webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h
Florent Castelli 8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00

31 lines
991 B
C++

/*
* 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_VIDEO_RTP_DEPACKETIZER_RAW_H_
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_RAW_H_
#include <optional>
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
class VideoRtpDepacketizerRaw : public VideoRtpDepacketizer {
public:
~VideoRtpDepacketizerRaw() override = default;
std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_RAW_H_