Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
BUG=3111
TEST=try bots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 444c665..d7e7cf8 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -180,7 +180,6 @@
'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h',
'remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
- 'remote_bitrate_estimator/rtp_to_ntp_unittest.cc',
'remote_bitrate_estimator/test/bwe_test_baselinefile.cc',
'remote_bitrate_estimator/test/bwe_test_baselinefile.h',
'remote_bitrate_estimator/test/bwe_test_fileutils.cc',
diff --git a/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h b/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h
deleted file mode 100644
index 08a4d46..0000000
--- a/webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2012 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.
- */
-
-#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_
-#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_
-
-#include <list>
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-namespace synchronization {
-
-struct RtcpMeasurement {
- RtcpMeasurement();
- RtcpMeasurement(uint32_t ntp_secs, uint32_t ntp_frac, uint32_t timestamp);
- uint32_t ntp_secs;
- uint32_t ntp_frac;
- uint32_t rtp_timestamp;
-};
-
-typedef std::list<RtcpMeasurement> RtcpList;
-
-// Updates |rtcp_list| with timestamps from the latest RTCP SR.
-// |new_rtcp_sr| will be set to true if these are the timestamps which have
-// never be added to |rtcp_list|.
-bool UpdateRtcpList(uint32_t ntp_secs,
- uint32_t ntp_frac,
- uint32_t rtp_timestamp,
- RtcpList* rtcp_list,
- bool* new_rtcp_sr);
-
-// Converts an RTP timestamp to the NTP domain in milliseconds using two
-// (RTP timestamp, NTP timestamp) pairs.
-bool RtpToNtpMs(int64_t rtp_timestamp, const RtcpList& rtcp,
- int64_t* timestamp_in_ms);
-
-// Returns 1 there has been a forward wrap around, 0 if there has been no wrap
-// around and -1 if there has been a backwards wrap around (i.e. reordering).
-int CheckForWrapArounds(uint32_t rtp_timestamp, uint32_t rtcp_rtp_timestamp);
-} // namespace synchronization
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_RTP_TO_NTP_H_
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
index 810da46..7292cae 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
@@ -21,10 +21,8 @@
'sources': [
'include/bwe_defines.h',
'include/remote_bitrate_estimator.h',
- 'include/rtp_to_ntp.h',
'rate_statistics.cc',
'rate_statistics.h',
- 'rtp_to_ntp.cc',
], # source
},
{
diff --git a/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc b/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc
deleted file mode 100644
index 775cd0d..0000000
--- a/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc
+++ /dev/null
@@ -1,153 +0,0 @@
-/*
- * Copyright (c) 2012 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/remote_bitrate_estimator/include/rtp_to_ntp.h"
-
-#include "webrtc/system_wrappers/interface/clock.h"
-
-#include <assert.h>
-
-namespace webrtc {
-
-namespace synchronization {
-
-RtcpMeasurement::RtcpMeasurement()
- : ntp_secs(0), ntp_frac(0), rtp_timestamp(0) {}
-
-RtcpMeasurement::RtcpMeasurement(uint32_t ntp_secs, uint32_t ntp_frac,
- uint32_t timestamp)
- : ntp_secs(ntp_secs), ntp_frac(ntp_frac), rtp_timestamp(timestamp) {}
-
-// Calculates the RTP timestamp frequency from two pairs of NTP and RTP
-// timestamps.
-bool CalculateFrequency(
- int64_t rtcp_ntp_ms1,
- uint32_t rtp_timestamp1,
- int64_t rtcp_ntp_ms2,
- uint32_t rtp_timestamp2,
- double* frequency_khz) {
- if (rtcp_ntp_ms1 <= rtcp_ntp_ms2) {
- return false;
- }
- *frequency_khz = static_cast<double>(rtp_timestamp1 - rtp_timestamp2) /
- static_cast<double>(rtcp_ntp_ms1 - rtcp_ntp_ms2);
- return true;
-}
-
-// Detects if there has been a wraparound between |old_timestamp| and
-// |new_timestamp|, and compensates by adding 2^32 if that is the case.
-bool CompensateForWrapAround(uint32_t new_timestamp,
- uint32_t old_timestamp,
- int64_t* compensated_timestamp) {
- assert(compensated_timestamp);
- int64_t wraps = synchronization::CheckForWrapArounds(new_timestamp,
- old_timestamp);
- if (wraps < 0) {
- // Reordering, don't use this packet.
