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Revert "Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator"

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This reverts commit 550b666e20a13f9c22effce878a8e0078a0f7bad.

Reason for revert: breaks downstream projects.

Original change's description:
> Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator
> 
> If Webrtc-ClockEstimation experiment is enabled, median filtering is
> applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth
> out random errors introduced by incorrect RTCP SR reports and networking
> delays.
> 
> Bug: webrtc:8468
> Change-Id: Iec6d094d2809d1efeb0b9483059167d9a03880da
> Reviewed-on: https://webrtc-review.googlesource.com/22682
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20682}

TBR=ilnik@webrtc.org,asapersson@webrtc.org,perkj@webrtc.org

Change-Id: I17345d912bbaf635612c9b399d5f9166de818190
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8468
Reviewed-on: https://webrtc-review.googlesource.com/23320
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20689}
This commit is contained in:
Mirko Bonadei
2017-11-15 16:14:28 +00:00
committed by Commit Bot
parent e403212ede
commit f6703c4dcb
7 changed files with 31 additions and 179 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
modules/rtp_rtcp/BUILD.gn
View File

@ -197,7 +197,6 @@ rtc_static_library("rtp_rtcp") {
"../../logging:rtc_event_log_api",
"../../rtc_base:gtest_prod",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../rtc_base:sequenced_task_checker",
"../../system_wrappers",
"../audio_coding:audio_format_conversion",

3
modules/rtp_rtcp/include/remote_ntp_time_estimator.h
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@ -14,7 +14,6 @@
#include <memory>
#include "rtc_base/constructormagic.h"
#include "rtc_base/numerics/moving_median_filter.h"
#include "system_wrappers/include/rtp_to_ntp_estimator.h"
namespace webrtc {
@ -44,10 +43,8 @@ class RemoteNtpTimeEstimator {
private:
Clock* clock_;
std::unique_ptr<TimestampExtrapolator> ts_extrapolator_;
MovingMedianFilter<int64_t> ntp_clocks_offset_estimator_;
RtpToNtpEstimator rtp_to_ntp_;
int64_t last_timing_log_ms_;
const bool is_experiment_enabled_;
RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
};

39
modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
View File

@ -12,28 +12,19 @@
#include "rtc_base/logging.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/timestamp_extrapolator.h"
namespace webrtc {
namespace {
static const int kTimingLogIntervalMs = 10000;
static const int kClocksOffsetSmoothingWindow = 100;
bool IsClockEstimationExperimentEnabled() {
return webrtc::field_trial::IsEnabled("WebRTC-ClockEstimation");
}
} // namespace
// 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())),
ntp_clocks_offset_estimator_(kClocksOffsetSmoothingWindow),
last_timing_log_ms_(-1),
is_experiment_enabled_(IsClockEstimationExperimentEnabled()) {}
last_timing_log_ms_(-1) {
}
RemoteNtpTimeEstimator::~RemoteNtpTimeEstimator() {}
@ -50,18 +41,12 @@ bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(int64_t rtt,
// 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);
int64_t sender_arrival_time_ms = sender_send_time_ms + rtt / 2;
int64_t remote_to_local_clocks_offset =
receiver_arrival_time_ms - sender_arrival_time_ms;
ntp_clocks_offset_estimator_.Insert(remote_to_local_clocks_offset);
return true;
}
@ -70,21 +55,13 @@ int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) {
if (!rtp_to_ntp_.Estimate(rtp_timestamp, &sender_capture_ntp_ms)) {
return -1;
}
int64_t receiver_capture_ms;
if (is_experiment_enabled_) {
int64_t remote_to_local_clocks_offset =
ntp_clocks_offset_estimator_.GetFilteredValue();
receiver_capture_ms = sender_capture_ntp_ms + remote_to_local_clocks_offset;
} else {
uint32_t timestamp = sender_capture_ntp_ms * 90;
receiver_capture_ms = ts_extrapolator_->ExtrapolateLocalTime(timestamp);
}
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t ntp_offset = clock_->CurrentNtpInMilliseconds() - now_ms;
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) {
RTC_LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp
<< " in NTP clock: " << sender_capture_ntp_ms

