| search_documentationD | – |
| get_voicesA | Get a list of available voices for text-to-speech using V2 API Args:
model: Optional filter for specific TTS models (ssfm-v21 or ssfm-v30).
gender: Optional filter for voice gender (male or female).
age: Optional filter for voice age group (child, teen, young_adult, middle_aged, senior).
use_cases: Optional filter for voice use case (e.g. 'audiobook', 'narration', 'documentary').
Pass a single use case string supported by the V2 voices endpoint.
Returns:
List of available voices with enhanced metadata including gender, age, and use cases.
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| recommend_voicesA | Recommend voices that match a natural-language text description. The recommendation API returns only voice_id, voice_name, and score. Call
get_voices or get_voice with the returned IDs when you need metadata such as
supported models, emotions, gender, age, or use cases before making a TTS
request.
Args:
query: Text description of the desired style, mood, language, use case,
or content context.
count: Maximum number of recommendations to return. Must be 1-10.
Returns:
Recommended voice candidates sorted by relevance score.
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| get_voiceA | Get detailed information for a specific voice by ID using V2 API Args:
voice_id: The voice ID (e.g., 'tc_672c5f5ce59fac2a48faeaee')
Returns:
Voice information with enhanced metadata including gender, age, use cases, and supported models with emotions.
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| clone_voiceA | Create a quick-cloned custom voice. Calls POST /v1/voices/clone with multipart form data. Use the returned
voice_id with text_to_speech, text_to_speech_stream, or
text_to_speech_with_timestamps. Delete temporary cloned voices with
delete_cloned_voice when they are no longer needed.
Args:
name: Display name for the cloned voice. Must be 1-30 characters.
audio_file_path: Local WAV or MP3 sample path. Maximum file size is 25 MB.
model: Voice cloning model. Default: ssfm-v30.
audio_base64: Base64-encoded WAV or MP3 sample for a remote MCP server.
audio_filename: Filename with .wav or .mp3 extension for audio_base64.
Returns:
Dict returned by the Typecast API plus normalized handoff fields:
voice_id, cloned_voice_id, next_step_voice_id, next_step_model.
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| delete_cloned_voiceA | Delete a quick-cloned custom voice. Args:
voice_id: Cloned voice ID returned by clone_voice. Must start with uc_.
Returns:
Dict with success=true and the deleted voice_id.
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| text_to_speechA | Convert text to speech using the specified voice and parameters Args:
voice_id: ID of the voice to use
text: Text to convert to speech
model: TTS model to use (ssfm-v21 or ssfm-v30, default: ssfm-v30)
emotion_type: For ssfm-v30: 'preset' for explicit emotion or 'smart' for context-aware inference (default: preset)
emotion_preset: Emotion preset type. v21: normal/happy/sad/angry. v30: adds whisper/toneup/tonedown (default: normal)
emotion_intensity: Intensity of the emotion, between 0.0 and 2.0 (default: 1.0)
previous_text: For smart mode - previous context text for emotion inference
next_text: For smart mode - next context text for emotion inference
volume: Audio volume level, between 0 and 200 (default: 100)
audio_pitch: Audio pitch adjustment, between -12 and 12 (default: 0)
audio_tempo: Audio playback speed, between 0.5 and 2.0 (default: 1.0)
audio_format: Audio format, either 'wav' or 'mp3' (default: wav)
target_lufs: Optional absolute loudness normalization target in LUFS (-70.0 ~ 0.0).
Mutually exclusive with a custom volume value on this non-streaming endpoint.
Returns:
Local mode: path to the saved audio file.
Remote mode: dict with audio_url and expires_in_seconds.
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| play_audioB | Play the audio file at the specified path Args:
file_path: Path to the audio file to play
Returns:
Status message
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| text_to_speech_streamA | Convert text to speech via the streaming endpoint and save the result. Calls POST /v1/text-to-speech/stream which returns chunked audio data
in real time. The chunks are concatenated and saved as a single file.
Note: the streaming endpoint does not accept volume, but supports
target_lufs for absolute loudness normalization.
Args:
voice_id: ID of the voice to use
text: Text to convert to speech
model: TTS model (ssfm-v21 or ssfm-v30, default: ssfm-v30)
emotion_type: For ssfm-v30: 'preset' or 'smart' (default: preset)
emotion_preset: Emotion preset name (default: normal)
emotion_intensity: Emotion intensity, 0.0 ~ 2.0 (default: 1.0)
previous_text: For smart mode - previous context text
next_text: For smart mode - next context text
audio_pitch: -12 ~ 12 (default: 0)
audio_tempo: 0.5 ~ 2.0 (default: 1.0)
audio_format: 'wav' or 'mp3' (default: wav)
target_lufs: Optional absolute loudness normalization target in LUFS (-70.0 ~ 0.0)
Returns:
Local mode: path to the saved audio file.
Remote mode: dict with audio_url and expires_in_seconds.
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| get_my_subscriptionA | Get the authenticated user's subscription information. Calls GET /v1/users/me/subscription and returns the plan tier, credits
(used / total), and concurrency limit.
Returns:
Dict with this shape:
{
"plan": "free" | "lite" | "plus" | "custom",
"credits": {"plan_credits": int, "used_credits": int},
"limits": {"concurrency_limit": int}
}
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| text_to_speech_with_timestampsA | Convert text to speech and return timestamp alignment for caption generation. Calls POST /v1/text-to-speech/with-timestamps. Saves the audio file and
returns the file path together with the raw alignment payload (words and
characters arrays as returned by the server).
For non-whitespace languages such as jpn or zho, pass granularity='char'
or 'both'. With 'word' on those languages the server collapses the entire
sentence into a single word segment.
Args:
voice_id: ID of the voice to use
text: Text to convert to speech
model: TTS model (default: ssfm-v30)
granularity: 'word', 'char', or 'both'. None lets the server use its
default (word). For jpn/zho prefer 'char' or 'both'.
emotion_type, emotion_preset, emotion_intensity, previous_text,
next_text, language, volume, audio_pitch, audio_tempo, audio_format:
same shape as text_to_speech.
Returns:
Dict:
- local mode: 'audio_path' — path to the saved audio file
- remote mode: 'audio_url' and 'expires_in_seconds'
- 'words': list | None — word-level alignment when available
- 'characters': list | None — character-level alignment when available
- 'raw': dict — full server response with the audio bytes stripped
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