| /* |
| * 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_RECEIVER_H_ |
| #define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_RECEIVER_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 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: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/ |
| // |
| class AbsoluteCaptureTimeReceiver { |
| public: |
| static constexpr TimeDelta kInterpolationMaxInterval = |
| TimeDelta::Millis<5000>(); |
| |
| 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); |
| |
| private: |
| 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 |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| |
| absl::optional<int64_t> AdjustEstimatedCaptureClockOffset( |
| absl::optional<int64_t> received_value) const |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| |
| 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_ |
| RTC_GUARDED_BY(crit_); |
| }; // AbsoluteCaptureTimeReceiver |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_RECEIVER_H_ |