67
modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
View File

@ -8,10 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
@ -32,7 +31,7 @@ class RemoteNtpTimeEstimatorTest : public ::testing::Test {
RemoteNtpTimeEstimatorTest()
: local_clock_(kLocalClockInitialTimeMs * 1000),
remote_clock_(kRemoteClockInitialTimeMs * 1000),
estimator_(new RemoteNtpTimeEstimator(&local_clock_)) {}
estimator_(&local_clock_) {}
~RemoteNtpTimeEstimatorTest() {}
void AdvanceTimeMilliseconds(int64_t ms) {
@ -53,21 +52,11 @@ class RemoteNtpTimeEstimatorTest : public ::testing::Test {
ReceiveRtcpSr(kTestRtt, rtcp_timestamp, ntp.seconds(), ntp.fractions());
}
void SendRtcpSrInaccurately(int64_t ntp_error_ms,
int64_t networking_delay_ms) {
uint32_t rtcp_timestamp = GetRemoteTimestamp();
int64_t ntp_error_fractions =
ntp_error_ms * NtpTime::kFractionsPerSecond / 1000;
NtpTime ntp(static_cast<uint64_t>(remote_clock_.CurrentNtpTime()) +
ntp_error_fractions);
AdvanceTimeMilliseconds(kTestRtt / 2 + networking_delay_ms);
ReceiveRtcpSr(kTestRtt, rtcp_timestamp, ntp.seconds(), ntp.fractions());
}
void UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
uint32_t rtp_timestamp, bool expected_result) {
EXPECT_EQ(expected_result, estimator_->UpdateRtcpTimestamp(
rtt, ntp_secs, ntp_frac, rtp_timestamp));
EXPECT_EQ(expected_result,
estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac,
rtp_timestamp));
}
void ReceiveRtcpSr(int64_t rtt,
@ -79,7 +68,7 @@ class RemoteNtpTimeEstimatorTest : public ::testing::Test {
SimulatedClock local_clock_;
SimulatedClock remote_clock_;
std::unique_ptr<RemoteNtpTimeEstimator> estimator_;
RemoteNtpTimeEstimator estimator_;
};
TEST_F(RemoteNtpTimeEstimatorTest, Estimate) {
@ -97,54 +86,14 @@ TEST_F(RemoteNtpTimeEstimatorTest, Estimate) {
// Local peer needs at least 2 RTCP SR to calculate the capture time.
const int64_t kNotEnoughRtcpSr = -1;
EXPECT_EQ(kNotEnoughRtcpSr, estimator_->Estimate(rtp_timestamp));
EXPECT_EQ(kNotEnoughRtcpSr, estimator_.Estimate(rtp_timestamp));
AdvanceTimeMilliseconds(800);
// Remote sends second RTCP SR.
SendRtcpSr();
// Local peer gets enough RTCP SR to calculate the capture time.
EXPECT_EQ(capture_ntp_time_ms, estimator_->Estimate(rtp_timestamp));
}
TEST_F(RemoteNtpTimeEstimatorTest, AveragesErrorsOut) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-ClockEstimation/Enabled/");
// Reset estimator_ because it checks experiment status during construction.
estimator_.reset(new RemoteNtpTimeEstimator(&local_clock_));
// Remote peer sends first 5 RTCP SR without errors.
AdvanceTimeMilliseconds(1000);
SendRtcpSr();
AdvanceTimeMilliseconds(1000);
SendRtcpSr();
AdvanceTimeMilliseconds(1000);
SendRtcpSr();
AdvanceTimeMilliseconds(1000);
SendRtcpSr();
AdvanceTimeMilliseconds(1000);
SendRtcpSr();
AdvanceTimeMilliseconds(15);
uint32_t rtp_timestamp = GetRemoteTimestamp();
int64_t capture_ntp_time_ms = local_clock_.CurrentNtpInMilliseconds();
// Local peer gets enough RTCP SR to calculate the capture time.
EXPECT_EQ(capture_ntp_time_ms, estimator_->Estimate(rtp_timestamp));
// Remote sends corrupted RTCP SRs
AdvanceTimeMilliseconds(1000);
SendRtcpSrInaccurately(10, 10);
AdvanceTimeMilliseconds(1000);
SendRtcpSrInaccurately(-20, 5);
// New RTP packet to estimate timestamp.
AdvanceTimeMilliseconds(150);
rtp_timestamp = GetRemoteTimestamp();
capture_ntp_time_ms = local_clock_.CurrentNtpInMilliseconds();
// Errors should be averaged out.
EXPECT_EQ(capture_ntp_time_ms, estimator_->Estimate(rtp_timestamp));
EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp));
}
} // namespace webrtc

