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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}
56 lines
2 KiB
C++
56 lines
2 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 "api/rtp_packet_info.h"
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#include <algorithm>
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#include <utility>
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namespace webrtc {
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RtpPacketInfo::RtpPacketInfo()
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: ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {}
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RtpPacketInfo::RtpPacketInfo(uint32_t ssrc,
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std::vector<uint32_t> csrcs,
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uint32_t rtp_timestamp,
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Timestamp receive_time)
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: ssrc_(ssrc),
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csrcs_(std::move(csrcs)),
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rtp_timestamp_(rtp_timestamp),
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receive_time_(receive_time) {}
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RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
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Timestamp receive_time)
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: ssrc_(rtp_header.ssrc),
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rtp_timestamp_(rtp_header.timestamp),
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receive_time_(receive_time) {
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const auto& extension = rtp_header.extension;
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const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize);
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csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]);
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if (extension.audio_level()) {
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audio_level_ = extension.audio_level()->level();
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}
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absolute_capture_time_ = extension.absolute_capture_time;
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}
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bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
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return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
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(lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
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(lhs.receive_time() == rhs.receive_time()) &&
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(lhs.audio_level() == rhs.audio_level()) &&
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(lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
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(lhs.local_capture_clock_offset() == rhs.local_capture_clock_offset());
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}
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} // namespace webrtc
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