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https://github.com/mollyim/webrtc.git
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5075cb4a60
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead. The old fields are preserved for compatibility with downstream projects, but will be removed in the future. Bug: webrtc:15788 Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41947}
79 lines
3.2 KiB
C++
79 lines
3.2 KiB
C++
/*
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* Copyright (c) 2017 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_packet_received.h"
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#include <stddef.h>
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#include <cstdint>
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#include <vector>
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "rtc_base/numerics/safe_conversions.h"
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namespace webrtc {
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RtpPacketReceived::RtpPacketReceived() = default;
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RtpPacketReceived::RtpPacketReceived(
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const ExtensionManager* extensions,
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webrtc::Timestamp arrival_time /*= webrtc::Timestamp::MinusInfinity()*/)
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: RtpPacket(extensions), arrival_time_(arrival_time) {}
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RtpPacketReceived::RtpPacketReceived(const RtpPacketReceived& packet) = default;
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RtpPacketReceived::RtpPacketReceived(RtpPacketReceived&& packet) = default;
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RtpPacketReceived& RtpPacketReceived::operator=(
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const RtpPacketReceived& packet) = default;
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RtpPacketReceived& RtpPacketReceived::operator=(RtpPacketReceived&& packet) =
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default;
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RtpPacketReceived::~RtpPacketReceived() {}
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void RtpPacketReceived::GetHeader(RTPHeader* header) const {
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header->markerBit = Marker();
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header->payloadType = PayloadType();
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header->sequenceNumber = SequenceNumber();
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header->timestamp = Timestamp();
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header->ssrc = Ssrc();
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std::vector<uint32_t> csrcs = Csrcs();
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header->numCSRCs = rtc::dchecked_cast<uint8_t>(csrcs.size());
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for (size_t i = 0; i < csrcs.size(); ++i) {
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header->arrOfCSRCs[i] = csrcs[i];
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}
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header->paddingLength = padding_size();
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header->headerLength = headers_size();
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header->extension.hasTransmissionTimeOffset =
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GetExtension<TransmissionOffset>(
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&header->extension.transmissionTimeOffset);
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header->extension.hasAbsoluteSendTime =
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GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime);
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header->extension.absolute_capture_time =
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GetExtension<AbsoluteCaptureTimeExtension>();
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header->extension.hasTransportSequenceNumber =
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GetExtension<TransportSequenceNumberV2>(
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&header->extension.transportSequenceNumber,
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&header->extension.feedback_request) ||
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GetExtension<TransportSequenceNumber>(
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&header->extension.transportSequenceNumber);
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header->extension.set_audio_level(GetExtension<AudioLevelExtension>());
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header->extension.hasVideoRotation =
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GetExtension<VideoOrientation>(&header->extension.videoRotation);
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header->extension.hasVideoContentType =
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GetExtension<VideoContentTypeExtension>(
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&header->extension.videoContentType);
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header->extension.has_video_timing =
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GetExtension<VideoTimingExtension>(&header->extension.video_timing);
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GetExtension<RtpStreamId>(&header->extension.stream_id);
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GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
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GetExtension<RtpMid>(&header->extension.mid);
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GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
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header->extension.color_space = GetExtension<ColorSpaceExtension>();
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}
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} // namespace webrtc
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