d06588a758
Bug: webrtc:11152 Change-Id: I322115263f60439bee36277157a0acef9bd28e3e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30260}
71 lines
2.5 KiB
C++
71 lines
2.5 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
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#include <memory>
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#include "absl/memory/memory.h"
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/copy_on_write_buffer.h"
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namespace webrtc {
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namespace {
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// Wrapper over legacy RtpDepacketizer interface.
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// TODO(bugs.webrtc.org/11152): Delete when all RtpDepacketizers updated to
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// the VideoRtpDepacketizer interface.
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template <typename Depacketizer>
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class Legacy : public VideoRtpDepacketizer {
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public:
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absl::optional<ParsedRtpPayload> Parse(
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rtc::CopyOnWriteBuffer rtp_payload) override {
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Depacketizer depacketizer;
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RtpDepacketizer::ParsedPayload parsed_payload;
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if (!depacketizer.Parse(&parsed_payload, rtp_payload.cdata(),
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rtp_payload.size())) {
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return absl::nullopt;
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}
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absl::optional<ParsedRtpPayload> result(absl::in_place);
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result->video_header = parsed_payload.video;
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result->video_payload.SetData(parsed_payload.payload,
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parsed_payload.payload_length);
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return result;
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}
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};
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} // namespace
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std::unique_ptr<VideoRtpDepacketizer> CreateVideoRtpDepacketizer(
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VideoCodecType codec) {
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switch (codec) {
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case kVideoCodecH264:
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return std::make_unique<Legacy<RtpDepacketizerH264>>();
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case kVideoCodecVP8:
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return std::make_unique<VideoRtpDepacketizerVp8>();
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case kVideoCodecVP9:
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return std::make_unique<VideoRtpDepacketizerVp9>();
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case kVideoCodecAV1:
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return std::make_unique<VideoRtpDepacketizerAv1>();
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case kVideoCodecGeneric:
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case kVideoCodecMultiplex:
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return std::make_unique<VideoRtpDepacketizerGeneric>();
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}
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}
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} // namespace webrtc
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