Audio models

Speech, sound effects, and music (model input schemas), plus local audio processing (our ffmpeg implementation, free).

Generations are charged in credits (see Credits & plans). Every generation model also accepts mock: true for a free placeholder result.

ElevenLabs TTS v3 elevenlabs_tts_v3

Expressive text-to-speech with inline audio-tag emotional control and 70+ language support, powered by ElevenLabs' Eleven v3 model.

Call it via — audio(action: "speak") (MCP audio tool) · raw: POST /v1/jobs/elevenlabs_tts_v3

Cost	20 cr per 1,000 characters
Mode / timeout	sync / 60s

Parameters — the model's input schema:

Param	Type	Required	Default	Allowed / range	Description
text	string	✓	—	—	Text to convert to speech. Supports inline audio tags like [laughs], [whispers], [excited].
voice	string		Rachel	e.g. Aria, Roger, Sarah, Laura, Charlie, George, Callum, River, Liam, Charlotte, Alice, Matilda, Will, Jessica, Eric, Chris, Brian, Daniel, Lily, Bill (or a voice ID)	Voice name or ID.
stability	float		0.5	0–1	Voice stability. Lower = more expressive variation; higher = more consistent delivery.
similarity_boost	float		0.75	0–1	How closely the output matches the reference voice.
speed	float		1	—	Playback speed multiplier.
language_code	string		—	ISO 639-1 (e.g. en, ru, es, fr, de, ja, ko, zh)	Forces a specific output language.
apply_text_normalization	enum		auto	auto, on, off	Controls spelling-out of numbers, abbreviations, etc.
seed	int		—	—	Random seed for reproducibility.
timestamps	bool		false	—	When true, returns per-word timestamps in the response.
output_format	enum		mp3_44100_128	mp3_22050_32, mp3_44100_32, mp3_44100_64, mp3_44100_96, mp3_44100_128, mp3_44100_192, pcm_8000, pcm_16000, pcm_22050, pcm_24000, pcm_44100, pcm_48000, ulaw_8000, alaw_8000, opus_48000_32, opus_48000_64, opus_48000_96, opus_48000_128, opus_48000_192	Output codec, sample rate, and bitrate.

Our wrapper params (not part of the model schema): out (required — workdir-relative output path, .mp3) and mock (optional — test placeholder, no real generation). This model does not use the format→size mapping (format_field is empty).

Limits — Pricing is 20 cr per 1,000 characters (a 500-char paragraph = 10 cr; a 10,000-char story = 200 cr). Supported output formats: MP3 (22.05/44.1 kHz, 32–192 kbps), PCM (8–48 kHz), µ-law/A-law 8 kHz, Opus 48 kHz (32–192 kbps). 70+ languages supported. No hard maximum character count is published.

ElevenLabs TTS (direct) elevenlabs_tts_direct

Converts text into speech using a chosen ElevenLabs voice_id (cloned, linked, or library voice) and returns an audio file.

Call it via — audio(speak, actor_id=…) (routes a configured actor's voice through this model; plain audio(speak) without actor_id uses elevenlabs_tts_v3 instead). Also used internally by video(scene) for per-line narration. · raw: POST /v1/jobs/elevenlabs_tts_direct

Cost	20 cr per call
Mode / timeout	sync / 60s

Parameters — the model's input schema (voice_id is a path parameter; the rest are request-body fields):

