webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h
Danil Chapovalov 61d6471912 Change H264 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: If5169f47d85918356fa66e2bf3422d722044aa1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165581
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30264}
2020-01-15 12:26:55 +00:00

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/*
* Copyright (c) 2020 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_H264_H_
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_
#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
class VideoRtpDepacketizerH264 : public VideoRtpDepacketizer {
public:
~VideoRtpDepacketizerH264() override = default;
absl::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_