- return false;
- }
- *compensated_timestamp = new_timestamp + (wraps << 32);
- return true;
-}
-
-bool UpdateRtcpList(uint32_t ntp_secs,
- uint32_t ntp_frac,
- uint32_t rtp_timestamp,
- RtcpList* rtcp_list,
- bool* new_rtcp_sr) {
- *new_rtcp_sr = false;
- if (ntp_secs == 0 && ntp_frac == 0) {
- return false;
- }
-
- RtcpMeasurement measurement;
- measurement.ntp_secs = ntp_secs;
- measurement.ntp_frac = ntp_frac;
- measurement.rtp_timestamp = rtp_timestamp;
-
- for (RtcpList::iterator it = rtcp_list->begin();
- it != rtcp_list->end(); ++it) {
- if (measurement.ntp_secs == (*it).ntp_secs &&
- measurement.ntp_frac == (*it).ntp_frac) {
- // This RTCP has already been added to the list.
- return true;
- }
- }
-
- // We need two RTCP SR reports to map between RTP and NTP. More than two will
- // not improve the mapping.
- if (rtcp_list->size() == 2) {
- rtcp_list->pop_back();
- }
- rtcp_list->push_front(measurement);
- *new_rtcp_sr = true;
- return true;
-}
-
-// Converts |rtp_timestamp| to the NTP time base using the NTP and RTP timestamp
-// pairs in |rtcp|. The converted timestamp is returned in
-// |rtp_timestamp_in_ms|. This function compensates for wrap arounds in RTP
-// timestamps and returns false if it can't do the conversion due to reordering.
-bool RtpToNtpMs(int64_t rtp_timestamp,
- const synchronization::RtcpList& rtcp,
- int64_t* rtp_timestamp_in_ms) {
- assert(rtcp.size() == 2);
- int64_t rtcp_ntp_ms_new = Clock::NtpToMs(rtcp.front().ntp_secs,
- rtcp.front().ntp_frac);
- int64_t rtcp_ntp_ms_old = Clock::NtpToMs(rtcp.back().ntp_secs,
- rtcp.back().ntp_frac);
- int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp;
- int64_t rtcp_timestamp_old = rtcp.back().rtp_timestamp;
- if (!CompensateForWrapAround(rtcp_timestamp_new,
- rtcp_timestamp_old,
- &rtcp_timestamp_new)) {
- return false;
- }
- double freq_khz;
- if (!CalculateFrequency(rtcp_ntp_ms_new,
- rtcp_timestamp_new,
- rtcp_ntp_ms_old,
- rtcp_timestamp_old,
- &freq_khz)) {
- return false;
- }
- double offset = rtcp_timestamp_new - freq_khz * rtcp_ntp_ms_new;
- int64_t rtp_timestamp_unwrapped;
- if (!CompensateForWrapAround(rtp_timestamp, rtcp_timestamp_old,
- &rtp_timestamp_unwrapped)) {
- return false;
- }
- double rtp_timestamp_ntp_ms = (static_cast<double>(rtp_timestamp_unwrapped) -
- offset) / freq_khz + 0.5f;
- if (rtp_timestamp_ntp_ms < 0) {
- return false;
- }
- *rtp_timestamp_in_ms = rtp_timestamp_ntp_ms;
- return true;
-}
-
-int CheckForWrapArounds(uint32_t new_timestamp, uint32_t old_timestamp) {
- if (new_timestamp < old_timestamp) {
- // This difference should be less than -2^31 if we have had a wrap around
- // (e.g. |new_timestamp| = 1, |rtcp_rtp_timestamp| = 2^32 - 1). Since it is
- // cast to a int32_t, it should be positive.
- if (static_cast<int32_t>(new_timestamp - old_timestamp) > 0) {
- // Forward wrap around.
- return 1;
- }
- } else if (static_cast<int32_t>(old_timestamp - new_timestamp) > 0) {
- // This difference should be less than -2^31 if we have had a backward wrap
- // around. Since it is cast to a int32_t, it should be positive.