Param	Type	Required	Default	Allowed / range	Description
voice_id	string	✓	—	—	Path param. ID of the voice to use (from Get Voices).
text	string	✓	—	—	The text that will be converted into speech.
model_id	string		eleven_multilingual_v2	any TTS-capable model id	Model identifier; must support text-to-speech.
language_code	string | null		null	ISO 639-1	Enforces a language for the model and text normalization.
voice_settings	object | null		null	see sub-properties	Per-request overrides of the voice's stored settings.
voice_settings.stability	number		0.5	0.0–1.0	How stable the voice is / randomness between generations.
voice_settings.similarity_boost	number		0.75	0.0–1.0	How closely the AI adheres to the original voice.
voice_settings.style	number		0	0.0–1.0	Style exaggeration of the voice.
voice_settings.use_speaker_boost	boolean		true	true/false	Boosts similarity to the original speaker.
voice_settings.speed	number		1.0	~0.7–1.2	Playback speed; <1 slows, >1 speeds up.
seed	integer | null		null	0–4294967295	Best-effort deterministic sampling.
previous_text	string | null		null	—	Text preceding this request, for continuity.
next_text	string | null		null	—	Text following this request, for continuity.
previous_request_ids	string[] | null		null	max 3	Request ids of prior samples, for continuity.
next_request_ids	string[] | null		null	max 3	Request ids of later samples, for continuity.
pronunciation_dictionary_locators	object[] | null		null	max 3	Pronunciation dictionary locators (id, version_id).
apply_text_normalization	enum		auto	auto, on, off	Controls number/date spell-out normalization.
apply_language_text_normalization	boolean		false	true/false	Language-specific normalization (Japanese only; raises latency).
output_format	enum (query)		mp3_44100_128	mp3_22050_32, mp3_44100_32/64/96/128/192, pcm_8000/16000/22050/24000/44100, ulaw_8000, alaw_8000, opus_48000_*, etc. (28 values)	Query param. codec_samplerate_bitrate; mp3_192 needs Creator+, pcm/wav 44.1kHz needs Pro+.
enable_logging	boolean (query)		true	true/false	Query param. false = zero-retention mode (enterprise only).

Our wrapper params (not part of the model schema): out (required — output audio filename, mp3) and mock (optional — test placeholder). This model has no format→size mapping (format_field is empty in our YAML).

Limits — model limits: seed 0–4294967295; up to 3 pronunciation_dictionary_locators, 3 previous_request_ids, 3 next_request_ids per request; output formats limited to the 28 output_format enum values (mp3 192kbps requires Creator tier or above; PCM/WAV at 44.1kHz requires Pro tier or above). No hard maximum text length is published for this endpoint, so no character cap is asserted here (our YAML's "keep under 5000 characters" is guidance, not a confirmed limit).

ElevenLabs Sound Effects elevenlabs_sfx

Generate sound effects (foley, ambience, UI, impacts) from a text description using ElevenLabs' Sound Effects V2 model.

Call it via — audio(sfx) (the audio MCP tool with action: "sfx"; pass your description in prompt, which the worker maps to the model's text field) · raw: POST /v1/jobs/elevenlabs_sfx

Cost	Billed per second of audio
Mode / timeout	sync / 60s

Parameters — the model's input schema:

Param	Type	Required	Default	Allowed / range	Description
text	string	✓	—	max 450 characters	The text describing the sound effect to generate.
duration_seconds	number		none (model decides)	0.5–22 (nullable)	Duration in seconds. If omitted/null, optimal duration is determined from the prompt.
prompt_influence	number		0.3	0–1	How closely to follow the prompt. Higher values mean less variation.
output_format	string (enum)		mp3_44100_128	mp3_22050_32, mp3_44100_32, mp3_44100_64, mp3_44100_96, mp3_44100_128, mp3_44100_192, pcm_8000, pcm_16000, pcm_22050, pcm_24000, pcm_44100, pcm_48000, ulaw_8000, alaw_8000, opus_48000_32, opus_48000_64, opus_48000_96, opus_48000_128, opus_48000_192	Output audio format, as codec_sampleRate_bitrate.
loop	boolean		false	true / false	Whether to create a sound effect that loops smoothly.

Our wrapper params (not part of the model schema): out (required — workdir-relative output path, e.g. .mp3) and mock (optional — test placeholder). No format mapping applies to this model (format_field is empty).

Limits — model limits:

• text: max 450 characters.
• duration_seconds: 0.5–22 seconds.
• prompt_influence: 0–1.
• Output codecs: MP3 (22.05/44.1 kHz, 32–192 kbps), PCM (8–48 kHz), μ-law/A-law 8 kHz, Opus 48 kHz (32–192 kbps).

Minimax Music v2.6 minimax_music

MiniMax Music 2.6 creates complete tracks with singing, backing music, and detailed arrangements from a style description and optional lyrics.

