2fa4774067
This reverts commit e6ee8fab7eac915b2b6abc9b71b6d33ad086f3d1. Reason for revert: Breaks downstream test Original change's description: > Deprecate microsecond timestamps in RTC event log. > > (Microsecond timestamps are only used in the legacy wire-format, > and the clocks only have microsecond resolution on some platforms.) > > Also convert structs on the parsing side to use a Timestamp instead > of a uint64_t to represent the log time. > > Bug: webrtc:11933 > Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467 > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34097} TBR=terelius@webrtc.org,srte@webrtc.org,crodbro@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com Change-Id: I478c9a4a1664b984891c4fcfc78f0ce9a51fe4c0 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11933 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219636 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34100}
58 lines
2.4 KiB
C++
58 lines
2.4 KiB
C++
/*
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* Copyright 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 "logging/rtc_event_log/logged_events.h"
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namespace webrtc {
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LoggedPacketInfo::LoggedPacketInfo(const LoggedRtpPacket& rtp,
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LoggedMediaType media_type,
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bool rtx,
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Timestamp capture_time)
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: ssrc(rtp.header.ssrc),
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stream_seq_no(rtp.header.sequenceNumber),
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size(static_cast<uint16_t>(rtp.total_length)),
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payload_size(static_cast<uint16_t>(rtp.total_length -
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rtp.header.paddingLength -
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rtp.header.headerLength)),
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padding_size(static_cast<uint16_t>(rtp.header.paddingLength)),
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payload_type(rtp.header.payloadType),
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media_type(media_type),
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rtx(rtx),
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marker_bit(rtp.header.markerBit),
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has_transport_seq_no(rtp.header.extension.hasTransportSequenceNumber),
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transport_seq_no(static_cast<uint16_t>(
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has_transport_seq_no ? rtp.header.extension.transportSequenceNumber
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: 0)),
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capture_time(capture_time),
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log_packet_time(Timestamp::Micros(rtp.log_time_us())),
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reported_send_time(rtp.header.extension.hasAbsoluteSendTime
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? rtp.header.extension.GetAbsoluteSendTimestamp()
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: Timestamp::MinusInfinity()) {}
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LoggedPacketInfo::LoggedPacketInfo(const LoggedPacketInfo&) = default;
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LoggedPacketInfo::~LoggedPacketInfo() {}
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LoggedRtcpPacket::LoggedRtcpPacket(int64_t timestamp_us,
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const std::vector<uint8_t>& packet)
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: timestamp_us(timestamp_us), raw_data(packet) {}
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LoggedRtcpPacket::LoggedRtcpPacket(int64_t timestamp_us,
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const std::string& packet)
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: timestamp_us(timestamp_us), raw_data(packet.size()) {
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memcpy(raw_data.data(), packet.data(), packet.size());
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}
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LoggedRtcpPacket::LoggedRtcpPacket(const LoggedRtcpPacket& rhs) = default;
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LoggedRtcpPacket::~LoggedRtcpPacket() = default;
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} // namespace webrtc
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