- return -1;
- }
- return 0;
-}
-} // namespace synchronization
-} // namespace webrtc
diff --git a/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc b/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc
deleted file mode 100644
index aff314a..0000000
--- a/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp_unittest.cc
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- * Copyright (c) 2012 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 "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
-
-namespace webrtc {
-
-TEST(WrapAroundTests, NoWrap) {
- EXPECT_EQ(0, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0xFFFFFFFE));
- EXPECT_EQ(0, synchronization::CheckForWrapArounds(1, 0));
- EXPECT_EQ(0, synchronization::CheckForWrapArounds(0x00010000, 0x0000FFFF));
-}
-
-TEST(WrapAroundTests, ForwardWrap) {
- EXPECT_EQ(1, synchronization::CheckForWrapArounds(0, 0xFFFFFFFF));
- EXPECT_EQ(1, synchronization::CheckForWrapArounds(0, 0xFFFF0000));
- EXPECT_EQ(1, synchronization::CheckForWrapArounds(0x0000FFFF, 0xFFFFFFFF));
- EXPECT_EQ(1, synchronization::CheckForWrapArounds(0x0000FFFF, 0xFFFF0000));
-}
-
-TEST(WrapAroundTests, BackwardWrap) {
- EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0));
- EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFF0000, 0));
- EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFFFFFF, 0x0000FFFF));
- EXPECT_EQ(-1, synchronization::CheckForWrapArounds(0xFFFF0000, 0x0000FFFF));
-}
-
-TEST(WrapAroundTests, OldRtcpWrapped) {
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0;
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- // This expected to fail since it's highly unlikely that the older RTCP
- // has a much smaller RTP timestamp than the newer.
- EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms));
-}
-
-TEST(WrapAroundTests, NewRtcpWrapped) {
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0xFFFFFFFF;
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- int64_t timestamp_in_ms = -1;
- EXPECT_TRUE(synchronization::RtpToNtpMs(rtcp.back().rtp_timestamp, rtcp,
- ×tamp_in_ms));
- // Since this RTP packet has the same timestamp as the RTCP packet constructed
- // at time 0 it should be mapped to 0 as well.
- EXPECT_EQ(0, timestamp_in_ms);
-}
-
-TEST(WrapAroundTests, RtpWrapped) {
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0xFFFFFFFF - 2 * kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- EXPECT_TRUE(synchronization::RtpToNtpMs(timestamp, rtcp,
- ×tamp_in_ms));
- // Since this RTP packet has the same timestamp as the RTCP packet constructed
- // at time 0 it should be mapped to 0 as well.
- EXPECT_EQ(2, timestamp_in_ms);
-}
-
-TEST(WrapAroundTests, OldRtp_RtcpsWrapped) {
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= 2*kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp,
- ×tamp_in_ms));
-}
-
-TEST(WrapAroundTests, OldRtp_NewRtcpWrapped) {
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0xFFFFFFFF;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- EXPECT_TRUE(synchronization::RtpToNtpMs(timestamp, rtcp,
- ×tamp_in_ms));
- // Constructed at the same time as the first RTCP and should therefore be
- // mapped to zero.
