| /* |
| * Copyright (c) 2020 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_H264_H_ |
| #define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_ |
| |
| #include <optional> |
| |
| #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| |
| namespace webrtc { |
| class VideoRtpDepacketizerH264 : public VideoRtpDepacketizer { |
| public: |
| ~VideoRtpDepacketizerH264() override = default; |
| |
| std::optional<ParsedRtpPayload> Parse( |
| rtc::CopyOnWriteBuffer rtp_payload) override; |
| }; |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_ |