From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../rtp_rtcp/source/video_rtp_depacketizer_h264.h | 28 ++++++++++++++++++++++ 1 file changed, 28 insertions(+) create mode 100644 third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h') diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h new file mode 100644 index 0000000000..cbea860049 --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h @@ -0,0 +1,28 @@ +/* + * 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 Parse( + rtc::CopyOnWriteBuffer rtp_payload) override; +}; +} // namespace webrtc + +#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_ -- cgit v1.2.3