Move NTP functionality to Clock. BUG=1563 TEST=trybots and vie_auto_test --automated Review URL: https://webrtc-codereview.appspot.com/1313005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3842 4adac7df-926f-26a2-2b94-8c16560cd09d
122 lines
4.5 KiB
C++
122 lines
4.5 KiB
C++
/*
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* Copyright (c) 2012 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 "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
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#include "webrtc/system_wrappers/interface/clock.h"
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#include <assert.h>
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namespace webrtc {
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namespace synchronization {
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const double kNtpFracPerMs = 4.294967296E6;
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RtcpMeasurement::RtcpMeasurement()
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: ntp_secs(0), ntp_frac(0), rtp_timestamp(0) {}
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RtcpMeasurement::RtcpMeasurement(uint32_t ntp_secs, uint32_t ntp_frac,
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uint32_t timestamp)
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: ntp_secs(ntp_secs), ntp_frac(ntp_frac), rtp_timestamp(timestamp) {}
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// Calculates the RTP timestamp frequency from two pairs of NTP and RTP
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// timestamps.
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bool CalculateFrequency(
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int64_t rtcp_ntp_ms1,
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uint32_t rtp_timestamp1,
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int64_t rtcp_ntp_ms2,
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uint32_t rtp_timestamp2,
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double* frequency_khz) {
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if (rtcp_ntp_ms1 <= rtcp_ntp_ms2) {
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return false;
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}
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*frequency_khz = static_cast<double>(rtp_timestamp1 - rtp_timestamp2) /
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static_cast<double>(rtcp_ntp_ms1 - rtcp_ntp_ms2);
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return true;
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}
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// Detects if there has been a wraparound between |old_timestamp| and
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// |new_timestamp|, and compensates by adding 2^32 if that is the case.
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bool CompensateForWrapAround(uint32_t new_timestamp,
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uint32_t old_timestamp,
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int64_t* compensated_timestamp) {
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assert(compensated_timestamp);
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int64_t wraps = synchronization::CheckForWrapArounds(new_timestamp,
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old_timestamp);
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if (wraps < 0) {
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// Reordering, don't use this packet.
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return false;
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}
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*compensated_timestamp = new_timestamp + (wraps << 32);
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return true;
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}
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// Converts |rtp_timestamp| to the NTP time base using the NTP and RTP timestamp
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// pairs in |rtcp|. The converted timestamp is returned in
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// |rtp_timestamp_in_ms|. This function compensates for wrap arounds in RTP
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// timestamps and returns false if it can't do the conversion due to reordering.
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bool RtpToNtpMs(int64_t rtp_timestamp,
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const synchronization::RtcpList& rtcp,
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int64_t* rtp_timestamp_in_ms) {
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assert(rtcp.size() == 2);
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int64_t rtcp_ntp_ms_new = Clock::NtpToMs(rtcp.front().ntp_secs,
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rtcp.front().ntp_frac);
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int64_t rtcp_ntp_ms_old = Clock::NtpToMs(rtcp.back().ntp_secs,
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rtcp.back().ntp_frac);
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int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp;
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int64_t rtcp_timestamp_old = rtcp.back().rtp_timestamp;
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if (!CompensateForWrapAround(rtcp_timestamp_new,
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rtcp_timestamp_old,
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&rtcp_timestamp_new)) {
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return false;
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}
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double freq_khz;
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if (!CalculateFrequency(rtcp_ntp_ms_new,
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rtcp_timestamp_new,
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rtcp_ntp_ms_old,
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rtcp_timestamp_old,
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&freq_khz)) {
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return false;
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}
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double offset = rtcp_timestamp_new - freq_khz * rtcp_ntp_ms_new;
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int64_t rtp_timestamp_unwrapped;
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if (!CompensateForWrapAround(rtp_timestamp, rtcp_timestamp_old,
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&rtp_timestamp_unwrapped)) {
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return false;
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}
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double rtp_timestamp_ntp_ms = (static_cast<double>(rtp_timestamp_unwrapped) -
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offset) / freq_khz + 0.5f;
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if (rtp_timestamp_ntp_ms < 0) {
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return false;
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}
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*rtp_timestamp_in_ms = rtp_timestamp_ntp_ms;
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return true;
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}
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int CheckForWrapArounds(uint32_t new_timestamp, uint32_t old_timestamp) {
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if (new_timestamp < old_timestamp) {
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// This difference should be less than -2^31 if we have had a wrap around
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// (e.g. |new_timestamp| = 1, |rtcp_rtp_timestamp| = 2^32 - 1). Since it is
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// cast to a int32_t, it should be positive.
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if (static_cast<int32_t>(new_timestamp - old_timestamp) > 0) {
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// Forward wrap around.
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return 1;
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}
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} else if (static_cast<int32_t>(old_timestamp - new_timestamp) > 0) {
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// This difference should be less than -2^31 if we have had a backward wrap
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// around. Since it is cast to a int32_t, it should be positive.
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return -1;
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}
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return 0;
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}
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} // namespace synchronization
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} // namespace webrtc
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