Call it via — audio(music) MCP tool · raw: POST /v1/jobs/minimax_music

Cost	30 cr per call
Mode / timeout	webhook / 8m (from our YAML)

Parameters — the model's input schema:

Param	Type	Required	Default	Allowed / range	Description
prompt	string	✓	—	10–2000 chars	Description of the music style, mood, genre, and scenario.
lyrics	string		""	max 3500 chars	Song lyrics. Use \n to separate lines. Supports structure tags: [Intro], [Verse], [Pre Chorus], [Chorus], [Post Chorus], [Hook], [Bridge], [Interlude], [Transition], [Build Up], [Break], [Inst], [Solo], [Outro]. Required when is_instrumental is false.
lyrics_optimizer	boolean		false	true / false	When true and lyrics is empty, auto-generates lyrics from the prompt.
is_instrumental	boolean		false	true / false	When true, generates vocal-free instrumental music.
audio_setting	object		—	see below	Audio configuration settings (object).
audio_setting.sample_rate	integer		44100	16000, 24000, 32000, 44100	Sample rate of generated audio (Hz).
audio_setting.bitrate	integer		256000	32000, 64000, 128000, 256000	Bitrate of generated audio (bps).
audio_setting.format	string		mp3	mp3, wav, pcm	Output audio format.

Our wrapper params (not part of the model schema): out (required — workdir-relative output path, e.g. .mp3), mock (optional — test placeholder). This model has no format_field, so our format wrapper is not used here.

Limits — model limits: prompt 10–2000 characters; lyrics max 3500 characters; output formats mp3 / wav / pcm; sample rate up to 44100 Hz; bitrate up to 256000 bps. Lyrics are required when is_instrumental is false.

Audio Concat audio_concat

Field	Value
Category	audio_process
Mode	sync
Timeout	30s
Cost	Free (cost_per_unit: 0)
Handler	execAudioConcat → AudioConcat (internal/ffmpeg/audio_concat.go)
MCP route	audio(action: "concat") — maps the tool's tracks[] arg to the model's files field

Description: Concatenate multiple audio files in order. Accepts a mix of input formats — every input is decoded and re-encoded to the target output format, then joined with ffmpeg's concat demuxer (-c copy, no second re-encode).

Parameters (from YAML input_schema, cross-checked against handler):

Param	Type	Required	Default	Notes
files	array of string	yes	—	Ordered list of audio paths (any mix of mp3/wav/aac/flac/ogg). Handler errors if empty; non-string entries rejected.
out	string	yes	—	Output audio path.
silence_between	number	no	0	Seconds of silence inserted between files (not after the last). Implemented via generated anullsrc mono 44.1 kHz segments.
output_format	string	no	inferred from out ext, else mp3	enum: mp3, aac, wav, flac, ogg. Read by handler ✓.
sample_rate	integer	no	source rate	Target Hz; applied via -ar. Read by handler ✓.

Behaviour notes:

• Single-file fast path: with one file and silence_between <= 0, if input/output extensions match and no sample_rate is given, it byte-copies the file (acts as a pass-through). Otherwise it delegates to AudioConvert — i.e. a single file makes this a format converter.
• Codec mapping (via outputCodecArgs): wav→pcm_s16le, flac→flac, ogg→libvorbis 192k, aac→aac 192k, default→libmp3lame 192k.
• Concat-list injection is guarded: a file path containing a quote or newline is rejected.
• Returns outputs.audio / outputs.local_path plus metrics (num_files, total_duration_sec, silence_between).

---

Audio-Only Mix audio_only_mix

Field	Value
Category	audio_process
Mode	sync
Timeout	2m
Cost	Free (cost_per_unit: 0)
Handler	execAudioOnlyMix → AudioOnlyMix (internal/ffmpeg/audio_only_mix.go)
MCP route	audio(action: "mix") — passes tracks[] (and the optional music / music_level) through

Description: Mix audio files into a single audio file. Two modes: a flat mix of 2+ tracks with ffmpeg's amix filter, or — when the optional music bed is set — a music-under-voice mix where tracks are the primary program (1+ allowed) and the bed is auto-fit to their length and ducked under them. Unlike video_audio_mix (which overlays audio onto a video), this produces a pure audio file with no video track.

Parameters:

Param	Type	Required	Default	Notes
tracks	array of string	yes	—	Audio paths. Flat mix: min 2, all at equal level. With music: the primary program (e.g. voiceover), min 1.
music	string	no	—	Optional background music bed. When set, the bed is auto-fit to the tracks' length (trimmed if longer, looped if shorter) and ducked under them.
music_level	number	no	-18	Music bed level in dB relative to the voice (used only with music).
out	string	yes	—	Output audio path.

Behaviour notes (code-only, not exposed as params):

• Flat mix: all tracks are mixed at equal levels; output is normalized (amix=...:normalize=1) to prevent clipping; output duration equals the longest input.
• Music-under-voice: the bed never runs past the voice and never drowns it (ducked at music_level dB).
• Output is forced to stereo (-ac 2).
• For per-layer volume / timing offsets onto a video, use video_audio_mix instead.

