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e7c015e112
Bug: webrtc:11567 Change-Id: I4c71f3a28ef875af2c232b1b553840d6e21649d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178804 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31645}
100 lines
3.8 KiB
C++
100 lines
3.8 KiB
C++
/*
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* Copyright (c) 2019 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_RECEIVER_H_
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#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_RECEIVER_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 receiving 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 AbsoluteCaptureTimeReceiver {
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public:
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static constexpr TimeDelta kInterpolationMaxInterval =
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TimeDelta::Millis(5000);
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explicit AbsoluteCaptureTimeReceiver(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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// Sets the NTP clock offset between the sender system (which may be different
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// from the capture system) and the local system. This information is normally
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// provided by passing half the value of the Round-Trip Time estimation given
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// by RTCP sender reports (see DLSR/DLRR).
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//
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// Note that the value must be in Q32.32-formatted fixed-point seconds.
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void SetRemoteToLocalClockOffset(absl::optional<int64_t> value_q32x32);
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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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uint32_t rtp_clock_frequency,
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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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uint32_t rtp_clock_frequency,
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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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uint32_t rtp_clock_frequency) const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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absl::optional<int64_t> AdjustEstimatedCaptureClockOffset(
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absl::optional<int64_t> received_value) 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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absl::optional<int64_t> remote_to_local_clock_offset_ RTC_GUARDED_BY(mutex_);
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Timestamp last_receive_time_ RTC_GUARDED_BY(mutex_);
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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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uint32_t last_rtp_clock_frequency_ RTC_GUARDED_BY(mutex_);
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uint64_t last_absolute_capture_timestamp_ RTC_GUARDED_BY(mutex_);
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absl::optional<int64_t> last_estimated_capture_clock_offset_
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RTC_GUARDED_BY(mutex_);
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}; // AbsoluteCaptureTimeReceiver
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_RECEIVER_H_
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