- EXPECT_EQ(0, timestamp_in_ms);
-}
-
-TEST(WrapAroundTests, OldRtp_OldRtcpWrapped) {
- const uint32_t kOneMsInNtpFrac = 4294967;
- const uint32_t kTimestampTicksPerMs = 90;
- synchronization::RtcpList rtcp;
- uint32_t ntp_sec = 0;
- uint32_t ntp_frac = 0;
- uint32_t timestamp = 0;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp -= kTimestampTicksPerMs;
- rtcp.push_front(synchronization::RtcpMeasurement(ntp_sec, ntp_frac,
- timestamp));
- ntp_frac += kOneMsInNtpFrac;
- timestamp += 2*kTimestampTicksPerMs;
- int64_t timestamp_in_ms = -1;
- EXPECT_FALSE(synchronization::RtpToNtpMs(timestamp, rtcp,
- ×tamp_in_ms));
-}
-}; // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/Android.mk b/webrtc/modules/video_coding/main/source/Android.mk
index 9ebdbed..a8cf2d0 100644
--- a/webrtc/modules/video_coding/main/source/Android.mk
+++ b/webrtc/modules/video_coding/main/source/Android.mk
@@ -37,7 +37,6 @@
receiver.cc \
rtt_filter.cc \
session_info.cc \
- timestamp_extrapolator.cc \
timestamp_map.cc \
timing.cc \
video_coding_impl.cc
@@ -56,7 +55,7 @@
$(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
$(LOCAL_PATH)/../../../../common_video/interface \
$(LOCAL_PATH)/../../utility/include \
- $(LOCAL_PATH)/../../../../system_wrappers/interface
+ $(LOCAL_PATH)/../../../../system_wrappers/interface
LOCAL_SHARED_LIBRARIES := \
libcutils \
diff --git a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc b/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc
deleted file mode 100644
index bde2f0a..0000000
--- a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc
+++ /dev/null
@@ -1,231 +0,0 @@
-/*
- * Copyright (c) 2011 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/video_coding/main/source/timestamp_extrapolator.h"
-
-#include <algorithm>
-
-namespace webrtc {
-
-VCMTimestampExtrapolator::VCMTimestampExtrapolator(int64_t start_ms)
- : _rwLock(RWLockWrapper::CreateRWLock()),
- _startMs(0),
- _firstTimestamp(0),
- _wrapArounds(0),
- _prevUnwrappedTimestamp(-1),
- _prevWrapTimestamp(-1),
- _lambda(1),
- _firstAfterReset(true),
- _packetCount(0),
- _startUpFilterDelayInPackets(2),
- _detectorAccumulatorPos(0),
- _detectorAccumulatorNeg(0),
- _alarmThreshold(60e3),
- _accDrift(6600), // in timestamp ticks, i.e. 15 ms
- _accMaxError(7000),
- _P11(1e10) {
- Reset(start_ms);
-}
-
-VCMTimestampExtrapolator::~VCMTimestampExtrapolator()
-{
- delete _rwLock;
-}
-
-void
-VCMTimestampExtrapolator::Reset(int64_t start_ms)
-{
- WriteLockScoped wl(*_rwLock);
- _startMs = start_ms;
- _prevMs = _startMs;
- _firstTimestamp = 0;
- _w[0] = 90.0;
- _w[1] = 0;
- _P[0][0] = 1;
- _P[1][1] = _P11;
- _P[0][1] = _P[1][0] = 0;
- _firstAfterReset = true;
- _prevUnwrappedTimestamp = -1;
- _prevWrapTimestamp = -1;
- _wrapArounds = 0;
- _packetCount = 0;
- _detectorAccumulatorPos = 0;
- _detectorAccumulatorNeg = 0;
-}
-
-void
-VCMTimestampExtrapolator::Update(int64_t tMs, uint32_t ts90khz)
-{
-
- _rwLock->AcquireLockExclusive();
- if (tMs - _prevMs > 10e3)
- {
- // Ten seconds without a complete frame.
- // Reset the extrapolator
- _rwLock->ReleaseLockExclusive();
- Reset(tMs);
- _rwLock->AcquireLockExclusive();
- }
- else
- {
- _prevMs = tMs;
- }
-
- // Remove offset to prevent badly scaled matrices
- tMs -= _startMs;
-
- CheckForWrapArounds(ts90khz);
-
- int64_t unwrapped_ts90khz = static_cast<int64_t>(ts90khz) +
- _wrapArounds * ((static_cast<int64_t>(1) << 32) - 1);
-
- if (_prevUnwrappedTimestamp >= 0 &&
- unwrapped_ts90khz < _prevUnwrappedTimestamp)
- {
- // Drop reordered frames.
- _rwLock->ReleaseLockExclusive();
- return;
- }
-
- if (_firstAfterReset)
- {
- // Make an initial guess of the offset,
- // should be almost correct since tMs - _startMs
- // should about zero at this time.
- _w[1] = -_w[0] * tMs;
- _firstTimestamp = unwrapped_ts90khz;
- _firstAfterReset = false;
- }
-
- double residual =
- (static_cast<double>(unwrapped_ts90khz) - _firstTimestamp) -
- static_cast<double>(tMs) * _w[0] - _w[1];
- if (DelayChangeDetection(residual) &&
- _packetCount >= _startUpFilterDelayInPackets)
- {
- // A sudden change of average network delay has been detected.
- // Force the filter to adjust its offset parameter by changing
- // the offset uncertainty. Don't do this during startup.
