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# Copyright (C) 2023, Xorbits Inc.
# This file is distributed under the same license as the Xinference package.
# FIRST AUTHOR <EMAIL@ADDRESS>, 2024.
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msgid ""
msgstr ""
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"Generated-By: Babel 2.18.0\n"

#: ../../source/models/model_abilities/audio.rst:5
msgid "Audio"
msgstr "音頻"

#: ../../source/models/model_abilities/audio.rst:7
msgid "Learn how to turn audio into text or text into audio with Xinference."
msgstr "學習如何使用 Xinference 將音訊轉換為文字或將文字轉換為音訊。"

#: ../../source/models/model_abilities/audio.rst:11
msgid "Introduction"
msgstr "您好！请问您需要我介绍什么内容呢？比如产品、服务、技术，或者其他方面？请提供更多信息，以便我为您提供准确的介绍。"

#: ../../source/models/model_abilities/audio.rst:14
msgid "The Audio API provides three methods for interacting with audio:"
msgstr "Audio API 提供了三種與音訊互動的方法："

#: ../../source/models/model_abilities/audio.rst:17
msgid "The transcriptions endpoint transcribes audio into the input language."
msgstr "轉錄終端將音訊轉錄為輸入語言。"

#: ../../source/models/model_abilities/audio.rst:18
msgid "The translations endpoint translates audio into English."
msgstr "翻譯端點將音訊轉換為英文。"

#: ../../source/models/model_abilities/audio.rst:19
msgid "The speech endpoint generates audio from the input text."
msgstr "轉錄終端將音頻轉錄為輸入語言。"

#: ../../source/models/model_abilities/audio.rst:26
msgid "API ENDPOINT"
msgstr "API 端點"

#: ../../source/models/model_abilities/audio.rst:27
msgid "OpenAI-compatible ENDPOINT"
msgstr "OpenAI 相容端點"

#: ../../source/models/model_abilities/audio.rst:29
msgid "Transcription API"
msgstr "轉錄 API"

#: ../../source/models/model_abilities/audio.rst:30
msgid "/v1/audio/transcriptions"
msgstr "/v1/audio/transcriptions"

#: ../../source/models/model_abilities/audio.rst:32
msgid "Translation API"
msgstr "API"

#: ../../source/models/model_abilities/audio.rst:33
msgid "/v1/audio/translations"
msgstr "/v1/audio/translations"

#: ../../source/models/model_abilities/audio.rst:35
msgid "Speech API"
msgstr "語音 API"

#: ../../source/models/model_abilities/audio.rst:36
msgid "/v1/audio/speech"
msgstr "/v1/audio/speech"

#: ../../source/models/model_abilities/audio.rst:40
msgid "Supported models"
msgstr "支援的模型列表"

#: ../../source/models/model_abilities/audio.rst:42
msgid "The audio API is supported with the following models in Xinference:"
msgstr "在Xinference中，以下模型支援音訊API："

#: ../../source/models/model_abilities/audio.rst:45
msgid "Audio to text"
msgstr "語音轉文字"

#: ../../source/models/model_abilities/audio.rst:47
msgid ":ref:`whisper-tiny <models_builtin_whisper-tiny>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:48
msgid ":ref:`whisper-tiny.en <models_builtin_whisper-tiny.en>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:49
msgid ":ref:`whisper-base <models_builtin_whisper-base>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:50
msgid ":ref:`whisper-base.en <models_builtin_whisper-base.en>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:51
msgid ":ref:`whisper-medium <models_builtin_whisper-medium>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:52
msgid ":ref:`whisper-medium.en <models_builtin_whisper-medium.en>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:53
msgid ":ref:`whisper-large-v3 <models_builtin_whisper-large-v3>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:54
msgid ":ref:`whisper-large-v3-turbo <models_builtin_whisper-large-v3-turbo>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:55
msgid ""
":ref:`Belle-distilwhisper-large-v2-zh <models_builtin_belle-distilwhisper-"
"large-v2-zh>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:56
msgid ""
":ref:`Belle-whisper-large-v2-zh <models_builtin_belle-whisper-large-v2-zh>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:57
msgid ""
":ref:`Belle-whisper-large-v3-zh <models_builtin_belle-whisper-large-v3-zh>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:58
msgid ":ref:`SenseVoiceSmall <models_builtin_sensevoicesmall>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:59
msgid ":ref:`Paraformer-zh <models_builtin_paraformer-zh>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:61
#: ../../source/models/model_abilities/audio.rst:99
msgid "For Mac M-series chips only:"
msgstr "僅適用於 Mac M 系列晶片："

