| /* |
| * 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 "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
| |
| #include "webrtc/rtc_base/logging.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/system_wrappers/include/timestamp_extrapolator.h" |
| |
| namespace webrtc { |
| |
| static const int kTimingLogIntervalMs = 10000; |
| |
| // TODO(wu): Refactor this class so that it can be shared with |
| // vie_sync_module.cc. |
| RemoteNtpTimeEstimator::RemoteNtpTimeEstimator(Clock* clock) |
| : clock_(clock), |
| ts_extrapolator_(new TimestampExtrapolator(clock_->TimeInMilliseconds())), |
| last_timing_log_ms_(-1) { |
| } |
| |
| RemoteNtpTimeEstimator::~RemoteNtpTimeEstimator() {} |
| |
| bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(int64_t rtt, |
| uint32_t ntp_secs, |
| uint32_t ntp_frac, |
| uint32_t rtcp_timestamp) { |
| bool new_rtcp_sr = false; |
| if (!rtp_to_ntp_.UpdateMeasurements(ntp_secs, ntp_frac, rtcp_timestamp, |
| &new_rtcp_sr)) { |
| return false; |
| } |
| if (!new_rtcp_sr) { |
| // No new RTCP SR since last time this function was called. |
| return true; |
| } |
| // Update extrapolator with the new arrival time. |
| // The extrapolator assumes the TimeInMilliseconds time. |
| int64_t receiver_arrival_time_ms = clock_->TimeInMilliseconds(); |
| int64_t sender_send_time_ms = Clock::NtpToMs(ntp_secs, ntp_frac); |
| int64_t sender_arrival_time_90k = (sender_send_time_ms + rtt / 2) * 90; |
| ts_extrapolator_->Update(receiver_arrival_time_ms, sender_arrival_time_90k); |
| return true; |
| } |
| |
| int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) { |
| int64_t sender_capture_ntp_ms = 0; |
| if (!rtp_to_ntp_.Estimate(rtp_timestamp, &sender_capture_ntp_ms)) { |
| return -1; |
| } |
| uint32_t timestamp = sender_capture_ntp_ms * 90; |
| int64_t receiver_capture_ms = |
| ts_extrapolator_->ExtrapolateLocalTime(timestamp); |
| int64_t ntp_offset = |
| clock_->CurrentNtpInMilliseconds() - clock_->TimeInMilliseconds(); |
| int64_t receiver_capture_ntp_ms = receiver_capture_ms + ntp_offset; |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| if (now_ms - last_timing_log_ms_ > kTimingLogIntervalMs) { |
| LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp |
| << " in NTP clock: " << sender_capture_ntp_ms |
| << " estimated time in receiver clock: " << receiver_capture_ms |
| << " converted to NTP clock: " << receiver_capture_ntp_ms; |
| last_timing_log_ms_ = now_ms; |
| } |
| return receiver_capture_ntp_ms; |
| } |
| |
| } // namespace webrtc |