path: root/l10n-ja/toolkit/toolkit/about/aboutWebrtc.ftl

# 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 内部情報
# "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 = about:webrtc を名前を付けて保存

## These labels are for a disclosure which contains the information for closed PeerConnection sections

about-webrtc-closed-peerconnection-disclosure-show-msg = 閉じた PeerConnection を表示
about-webrtc-closed-peerconnection-disclosure-hide-msg = 閉じた PeerConnection を隠す

## AEC is an abbreviation for Acoustic Echo Cancellation.

about-webrtc-aec-logging-msg-label = AEC ログ記録
about-webrtc-aec-logging-off-state-label = AEC ログ記録を開始
about-webrtc-aec-logging-on-state-label = AEC ログ記録を停止
about-webrtc-aec-logging-on-state-msg = AEC ログ記録が有効です (数分間、通話相手と会話してから停止してください)
about-webrtc-aec-logging-toggled-on-state-msg = AEC ログ記録が有効です (数分間、通話相手と会話してから停止してください)
about-webrtc-aec-logging-unavailable-sandbox = AEC ログをエクスポートするには環境変数 MOZ_DISABLE_CONTENT_SANDBOX=1 が必要です。この環境変数は潜在的な危険性を理解している場合のみ設定してください。
# Variables:
#  $path (String) - The path to which the aec log file is saved.
about-webrtc-aec-logging-toggled-off-state-msg = 記録したログファイルの保存場所: { $path }

##

# The autorefresh checkbox causes a stats section to autorefresh its content when checked
about-webrtc-auto-refresh-label = 自動更新する
# A button which forces a refresh of displayed statistics
about-webrtc-force-refresh-button = 更新
# "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 = 開いているデータチャンネル:
# The number of once open DataChannels that a PeerConnection has closed
about-webrtc-data-channels-closed-label = 閉じているデータチャンネル:

## "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 = ローカル SDP
about-webrtc-local-sdp-heading-offer = ローカル SDP (オファー)
about-webrtc-local-sdp-heading-answer = ローカル SDP (アンサー)
about-webrtc-remote-sdp-heading = リモート SDP
about-webrtc-remote-sdp-heading-offer = リモート SDP (オファー)
about-webrtc-remote-sdp-heading-answer = リモート SDP (アンサー)
about-webrtc-sdp-history-heading = SDP 履歴
about-webrtc-sdp-parsing-errors-heading = SDP パースエラー

##

# "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 統計
# "Stats" is an abbreviation for Statistics.
about-webrtc-ice-stats-heading = ICE 統計
about-webrtc-ice-restart-count-label = ICE 再起動:
about-webrtc-ice-rollback-count-label = ICE ロールバック:
about-webrtc-ice-pair-bytes-sent = 送信バイト数:
about-webrtc-ice-pair-bytes-received = 受信バイト数:
about-webrtc-ice-component-id = コンポーネント 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 = ローカル
about-webrtc-type-remote = リモート

##

# 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 = ノミネート
# 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 = 選択
about-webrtc-save-page-label = ページを保存
about-webrtc-debug-mode-msg-label = デバッグモード
about-webrtc-debug-mode-off-state-label = デバッグモードを開始
about-webrtc-debug-mode-on-state-label = デバッグモードを停止
about-webrtc-stats-heading = セッション統計
about-webrtc-enable-logging-label = WebRTC ログのプリセットを有効化
about-webrtc-peerconnections-section-heading = RTCPeerConnection 統計
about-webrtc-peerconnections-section-show-msg = RTCPeerConnection 統計を表示
about-webrtc-peerconnections-section-hide-msg = RTCPeerConnection 統計を隠す
about-webrtc-stats-clear = 履歴を消去
about-webrtc-log-heading = 接続ログ
about-webrtc-log-clear = ログを消去
about-webrtc-log-show-msg = ログを表示
    .title = クリックしてセクションを展開します
about-webrtc-log-hide-msg = ログを隠す
    .title = クリックしてセクションを折りたたみます
about-webrtc-log-section-show-msg = ログを表示
    .title = クリックしてセクションを展開します
about-webrtc-log-section-hide-msg = ログを隠す
    .title = クリックしてセクションを折りたたみます
about-webrtc-copy-report-button = レポートをコピー
about-webrtc-copy-report-history-button = レポート履歴をコピー

## 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 } (切断) { $now }

## These are used to indicate what direction media is flowing.
## Variables:
##  $codecs - a list of media codecs

about-webrtc-short-send-receive-direction = 送信 / 受信: { $codecs }
about-webrtc-short-send-direction = 送信: { $codecs }
about-webrtc-short-receive-direction = 受信: { $codecs }

