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/*
* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
//
// Helper class for sending the |AbsoluteCaptureTime| header extension.
//
// Supports the "timestamp interpolation" optimization:
// A sender SHOULD save bandwidth by not sending abs-capture-time with every
// RTP packet. It SHOULD still send them at regular intervals (e.g. every
// second) to help mitigate the impact of clock drift and packet loss. Mixers
// SHOULD always send abs-capture-time with the first RTP packet after
// changing capture system.
//
// Timestamp interpolation works fine as long as there’s reasonably low
// NTP/RTP clock drift. This is not always true. Senders that detect “jumps”
// between its NTP and RTP clock mappings SHOULD send abs-capture-time with
// the first RTP packet after such a thing happening.
//
// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
//
class AbsoluteCaptureTimeSender {
public:
static constexpr TimeDelta kInterpolationMaxInterval =
TimeDelta::Millis(1000);
static constexpr TimeDelta kInterpolationMaxError = TimeDelta::Millis(1);
explicit AbsoluteCaptureTimeSender(Clock* clock);
// Returns the source (i.e. SSRC or CSRC) of the capture system.
static uint32_t GetSource(uint32_t ssrc,
rtc::ArrayView<const uint32_t> csrcs);
// Returns a header extension to be sent, or |absl::nullopt| if the header
// extension shouldn't be sent.
absl::optional<AbsoluteCaptureTime> OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
uint64_t absolute_capture_timestamp,
absl::optional<int64_t> estimated_capture_clock_offset);
private:
bool ShouldSendExtension(
Timestamp send_time,
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
uint64_t absolute_capture_timestamp,
absl::optional<int64_t> estimated_capture_clock_offset) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
Clock* const clock_;
rtc::CriticalSection crit_;
Timestamp last_send_time_ RTC_GUARDED_BY(crit_);
uint32_t last_source_ RTC_GUARDED_BY(crit_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(crit_);
uint32_t last_rtp_clock_frequency_ RTC_GUARDED_BY(crit_);
uint64_t last_absolute_capture_timestamp_ RTC_GUARDED_BY(crit_);
absl::optional<int64_t> last_estimated_capture_clock_offset_
RTC_GUARDED_BY(crit_);
}; // AbsoluteCaptureTimeSender
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_