summaryrefslogtreecommitdiffstats
path: root/thunderbird-l10n/en-CA/localization/en-CA/toolkit/about/aboutWebrtc.ftl
diff options
context:
space:
mode:
Diffstat (limited to 'thunderbird-l10n/en-CA/localization/en-CA/toolkit/about/aboutWebrtc.ftl')
-rw-r--r--thunderbird-l10n/en-CA/localization/en-CA/toolkit/about/aboutWebrtc.ftl316
1 files changed, 316 insertions, 0 deletions
diff --git a/thunderbird-l10n/en-CA/localization/en-CA/toolkit/about/aboutWebrtc.ftl b/thunderbird-l10n/en-CA/localization/en-CA/toolkit/about/aboutWebrtc.ftl
new file mode 100644
index 0000000000..b9c23aa974
--- /dev/null
+++ b/thunderbird-l10n/en-CA/localization/en-CA/toolkit/about/aboutWebrtc.ftl
@@ -0,0 +1,316 @@
+# 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
+
+## These labels are for a disclosure which contains the information for closed PeerConnection sections
+
+about-webrtc-closed-peerconnection-disclosure-show-msg = Show Closed PeerConnections
+about-webrtc-closed-peerconnection-disclosure-hide-msg = Hide Closed PeerConnections
+
+## 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)
+about-webrtc-aec-logging-toggled-on-state-msg = AEC logging active (speak with the caller for a few minutes and then stop the capture)
+about-webrtc-aec-logging-unavailable-sandbox = The environment variable MOZ_DISABLE_CONTENT_SANDBOX=1 is required to export AEC logs. Only set this variable if you understand the possible risks.
+# Variables:
+# $path (String) - The path to which the aec log file is saved.
+about-webrtc-aec-logging-toggled-off-state-msg = Captured log files can be found in: { $path }
+
+##
+
+# The autorefresh checkbox causes a stats section to autorefresh its content when checked
+about-webrtc-auto-refresh-label = Auto Refresh
+# Determines the default state of the Auto Refresh check boxes
+about-webrtc-auto-refresh-default-label = Auto Refresh By Default
+# A button which forces a refresh of displayed statistics
+about-webrtc-force-refresh-button = 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:
+# The number of DataChannels that a PeerConnection has opened
+about-webrtc-data-channels-opened-label = Data Channels Opened:
+# The number of once open DataChannels that a PeerConnection has closed
+about-webrtc-data-channels-closed-label = Data Channels Closed:
+
+## "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-enable-logging-label = Enable WebRTC Log Preset
+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
+about-webrtc-log-section-show-msg = Show Log
+ .title = Click to expand this section
+about-webrtc-log-section-hide-msg = Hide Log
+ .title = Click to collapse this section
+about-webrtc-copy-report-button = Copy Report
+about-webrtc-copy-report-history-button = Copy Report History
+
+## 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 }
+
+## These are used to indicate what direction media is flowing.
+## Variables:
+## $codecs - a list of media codecs
+
+about-webrtc-short-send-receive-direction = Send / Receive: { $codecs }
+about-webrtc-short-send-direction = Send: { $codecs }
+about-webrtc-short-receive-direction = Receive: { $codecs }
+
+##
+
+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-raw-cand-section-show-msg = Show Raw Candidates
+ .title = Click to expand this section
+about-webrtc-raw-cand-section-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-fold-default-show-msg = Show Details
+ .title = Click to expand this section
+about-webrtc-fold-default-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 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 = Page saved to: { $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
+
+##
+
+
+## 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 = Show Media Context
+about-webrtc-media-context-hide-msg = Hide Media Context
+about-webrtc-media-context-heading = Media Context
+
+##
+