- _P[1][1] = _P11;
- }
- //T = [t(k) 1]';
- //that = T'*w;
- //K = P*T/(lambda + T'*P*T);
- double K[2];
- K[0] = _P[0][0] * tMs + _P[0][1];
- K[1] = _P[1][0] * tMs + _P[1][1];
- double TPT = _lambda + tMs * K[0] + K[1];
- K[0] /= TPT;
- K[1] /= TPT;
- //w = w + K*(ts(k) - that);
- _w[0] = _w[0] + K[0] * residual;
- _w[1] = _w[1] + K[1] * residual;
- //P = 1/lambda*(P - K*T'*P);
- double p00 = 1 / _lambda * (_P[0][0] - (K[0] * tMs * _P[0][0] + K[0] * _P[1][0]));
- double p01 = 1 / _lambda * (_P[0][1] - (K[0] * tMs * _P[0][1] + K[0] * _P[1][1]));
- _P[1][0] = 1 / _lambda * (_P[1][0] - (K[1] * tMs * _P[0][0] + K[1] * _P[1][0]));
- _P[1][1] = 1 / _lambda * (_P[1][1] - (K[1] * tMs * _P[0][1] + K[1] * _P[1][1]));
- _P[0][0] = p00;
- _P[0][1] = p01;
- _prevUnwrappedTimestamp = unwrapped_ts90khz;
- if (_packetCount < _startUpFilterDelayInPackets)
- {
- _packetCount++;
- }
- _rwLock->ReleaseLockExclusive();
-}
-
-int64_t
-VCMTimestampExtrapolator::ExtrapolateLocalTime(uint32_t timestamp90khz)
-{
- ReadLockScoped rl(*_rwLock);
- int64_t localTimeMs = 0;
- CheckForWrapArounds(timestamp90khz);
- double unwrapped_ts90khz = static_cast<double>(timestamp90khz) +
- _wrapArounds * ((static_cast<int64_t>(1) << 32) - 1);
- if (_packetCount == 0)
- {
- localTimeMs = -1;
- }
- else if (_packetCount < _startUpFilterDelayInPackets)
- {
- localTimeMs = _prevMs + static_cast<int64_t>(
- static_cast<double>(unwrapped_ts90khz - _prevUnwrappedTimestamp) /
- 90.0 + 0.5);
- }
- else
- {
- if (_w[0] < 1e-3)
- {
- localTimeMs = _startMs;
- }
- else
- {
- double timestampDiff = unwrapped_ts90khz -
- static_cast<double>(_firstTimestamp);
- localTimeMs = static_cast<int64_t>(
- static_cast<double>(_startMs) + (timestampDiff - _w[1]) /
- _w[0] + 0.5);
- }
- }
- return localTimeMs;
-}
-
-// Investigates if the timestamp clock has overflowed since the last timestamp and
-// keeps track of the number of wrap arounds since reset.
-void
-VCMTimestampExtrapolator::CheckForWrapArounds(uint32_t ts90khz)
-{
- if (_prevWrapTimestamp == -1)
- {
- _prevWrapTimestamp = ts90khz;
- return;
- }
- if (ts90khz < _prevWrapTimestamp)
- {
- // This difference will probably be less than -2^31 if we have had a wrap around
- // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is casted to a Word32,
- // it should be positive.
- if (static_cast<int32_t>(ts90khz - _prevWrapTimestamp) > 0)
- {
- // Forward wrap around
- _wrapArounds++;
- }
- }
- // This difference will probably be less than -2^31 if we have had a backward wrap around.
- // Since it is casted to a Word32, it should be positive.