#: ../../source/models/model_abilities/audio.rst:63
msgid ":ref:`whisper-tiny-mlx <models_builtin_whisper-tiny-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:64
msgid ":ref:`whisper-tiny.en-mlx <models_builtin_whisper-tiny.en-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:65
msgid ":ref:`whisper-base-mlx <models_builtin_whisper-base-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:66
msgid ":ref:`whisper-base.en-mlx <models_builtin_whisper-base.en-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:67
msgid ":ref:`whisper-medium-mlx <models_builtin_whisper-medium-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:68
msgid ":ref:`whisper-medium.en-mlx <models_builtin_whisper-medium.en-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:69
msgid ":ref:`whisper-large-v3-mlx <models_builtin_whisper-large-v3-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:70
msgid ""
":ref:`whisper-large-v3-turbo-mlx <models_builtin_whisper-large-v3-turbo-"
"mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:74
msgid "Text to audio (TTS)"
msgstr "文字轉語音（TTS）"

#: ../../source/models/model_abilities/audio.rst:76
msgid ""
"**Models supporting zero-shot** (direct synthesis without reference audio):"
msgstr "**支援zero-shot的模型** （無需參考音訊）"

#: ../../source/models/model_abilities/audio.rst:78
msgid ":ref:`ChatTTS <models_builtin_chattts>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:79
msgid ":ref:`CosyVoice-300M-SFT <models_builtin_cosyvoice-300m-sft>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:80
msgid ""
":ref:`CosyVoice-300M-Instruct <models_builtin_cosyvoice-300m-instruct>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:81
msgid "MeloTTS series"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:82
msgid ":ref:`Kokoro-82M <models_builtin_kokoro-82m>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:83
#: ../../source/models/model_abilities/audio.rst:102
msgid ":ref:`Kokoro-82M-MLX <models_builtin_kokoro-82m-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:84
msgid ":ref:`MegaTTS3 <models_builtin_megatts3>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:86
msgid "**Models supporting voice cloning** (requires reference audio):"
msgstr "**支援語音克隆的模型**（需參考音訊）"

#: ../../source/models/model_abilities/audio.rst:88
msgid ":ref:`CosyVoice-300M <models_builtin_cosyvoice-300m>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:89
msgid ":ref:`CosyVoice 2.0 <models_builtin_cosyvoice2-0.5b>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:90
msgid ":ref:`FishSpeech-1.5 <models_builtin_fishspeech-1.5>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:91
msgid ":ref:`F5-TTS <models_builtin_f5-tts>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:92
#: ../../source/models/model_abilities/audio.rst:101
msgid ":ref:`F5-TTS-MLX <models_builtin_f5-tts-mlx>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:93
#: ../../source/models/model_abilities/audio.rst:97
msgid ":ref:`IndexTTS2 <models_builtin_indextts2>`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:95
msgid "**Models supporting emotion control**:"
msgstr "**支援情感控制的模型**"

#: ../../source/models/model_abilities/audio.rst:105
msgid "Quickstart"
msgstr "快速入門"

#: ../../source/models/model_abilities/audio.rst:108
msgid "Transcription"
msgstr "轉錄"

#: ../../source/models/model_abilities/audio.rst:110
msgid ""
"The Transcription API mimics OpenAI's `create transcriptions API "
"<https://platform.openai.com/docs/api-"
"reference/audio/createTranscription>`_. We can try Transcription API out "
"either via cURL, OpenAI Client, or Xinference's python client:"
msgstr ""
"Transcription API 模仿了 OpenAI 的 `create transcriptions API "
"<https://platform.openai.com/docs/api-"
"reference/audio/createTranscription>`_。你可以透過 cURL、OpenAI Client 或 Xinference"
" 的 Python 客戶端來嘗試 Transcription API："

#: ../../source/models/model_abilities/audio.rst:161
msgid "Translation"
msgstr "请提供需要翻译的文本。"

#: ../../source/models/model_abilities/audio.rst:163
msgid ""
"The Translation API mimics OpenAI's `create translations API "
"<https://platform.openai.com/docs/api-reference/audio/createTranslation>`_. "
"We can try Translation API out either via cURL, OpenAI Client, or "
"Xinference's python client:"
msgstr ""
"Translation API 模仿了 OpenAI 的 `create translations API "
"<https://platform.openai.com/docs/api-"
"reference/audio/createTranslation>`_。你可以透過 cURL、OpenAI Client 或 Xinference 的"
" Python 客戶端來嘗試使用 Translation API："

