/* * Copyright (c) 2017 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. */ // Syncable is used by RtpStreamsSynchronizer in VideoReceiveStream, and // implemented by AudioReceiveStream. #ifndef CALL_SYNCABLE_H_ #define CALL_SYNCABLE_H_ #include #include "absl/types/optional.h" namespace webrtc { class Syncable { public: struct Info { int64_t latest_receive_time_ms = 0; uint32_t latest_received_capture_timestamp = 0; uint32_t capture_time_ntp_secs = 0; uint32_t capture_time_ntp_frac = 0; uint32_t capture_time_source_clock = 0; int current_delay_ms = 0; }; virtual ~Syncable(); virtual uint32_t id() const = 0; virtual absl::optional GetInfo() const = 0; virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp, int64_t* time_ms) const = 0; virtual bool SetMinimumPlayoutDelay(int delay_ms) = 0; virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms, int64_t time_ms) = 0; }; } // namespace webrtc #endif // CALL_SYNCABLE_H_