| /* |
| * 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 |