#: ../../source/models/model_abilities/audio.rst:213
msgid "Speech"
msgstr "語音"

#: ../../source/models/model_abilities/audio.rst:217
msgid ""
"The Speech API mimics OpenAI's `create speech API "
"<https://platform.openai.com/docs/api-reference/audio/createSpeech>`_. We "
"can try Speech API out either via cURL, OpenAI Client, or Xinference's "
"python client:"
msgstr ""
"Transcription API 模仿了 OpenAI 的 `create speech API "
"<https://platform.openai.com/docs/api-reference/audio/createSpeech>`_。你可以透過 "
"cURL、OpenAI Client 或 Xinference 的 Python 用戶端來嘗試 Speech API："

#: ../../source/models/model_abilities/audio.rst:220
msgid "Speech API use non-stream by default as"
msgstr "Speech API 預設使用非串流"

#: ../../source/models/model_abilities/audio.rst:222
msgid ""
"The stream output of ChatTTS is not as good as the non-stream output, please"
" refer to: https://github.com/2noise/ChatTTS/pull/564"
msgstr "ChatTTS 的串流輸出不如非串流的效果好，參考：https://github.com/2noise/ChatTTS/pull/564"

#: ../../source/models/model_abilities/audio.rst:223
msgid ""
"The stream requires ffmpeg<7: "
"https://pytorch.org/audio/stable/installation.html#optional-dependencies"
msgstr ""
"ffmpeg<7 的串流需求：https://pytorch.org/audio/stable/installation.html#optional-"
"dependencies"

#: ../../source/models/model_abilities/audio.rst:275
msgid "ChatTTS Usage"
msgstr "ChatTTS 使用"

#: ../../source/models/model_abilities/audio.rst:277
#: ../../source/models/model_abilities/audio.rst:480
msgid "Basic usage, refer to :ref:`audio speech usage <audio_speech>`."
msgstr "基本使用，參考 :ref:`語音使用章節 <audio_speech>`。"

#: ../../source/models/model_abilities/audio.rst:279
msgid ""
"Fixed tone color. We can use fixed tone color provided by "
"https://github.com/6drf21e/ChatTTS_Speaker, Download the "
"`evaluation_result.csv "
"<https://github.com/6drf21e/ChatTTS_Speaker/blob/main/evaluation_results.csv>`_"
" , take ``seed_2155`` as example, we get the ``emb_data`` of it."
msgstr ""
"固定音色。我們可以使用由 https://github.com/6drf21e/ChatTTS_Speaker 提供的固定音色，下載 "
"`evaluation_result.csv "
"<https://github.com/6drf21e/ChatTTS_Speaker/blob/main/evaluation_results.csv>`_"
" ，以 ``seed_2155`` 音色作為範例，我們使用 ``emb_data`` 欄位的資料。"

#: ../../source/models/model_abilities/audio.rst:292
msgid "Use the fixed tone color of ``seed_2155`` to generate speech."
msgstr "使用 ``seed_2155`` 固定音色來建立語音。"

#: ../../source/models/model_abilities/audio.rst:308
msgid "CosyVoice Usage"
msgstr "CosyVoice 模型使用"

#: ../../source/models/model_abilities/audio.rst:310
msgid ""
"CosyVoice has two versions: CosyVoice 1.0 and CosyVoice 2.0. CosyVoice 1.0 "
"has three different models:"
msgstr ""
"CosyVoice 有兩個版本：CosyVoice 1.0 和 CosyVoice 2.0。CosyVoice 1.0 有 3 個不同模型："

#: ../../source/models/model_abilities/audio.rst:312
msgid ""
"**CosyVoice-300M-SFT**: Choose this model if you just want to convert text "
"to audio. There are pretrained voices available: ['中文女', '中文男', '日语男', "
"'粤语女', '英文女', '英文男', '韩语女']"
msgstr ""
"**CosyVoice-300M-SFT**: 如果你只想把文字轉換為語音，選擇這個模型。它提供了一些預訓練的音色: ['中文女', '中文男', "
"'日語男', '粵語女', '英文女', '英文男', '韓語女']"

