f7f8a1f176
merge construction and call to SetPayloadData Add NumPackets instead of SetPayloadData Remove virtual ToString() as unused move CHECK(rtp_video_header) from RtpPacketizer::Create to RtpSenderVideo::SendVideo Bug: webrtc:9680 Change-Id: I074644e048c797eb836f79979df363fe1ea0075e Reviewed-on: https://webrtc-review.googlesource.com/96543 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24474}
80 lines
2.9 KiB
C++
80 lines
2.9 KiB
C++
/*
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* Copyright (c) 2014 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/rtp_format.h"
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#include <utility>
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#include "absl/memory/memory.h"
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#include "modules/rtp_rtcp/source/rtp_format_h264.h"
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#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
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#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
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#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
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namespace webrtc {
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std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
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VideoCodecType type,
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rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits,
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// Codec-specific details.
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const RTPVideoHeader& rtp_video_header,
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FrameType frame_type,
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const RTPFragmentationHeader* fragmentation) {
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switch (type) {
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case kVideoCodecH264: {
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const auto& h264 =
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absl::get<RTPVideoHeaderH264>(rtp_video_header.video_type_header);
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auto packetizer = absl::make_unique<RtpPacketizerH264>(
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limits.max_payload_len, limits.last_packet_reduction_len,
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h264.packetization_mode);
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packetizer->SetPayloadData(payload.data(), payload.size(), fragmentation);
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return std::move(packetizer);
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}
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case kVideoCodecVP8: {
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const auto& vp8 =
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absl::get<RTPVideoHeaderVP8>(rtp_video_header.video_type_header);
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auto packetizer = absl::make_unique<RtpPacketizerVp8>(
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vp8, limits.max_payload_len, limits.last_packet_reduction_len);
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packetizer->SetPayloadData(payload.data(), payload.size(), nullptr);
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return std::move(packetizer);
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}
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case kVideoCodecVP9: {
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const auto& vp9 =
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absl::get<RTPVideoHeaderVP9>(rtp_video_header.video_type_header);
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auto packetizer = absl::make_unique<RtpPacketizerVp9>(
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vp9, limits.max_payload_len, limits.last_packet_reduction_len);
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packetizer->SetPayloadData(payload.data(), payload.size(), nullptr);
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return std::move(packetizer);
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}
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default: {
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auto packetizer = absl::make_unique<RtpPacketizerGeneric>(
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rtp_video_header, frame_type, limits.max_payload_len,
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limits.last_packet_reduction_len);
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packetizer->SetPayloadData(payload.data(), payload.size(), nullptr);
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return std::move(packetizer);
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}
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}
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}
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RtpDepacketizer* RtpDepacketizer::Create(VideoCodecType type) {
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switch (type) {
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case kVideoCodecH264:
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return new RtpDepacketizerH264();
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case kVideoCodecVP8:
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return new RtpDepacketizerVp8();
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case kVideoCodecVP9:
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return new RtpDepacketizerVp9();
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default:
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return new RtpDepacketizerGeneric();
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}
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}
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} // namespace webrtc
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