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27f4d325ad
Bug: webrtc:11152 Change-Id: I27d6a62093d36a4e77dd35d4c115af8cdcc0178a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162202 Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30160}
30 lines
1 KiB
C++
30 lines
1 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_GENERIC_H_
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#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_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 VideoRtpDepacketizerGeneric : public VideoRtpDepacketizer {
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public:
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~VideoRtpDepacketizerGeneric() 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_GENERIC_H_
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