##

about-webrtc-local-candidate = ローカル通信情報
about-webrtc-remote-candidate = リモート通信情報
about-webrtc-raw-candidates-heading = すべての生通信情報
about-webrtc-raw-local-candidate = ローカルの生通信情報
about-webrtc-raw-remote-candidate = リモートの生通信情報
about-webrtc-raw-cand-show-msg = 生通信情報を表示
    .title = クリックしてセクションを展開します
about-webrtc-raw-cand-hide-msg = 生通信情報を隠す
    .title = クリックしてセクションを折りたたみます
about-webrtc-raw-cand-section-show-msg = 生通信情報を表示
    .title = クリックしてセクションを展開します
about-webrtc-raw-cand-section-hide-msg = 生通信情報を隠す
    .title = クリックしてセクションを折りたたみます
about-webrtc-priority = 優先度
about-webrtc-fold-show-msg = 詳細を表示
    .title = クリックしてセクションを展開します
about-webrtc-fold-hide-msg = 詳細を隠す
    .title = クリックしてセクションを折りたたみます
about-webrtc-fold-default-show-msg = 詳細を表示
    .title = クリックしてセクションを展開します
about-webrtc-fold-default-hide-msg = 詳細を隠す
    .title = クリックしてセクションを折りたたみます
about-webrtc-dropped-frames-label = ドロップフレーム:
about-webrtc-discarded-packets-label = 破棄パケット:
about-webrtc-decoder-label = デコーダー
about-webrtc-encoder-label = エンコーダー
about-webrtc-show-tab-label = タブを表示
about-webrtc-current-framerate-label = フレームレート
about-webrtc-width-px = 幅 (px)
about-webrtc-height-px = 高さ (px)
about-webrtc-consecutive-frames = 連続フレーム数
about-webrtc-time-elapsed = 経過時間 (秒)
about-webrtc-estimated-framerate = 予測フレームレート
about-webrtc-rotation-degrees = 回転 (度)
about-webrtc-first-frame-timestamp = 先頭フレームの受信時刻
about-webrtc-last-frame-timestamp = 末尾フレームの受信時刻

## 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 = ローカル受信 SSRC
# This is an SSRC on the remote side of the connection that is sending RTP
about-webrtc-remote-send-ssrc = リモート送信 SSRC

## These are displayed on the button that shows or hides the
## PeerConnection configuration disclosure

about-webrtc-pc-configuration-show-msg = 構成を表示
about-webrtc-pc-configuration-hide-msg = 構成を隠す

##

# An option whose value will not be displayed but instead noted as having been
# provided
about-webrtc-configuration-element-provided = 提供済み
# An option whose value will not be displayed but instead noted as having not
# been provided
about-webrtc-configuration-element-not-provided = 未提供
# The options set by the user in about:config that could impact a WebRTC call
about-webrtc-custom-webrtc-configuration-heading = ユーザー設定の WebRTC オプション
# The options set by the user in about:config that could impact a WebRTC call
about-webrtc-user-modified-configuration-heading = 変更された 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 = 変更された設定を表示
about-webrtc-user-modified-configuration-hide-msg = 変更された設定を隠す

##

# Section header for estimated bandwidths of WebRTC media flows
about-webrtc-bandwidth-stats-heading = 推定帯域幅
# The ID of the MediaStreamTrack
about-webrtc-track-identifier = トラック識別子
# The estimated bandwidth available for sending WebRTC media in bytes per second
about-webrtc-send-bandwidth-bytes-sec = 送信帯域幅 (バイト/秒)
# The estimated bandwidth available for receiving WebRTC media in bytes per second
about-webrtc-receive-bandwidth-bytes-sec = 受信帯域幅 (バイト/秒)
# Maximum number of bytes per second that will be padding zeros at the ends of packets
about-webrtc-max-padding-bytes-sec = ゼロ埋め最大 (バイト/秒)
# The amount of time inserted between packets to keep them spaced out
about-webrtc-pacer-delay-ms = 遅延挿入 (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 = 動画フレーム統計 - 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 = ページを保存しました: { $path }
about-webrtc-debug-mode-off-state-msg = トレースログの保存場所: { $path }
about-webrtc-debug-mode-on-state-msg = デバッグモードが有効です。トレースログの保存場所: { $path }
about-webrtc-aec-logging-off-state-msg = 記録したログファイルの保存場所: { $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 = ページを保存しました: { $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 } フレーム
       *[other] { $frames } フレーム
    }
# 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 } チャンネル
       *[other] { $channels } チャンネル
    }
# 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] { $packets } パケット受信
       *[other] { $packets } パケット受信
    }
# 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] { $packets } パケット損失
       *[other] { $packets } パケット損失
    }
# 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] { $packets } パケット送信
       *[other] { $packets } パケット送信
    }
# 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 = ジッター { $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 通信情報 (アンサー後の着信) は青色で強調されます

## "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:
#  $timestamp (Number) - The Unix Epoch time at which the SDP was set.
about-webrtc-sdp-set-at-timestamp-local = 時刻 { NUMBER($timestamp, useGrouping: "false") } に ローカル SDP を設定
# This is used as a header for remote SDP.
# Variables:
#  $timestamp (Number) - The Unix Epoch time at which the SDP was set.
about-webrtc-sdp-set-at-timestamp-remote = 時刻 { NUMBER($timestamp, useGrouping: "false") } に リモート SDP を設定
# 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.
#  $relative-timestamp (Number) - The timestamp relative to the timestamp of the earliest received SDP.
about-webrtc-sdp-set-timestamp = タイムスタンプ { NUMBER($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 = SDP を表示
about-webrtc-hide-msg-sdp = SDP を隠す

## 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 = メディアコンテキストを表示
about-webrtc-media-context-hide-msg = メディアコンテキストを隠す
about-webrtc-media-context-heading = メディアコンテキスト

##