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Audio Trim audio_trim

Field	Value
Category	audio_process
Mode	sync
Timeout	1m
Cost	Free (cost_per_unit: 0)
MCP route	audio(action: "trim") — maps the tool's audio arg to the model's in field

Description: Cut an audio file to a start time and optional duration — e.g. shorten a long music bed before mixing, or drop a lead-in/lead-out. Output timestamps are rebased to 0, so the result is a clean seekable clip.

Parameters:

Param	Type	Required	Default	Notes
in	string	yes	—	Input audio path (the MCP trim action's audio argument).
out	string	yes	—	Output audio path.
start_sec	number	no	0	Where the kept window starts, in seconds (≥ 0).
duration_sec	number	no	—	Length of the kept window. Omit (or ≤ 0) to keep everything from start_sec to the end.

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Audio Convert audio_convert

Field	Value
Category	audio_process
Mode	sync
Timeout	30s
Cost	Free (cost_per_unit: 0)
Handler	execAudioConvert → AudioConvert (internal/ffmpeg/audio_convert.go)
MCP route	None — internal-only (REST POST /v1/jobs/audio_convert or pipeline step). No audio(...) action routes here.

Description: Convert an audio file between formats, change sample rate, and/or adjust bitrate. Input format is auto-detected; output is chosen by the format key (see mismatch below) or inferred from the out extension.

Parameters (from YAML — see mismatch flag):

Param	Type	Required	Default	Notes
in	string	yes	—	Input audio path.
out	string	yes	—	Output audio path; format inferred from extension if no format key set.
output_format	string	no	inferred from out ext	enum: mp3, mp3_128, mp3_320, aac, aac_256, wav, wav_48k, flac, ogg, opus. ⚠ See mismatch.
sample_rate	integer	no	original	Target Hz (e.g. 44100, 48000); applied via -ar. Read by handler ✓.

⚠ YAML ↔ handler mismatch (important): The YAML declares the format selector as output_format, but execAudioConvert reads inputs["format"] (executor.go:250), not output_format. Consequences:

• A caller passing output_format exactly as the YAML documents will have it silently ignored; the handler falls back to inferring the format from the out file extension.
• The extended enum values that have no matching extension — mp3_128, mp3_320, aac_256, wav_48k, opus — are only reachable by passing the undocumented key format (e.g. format: "mp3_320"). Format/bitrate table (handler audioCodecs): mp3=192k, mp3_128=128k, mp3_320=320k, aac=192k, aac_256=256k, wav/wav_48k=pcm_s16le (wav_48k forces -ar 48000), flac=lossless, ogg=libvorbis 192k, opus=libopus 128k.
• Recommendation: either rename the YAML field to format, or update the handler to also read output_format (as audio_concat does), or have the MCP/handler alias the two keys.

Behaviour notes: Unknown format → error listing valid keys. Returns outputs.audio / outputs.local_path plus metrics (input_duration_sec, output_duration_sec, format, codec).

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Audio Tail Fade tail_fade

Field	Value
Category	audio_process
Mode	sync
Timeout	30s
Cost	Free (cost_per_unit: 0)
Handler	execTailFade → TailFade (internal/ffmpeg/tail_fade.go)
MCP route	None — internal-only (REST POST /v1/jobs/tail_fade or pipeline step). No audio(...) action routes here.

Description: Add a silence pad and a fade-out at the end of an audio file to prevent an abrupt ending (the "audio cuts off" bug). Intended to run after voiceover generation, before assembly. Purely parameter-driven — no prompt.

Parameters:

Param	Type	Required	Default	Notes
in	string	yes	—	Input audio path (workdir-relative).
out	string	yes	—	Output audio path.
pad_sec	number	no	0.8	Seconds of trailing silence added (ffmpeg apad=pad_dur).
fade_sec	number	no	0.6	Fade-out duration (ffmpeg afade=t=out).

Behaviour notes:

• Defaults are applied when the value is <= 0, so passing 0 yields the default (0.8 / 0.6), not a true zero. To disable padding/fade you cannot use this model with 0.
• The fade start point is computed internally as input_duration + 0.1s — it is not a parameter.
• Output encoded with -q:a 2 (VBR ~190 kbps mp3-class quality, format from out ext).
• Returns outputs.audio / outputs.local_path plus metrics (input_duration_sec, output_duration_sec, pad_sec, fade_sec, fade_start_sec).