代码拉取完成,页面将自动刷新
同步操作将从 egege/webrtc-src 强制同步,此操作会覆盖自 Fork 仓库以来所做的任何修改,且无法恢复!!!
确定后同步将在后台操作,完成时将刷新页面,请耐心等待。
/*
* 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 VideoReceiveStreamInterface,
// and implemented by AudioReceiveStreamInterface.
#ifndef CALL_SYNCABLE_H_
#define CALL_SYNCABLE_H_
#include <stdint.h>
#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<Info> 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_
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