Introduce VideoRtpDepacketizer interface to replace RtpDepacketizer
Bug: webrtc:11152
Change-Id: I20fd81233080d45d8978e5e57d0be6b592f44f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161322
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30018}
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 2bb12ce..d0f4ce8 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -210,6 +210,7 @@
"source/ulpfec_header_reader_writer.h",
"source/ulpfec_receiver_impl.cc",
"source/ulpfec_receiver_impl.h",
+ "source/video_rtp_depacketizer.h",
]
if (rtc_enable_bwe_test_logging) {
diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h
index 1c49811..2093bfa 100644
--- a/modules/rtp_rtcp/source/rtp_format.h
+++ b/modules/rtp_rtcp/source/rtp_format.h
@@ -60,7 +60,7 @@
const PayloadSizeLimits& limits);
};
-// TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
+// TODO(bugs.webrtc.org/11152): Update the depacketizer to return a copy
// of the parsed payload, rather than just a pointer into the incoming buffer.
// This way we can move some parsing out from the jitter buffer into here, and
// the jitter buffer can just store that pointer rather than doing a copy there.
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer.h b/modules/rtp_rtcp/source/video_rtp_depacketizer.h
new file mode 100644
index 0000000..0420e4e
--- /dev/null
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer.h
@@ -0,0 +1,34 @@
+/*
+ * 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_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class VideoRtpDepacketizer {
+ public:
+ struct ParsedRtpPayload {
+ RTPVideoHeader video_header;
+ rtc::CopyOnWriteBuffer video_payload;
+ };
+
+ virtual ~VideoRtpDepacketizer() = default;
+ virtual absl::optional<ParsedRtpPayload> Parse(
+ rtc::CopyOnWriteBuffer rtp_payload) = 0;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
