| /* |
| * Copyright (c) 2014 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/rtp_format.h" |
| |
| #include <utility> |
| |
| #include "absl/memory/memory.h" |
| #include "modules/rtp_rtcp/source/rtp_format_h264.h" |
| #include "modules/rtp_rtcp/source/rtp_format_video_generic.h" |
| #include "modules/rtp_rtcp/source/rtp_format_vp8.h" |
| #include "modules/rtp_rtcp/source/rtp_format_vp9.h" |
| |
| namespace webrtc { |
| |
| std::unique_ptr<RtpPacketizer> RtpPacketizer::Create( |
| VideoCodecType type, |
| rtc::ArrayView<const uint8_t> payload, |
| PayloadSizeLimits limits, |
| // Codec-specific details. |
| const RTPVideoHeader& rtp_video_header, |
| FrameType frame_type, |
| const RTPFragmentationHeader* fragmentation) { |
| switch (type) { |
| case kVideoCodecH264: { |
| const auto& h264 = |
| absl::get<RTPVideoHeaderH264>(rtp_video_header.video_type_header); |
| auto packetizer = absl::make_unique<RtpPacketizerH264>( |
| limits.max_payload_len, limits.last_packet_reduction_len, |
| h264.packetization_mode); |
| packetizer->SetPayloadData(payload.data(), payload.size(), fragmentation); |
| return std::move(packetizer); |
| } |
| case kVideoCodecVP8: { |
| const auto& vp8 = |
| absl::get<RTPVideoHeaderVP8>(rtp_video_header.video_type_header); |
| return absl::make_unique<RtpPacketizerVp8>(payload, limits, vp8); |
| } |
| case kVideoCodecVP9: { |
| const auto& vp9 = |
| absl::get<RTPVideoHeaderVP9>(rtp_video_header.video_type_header); |
| auto packetizer = absl::make_unique<RtpPacketizerVp9>( |
| vp9, limits.max_payload_len, limits.last_packet_reduction_len); |
| packetizer->SetPayloadData(payload.data(), payload.size(), nullptr); |
| return std::move(packetizer); |
| } |
| default: { |
| auto packetizer = absl::make_unique<RtpPacketizerGeneric>( |
| rtp_video_header, frame_type, limits.max_payload_len, |
| limits.last_packet_reduction_len); |
| packetizer->SetPayloadData(payload.data(), payload.size(), nullptr); |
| return std::move(packetizer); |
| } |
| } |
| } |
| |
| RtpDepacketizer* RtpDepacketizer::Create(VideoCodecType type) { |
| switch (type) { |
| case kVideoCodecH264: |
| return new RtpDepacketizerH264(); |
| case kVideoCodecVP8: |
| return new RtpDepacketizerVp8(); |
| case kVideoCodecVP9: |
| return new RtpDepacketizerVp9(); |
| default: |
| return new RtpDepacketizerGeneric(); |
| } |
| } |
| } // namespace webrtc |