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cfea2182f8
Bug: webrtc:12338 Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34696}
88 lines
3.2 KiB
C++
88 lines
3.2 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_SENDER_H_
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#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_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 sending the `AbsoluteCaptureTime` header extension.
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//
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// Supports the "timestamp interpolation" optimization:
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// A sender SHOULD save bandwidth by not sending abs-capture-time with every
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// RTP packet. It SHOULD still send them at regular intervals (e.g. every
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// second) to help mitigate the impact of clock drift and packet loss. Mixers
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// SHOULD always send abs-capture-time with the first RTP packet after
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// changing capture system.
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//
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// Timestamp interpolation works fine as long as there’s reasonably low
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// NTP/RTP clock drift. This is not always true. Senders that detect “jumps”
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// between its NTP and RTP clock mappings SHOULD send abs-capture-time with
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// the first RTP packet after such a thing happening.
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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 AbsoluteCaptureTimeSender {
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public:
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static constexpr TimeDelta kInterpolationMaxInterval =
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TimeDelta::Millis(1000);
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static constexpr TimeDelta kInterpolationMaxError = TimeDelta::Millis(1);
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explicit AbsoluteCaptureTimeSender(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 header extension to be sent, or `absl::nullopt` if the header
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// extension shouldn't be sent.
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absl::optional<AbsoluteCaptureTime> OnSendPacket(
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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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uint64_t absolute_capture_timestamp,
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absl::optional<int64_t> estimated_capture_clock_offset);
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private:
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bool ShouldSendExtension(
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Timestamp send_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,
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uint64_t absolute_capture_timestamp,
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absl::optional<int64_t> estimated_capture_clock_offset) 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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Timestamp last_send_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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}; // AbsoluteCaptureTimeSender
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
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