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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
81 lines
2.7 KiB
C++
81 lines
2.7 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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#ifndef SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_
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#define SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_
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#include <stdint.h>
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#include <list>
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#include "absl/types/optional.h"
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#include "modules/include/module_common_types_public.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/moving_median_filter.h"
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#include "system_wrappers/include/ntp_time.h"
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namespace webrtc {
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// Class for converting an RTP timestamp to the NTP domain in milliseconds.
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// The class needs to be trained with (at least 2) RTP/NTP timestamp pairs from
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// RTCP sender reports before the convertion can be done.
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class RtpToNtpEstimator {
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public:
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RtpToNtpEstimator();
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~RtpToNtpEstimator();
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// RTP and NTP timestamp pair from a RTCP SR report.
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struct RtcpMeasurement {
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RtcpMeasurement(uint32_t ntp_secs,
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uint32_t ntp_frac,
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int64_t unwrapped_timestamp);
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bool IsEqual(const RtcpMeasurement& other) const;
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NtpTime ntp_time;
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int64_t unwrapped_rtp_timestamp;
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};
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// Estimated parameters from RTP and NTP timestamp pairs in |measurements_|.
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struct Parameters {
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Parameters() : frequency_khz(0.0), offset_ms(0.0) {}
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Parameters(double frequency_khz, double offset_ms)
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: frequency_khz(frequency_khz), offset_ms(offset_ms) {}
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double frequency_khz;
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double offset_ms;
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};
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// Updates measurements with RTP/NTP timestamp pair from a RTCP sender report.
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// |new_rtcp_sr| is set to true if a new report is added.
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bool UpdateMeasurements(uint32_t ntp_secs,
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uint32_t ntp_frac,
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uint32_t rtp_timestamp,
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bool* new_rtcp_sr);
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// Converts an RTP timestamp to the NTP domain in milliseconds.
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// Returns true on success, false otherwise.
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bool Estimate(int64_t rtp_timestamp, int64_t* ntp_timestamp_ms) const;
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// Returns estimated rtp to ntp linear transform parameters.
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const absl::optional<Parameters> params() const;
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static const int kMaxInvalidSamples = 3;
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private:
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void UpdateParameters();
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int consecutive_invalid_samples_;
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std::list<RtcpMeasurement> measurements_;
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absl::optional<Parameters> params_;
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mutable TimestampUnwrapper unwrapper_;
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};
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} // namespace webrtc
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#endif // SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_
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