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853a407273
This reverts commit a154a15c97.
Reason for revert: breaks downstream tests
Original change's description:
> Migrate RemoteNtpTimeEstimator to more precise time representations
>
> Bug: webrtc:13757
> Change-Id: I880ab3cc6e4f72da587ae42ddca051332907c07f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261311
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36817}
Bug: webrtc:13757
Change-Id: Id21edb1378e6e944b24955396250ddc33fa70663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261722
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36819}
97 lines
3.3 KiB
C++
97 lines
3.3 KiB
C++
/*
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* Copyright (c) 2014 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 "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
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#include <cstdint>
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#include "modules/rtp_rtcp/source/time_util.h"
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#include "rtc_base/logging.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/ntp_time.h"
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namespace webrtc {
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namespace {
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constexpr int kMinimumNumberOfSamples = 2;
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constexpr int kTimingLogIntervalMs = 10000;
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constexpr int kClocksOffsetSmoothingWindow = 100;
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} // namespace
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// TODO(wu): Refactor this class so that it can be shared with
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// vie_sync_module.cc.
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RemoteNtpTimeEstimator::RemoteNtpTimeEstimator(Clock* clock)
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: clock_(clock),
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ntp_clocks_offset_estimator_(kClocksOffsetSmoothingWindow),
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last_timing_log_ms_(-1) {}
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RemoteNtpTimeEstimator::~RemoteNtpTimeEstimator() {}
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bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(int64_t rtt,
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uint32_t ntp_secs,
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uint32_t ntp_frac,
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uint32_t rtp_timestamp) {
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NtpTime sender_send_time(ntp_secs, ntp_frac);
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switch (rtp_to_ntp_.UpdateMeasurements(sender_send_time, rtp_timestamp)) {
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case RtpToNtpEstimator::kInvalidMeasurement:
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return false;
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case RtpToNtpEstimator::kSameMeasurement:
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// No new RTCP SR since last time this function was called.
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return true;
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case RtpToNtpEstimator::kNewMeasurement:
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break;
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}
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// Update extrapolator with the new arrival time.
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// The extrapolator assumes the ntp time.
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int64_t receiver_arrival_time_ms = clock_->CurrentNtpInMilliseconds();
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int64_t sender_arrival_time_ms = sender_send_time.ToMs() + rtt / 2;
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int64_t remote_to_local_clocks_offset =
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receiver_arrival_time_ms - sender_arrival_time_ms;
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ntp_clocks_offset_estimator_.Insert(remote_to_local_clocks_offset);
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return true;
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}
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int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) {
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NtpTime sender_capture = rtp_to_ntp_.Estimate(rtp_timestamp);
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if (!sender_capture.Valid()) {
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return -1;
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}
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int64_t sender_capture_ntp_ms = sender_capture.ToMs();
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int64_t remote_to_local_clocks_offset =
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ntp_clocks_offset_estimator_.GetFilteredValue();
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int64_t receiver_capture_ntp_ms =
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sender_capture_ntp_ms + remote_to_local_clocks_offset;
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int64_t now_ms = clock_->TimeInMilliseconds();
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if (now_ms - last_timing_log_ms_ > kTimingLogIntervalMs) {
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RTC_LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp
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<< " in NTP clock: " << sender_capture_ntp_ms
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<< " estimated time in receiver NTP clock: "
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<< receiver_capture_ntp_ms;
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last_timing_log_ms_ = now_ms;
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}
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return receiver_capture_ntp_ms;
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}
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absl::optional<int64_t>
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RemoteNtpTimeEstimator::EstimateRemoteToLocalClockOffsetMs() {
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if (ntp_clocks_offset_estimator_.GetNumberOfSamplesStored() <
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kMinimumNumberOfSamples) {
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return absl::nullopt;
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}
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return ntp_clocks_offset_estimator_.GetFilteredValue();
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}
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} // namespace webrtc
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