Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 30 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h
new file mode 100644
index 0000000000..27056da481
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h
@@ -0,0 +1,30 @@
+/*
+ *  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_GENERIC_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_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 VideoRtpDepacketizerGeneric : public VideoRtpDepacketizer {
+ public:
+  ~VideoRtpDepacketizerGeneric() override = default;
+
+  absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_H_