mirror of
https://github.com/mollyim/webrtc.git
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6634c91194
Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023}
87 lines
3.2 KiB
C++
87 lines
3.2 KiB
C++
/*
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* Copyright (c) 2021 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 MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
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#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
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#include "api/array_view.h"
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#include "api/rtp_headers.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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//
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// Helper class for interpolating the `AbsoluteCaptureTime` header extension.
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//
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// Supports the "timestamp interpolation" optimization:
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// A receiver SHOULD memorize the capture system (i.e. CSRC/SSRC), capture
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// timestamp, and RTP timestamp of the most recently received abs-capture-time
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// packet on each received stream. It can then use that information, in
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// combination with RTP timestamps of packets without abs-capture-time, to
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// extrapolate missing capture timestamps.
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//
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// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
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//
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class AbsoluteCaptureTimeInterpolator {
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public:
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static constexpr TimeDelta kInterpolationMaxInterval = TimeDelta::Seconds(5);
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explicit AbsoluteCaptureTimeInterpolator(Clock* clock);
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// Returns the source (i.e. SSRC or CSRC) of the capture system.
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static uint32_t GetSource(uint32_t ssrc,
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rtc::ArrayView<const uint32_t> csrcs);
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// Returns a received header extension, an interpolated header extension, or
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// `absl::nullopt` if it's not possible to interpolate a header extension.
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absl::optional<AbsoluteCaptureTime> OnReceivePacket(
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uint32_t source,
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uint32_t rtp_timestamp,
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int rtp_clock_frequency_hz,
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const absl::optional<AbsoluteCaptureTime>& received_extension);
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private:
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friend class AbsoluteCaptureTimeSender;
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static uint64_t InterpolateAbsoluteCaptureTimestamp(
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uint32_t rtp_timestamp,
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int rtp_clock_frequency_hz,
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uint32_t last_rtp_timestamp,
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uint64_t last_absolute_capture_timestamp);
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bool ShouldInterpolateExtension(Timestamp receive_time,
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uint32_t source,
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uint32_t rtp_timestamp,
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int rtp_clock_frequency_hz) const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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Clock* const clock_;
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Mutex mutex_;
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// Time of the last received header extension eligible for interpolation,
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// MinusInfinity() if no extension was received, or last received one is
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// not eligible for interpolation.
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Timestamp last_receive_time_ RTC_GUARDED_BY(mutex_) =
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Timestamp::MinusInfinity();
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uint32_t last_source_ RTC_GUARDED_BY(mutex_);
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uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_);
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int last_rtp_clock_frequency_hz_ RTC_GUARDED_BY(mutex_);
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AbsoluteCaptureTime last_received_extension_ RTC_GUARDED_BY(mutex_);
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
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