#: ../../source/models/model_abilities/audio.rst:313
msgid ""
"**CosyVoice-300M**: Choose this model if you want to clone voice or convert "
"text to audio in different languages. The ``prompt_speech`` is always "
"required and should be a WAV file. For optimal performance, use a sample "
"rate of 16,000 Hz."
msgstr ""
"**CosyVoice-300M**: 如果你想複製聲音或將文字轉換為另一種語言的語音，請選擇這個模型。使用此模型時，你必須提供 "
"``prompt_speech`` WAV格式的音頻檔案，建議使用 16,000 Hz 取樣率以獲得更好的效能。"

#: ../../source/models/model_abilities/audio.rst:314
msgid ""
"**CosyVoice-300M-Instruct**: Choose this model If you need precise control "
"over the tone and pitch."
msgstr "**CosyVoice-300M-Instruct**：如要精準控制音調與音色，請選擇此模型。"

#: ../../source/models/model_abilities/audio.rst:316
msgid "Basic usage, launch model ``CosyVoice-300M-SFT``."
msgstr "基本使用，載入模型 ``CosyVoice-300M-SFT``。"

#: ../../source/models/model_abilities/audio.rst:365
msgid "Clone voice, launch model ``CosyVoice-300M``."
msgstr "複製聲音，載入模型 ``CosyVoice-300M``。"

#: ../../source/models/model_abilities/audio.rst:395
msgid "Cross lingual usage, launch model ``CosyVoice-300M``."
msgstr "跨語言使用，載入模型 ``CosyVoice-300M``。"

#: ../../source/models/model_abilities/audio.rst:418
msgid "Instruction based, launch model ``CosyVoice-300M-Instruct``."
msgstr "基於指令的聲音合成，載入模型 ``CosyVoice-300M-Instruct``。"

#: ../../source/models/model_abilities/audio.rst:435
msgid ""
"CosyVoice 2.0 only has one model, it provides all the capabilities of the "
"three CosyVoice models. The usage is the same as CosyVoice."
msgstr "CosyVoice 2.0 只有一個模型，但它包含了 CosyVoice 三個模型的所有能力。使用方法與 CosyVoice 一樣。"

#: ../../source/models/model_abilities/audio.rst:437
msgid "CosyVoice 2.0 stream usage, launch model ``CosyVoice2-0.5B``."
msgstr "CosyVoice 2.0 流式使用，載入模型 ``CosyVoice2-0.5B``。"

#: ../../source/models/model_abilities/audio.rst:474
msgid ""
"More instructions and examples, could be found at https://fun-audio-"
"llm.github.io/ ."
msgstr "更多指令和範例，請參考 https://fun-audio-llm.github.io/ 。"

#: ../../source/models/model_abilities/audio.rst:478
msgid "FishSpeech Usage"
msgstr "FishSpeech 模型使用"

#: ../../source/models/model_abilities/audio.rst:482
msgid ""
"Clone voice, launch model ``FishSpeech-1.5``. Please use `prompt_speech` "
"instead of `reference_audio` and `prompt_text` instead of `reference_text` "
"to clone voice from the reference audio for the FishSpeech model. This "
"arguments is aligned to voice cloning of CosyVoice."
msgstr ""
"克隆語音，啟動模型 ``FishSpeech-1.5``。請使用 `prompt_speech` 而不是 `reference_audio` 以及 "
"`prompt_text` 而不是 `reference_text` 來為 FishSpeech 模型提供參考音頻。這個參數和 CosyVoice "
"的語音克隆保持一致。"

#: ../../source/models/model_abilities/audio.rst:509
msgid "Paraformer Usage"
msgstr "Paraformer 使用說明"

#: ../../source/models/model_abilities/audio.rst:512
msgid "model"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:512
msgid "vad"
msgstr "語音活動檢測（vad）"

#: ../../source/models/model_abilities/audio.rst:512
msgid "punc"
msgstr "標點恢復（punc）"

#: ../../source/models/model_abilities/audio.rst:512
msgid "timestamp"
msgstr "時間戳"

#: ../../source/models/model_abilities/audio.rst:512
msgid "speaker"
msgstr "說話人"

#: ../../source/models/model_abilities/audio.rst:512
msgid "hotword"
msgstr "熱詞"

#: ../../source/models/model_abilities/audio.rst:514
msgid ":ref:`models_builtin_paraformer-zh`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:514
#: ../../source/models/model_abilities/audio.rst:516
#: ../../source/models/model_abilities/audio.rst:518
#: ../../source/models/model_abilities/audio.rst:520
#: ../../source/models/model_abilities/audio.rst:522
msgid "yes"
msgstr "是"

