diff options
Diffstat (limited to 'thunderbird-l10n/zh-TW/localization/zh-TW/toolkit/about/aboutWebrtc.ftl')
-rw-r--r-- | thunderbird-l10n/zh-TW/localization/zh-TW/toolkit/about/aboutWebrtc.ftl | 322 |
1 files changed, 322 insertions, 0 deletions
diff --git a/thunderbird-l10n/zh-TW/localization/zh-TW/toolkit/about/aboutWebrtc.ftl b/thunderbird-l10n/zh-TW/localization/zh-TW/toolkit/about/aboutWebrtc.ftl new file mode 100644 index 0000000000..76707e2f1d --- /dev/null +++ b/thunderbird-l10n/zh-TW/localization/zh-TW/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 內部資訊 +# "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 = 將 about:webrtc 儲存至 + +## These labels are for a disclosure which contains the information for closed PeerConnection sections + +about-webrtc-closed-peerconnection-disclosure-show-msg = 顯示關閉的 PeerConnections +about-webrtc-closed-peerconnection-disclosure-hide-msg = 隱藏關閉的 PeerConnections + +## AEC is an abbreviation for Acoustic Echo Cancellation. + +about-webrtc-aec-logging-msg-label = AEC 記錄 +about-webrtc-aec-logging-off-state-label = 開始 AEC 記錄 +about-webrtc-aec-logging-on-state-label = 停止 AEC 記錄 +about-webrtc-aec-logging-on-state-msg = AEC 紀錄中(請與來電者交談幾分鐘後再停止捕捉) +about-webrtc-aec-logging-toggled-on-state-msg = AEC 紀錄中(請與來電者交談幾分鐘後再停止捕捉) +about-webrtc-aec-logging-unavailable-sandbox = 需要設定環境變數 MOZ_DISABLE_CONTENT_SANDBOX=1 才可以匯出 AEC 紀錄。請務必先理解可能造成的風險,再設定此環境變數。 +# Variables: +# $path (String) - The path to which the aec log file is saved. +about-webrtc-aec-logging-toggled-off-state-msg = 捕捉到的記錄檔位於: { $path } + +## + +# The autorefresh checkbox causes a stats section to autorefresh its content when checked +about-webrtc-auto-refresh-label = 自動重新整理 +# Determines the default state of the Auto Refresh check boxes +about-webrtc-auto-refresh-default-label = 預設自動重新整理 +# A button which forces a refresh of displayed statistics +about-webrtc-force-refresh-button = 重新整理 +# "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 = 資料頻道開啟數量: +# The number of once open DataChannels that a PeerConnection has closed +about-webrtc-data-channels-closed-label = 資料頻道關閉數量: + +## "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 (接聽) +about-webrtc-remote-sdp-heading = 遠端 SDP +about-webrtc-remote-sdp-heading-offer = 遠端 SDP (提供) +about-webrtc-remote-sdp-heading-answer = 遠端 SDP (接聽) +about-webrtc-sdp-history-heading = SDP 歷史 +about-webrtc-sdp-parsing-errors-heading = SDP 剖析錯誤 + +## + +# "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 統計 + +## "ICE" is an abbreviation for Interactive Connectivity Establishment, which +## is an IETF protocol, and should not normally be translated. + +about-webrtc-ice-state = ICE 狀態 +# "Stats" is an abbreviation for Statistics. +about-webrtc-ice-stats-heading = ICE 統計 +about-webrtc-ice-restart-count-label = ICE 重新啟動: +about-webrtc-ice-rollback-count-label = ICE rollback: +about-webrtc-ice-pair-bytes-sent = 位元組已送出: +about-webrtc-ice-pair-bytes-received = 位元組已接收: +about-webrtc-ice-component-id = 元件 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 = 遠端 + +## + +# 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 = 已指定 +# 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 = 已選取 +about-webrtc-save-page-label = 儲存本頁 +about-webrtc-debug-mode-msg-label = 除錯模式 +about-webrtc-debug-mode-off-state-label = 開始除錯模式 +about-webrtc-debug-mode-on-state-label = 停止除錯模式 +about-webrtc-enable-logging-label = 開啟 WebRTC 保留紀錄 +about-webrtc-stats-heading = 使用階段統計 +about-webrtc-peerconnections-section-heading = RTCPeerConnection 統計資訊 +about-webrtc-peerconnections-section-show-msg = 顯示 RTCPeerConnection 統計資訊 +about-webrtc-peerconnections-section-hide-msg = 隱藏 RTCPeerConnection 統計資訊 +about-webrtc-stats-clear = 清除紀錄 +about-webrtc-log-heading = 連線記錄 +about-webrtc-log-clear = 清除紀錄 +about-webrtc-log-show-msg = 顯示紀錄 + .title = 點擊展開此段落 +about-webrtc-log-hide-msg = 隱藏紀錄 + .title = 點擊摺疊此段落 +about-webrtc-log-section-show-msg = 顯示紀錄 + .title = 點擊展開此段落 +about-webrtc-log-section-hide-msg = 隱藏紀錄 + .title = 點擊摺疊此段落 +about-webrtc-copy-report-button = 複製報告 +about-webrtc-copy-report-history-button = 複製報告紀錄 + +## 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 } (已關閉) { $now } + +## These are used to indicate what direction media is flowing. +## Variables: +## $codecs - a list of media codecs + +about-webrtc-short-send-receive-direction = 傳送/接收:{ $codecs } +about-webrtc-short-send-direction = 傳送:{ $codecs } +about-webrtc-short-receive-direction = 接收:{ $codecs } + +## + +about-webrtc-local-candidate = 本地候選 +about-webrtc-remote-candidate = 遠端候選 +about-webrtc-raw-candidates-heading = 所有原始候選 +about-webrtc-raw-local-candidate = 原始本地候選 +about-webrtc-raw-remote-candidate = 原始遠端候選 +about-webrtc-raw-cand-show-msg = 顯示原始候選 + .title = 點擊展開此段落 +about-webrtc-raw-cand-hide-msg = 隱藏原始候選 + .title = 點擊摺疊此段落 +about-webrtc-raw-cand-section-show-msg = 顯示原始候選 + .title = 點擊展開此段落 +about-webrtc-raw-cand-section-hide-msg = 隱藏原始候選 + .title = 點擊摺疊此段落 +about-webrtc-priority = 重要性 +about-webrtc-fold-show-msg = 顯示詳細資訊 + .title = 點擊展開此段落 +about-webrtc-fold-hide-msg = 隱藏詳細資訊 + .title = 點擊摺疊此段落 +about-webrtc-fold-default-show-msg = 顯示詳細資訊 + .title = 點擊展開此段落 +about-webrtc-fold-default-hide-msg = 隱藏詳細資訊 + .title = 點擊摺疊此段落 +about-webrtc-dropped-frames-label = 捨棄的畫框數: +about-webrtc-discarded-packets-label = 捨棄的封包數: +about-webrtc-decoder-label = 解碼器 +about-webrtc-encoder-label = 編碼器 +about-webrtc-show-tab-label = 顯示分頁 +about-webrtc-current-framerate-label = 畫框率 +about-webrtc-width-px = 寬度(像素) +about-webrtc-height-px = 高度(像素) +about-webrtc-consecutive-frames = 連續畫框 +about-webrtc-time-elapsed = 經過時間(秒) +about-webrtc-estimated-framerate = 估計畫框率 +about-webrtc-rotation-degrees = 旋轉(度) +about-webrtc-first-frame-timestamp = 接收到第一個畫框的時間戳記 +about-webrtc-last-frame-timestamp = 接收到最後一個畫框的時間戳記 + +## 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 = 遠端發送 SSRC + +## These are displayed on the button that shows or hides the +## PeerConnection configuration disclosure + +about-webrtc-pc-configuration-show-msg = 顯示設定 +about-webrtc-pc-configuration-hide-msg = 隱藏設定 + +## + +# 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 = 使用者設定的 WebRTC 偏好設定 +# The options set by the user in about:config that could impact a WebRTC call +about-webrtc-user-modified-configuration-heading = 使用者修改的 WebRTC 設定 + +## These are displayed on the button that shows or hides the +## user modified configuration disclosure + +about-webrtc-user-modified-configuration-show-msg = 顯示使用者修改的 WebRTC 設定 +about-webrtc-user-modified-configuration-hide-msg = 隱藏使用者修改的 WebRTC 設定 + +## + +# Section header for estimated bandwidths of WebRTC media flows +about-webrtc-bandwidth-stats-heading = 估計頻寬 +# The ID of the MediaStreamTrack +about-webrtc-track-identifier = 軌道識別符 +# The estimated bandwidth available for sending WebRTC media in bytes per second +about-webrtc-send-bandwidth-bytes-sec = 傳送頻寬(位元組/秒) +# The estimated bandwidth available for receiving WebRTC media in bytes per second +about-webrtc-receive-bandwidth-bytes-sec = 接收頻寬(位元組/秒) +# Maximum number of bytes per second that will be padding zeros at the ends of packets +about-webrtc-max-padding-bytes-sec = 封包填充資料(位元組/秒) +# The amount of time inserted between packets to keep them spaced out +about-webrtc-pacer-delay-ms = 間隔時間(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 = 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 = 畫框統計資訊 - 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 = 已將頁面儲存至: { $path } +about-webrtc-debug-mode-off-state-msg = 追蹤紀錄位於: { $path } +about-webrtc-debug-mode-on-state-msg = 已進入除錯模式,追蹤紀錄位於: { $path } +about-webrtc-aec-logging-off-state-msg = 捕捉到的記錄檔位於: { $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 = 已將頁面儲存至: { $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 } 畫框 + } +# 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 } 頻道 + } +# 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 } 個封包 + } +# 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] 已捨棄 { $packets } 個封包 + } +# 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] 已送出 { $packets } 個封包 + } +# 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 = 抖動 { $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 = 使用 藍色 強調太晚抵達的候選(接聽後才抵達) + +## "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: +# $timestamp (Number) - The Unix Epoch time at which the SDP was set. +about-webrtc-sdp-set-at-timestamp-local = 已將本地 SDP 時間戳記設為 { NUMBER($timestamp, useGrouping: "false") } +# This is used as a header for remote SDP. +# Variables: +# $timestamp (Number) - The Unix Epoch time at which the SDP was set. +about-webrtc-sdp-set-at-timestamp-remote = 已將遠端 SDP 時間戳記設為 { NUMBER($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. +# $relative-timestamp (Number) - The timestamp relative to the timestamp of the earliest received SDP. +about-webrtc-sdp-set-timestamp = 時間戳記 { NUMBER($timestamp, useGrouping: "false") }(+ { $relative-timestamp } ms) + +## These are displayed on the button that shows or hides the SDP information disclosure + +about-webrtc-show-msg-sdp = 顯示 SDP +about-webrtc-hide-msg-sdp = 隱藏 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 = 顯示媒體內容環境 +about-webrtc-media-context-hide-msg = 隱藏媒體內容環境 +about-webrtc-media-context-heading = 媒體內容環境 + +## + |