f43e8ebab9
1. Depacketize single nalu packet/AP/FU 2. Insert start code before each nalu Bug: webrtc:13485 Change-Id: I8346f9c31e61e5d3c2c7e1bf5fdaae4018a1ff78 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325660 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41628}
29 lines
1007 B
C++
29 lines
1007 B
C++
/*
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* Copyright (c) 2024 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_H265_H_
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#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H265_H_
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
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#include "rtc_base/copy_on_write_buffer.h"
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namespace webrtc {
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class VideoRtpDepacketizerH265 : public VideoRtpDepacketizer {
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public:
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~VideoRtpDepacketizerH265() override = default;
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absl::optional<ParsedRtpPayload> Parse(
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rtc::CopyOnWriteBuffer rtp_payload) override;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H265_H_
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