs; +} + +} // namespace webrtc |