#: ../../source/models/model_abilities/audio.rst:514
#: ../../source/models/model_abilities/audio.rst:516
#: ../../source/models/model_abilities/audio.rst:518
#: ../../source/models/model_abilities/audio.rst:520
msgid "no"
msgstr "否"

#: ../../source/models/model_abilities/audio.rst:516
msgid ":ref:`models_builtin_paraformer-zh-hotword`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:518
msgid ":ref:`models_builtin_paraformer-zh-spk`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:520
msgid ":ref:`models_builtin_paraformer-zh-long`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:522
msgid ":ref:`models_builtin_seaco-paraformer-zh` (recommend)"
msgstr ":ref:`models_builtin_seaco-paraformer-zh` （推薦）"

#: ../../source/models/model_abilities/audio.rst:525
msgid "**VAD & Punctuation Usage**"
msgstr "**VAD 與標點符號的使用**"

#: ../../source/models/model_abilities/audio.rst:527
msgid "All Paraformer models support VAD and punctuation."
msgstr "所有 Paraformer 模型均支援 VAD 與標點功能。"

#: ../../source/models/model_abilities/audio.rst:529
msgid "**Timestamp & Speaker Usage**"
msgstr "**時間戳記與說話人識別使用說明**"

#: ../../source/models/model_abilities/audio.rst:531
msgid "Only the following models support `timestamp` and `speaker`:"
msgstr "僅以下模型支援 `時間戳` 和 `說話人` 辨識："

#: ../../source/models/model_abilities/audio.rst:533
msgid "`paraformer-zh-spk`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:534
msgid "`paraformer-zh-long`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:535
#: ../../source/models/model_abilities/audio.rst:560
msgid "`seaco-paraformer-zh`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:537
msgid ""
"Among them, only `paraformer-zh-spk` enables **speaker info by default**."
msgstr "其中，僅 `paraformer-zh-spk` 預設啟用說話人識別功能。"

#: ../../source/models/model_abilities/audio.rst:539
msgid ""
"If you need speaker info when using `paraformer-zh-long` or `seaco-"
"paraformer-zh`:"
msgstr "如果你使用的是 `paraformer-zh-long` 或 `seaco-paraformer-zh`，且需要啟用說話人識別功能："

#: ../../source/models/model_abilities/audio.rst:541
msgid ""
"In Web UI: add an extra parameter with key ``spk_model`` and value ``cam++``"
msgstr "在 Web UI 中：添加名為 ``spk_model``、值為 ``cam++`` 的參數"

#: ../../source/models/model_abilities/audio.rst:542
msgid "In command line: add the option ``--spk_model cam++``"
msgstr "在命令列中：添加參數 ``--spk_model cam++``"

#: ../../source/models/model_abilities/audio.rst:544
#: ../../source/models/model_abilities/audio.rst:562
msgid "Example:"
msgstr "示例："

#: ../../source/models/model_abilities/audio.rst:555
msgid "**Hotword Usage**"
msgstr "**熱詞功能使用說明**"

#: ../../source/models/model_abilities/audio.rst:557
msgid "Only the following models support `hotword`:"
msgstr "僅以下模型支援 `hotword` （熱詞功能）："

#: ../../source/models/model_abilities/audio.rst:559
msgid "`paraformer-zh-hotword`"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:576
msgid "SenseVoiceSmall Offline Usage"
msgstr "SenseVoiceSmall 離線使用"

#: ../../source/models/model_abilities/audio.rst:578
msgid ""
"Now SenseVoiceSmall use a small vad model ``fsmn-vad``, it will be "
"downloaded thus network required."
msgstr "現在 SenseVoiceSmall 使用一個小的 VAD 模型 ``fsmn-vad``，因此它需要網路來下載。"

#: ../../source/models/model_abilities/audio.rst:580
msgid "For offline environment, you can download the vad model in advance."
msgstr "對於離線環境，你可以事先下載這個 VAD 模型。"

#: ../../source/models/model_abilities/audio.rst:582
msgid ""
"Download from `huggingface <https://huggingface.co/funasr/fsmn-vad>`_ or "
"`modelscope <https://modelscope.cn/models/iic/speech_fsmn_vad_zh-"
"cn-16k-common-pytorch/files>`_. Assume downloaded to ``/path/to/fsmn-vad``."
msgstr ""
"從 `huggingface <https://huggingface.co/funasr/fsmn-vad>`_ 或者 `modelscope "
"<https://modelscope.cn/models/iic/speech_fsmn_vad_zh-cn-16k-common-"
"pytorch/files>`_ 下載。假設下載到 ``/path/to/fsmn-vad``。"