5
system_wrappers/BUILD.gn
View File

@ -101,10 +101,7 @@ rtc_static_library("system_wrappers") {
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps += [
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_numerics",
]
deps += [ "../rtc_base:rtc_base_approved" ]
}
rtc_source_set("cpu_features_api") {

27
system_wrappers/include/rtp_to_ntp_estimator.h
View File

@ -15,7 +15,6 @@
#include "api/optional.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/numerics/moving_median_filter.h"
#include "system_wrappers/include/ntp_time.h"
#include "typedefs.h" // NOLINT(build/include)
@ -41,23 +40,8 @@ class RtpToNtpEstimator {
// Estimated parameters from RTP and NTP timestamp pairs in |measurements_|.
struct Parameters {
// Implicit conversion from int because MovingMedianFilter returns 0
// internally if no samples are present. However, it should never happen as
// we don't ask smoothing_filter_ to return anything if there were no
// samples.
Parameters(const int& value) { // NOLINT
RTC_NOTREACHED();
}
Parameters() : frequency_khz(0.0), offset_ms(0.0) {}
double frequency_khz;
double offset_ms;
// Needed to make it work inside MovingMedianFilter
bool operator<(const Parameters& other) const;
bool operator==(const Parameters& other) const;
bool operator<=(const Parameters& other) const;
bool operator!=(const Parameters& other) const;
};
// Updates measurements with RTP/NTP timestamp pair from a RTCP sender report.
@ -71,8 +55,13 @@ class RtpToNtpEstimator {
// Returns true on success, false otherwise.
bool Estimate(int64_t rtp_timestamp, int64_t* rtp_timestamp_ms) const;
// Returns estimated rtp to ntp linear transform parameters.
const rtc::Optional<Parameters> params() const;
const rtc::Optional<Parameters> params() const {
rtc::Optional<Parameters> res;
if (params_calculated_) {
res.emplace(params_);
}
return res;
}
static const int kMaxInvalidSamples = 3;
@ -82,10 +71,8 @@ class RtpToNtpEstimator {
int consecutive_invalid_samples_;
std::list<RtcpMeasurement> measurements_;
Parameters params_;
MovingMedianFilter<Parameters> smoothing_filter_;
bool params_calculated_;
mutable TimestampUnwrapper unwrapper_;
const bool is_experiment_enabled_;
};
} // namespace webrtc

