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9aeaa25d4c
* Merge in branch-heads/4103 (M83) * Disable legacy DTLS protocols (before 1.2) * Update sdk/objc modifications for upstream changes * Update ios and mac deployment targets Co-authored-by: Jim Gustafson <jim@signal.org>
41 lines
1.3 KiB
C++
41 lines
1.3 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
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#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/scoped_refptr.h"
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#include "api/video/encoded_image.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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#include "rtc_base/copy_on_write_buffer.h"
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namespace webrtc {
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class VideoRtpDepacketizer {
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public:
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struct ParsedRtpPayload {
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RTPVideoHeader video_header;
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rtc::CopyOnWriteBuffer video_payload;
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};
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virtual ~VideoRtpDepacketizer() = default;
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virtual absl::optional<ParsedRtpPayload> Parse(
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rtc::CopyOnWriteBuffer rtp_payload) = 0;
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virtual rtc::scoped_refptr<EncodedImageBuffer> AssembleFrame(
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rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads);
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
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