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Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h | 28 |
1 files changed, 28 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h new file mode 100644 index 0000000000..ed5290d1cb --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h @@ -0,0 +1,28 @@ +/* + * Copyright (c) 2024 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_H265_H_ +#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H265_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 VideoRtpDepacketizerH265 : public VideoRtpDepacketizer { + public: + ~VideoRtpDepacketizerH265() override = default; + + absl::optional<ParsedRtpPayload> Parse( + rtc::CopyOnWriteBuffer rtp_payload) override; +}; +} // namespace webrtc + +#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H265_H_ |