#: ../../source/models/model_abilities/audio.rst:585
msgid ""
"Then when launching SenseVoiceSmall with Web UI, you can add an additional "
"parameter with key ``vad_model`` and value ``/path/to/fsmn-vad`` which is "
"the downloaded path. When launching with command line, you can add an option"
" ``--vad_model /path/to/fsmn-vad``."
msgstr ""
"然後當用 Web UI 載入 SenseVoiceSmall 時，新增額外選項，key 是 ``vad_model``，值是之前的下載路徑 "
"``/path/to/fsmn-vad``。用命令列載入時，增加選項 ``--vad_model /path/to/fsmn-vad``。"

#: ../../source/models/model_abilities/audio.rst:590
msgid "Kokoro Usage"
msgstr "Kokoro 模型使用"

#: ../../source/models/model_abilities/audio.rst:592
msgid ""
"The Kokoro model supports multiple languages, but the default language is "
"English. If you want to use other languages, such as Chinese, you need to "
"install additional dependency packages and add an additional parameter when "
"starting the model."
msgstr "Kokoro模型支援多語言，預設為英文。若想使用非預設語言（例如中文），則需要安裝額外依賴套件，並在模型啟動時加入對應參數。"

#: ../../source/models/model_abilities/audio.rst:596
msgid "pip install misaki[zh]"
msgstr ""

#: ../../source/models/model_abilities/audio.rst:598
msgid ""
"Initialize the model with the parameter lang_code='z', For all available "
"``lang_code`` options, please refer to `kokoro source code "
"<https://github.com/hexgrad/kokoro/blob/main/kokoro/pipeline.py#L22>`_. If "
"the model is started through the web UI, an additional parameter needs to be"
" added, with the key as ``lang_code`` and the value as ``z``. If the model "
"is started through the xinference client, the parameters are passed via the "
"launch_model interface:"
msgstr ""
"使用 lang_code='z' 參數初始化模型，可以參考 `kokoro source code "
"<https://github.com/hexgrad/kokoro/blob/main/kokoro/pipeline.py#L22>`_ "
"查看所有支援的 lang_code。如果你是透過 Web UI 啟動的模型，則需要添加額外參數，key 是 ``lang_code``，value 是 "
"``z``。如果你是透過 xinference client 啟動的模型，則可以參考如下程式碼傳遞參數："

#: ../../source/models/model_abilities/audio.rst:615
msgid ""
"When inferring, the voice must start with 'z', for example: ``zf_xiaoyi``. "
"The currently supported voices are: "
"https://huggingface.co/hexgrad/Kokoro-82M/tree/main/voices. For example:"
msgstr ""
"當進行推理時，需使用 'z' 開頭的 voice，例如：``zf_xiaoyi``。目前支援的 voices 可參考 "
"https://huggingface.co/hexgrad/Kokoro-82M/tree/main/voices。使用方式如下："

#: ../../source/models/model_abilities/audio.rst:625
msgid "IndexTTS2 Usage"
msgstr "IndexTTS2 使用方式"

#: ../../source/models/model_abilities/audio.rst:627
msgid ""
"The IndexTTS2 model supports emotion control, you can use this feature by "
"specifying some additional parameters. Here are several examples of how to "
"use IndexTTS2:"
msgstr "IndexTTS2模型支援情感控制，你可以透過使用一些額外的參數來啟用這個功能。以下為IndexTTS2的使用方式："

#: ../../source/models/model_abilities/audio.rst:630
msgid ""
"Synthesize new speech with a single reference audio file (voice cloning):"
msgstr "單一參考音頻（音色克隆）："

#: ../../source/models/model_abilities/audio.rst:646
msgid ""
"Using a separate, emotional reference audio file to condition the speech "
"synthesis:"
msgstr "指定情感參考音頻："

#: ../../source/models/model_abilities/audio.rst:666
msgid ""
"When an emotional reference audio file is specified, you can optionally set "
"the ``emo_alpha`` to adjust how much it affects the output. Valid range is "
"``0.0 - 1.0`` , and the default value is ``1.0`` (100%):"
msgstr ""
"當指定情感參考音頻時，可以選擇設定 ``emo_alpha`` 參數以調整其對輸出的影響程度。有效範圍為 ``0.0 - 1.0``，預設值為 "
"``1.0`` (100%)。"

