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76b7f51842
This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I51dfe8879c28c91bd1c667fc47b4892373671e0f Reviewed-on: https://webrtc-review.googlesource.com/21540 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22569}
84 lines
3.1 KiB
C++
84 lines
3.1 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 "rtc_base/logging.h"
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#include "rtc_base/time/timestamp_extrapolator.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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static const int kTimingLogIntervalMs = 10000;
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static const 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 rtcp_timestamp) {
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bool new_rtcp_sr = false;
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if (!rtp_to_ntp_.UpdateMeasurements(ntp_secs, ntp_frac, rtcp_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 TimeInMilliseconds time.
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int64_t receiver_arrival_time_ms = clock_->TimeInMilliseconds();
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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_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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int64_t ntp_offset = clock_->CurrentNtpInMilliseconds() - now_ms;
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int64_t receiver_capture_ntp_ms = receiver_capture_ms + ntp_offset;
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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 clock: "
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<< receiver_capture_ms
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<< " converted to NTP clock: " << 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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} // namespace webrtc
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