diff options
Diffstat (limited to 'toolkit/locales/en-US/toolkit/about/aboutWebrtc.ftl')
| -rw-r--r-- | toolkit/locales/en-US/toolkit/about/aboutWebrtc.ftl | 282 |
1 files changed, 282 insertions, 0 deletions
diff --git a/toolkit/locales/en-US/toolkit/about/aboutWebrtc.ftl b/toolkit/locales/en-US/toolkit/about/aboutWebrtc.ftl new file mode 100644 index 0000000000..ae323fe9e8 --- /dev/null +++ b/toolkit/locales/en-US/toolkit/about/aboutWebrtc.ftl @@ -0,0 +1,282 @@ +# 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 Internals + +# "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 = save about:webrtc as + +## AEC is an abbreviation for Acoustic Echo Cancellation. + +about-webrtc-aec-logging-msg-label = AEC Logging +about-webrtc-aec-logging-off-state-label = Start AEC Logging +about-webrtc-aec-logging-on-state-label = Stop AEC Logging +about-webrtc-aec-logging-on-state-msg = AEC logging active (speak with the caller for a few minutes and then stop the capture) + +# The autorefresh checkbox causes the page to autorefresh its content when checked +about-webrtc-auto-refresh-label = Auto Refresh + +## + +# "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: + +## "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 = Local SDP +about-webrtc-local-sdp-heading-offer = Local SDP (Offer) +about-webrtc-local-sdp-heading-answer = Local SDP (Answer) +about-webrtc-remote-sdp-heading = Remote SDP +about-webrtc-remote-sdp-heading-offer = Remote SDP (Offer) +about-webrtc-remote-sdp-heading-answer = Remote SDP (Answer) +about-webrtc-sdp-history-heading = SDP History +about-webrtc-sdp-parsing-errors-heading = SDP Parsing Errors + +## + +# "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 Stats + +## "ICE" is an abbreviation for Interactive Connectivity Establishment, which +## is an IETF protocol, and should not normally be translated. + +about-webrtc-ice-state = ICE State +# "Stats" is an abbreviation for Statistics. +about-webrtc-ice-stats-heading = ICE Stats +about-webrtc-ice-restart-count-label = ICE restarts: +about-webrtc-ice-rollback-count-label = ICE rollbacks: +about-webrtc-ice-pair-bytes-sent = Bytes sent: +about-webrtc-ice-pair-bytes-received = Bytes received: +about-webrtc-ice-component-id = Component 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 = Local +about-webrtc-type-remote = Remote + +## + +# 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 = Nominated + +# 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 = Selected + +about-webrtc-save-page-label = Save Page +about-webrtc-debug-mode-msg-label = Debug Mode +about-webrtc-debug-mode-off-state-label = Start Debug Mode +about-webrtc-debug-mode-on-state-label = Stop Debug Mode +about-webrtc-stats-heading = Session Statistics +about-webrtc-stats-clear = Clear History +about-webrtc-log-heading = Connection Log +about-webrtc-log-clear = Clear Log +about-webrtc-log-show-msg = show log + .title = click to expand this section +about-webrtc-log-hide-msg = hide log + .title = click to collapse this section + +## 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 } (closed) { $now } + +## + +about-webrtc-local-candidate = Local Candidate +about-webrtc-remote-candidate = Remote Candidate +about-webrtc-raw-candidates-heading = All Raw Candidates +about-webrtc-raw-local-candidate = Raw Local Candidate +about-webrtc-raw-remote-candidate = Raw Remote Candidate +about-webrtc-raw-cand-show-msg = show raw candidates + .title = click to expand this section +about-webrtc-raw-cand-hide-msg = hide raw candidates + .title = click to collapse this section +about-webrtc-priority = Priority +about-webrtc-fold-show-msg = show details + .title = click to expand this section +about-webrtc-fold-hide-msg = hide details + .title = click to collapse this section +about-webrtc-dropped-frames-label = Dropped frames: +about-webrtc-discarded-packets-label = Discarded packets: +about-webrtc-decoder-label = Decoder +about-webrtc-encoder-label = Encoder +about-webrtc-show-tab-label = Show tab +about-webrtc-current-framerate-label = Framerate +about-webrtc-width-px = Width (px) +about-webrtc-height-px = Height (px) +about-webrtc-consecutive-frames = Consecutive Frames +about-webrtc-time-elapsed = Time Elapsed (s) +about-webrtc-estimated-framerate = Estimated Framerate +about-webrtc-rotation-degrees = Rotation (degrees) +about-webrtc-first-frame-timestamp = First Frame Reception Timestamp +about-webrtc-last-frame-timestamp = Last Frame Reception Timestamp + +## 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 = Local Receiving SSRC +# This is an SSRC on the remote side of the connection that is sending RTP +about-webrtc-remote-send-ssrc = Remote Sending SSRC + +## These are displayed on the button that shows or hides the +## PeerConnection configuration disclosure + +about-webrtc-pc-configuration-show-msg = Show Configuration +about-webrtc-pc-configuration-hide-msg = Hide Configuration + +## + +# An option whose value will not be displayed but instead noted as having been +# provided +about-webrtc-configuration-element-provided = Provided + +# An option whose value will not be displayed but instead noted as having not +# been provided +about-webrtc-configuration-element-not-provided = Not Provided + +# The options set by the user in about:config that could impact a WebRTC call +about-webrtc-custom-webrtc-configuration-heading = User Set WebRTC Preferences + +# Section header for estimated bandwidths of WebRTC media flows +about-webrtc-bandwidth-stats-heading = Estimated Bandwidth + +# The ID of the MediaStreamTrack +about-webrtc-track-identifier = Track Identifier + +# The estimated bandwidth available for sending WebRTC media in bytes per second +about-webrtc-send-bandwidth-bytes-sec = Send Bandwidth (bytes/sec) + +# The estimated bandwidth available for receiving WebRTC media in bytes per second +about-webrtc-receive-bandwidth-bytes-sec = Receive Bandwidth (bytes/sec) + +# Maximum number of bytes per second that will be padding zeros at the ends of packets +about-webrtc-max-padding-bytes-sec = Maximum Padding (bytes/sec) + +# The amount of time inserted between packets to keep them spaced out +about-webrtc-pacer-delay-ms = Pacer Delay 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 = Video Frame Statistics - 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 = page saved to: { $path } +about-webrtc-debug-mode-off-state-msg = trace log can be found at: { $path } +about-webrtc-debug-mode-on-state-msg = debug mode active, trace log at: { $path } +about-webrtc-aec-logging-off-state-msg = captured log files can be found in: { $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 -> + [one] { $frames } frame + *[other] { $frames } frames + } + +# 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 -> + [one] { $channels } channel + *[other] { $channels } channels + } + +# This is the total number of packets received on the PeerConnection. +# Variables: +# $packets (Number) - The number of packets received. +about-webrtc-received-label = + { $packets -> + [one] Received { $packets } packet + *[other] Received { $packets } packets + } + +# This is the total number of packets lost by the PeerConnection. +# Variables: +# $packets (Number) - The number of packets lost. +about-webrtc-lost-label = + { $packets -> + [one] Lost { $packets } packet + *[other] Lost { $packets } packets + } + +# This is the total number of packets sent by the PeerConnection. +# Variables: +# $packets (Number) - The number of packets sent. +about-webrtc-sent-label = + { $packets -> + [one] Sent { $packets } packet + *[other] Sent { $packets } packets + } + +# 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 = Jitter { $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 candidates (arriving after answer) are highlighted in blue + +## "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: +# $timestamp (Number) - The Unix Epoch time at which the SDP was set. +about-webrtc-sdp-set-at-timestamp-local = Set Local SDP at timestamp { NUMBER($timestamp, useGrouping: "false") } + +# This is used as a header for remote SDP. +# Variables: +# $timestamp (Number) - The Unix Epoch time at which the SDP was set. +about-webrtc-sdp-set-at-timestamp-remote = Set Remote SDP at timestamp { NUMBER($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. +# $relative-timestamp (Number) - The timestamp relative to the timestamp of the earliest received SDP. +about-webrtc-sdp-set-timestamp = Timestamp { NUMBER($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 = Show SDP +about-webrtc-hide-msg-sdp = Hide SDP + +## |