#: ../../source/models/model_abilities/audio.rst:689
msgid ""
"It's also possible to omit the emotional reference audio and instead provide"
" an 8-float list specifying the intensity of each emotion, in the following "
"order: ``[happy, angry, sad, afraid, disgusted, melancholic, surprised, "
"calm]`` . You can additionally use the ``use_random`` parameter to introduce"
" stochasticity during inference; the default is ``False`` , and setting it "
"to ``True`` enables randomness:"
msgstr ""
"可以省略情緒參考音頻，改為提供一個包含8個浮點數的列表，按以下順序指定每種情緒的強度：``[快樂, 憤怒, 悲傷, 恐懼, 厭惡, 憂鬱, 驚訝, "
"平靜]``。您還可以使用 ``use_random`` 參數在推理過程中引入隨機性情緒；預設值為 ``False``，設定為 ``True`` "
"即可啟用隨機性情緒。"

#: ../../source/models/model_abilities/audio.rst:712
msgid ""
"Alternatively, you can enable ``use_emo_text`` to guide the emotions based "
"on your provided ``text`` script. Your text script will then automatically "
"be converted into emotion vectors. It's recommended to use ``emo_alpha`` "
"around 0.6 (or lower) when using the text emotion modes, for more natural "
"sounding speech. You can introduce randomness with ``use_random`` (default: "
"``False``; ``True`` enables randomness):"
msgstr ""
"或者，您可以啟用 ``use_emo_text`` 功能，根據您提供的 ``text`` "
"腳本引導情感表達。您的文字腳本將自動轉換為情感向量。使用文字情感模式時，建議將 ``emo_alpha`` 設定為 0.6 "
"左右（或更低），以獲得更自然的語音效果。您可以透過 ``use_random`` 引入隨機性（預設值：``False`` ；``True`` "
"啟用隨機性）："

#: ../../source/models/model_abilities/audio.rst:737
msgid ""
"It's also possible to directly provide a specific text emotion description "
"via the ``emo_text`` parameter. Your emotion text will then automatically be"
" converted into emotion vectors. This gives you separate control of the text"
" script and the text emotion description:"
msgstr ""
"您也可以透過 ``emo_text`` 參數直接提供特定的文字情緒描述。您的情緒文字將自動轉換為情緒向量。這使您能夠分別控制文字腳本和文字情緒描述："

#: ../../source/models/model_abilities/audio.rst:761
msgid "IndexTTS2 Offline Usage"
msgstr "IndexTTS2 離線使用"

#: ../../source/models/model_abilities/audio.rst:763
msgid ""
"IndexTTS2 requires several small models that are downloaded automatically "
"during initialization. For offline environments, you can download these "
"models to a single directory and specify the directory path."
msgstr "IndexTTS2需要多個小型模型，這些模型會在初始化過程中自動下載。在離線環境中，您可以將這些模型下載到單一目錄，並指定該目錄路徑。"

#: ../../source/models/model_abilities/audio.rst:766
msgid "**Easy Setup Method**"
msgstr "**簡易設定方法**"

#: ../../source/models/model_abilities/audio.rst:768
msgid ""
"The simplest way to set up offline usage is to Use the `hf download` command"
" to download the small model in advance:"
msgstr "設定離線使用最簡單的方法是使用： `hf download` 指令去提前下載所有小模型。"

#: ../../source/models/model_abilities/audio.rst:781
msgid "The final directory structure should look like this:"
msgstr "最終的目錄結構應如下所示："

#: ../../source/models/model_abilities/audio.rst:791
msgid "**Required Models**"
msgstr "**支援的模型列表**"

#: ../../source/models/model_abilities/audio.rst:793
msgid "The small models are automatically mapped as follows:"
msgstr "小型模型將按以下方式自動映射："

#: ../../source/models/model_abilities/audio.rst:795
msgid ""
"**w2v-bert-2.0** (``models--facebook--w2v-bert-2.0``) - Feature extraction "
"model"
msgstr "**w2v-bert-2.0** (``models--facebook--w2v-bert-2.0``) - 特徵提取模型"

#: ../../source/models/model_abilities/audio.rst:796
msgid ""
"**campplus** (``models--funasr--campplus``) - Speaker recognition model"
msgstr "**campplus** (``models--funasr--campplus``) - 說話人辨識模型"

