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67dba30178
Bug: webrtc:11342 Change-Id: Ied155984794670ad08a663ac71f98719e96f8037 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168223 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#30474}
104 lines
3.8 KiB
C++
104 lines
3.8 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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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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bool new_rtcp_sr = false;
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if (!rtp_to_ntp_.UpdateMeasurements(ntp_secs, ntp_frac, rtp_timestamp,
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&new_rtcp_sr)) {
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return false;
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}
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if (!new_rtcp_sr) {
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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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}
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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 =
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clock_->TimeInMilliseconds() + NtpOffsetMs();
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int64_t sender_send_time_ms = Clock::NtpToMs(ntp_secs, ntp_frac);
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int64_t sender_arrival_time_ms = sender_send_time_ms + 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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int64_t sender_capture_ntp_ms = 0;
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if (!rtp_to_ntp_.Estimate(rtp_timestamp, &sender_capture_ntp_ms)) {
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return -1;
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}
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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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// TODO(bugs.webrtc.org/11327): Clock::CurrentNtpInMilliseconds() was
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// previously used to calculate the offset between the local and the remote
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// clock. However, rtc::TimeMillis() + NtpOffsetMs() is now used as the local
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// ntp clock value. To preserve the old behavior of this method, the return
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// value is adjusted with the difference between the two local ntp clocks.
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int64_t now_ms = clock_->TimeInMilliseconds();
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int64_t offset_between_local_ntp_clocks =
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clock_->CurrentNtpInMilliseconds() - now_ms - NtpOffsetMs();
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receiver_capture_ntp_ms += offset_between_local_ntp_clocks;
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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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