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diff --git a/l10n-zh-CN/toolkit/toolkit/about/aboutWebrtc.ftl b/l10n-zh-CN/toolkit/toolkit/about/aboutWebrtc.ftl new file mode 100644 index 0000000000..0a3ad6e3ea --- /dev/null +++ b/l10n-zh-CN/toolkit/toolkit/about/aboutWebrtc.ftl @@ -0,0 +1,322 @@ +# This Source Code Form is subject to the terms of the Mozilla Public +# License, v. 2.0. If a copy of the MPL was not distributed with this +# file, You can obtain one at http://mozilla.org/MPL/2.0/. + + +### Localization for about:webrtc, a troubleshooting and diagnostic page +### for WebRTC calls. See https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API. + +# The text "WebRTC" is a proper noun and should not be translated. +about-webrtc-document-title = WebRTC 内部 +# "about:webrtc" is a internal browser URL and should not be +# translated. This string is used as a title for a file save dialog box. +about-webrtc-save-page-dialog-title = 另存 about:webrtc 为 + +## These labels are for a disclosure which contains the information for closed PeerConnection sections + +about-webrtc-closed-peerconnection-disclosure-show-msg = 显示已关闭的 PeerConnection +about-webrtc-closed-peerconnection-disclosure-hide-msg = 隐藏已关闭的 PeerConnection + +## AEC is an abbreviation for Acoustic Echo Cancellation. + +about-webrtc-aec-logging-msg-label = AEC 正在记录 +about-webrtc-aec-logging-off-state-label = 开始 AEC 日志记录 +about-webrtc-aec-logging-on-state-label = 停止 AEC 日志记录 +about-webrtc-aec-logging-on-state-msg = AEC 日志正在记录（与呼叫者说几分钟话，然后停止捕捉） +about-webrtc-aec-logging-toggled-on-state-msg = AEC 日志正在记录（与呼叫者说几分钟话，然后停止捕捉） +about-webrtc-aec-logging-unavailable-sandbox = 导出AEC日志需要环境变量MOZ_DISABLE_CONTENT_SANDBOX=1。仅当您了解潜在的风险时才应设置此变量。 +# Variables: +# $path (String) - The path to which the aec log file is saved. +about-webrtc-aec-logging-toggled-off-state-msg = 捕捉到的日志文件在这里: { $path } + +## + +# The autorefresh checkbox causes a stats section to autorefresh its content when checked +about-webrtc-auto-refresh-label = 自动刷新 +# Determines the default state of the Auto Refresh check boxes +about-webrtc-auto-refresh-default-label = 默认自动刷新 +# A button which forces a refresh of displayed statistics +about-webrtc-force-refresh-button = 刷新 +# "PeerConnection" is a proper noun associated with the WebRTC module. "ID" is +# an abbreviation for Identifier. This string should not normally be translated +# and is used as a data label. +about-webrtc-peerconnection-id-label = PeerConnection ID: +# The number of DataChannels that a PeerConnection has opened +about-webrtc-data-channels-opened-label = 数据通道开启数： +# The number of once open DataChannels that a PeerConnection has closed +about-webrtc-data-channels-closed-label = 数据通道关闭数： + +## "SDP" is an abbreviation for Session Description Protocol, an IETF standard. +## See http://wikipedia.org/wiki/Session_Description_Protocol + +about-webrtc-sdp-heading = SDP +about-webrtc-local-sdp-heading = 本地 SDP +about-webrtc-local-sdp-heading-offer = 本地 SDP (提供) +about-webrtc-local-sdp-heading-answer = 本地 SDP (回答) +about-webrtc-remote-sdp-heading = 远程 SDP +about-webrtc-remote-sdp-heading-offer = 远程 SDP (提供) +about-webrtc-remote-sdp-heading-answer = 远程 SDP (回答) +about-webrtc-sdp-history-heading = SDP 历史 +about-webrtc-sdp-parsing-errors-heading = SDP 解析错误 + +## + +# "RTP" is an abbreviation for the Real-time Transport Protocol, an IETF +# specification, and should not normally be translated. "Stats" is an +# abbreviation for Statistics. +about-webrtc-rtp-stats-heading = RTP 状态 + +## "ICE" is