#: ../../source/models/model_abilities/audio.rst:797
msgid ""
"**bigvgan** (``models--nvidia--bigvgan_v2_22khz_80band_256x``) - Vocoder "
"model"
msgstr ""
"**bigvgan** (``models--nvidia--bigvgan_v2_22khz_80band_256x``) - 語音編碼器模型"

#: ../../source/models/model_abilities/audio.rst:798
msgid ""
"**semantic_codec** (``models--amphion--MaskGCT``) - Semantic "
"encoding/decoding model"
msgstr "**語義編解碼器** (``models--amphion--MaskGCT``) - 語義編碼/解碼模型"

#: ../../source/models/model_abilities/audio.rst:800
msgid "**Launching IndexTTS2 with Offline Models**"
msgstr "**使用離線模式啟動IndexTTS2**"

#: ../../source/models/model_abilities/audio.rst:802
msgid ""
"When launching IndexTTS2 with Web UI, you can add an additional parameter: -"
" ``small_models_dir`` - Path to directory containing all small models"
msgstr "在透過Web UI啟動IndexTTS2時，可新增額外參數：- ``small_models_dir`` - 包含所有小型模型的目錄路徑"

#: ../../source/models/model_abilities/audio.rst:805
msgid "When launching with command line, you can add the option:"
msgstr "在透過命令列啟動時，您可以加入以下選項："

#: ../../source/models/model_abilities/audio.rst:812
msgid "When launching with Python client:"
msgstr "使用 Python 客戶端啟動時："

#~ msgid "**random sampling**"
#~ msgstr ""

#~ msgid ""
#~ "Enabling random sampling will reduce the voice cloning fidelity of the "
#~ "speech synthesis."
#~ msgstr ""

#~ msgid ""
#~ "5.Alternatively, you can enable `use_emo_text` to guide the emotions based "
#~ "on"
#~ msgstr ""

#~ msgid ""
#~ "your provided `text` script. Your text script will then automatically be "
#~ "converted into emotion vectors. It's recommended to use `emo_alpha` around "
#~ "0.6 (or lower) when using the text emotion modes, for more natural sounding "
#~ "speech. You can introduce randomness with `use_random` (default: `False`; "
#~ "`True` enables randomness):"
#~ msgstr ""

#~ msgid ""
#~ "The required small models are: 1. **w2v-bert-2.0** - Feature extraction "
#~ "model (place in ``w2v-bert-2.0/`` subdirectory) 2. **semantic_codec** - "
#~ "Semantic encoding/decoding model (place in ``semantic_codec/`` subdirectory)"
#~ " 3. **campplus** - Speaker recognition model (place in ``campplus/`` "
#~ "subdirectory) 4. **bigvgan** - Vocoder model (place in ``bigvgan/`` "
#~ "subdirectory)"
#~ msgstr ""
#~ "所需的小型模型包括：1. **w2v-bert-2.0** - 特徵提取模型（放置於``w2v-bert-2.0/``子目錄）2. "
#~ "**semantic_codec** - 語意編碼/解碼模型（放置於``semantic_codec/``子目錄）3. **campplus** - "
#~ "說話人辨識模型（放置於``campplus/``子目錄）4. **bigvgan** - 聲碼器模型（放置於``bigvgan/``子目錄）"

#~ msgid ""
#~ "Assume downloaded to ``/path/to/small_models`` with the following structure:"
#~ msgstr "假設下載到``/path/to/small_models``目錄，其結構如下："

#~ msgid ""
#~ "**Find your Hugging Face cache directory** (usually "
#~ "``~/.cache/huggingface/hub/``)"
#~ msgstr "**查找您的Hugging Face快取目錄** （通常位於 ``~/.cache/huggingface/hub/`` ）"

#~ msgid "**Copy the required models** to your target directory:"
#~ msgstr "**將所需模型** 複製到目標目錄："

#~ msgid "**Note about Directory Structure**"
#~ msgstr "**關於目錄結構的說明**"

#~ msgid ""
#~ "The ``snapshots/`` directories contain version-specific model files with "
#~ "hash names. Xinference automatically detects and uses the correct snapshot "
#~ "directory, so you don't need to worry about the exact hash values."
#~ msgstr ""
#~ "``snapshots/`` 目錄包含具有雜湊名稱的特定版本模型檔案。Xinference會自動偵測並使用正確的快照目錄，因此您無需擔心精確的雜湊值。"
