webrtc/modules/rtp_rtcp/source/rtp_depacketizer_av1.h
Danil Chapovalov ccf12c6e97 Reland "Add AV1 RtpDepacketizer class"
This is a reland of 49470c2ac4
Tentative reland to rule-out bot flakiness.

Original change's description:
> Add AV1 RtpDepacketizer class
>
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
>
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}

TBR=saza@webrtc.org,philipel@webrtc.org

Bug: webrtc:11042
Change-Id: Ibd672ce685bcab86960500740465539ed70fcdf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159941
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29819}
2019-11-18 15:23:08 +00:00

34 lines
1.1 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_RTP_DEPACKETIZER_AV1_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_DEPACKETIZER_AV1_H_
#include <stddef.h>
#include <stdint.h>
#include "modules/rtp_rtcp/source/rtp_format.h"
namespace webrtc {
class RtpDepacketizerAv1 : public RtpDepacketizer {
public:
RtpDepacketizerAv1() = default;
RtpDepacketizerAv1(const RtpDepacketizerAv1&) = delete;
RtpDepacketizerAv1& operator=(const RtpDepacketizerAv1&) = delete;
~RtpDepacketizerAv1() override = default;
bool Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) override;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_DEPACKETIZER_AV1_H_