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/*
* 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 "absl/types/optional.h"
#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;
absl::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_H_
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