68
system_wrappers/source/rtp_to_ntp_estimator.cc
View File

@ -13,18 +13,11 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
// Number of RTCP SR reports to use to map between RTP and NTP.
const size_t kNumRtcpReportsToUse = 2;
// Number of parameters samples used to smooth.
const size_t kNumSamplesToSmooth = 20;
bool IsClockEstimationExperimentEnabled() {
return webrtc::field_trial::IsEnabled("WebRTC-ClockEstimation");
}
// Calculates the RTP timestamp frequency from two pairs of NTP/RTP timestamps.
bool CalculateFrequency(int64_t ntp_ms1,
@ -50,28 +43,6 @@ bool Contains(const std::list<RtpToNtpEstimator::RtcpMeasurement>& measurements,
}
} // namespace
bool RtpToNtpEstimator::Parameters::operator<(const Parameters& other) const {
if (frequency_khz < other.frequency_khz - 1e-6) {
return true;
} else if (frequency_khz > other.frequency_khz + 1e-6) {
return false;
} else {
return offset_ms < other.offset_ms - 1e-6;
}
}
bool RtpToNtpEstimator::Parameters::operator==(const Parameters& other) const {
return !(other < *this || *this < other);
}
bool RtpToNtpEstimator::Parameters::operator!=(const Parameters& other) const {
return other < *this || *this < other;
}
bool RtpToNtpEstimator::Parameters::operator<=(const Parameters& other) const {
return !(other < *this);
}
RtpToNtpEstimator::RtcpMeasurement::RtcpMeasurement(uint32_t ntp_secs,
uint32_t ntp_frac,
int64_t unwrapped_timestamp)
@ -88,18 +59,13 @@ bool RtpToNtpEstimator::RtcpMeasurement::IsEqual(
// Class for converting an RTP timestamp to the NTP domain.
RtpToNtpEstimator::RtpToNtpEstimator()
: consecutive_invalid_samples_(0),
smoothing_filter_(kNumSamplesToSmooth),
params_calculated_(false),
is_experiment_enabled_(IsClockEstimationExperimentEnabled()) {}
: consecutive_invalid_samples_(0), params_calculated_(false) {}
RtpToNtpEstimator::~RtpToNtpEstimator() {}
void RtpToNtpEstimator::UpdateParameters() {
if (measurements_.size() != kNumRtcpReportsToUse)
return;
Parameters params;
int64_t timestamp_new = measurements_.front().unwrapped_rtp_timestamp;
int64_t timestamp_old = measurements_.back().unwrapped_rtp_timestamp;
@ -107,16 +73,11 @@ void RtpToNtpEstimator::UpdateParameters() {
int64_t ntp_ms_old = measurements_.back().ntp_time.ToMs();
if (!CalculateFrequency(ntp_ms_new, timestamp_new, ntp_ms_old, timestamp_old,
&params.frequency_khz)) {
&params_.frequency_khz)) {
return;
}
params.offset_ms = timestamp_new - params.frequency_khz * ntp_ms_new;
params_.offset_ms = timestamp_new - params_.frequency_khz * ntp_ms_new;
params_calculated_ = true;
if (is_experiment_enabled_) {
smoothing_filter_.Insert(params);
} else {
params_ = params;
}
}
bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
@ -133,7 +94,6 @@ bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
// RTCP SR report already added.
return true;
}
if (!new_measurement.ntp_time.Valid())
return false;
@ -162,7 +122,6 @@ bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
RTC_LOG(LS_WARNING) << "Multiple consecutively invalid RTCP SR reports, "
"clearing measurements.";
measurements_.clear();
smoothing_filter_.Reset();
params_calculated_ = false;
}
consecutive_invalid_samples_ = 0;
@ -186,15 +145,12 @@ bool RtpToNtpEstimator::Estimate(int64_t rtp_timestamp,
int64_t rtp_timestamp_unwrapped = unwrapper_.Unwrap(rtp_timestamp);
Parameters params =
is_experiment_enabled_ ? smoothing_filter_.GetFilteredValue() : params_;
// params_calculated_ should not be true unless ms params.frequency_khz has
// params_calculated_ should not be true unless ms params_.frequency_khz has
// been calculated to something non zero.
RTC_DCHECK_NE(params.frequency_khz, 0.0);
RTC_DCHECK_NE(params_.frequency_khz, 0.0);
double rtp_ms =
(static_cast<double>(rtp_timestamp_unwrapped) - params.offset_ms) /
params.frequency_khz +
(static_cast<double>(rtp_timestamp_unwrapped) - params_.offset_ms) /
params_.frequency_khz +
0.5f;
if (rtp_ms < 0)
@ -203,14 +159,4 @@ bool RtpToNtpEstimator::Estimate(int64_t rtp_timestamp,
*rtp_timestamp_ms = rtp_ms;
return true;
}
const rtc::Optional<RtpToNtpEstimator::Parameters> RtpToNtpEstimator::params()
const {
rtc::Optional<Parameters> res;
if (params_calculated_) {
res.emplace(is_experiment_enabled_ ? smoothing_filter_.GetFilteredValue()
: params_);
}
return res;
}
} // namespace webrtc