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