blob: ea55ab4d221e5d54d4c0487718117d9b94c8bded [file] [log] [blame]
* 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.
#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 receiving 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 AbsoluteCaptureTimeReceiver {
static constexpr TimeDelta kInterpolationMaxInterval =
explicit AbsoluteCaptureTimeReceiver(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);
// Sets the NTP clock offset between the sender system (which may be different
// from the capture system) and the local system. This information is normally
// provided by passing half the value of the Round-Trip Time estimation given
// by RTCP sender reports (see DLSR/DLRR).
// Note that the value must be in Q32.32-formatted fixed-point seconds.
void SetRemoteToLocalClockOffset(absl::optional<int64_t> value_q32x32);
// 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
absl::optional<int64_t> AdjustEstimatedCaptureClockOffset(
absl::optional<int64_t> received_value) const
Clock* const clock_;
rtc::CriticalSection crit_;
absl::optional<int64_t> remote_to_local_clock_offset_ RTC_GUARDED_BY(crit_);
Timestamp last_receive_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_
}; // AbsoluteCaptureTimeReceiver
} // namespace webrtc