- else if (static_cast<int32_t>(_prevWrapTimestamp - ts90khz) > 0)
- {
- // Backward wrap around
- _wrapArounds--;
- }
- _prevWrapTimestamp = ts90khz;
-}
-
-bool
-VCMTimestampExtrapolator::DelayChangeDetection(double error)
-{
- // CUSUM detection of sudden delay changes
- error = (error > 0) ? std::min(error, _accMaxError) :
- std::max(error, -_accMaxError);
- _detectorAccumulatorPos =
- std::max(_detectorAccumulatorPos + error - _accDrift, (double)0);
- _detectorAccumulatorNeg =
- std::min(_detectorAccumulatorNeg + error + _accDrift, (double)0);
- if (_detectorAccumulatorPos > _alarmThreshold || _detectorAccumulatorNeg < -_alarmThreshold)
- {
- // Alarm
- _detectorAccumulatorPos = _detectorAccumulatorNeg = 0;
- return true;
- }
- return false;
-}
-
-}
diff --git a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h b/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h
deleted file mode 100644
index 151e4de..0000000
--- a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 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.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
-
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-class VCMTimestampExtrapolator
-{
-public:
- explicit VCMTimestampExtrapolator(int64_t start_ms);
- ~VCMTimestampExtrapolator();
- void Update(int64_t tMs, uint32_t ts90khz);
- int64_t ExtrapolateLocalTime(uint32_t timestamp90khz);
- void Reset(int64_t start_ms);
-
-private:
- void CheckForWrapArounds(uint32_t ts90khz);
- bool DelayChangeDetection(double error);
- RWLockWrapper* _rwLock;
- double _w[2];
- double _P[2][2];
- int64_t _startMs;
- int64_t _prevMs;
- uint32_t _firstTimestamp;
- int32_t _wrapArounds;
- int64_t _prevUnwrappedTimestamp;
- int64_t _prevWrapTimestamp;
- const double _lambda;
- bool _firstAfterReset;
- uint32_t _packetCount;
- const uint32_t _startUpFilterDelayInPackets;
-
- double _detectorAccumulatorPos;
- double _detectorAccumulatorNeg;
- const double _alarmThreshold;
- const double _accDrift;
- const double _accMaxError;
- const double _P11;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
diff --git a/webrtc/modules/video_coding/main/source/timing.cc b/webrtc/modules/video_coding/main/source/timing.cc
index dd82187..af0e35c 100644
--- a/webrtc/modules/video_coding/main/source/timing.cc
+++ b/webrtc/modules/video_coding/main/source/timing.cc
@@ -10,12 +10,10 @@
#include "webrtc/modules/video_coding/main/source/timing.h"
-
#include "webrtc/modules/video_coding/main/source/internal_defines.h"
#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/timestamp_extrapolator.h"
#include "webrtc/system_wrappers/interface/clock.h"
-
+#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
namespace webrtc {
@@ -35,8 +33,7 @@
prev_frame_timestamp_(0) {
if (master_timing == NULL) {
master_ = true;
- ts_extrapolator_ =
- new VCMTimestampExtrapolator(clock_->TimeInMilliseconds());
+ ts_extrapolator_ = new TimestampExtrapolator(clock_->TimeInMilliseconds());
} else {
ts_extrapolator_ = master_timing->ts_extrapolator_;
}
diff --git a/webrtc/modules/video_coding/main/source/timing.h b/webrtc/modules/video_coding/main/source/timing.h
index fcb0402..1dca5e6 100644
--- a/webrtc/modules/video_coding/main/source/timing.h
+++ b/webrtc/modules/video_coding/main/source/timing.h
@@ -18,7 +18,7 @@
namespace webrtc {
class Clock;
-class VCMTimestampExtrapolator;
+class TimestampExtrapolator;
class VCMTiming {
public:
@@ -101,7 +101,7 @@
CriticalSectionWrapper* crit_sect_;
Clock* clock_;
bool master_;
- VCMTimestampExtrapolator* ts_extrapolator_;
+ TimestampExtrapolator* ts_extrapolator_;
VCMCodecTimer codec_timer_;
uint32_t render_delay_ms_;
uint32_t min_playout_delay_ms_;
diff --git a/webrtc/modules/video_coding/main/source/video_coding.gypi b/webrtc/modules/video_coding/main/source/video_coding.gypi
index b4f6cb7..f19a585 100644
--- a/webrtc/modules/video_coding/main/source/video_coding.gypi
+++ b/webrtc/modules/video_coding/main/source/video_coding.gypi
@@ -48,7 +48,6 @@
'receiver.h',
'rtt_filter.h',
'session_info.h',
- 'timestamp_extrapolator.h',
'timestamp_map.h',
'timing.h',
'video_coding_impl.h',
@@ -72,7 +71,6 @@
'receiver.cc',
'rtt_filter.cc',
'session_info.cc',
- 'timestamp_extrapolator.cc',
'timestamp_map.cc',
'timing.cc',
'video_coding_impl.cc',
