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https://github.com/mollyim/webrtc.git
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355c47309d
Bug: none Change-Id: I4adca8b4cbf4ffa15172fabc1eaba8c2b65c6fb1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222650 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34306}
42 lines
1.5 KiB
C++
42 lines
1.5 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_VP9_H_
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#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP9_H_
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#include <cstdint>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.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 VideoRtpDepacketizerVp9 : public VideoRtpDepacketizer {
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public:
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VideoRtpDepacketizerVp9() = default;
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VideoRtpDepacketizerVp9(const VideoRtpDepacketizerVp9&) = delete;
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VideoRtpDepacketizerVp9& operator=(const VideoRtpDepacketizerVp9&) = delete;
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~VideoRtpDepacketizerVp9() override = default;
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// Parses vp9 rtp payload descriptor.
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// Returns zero on error or vp9 payload header offset on success.
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static int ParseRtpPayload(rtc::ArrayView<const uint8_t> rtp_payload,
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RTPVideoHeader* video_header);
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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_VP9_H_
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