webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.h
Danil Chapovalov 69679598e7 Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface.
Bug: None
Change-Id: I0498d9e82cbc876d54bebc7f3265e3ae6da61614
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171062
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30872}
2020-03-24 15:55:00 +00:00

41 lines
1.3 KiB
C++

/*
* 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_H_
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
#include <stdint.h>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
#include "api/video/encoded_image.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
class VideoRtpDepacketizer {
public:
struct ParsedRtpPayload {
RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer video_payload;
};
virtual ~VideoRtpDepacketizer() = default;
virtual absl::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) = 0;
virtual rtc::scoped_refptr<EncodedImageBuffer> AssembleFrame(
rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_