an abbreviation for Interactive Connectivity Establishment, which +## is an IETF protocol, and should not normally be translated. + +about-webrtc-ice-state = ICE 统计 +# "Stats" is an abbreviation for Statistics. +about-webrtc-ice-stats-heading = ICE 状态 +about-webrtc-ice-restart-count-label = ICE 重启: +about-webrtc-ice-rollback-count-label = ICE 回滚: +about-webrtc-ice-pair-bytes-sent = 已发送字节: +about-webrtc-ice-pair-bytes-received = 已接收字节: +about-webrtc-ice-component-id = 组件 ID + +## These adjectives are used to label a line of statistics collected for a peer +## connection. The data represents either the local or remote end of the +## connection. + +about-webrtc-type-local = 本地 +about-webrtc-type-remote = 远程 + +## + +# This adjective is used to label a table column. Cells in this column contain +# the localized javascript string representation of "true" or are left blank. +about-webrtc-nominated = 已提名 +# This adjective is used to label a table column. Cells in this column contain +# the localized javascript string representation of "true" or are left blank. +# This represents an attribute of an ICE candidate. +about-webrtc-selected = 已选定 +about-webrtc-save-page-label = 保存页面 +about-webrtc-debug-mode-msg-label = 调试模式 +about-webrtc-debug-mode-off-state-label = 开始调试模式 +about-webrtc-debug-mode-on-state-label = 停止调试模式 +about-webrtc-enable-logging-label = 启用 WebRTC 日志预设 +about-webrtc-stats-heading = 会话统计 +about-webrtc-peerconnections-section-heading = RTCPeerConnection 统计信息 +about-webrtc-peerconnections-section-show-msg = 显示 RTCPeerConnection 统计信息 +about-webrtc-peerconnections-section-hide-msg = 隐藏 RTCPeerConnection 统计信息 +about-webrtc-stats-clear = 清除历史记录 +about-webrtc-log-heading = 连接日志 +about-webrtc-log-clear = 清除日志 +about-webrtc-log-show-msg = 显示日志 + .title = 点击展开此段 +about-webrtc-log-hide-msg = 隐藏日志 + .title = 点击折叠此段 +about-webrtc-log-section-show-msg = 显示日志 + .title = 点击展开此段 +about-webrtc-log-section-hide-msg = 隐藏日志 + .title = 点击折叠此段 +about-webrtc-copy-report-button = 复制报告 +about-webrtc-copy-report-history-button = 复制报告历史 + +## These are used to display a header for a PeerConnection. +## Variables: +## $browser-id (Number) - A numeric id identifying the browser tab for the PeerConnection. +## $id (String) - A globally unique identifier for the PeerConnection. +## $url (String) - The url of the site which opened the PeerConnection. +## $now (Date) - The JavaScript timestamp at the time the report was generated. + +about-webrtc-connection-open = [ { $browser-id } | { $id } ] { $url } { $now } +about-webrtc-connection-closed = [ { $browser-id } | { $id } ] { $url } (已关闭) { $now } + +## These are used to indicate what direction media is flowing. +## Variables: +## $codecs - a list of media codecs + +about-webrtc-short-send-receive-direction = 发送/接收：{ $codecs } +about-webrtc-short-send-direction = 发送：{ $codecs } +about-webrtc-short-receive-direction = 接收：{ $codecs } + +## + +about-webrtc-local-candidate = 本地候选 +about-webrtc-remote-candidate = 远程候选 +about-webrtc-raw-candidates-heading = 所有原始候选者 +about-webrtc-raw-local-candidate = 原始本地候选者 +about-webrtc-raw-remote-candidate = 原始远程候选者 +about-webrtc-raw-cand-show-msg = 显示原始候选者 + .title = 点击展开此段 +about-webrtc-raw-cand-hide-msg = 隐藏原始候选者 + .title = 点击折叠此段 +about-webrtc-raw-cand-section-show-msg = 显示原始候选者 + .title = 点击展开此段 +about-webrtc-raw-cand-section-hide-msg = 隐藏原始候选者 + .title = 点击折叠此段 +about-webrtc-priority = 优先级 +about-webrtc-fold-show-msg = 显示详细信息 + .title = 点击展开此段 +about-webrtc-fold-hide-msg = 隐藏详细信息 + .title = 点击折叠此段 +about-webrtc-fold-default-show-msg = 显示详细信息 + .title = 点击展开此段 +about-webrtc-fold-default-hide-msg = 隐藏详细信息 + .title = 点击折叠此段 +about-webrtc-dropped-frames-label = 丢帧数： +about-webrtc-discarded-packets-label = 丢包数： +about-webrtc-decoder-label = 解码器 +about-webrtc-encoder-label = 编码器 +about-webrtc-show-tab-label = 显示标签页 +about-webrtc-current-framerate-label = 帧率 +about-webrtc-width-px = 宽度（像素） +about-webrtc-height-px = 高度（像素） +about-webrtc-consecutive-frames = 连续帧 +about-webrtc-time-elapsed = 已用时间（秒） +about-webrtc-estimated-framerate = 估计帧率 +about-webrtc-rotation-degrees = 旋转（度） +about-webrtc-first-frame-timestamp = 第一帧接收时间戳 +about-webrtc-last-frame-timestamp = 最后一帧接收时间戳 + +## SSRCs are identifiers that represent endpoints in an RTP stream + +# This is an SSRC on the local side of the connection that is receiving RTP +about-webrtc-local-receive-ssrc = 本地接收 SSRC +# This is an SSRC on the remote side of the connection that is sending RTP +about-webrtc-remote-send-ssrc = 远程发送 SSRC + +## These are displayed on the button that shows or hides the +## PeerConnection configuration disclosure + +about-webrtc-pc-configuration-show-msg = 显示配置 +about-webrtc-pc-configuration-hide-msg = 隐藏配置 + +## + +# An option whose value will not be displayed but instead noted as having been +# provided +about-webrtc-configuration-element-provided = 提供 +# An option whose value will not be displayed but instead noted as having not +# been provided +about-webrtc-configuration-element-not-provided = 不提供 +# The options set by the user in about:config that could impact a WebRTC call +about-webrtc-custom-webrtc-configuration-heading = WebRTC 用户设置项 +# The options set by the user in about:config that could impact a WebRTC call +about-webrtc-user-modified-configuration-heading = 用户修改过的 WebRTC 配置 + +## These are displayed on the button that shows or hides the +## user modified configuration disclosure + +about-webrtc-user-modified-configuration-show-msg = 显示用户修改过的配置 +about-webrtc-user-modified-configuration-hide-msg = 隐藏用户修改过的配置 + +## + +# Section header for estimated bandwidths of WebRTC media flows +about-webrtc-bandwidth-stats-heading = 估计带宽 +# The ID of the MediaStreamTrack +about-webrtc-track-identifier = 轨道标识符 +# The estimated bandwidth available for sending WebRTC media in bytes per second +about-webrtc-send-bandwidth-bytes-sec = 发送带宽（字节 / 秒） +# The estimated bandwidth available for receiving WebRTC media in bytes per second +about-webrtc-receive-bandwidth-bytes-sec = 接收带宽（字节 / 秒） +# Maximum number of bytes per second that will be padding zeros at the ends of packets +about-webrtc-max-padding-bytes-sec = 最大填补数据（字节 / 秒） +# The amount of time inserted between packets to keep them spaced out +about-webrtc-pacer-delay-ms = 间隔时间（ms） +# The amount of time it takes for a packet to travel from the local machine to the remote machine, +# and then have a packet return +about-webrtc-round-trip-time-ms = 往返时延（RTT | ms） +# This is a section heading for video frame statistics for a MediaStreamTrack. +# see https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack. +# Variables: +# $track-identifier (String) - The unique identifier for the MediaStreamTrack. +about-webrtc-frame-stats-heading = 视频帧统计信息 - MediaStreamTrack ID：{ $track-identifier } + +## These are paths used for saving the about:webrtc page or log files so +## they can be attached to bug reports. +## Variables: +## $path (String) - The path to which the file is saved. + +about-webrtc-save-page-msg = 页面已保存到: { $path } +about-webrtc-debug-mode-off-state-msg = 跟踪日志可以在这里找到: { $path } +about-webrtc-debug-mode-on-state-msg = 调试模式已激活，跟踪日志在: { $path } +about-webrtc-aec-logging-off-state-msg = 捕捉到的日志文件在这里: { $path } +# This path is used for saving the about:webrtc page so it can be attached to +# bug reports. +# Variables: +# $path (String) - The path to which the file is saved. +about-webrtc-save-page-complete-msg = 页面已保存到: { $path } +# This is the total number of frames encoded or decoded over an RTP stream. +# Variables: +# $frames (Number) - The number of frames encoded or decoded. +about-webrtc-frames = + { $frames -> + *[other] { $frames } 帧 + } +# This is the number of audio channels encoded or decoded over an RTP stream. +# Variables: +# $channels (Number) - The number of channels encoded or decoded. +about-webrtc-channels = + { $channels -> + *[other] { $channels } 频道 + } +# This is the total number of packets received on the PeerConnection. +# Variables: +# $packets (Number) - The number of packets received. +about-webrtc-received-label = + { $packets -> + *[other] 已收到 { $packets } 个包 + } +# This is the total number of packets lost by the PeerConnection. +# Variables: +# $packets (Number) - The number of packets lost. +about-webrtc-lost-label = + { $packets -> + *[other] 已丢弃 { $packets } 个包 + } +# This is the total number of packets sent by the PeerConnection. +# Variables: +# $packets (Number) - The number of packets sent. +about-webrtc-sent-label = + { $packets -> + *[other] 已发送 { $packets } 个包 + } +# Jitter is the variance in the arrival time of packets. +# See: https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-jitter +# Variables: +# $jitter (Number) - The jitter. +about-webrtc-jitter-label = 抖动 { $jitter } +# ICE candidates arriving after the remote answer arrives are considered trickled +# (an attribute of an ICE candidate). These are highlighted in the ICE stats +# table with light blue background. +about-webrtc-trickle-caption-msg = Trickled 候选者（回答后到达）已用 蓝色 高亮 + +## "SDP" is an abbreviation for Session Description Protocol, an IETF standard. +## See http://wikipedia.org/wiki/Session_Description_Protocol + +# This is used as a header for local SDP. +# Variables: +# $timestamp (Number) - The Unix Epoch time at which the SDP was set. +about-webrtc-sdp-set-at-timestamp-local = 已将 本地 SDP 时间戳设为 { NUMBER($timestamp, useGrouping: "false") } +# This is used as a header for remote SDP. +# Variables: +# $timestamp (Number) - The Unix Epoch time at which the SDP was set. +about-webrtc-sdp-set-at-timestamp-remote = 已将 远程 SDP 时间戳设为 { NUMBER($timestamp, useGrouping: "false") } +# This is used as a header for an SDP section contained in two columns allowing for side-by-side comparisons. +# Variables: +# $timestamp (Number) - The Unix Epoch time at which the SDP was set. +# $relative-timestamp (Number) - The timestamp relative to the timestamp of the earliest received SDP. +about-webrtc-sdp-set-timestamp = 时间戳 { NUMBER($timestamp, useGrouping: "false") }（+ { $relative-timestamp } 毫秒） + +## These are displayed on the button that shows or hides the SDP information disclosure + +about-webrtc-show-msg-sdp = 显示SDP +about-webrtc-hide-msg-sdp = 隐藏SDP + +## These are displayed on the button that shows or hides the Media Context information disclosure. +## The Media Context is the set of preferences and detected capabilities that informs +## the negotiated CODEC settings. + +about-webrtc-media-context-show-msg = 显示媒体内容 +about-webrtc-media-context-hide-msg = 隐藏媒体内容 +about-webrtc-media-context-heading = 媒体内容 + +## + |