| /* |
| * Copyright (c) 2014 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/rtp_rtcp/include/remote_ntp_time_estimator.h" |
| |
| #include <cstdint> |
| |
| #include "modules/rtp_rtcp/source/time_util.h" |
| #include "rtc_base/logging.h" |
| #include "system_wrappers/include/clock.h" |
| #include "system_wrappers/include/ntp_time.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| constexpr int kMinimumNumberOfSamples = 2; |
| constexpr TimeDelta kTimingLogInterval = TimeDelta::Seconds(10); |
| constexpr int kClocksOffsetSmoothingWindow = 100; |
| |
| // Subtracts two NtpTime values keeping maximum precision. |
| int64_t Subtract(NtpTime minuend, NtpTime subtrahend) { |
| uint64_t a = static_cast<uint64_t>(minuend); |
| uint64_t b = static_cast<uint64_t>(subtrahend); |
| return a >= b ? static_cast<int64_t>(a - b) : -static_cast<int64_t>(b - a); |
| } |
| |
| NtpTime Add(NtpTime lhs, int64_t rhs) { |
| uint64_t result = static_cast<uint64_t>(lhs); |
| if (rhs >= 0) { |
| result += static_cast<uint64_t>(rhs); |
| } else { |
| result -= static_cast<uint64_t>(-rhs); |
| } |
| return NtpTime(result); |
| } |
| |
| } // namespace |
| |
| // TODO(wu): Refactor this class so that it can be shared with |
| // vie_sync_module.cc. |
| RemoteNtpTimeEstimator::RemoteNtpTimeEstimator(Clock* clock) |
| : clock_(clock), |
| ntp_clocks_offset_estimator_(kClocksOffsetSmoothingWindow) {} |
| |
| bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(TimeDelta rtt, |
| NtpTime sender_send_time, |
| uint32_t rtp_timestamp) { |
| switch (rtp_to_ntp_.UpdateMeasurements(sender_send_time, rtp_timestamp)) { |
| case RtpToNtpEstimator::kInvalidMeasurement: |
| return false; |
| case RtpToNtpEstimator::kSameMeasurement: |
| // No new RTCP SR since last time this function was called. |
| return true; |
| case RtpToNtpEstimator::kNewMeasurement: |
| break; |
| } |
| |
| // Assume connection is symmetric and thus time to deliver the packet is half |
| // the round trip time. |
| int64_t deliver_time_ntp = ToNtpUnits(rtt) / 2; |
| |
| // Update extrapolator with the new arrival time. |
| NtpTime receiver_arrival_time = clock_->CurrentNtpTime(); |
| int64_t remote_to_local_clocks_offset = |
| Subtract(receiver_arrival_time, sender_send_time) - deliver_time_ntp; |
| ntp_clocks_offset_estimator_.Insert(remote_to_local_clocks_offset); |
| return true; |
| } |
| |
| NtpTime RemoteNtpTimeEstimator::EstimateNtp(uint32_t rtp_timestamp) { |
| NtpTime sender_capture = rtp_to_ntp_.Estimate(rtp_timestamp); |
| if (!sender_capture.Valid()) { |
| return sender_capture; |
| } |
| |
| int64_t remote_to_local_clocks_offset = |
| ntp_clocks_offset_estimator_.GetFilteredValue(); |
| NtpTime receiver_capture = Add(sender_capture, remote_to_local_clocks_offset); |
| |
| Timestamp now = clock_->CurrentTime(); |
| if (now - last_timing_log_ > kTimingLogInterval) { |
| RTC_LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp |
| << " in NTP clock: " << sender_capture.ToMs() |
| << " estimated time in receiver NTP clock: " |
| << receiver_capture.ToMs(); |
| last_timing_log_ = now; |
| } |
| |
| return receiver_capture; |
| } |
| |
| absl::optional<int64_t> |
| RemoteNtpTimeEstimator::EstimateRemoteToLocalClockOffset() { |
| if (ntp_clocks_offset_estimator_.GetNumberOfSamplesStored() < |
| kMinimumNumberOfSamples) { |
| return absl::nullopt; |
| } |
| return ntp_clocks_offset_estimator_.GetFilteredValue(); |
| } |
| |
| } // namespace webrtc |