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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/rtp_packet_info.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/rtp_packet_info.cc | 56 |
1 files changed, 56 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/rtp_packet_info.cc b/third_party/libwebrtc/api/rtp_packet_info.cc new file mode 100644 index 0000000000..cba274ec38 --- /dev/null +++ b/third_party/libwebrtc/api/rtp_packet_info.cc @@ -0,0 +1,56 @@ +/* + * Copyright (c) 2019 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 "api/rtp_packet_info.h" + +#include <algorithm> +#include <utility> + +namespace webrtc { + +RtpPacketInfo::RtpPacketInfo() + : ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {} + +RtpPacketInfo::RtpPacketInfo(uint32_t ssrc, + std::vector<uint32_t> csrcs, + uint32_t rtp_timestamp, + Timestamp receive_time) + : ssrc_(ssrc), + csrcs_(std::move(csrcs)), + rtp_timestamp_(rtp_timestamp), + receive_time_(receive_time) {} + +RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header, + Timestamp receive_time) + : ssrc_(rtp_header.ssrc), + rtp_timestamp_(rtp_header.timestamp), + receive_time_(receive_time) { + const auto& extension = rtp_header.extension; + const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize); + + csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]); + + if (extension.hasAudioLevel) { + audio_level_ = extension.audioLevel; + } + + absolute_capture_time_ = extension.absolute_capture_time; +} + +bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) { + return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) && + (lhs.rtp_timestamp() == rhs.rtp_timestamp()) && + (lhs.receive_time() == rhs.receive_time()) && + (lhs.audio_level() == rhs.audio_level()) && + (lhs.absolute_capture_time() == rhs.absolute_capture_time()) && + (lhs.local_capture_clock_offset() == rhs.local_capture_clock_offset()); +} + +} // namespace webrtc |