Add implementations of the VideoRtpDepacketizer interface

while suboptimal, these implementions are complete and allow to
swap code from using RtpDepacketizer interface to VideoRtpDepacketizer

Bug: webrtc:11152
Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30031}

modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc

@@ -0,0 +1,62 @@
+/*
+ *  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.
+ */
+
+#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
+
+#include <memory>
+
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+namespace {
+
+// Wrapper over legacy RtpDepacketizer interface.
+// TODO(bugs.webrtc.org/11152): Delete when all RtpDepacketizers updated to
+// the VideoRtpDepacketizer interface.
+class LegacyRtpDepacketizer : public VideoRtpDepacketizer {
+ public:
+  explicit LegacyRtpDepacketizer(VideoCodecType codec) : codec_(codec) {}
+  ~LegacyRtpDepacketizer() override = default;
+
+  absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) override {
+    auto depacketizer = absl::WrapUnique(RtpDepacketizer::Create(codec_));
+    RTC_CHECK(depacketizer);
+    RtpDepacketizer::ParsedPayload parsed_payload;
+    if (!depacketizer->Parse(&parsed_payload, rtp_payload.cdata(),
+                             rtp_payload.size())) {
+      return absl::nullopt;
+    }
+    absl::optional<ParsedRtpPayload> result(absl::in_place);
+    result->video_header = parsed_payload.video;
+    result->video_payload.SetData(parsed_payload.payload,
+                                  parsed_payload.payload_length);
+    return result;
+  }
+
+ private:
+  const VideoCodecType codec_;
+};
+
+}  // namespace
+
+std::unique_ptr<VideoRtpDepacketizer> CreateVideoRtpDepacketizer(
+    VideoCodecType codec) {
+  // TODO(bugs.webrtc.org/11152): switch on codec and create specialized
+  // VideoRtpDepacketizers when they are migrated to new interface.
+  return std::make_unique<LegacyRtpDepacketizer>(codec);
+}
+
+}  // namespace webrtc