summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
diff options
context:
space:
mode:
authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h')
-rw-r--r--third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h42
1 files changed, 42 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
new file mode 100644
index 0000000000..4bb358a15f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
@@ -0,0 +1,42 @@
+/*
+ * 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_VP9_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP9_H_
+
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class VideoRtpDepacketizerVp9 : public VideoRtpDepacketizer {
+ public:
+ VideoRtpDepacketizerVp9() = default;
+ VideoRtpDepacketizerVp9(const VideoRtpDepacketizerVp9&) = delete;
+ VideoRtpDepacketizerVp9& operator=(const VideoRtpDepacketizerVp9&) = delete;
+ ~VideoRtpDepacketizerVp9() override = default;
+
+ // Parses vp9 rtp payload descriptor.
+ // Returns zero on error or vp9 payload header offset on success.
+ static int ParseRtpPayload(rtc::ArrayView<const uint8_t> rtp_payload,
+ RTPVideoHeader* video_header);
+
+ absl::optional<ParsedRtpPayload> Parse(
+ rtc::CopyOnWriteBuffer rtp_payload) override;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP9_H_