webrtc/call/syncable.h
Danil Chapovalov b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:09:49 +00:00

43 lines
1.2 KiB
C++

/*
* 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 <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 int id() const = 0;
virtual absl::optional<Info> GetInfo() const = 0;
virtual uint32_t GetPlayoutTimestamp() const = 0;
virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
};
} // namespace webrtc
#endif // CALL_SYNCABLE_H_