blob: f5ec820dd54465b53bbee779dc747fbe0017c851 [file] [log] [blame]
* Copyright (c) 2021 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.
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
// Helper class for interpolating the `AbsoluteCaptureTime` header extension.
// Supports the "timestamp interpolation" optimization:
// A receiver SHOULD memorize the capture system (i.e. CSRC/SSRC), capture
// timestamp, and RTP timestamp of the most recently received abs-capture-time
// packet on each received stream. It can then use that information, in
// combination with RTP timestamps of packets without abs-capture-time, to
// extrapolate missing capture timestamps.
// See:
class AbsoluteCaptureTimeInterpolator {
static constexpr TimeDelta kInterpolationMaxInterval =
explicit AbsoluteCaptureTimeInterpolator(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 received header extension, an interpolated header extension, or
// `absl::nullopt` if it's not possible to interpolate a header extension.
absl::optional<AbsoluteCaptureTime> OnReceivePacket(
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
const absl::optional<AbsoluteCaptureTime>& received_extension);
friend class AbsoluteCaptureTimeSender;
static uint64_t InterpolateAbsoluteCaptureTimestamp(
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
uint32_t last_rtp_timestamp,
uint64_t last_absolute_capture_timestamp);
bool ShouldInterpolateExtension(Timestamp receive_time,
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency) const
Clock* const clock_;
Mutex mutex_;
Timestamp last_receive_time_ RTC_GUARDED_BY(mutex_);
uint32_t last_source_ RTC_GUARDED_BY(mutex_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_);
uint32_t last_rtp_clock_frequency_ RTC_GUARDED_BY(mutex_);
uint64_t last_absolute_capture_timestamp_ RTC_GUARDED_BY(mutex_);
absl::optional<int64_t> last_estimated_capture_clock_offset_
}; // AbsoluteCaptureTimeInterpolator
} // namespace webrtc