Translation Settings Breakdown

A complete reference for all configuration options of the set_task message.

What are translation settings?

Translation settings control how your audio is processed, transcribed, translated, and synthesized. They are included in the data field of a set_task message sent via the WebRTC data channel or the WebSocket connection.

You will use these settings when:

• Starting a new translation — send your initial set_task message to begin real-time translation.
• Updating an active translation — modify settings during an ongoing translation without interrupting the stream.
• Fine-tuning performance — adjust parameters to optimize quality, latency, or specific use cases.

For ready-to-use values, see Recommended settings.

Message envelope

{
  "message_type": "set_task",
  "data": {
    "input_stream":  { /* ... */ },
    "output_stream": { /* ... */ },
    "pipeline":      { /* ... */ }
  }
}

message_type

• Type: string
• Description: Identifies the type of command being sent:
  • "set_task" — create a new task or update an existing one.
  • "get_task" — return the current task.
  • "pause_task" — stop processing new audio but keep the task alive. Use set_task to resume.
  • "flush_task" — cancel processing of the current (ongoing) phrase.
  • "end_task" — end the task (you will be disconnected automatically).
  • "tts_task" — generate text-to-speech from text.
  • "input_audio_data" — WebSocket transport only; contains an audio data chunk.

data

• Type: object
• Description: The main task configuration. Contains five major sections:
  1. input_stream
  2. output_stream
  3. pipeline
  4. translation_queue_configs
  5. allowed_message_types

---

input_stream

Configures the input stream.

Field	Type	Description
content_type	string	Content type of the input stream. Currently only audio is supported.
source	object	How and from where the input audio is sourced. See below.

source

Field	Type	Description
type	string	Input audio transport: webrtc or ws. Must match the target transport in output_stream.
format	string	WebSocket transport only. Input audio format: opus, pcm_s16le, or wav.
sample_rate	integer	WebSocket transport only. Input sample rate. Allowed range: 16000–48000 Hz.
channels	integer	WebSocket transport only. Number of input channels: 1 or 2.

---

output_stream

Configures the output stream.

Field	Type	Description
content_type	string	Content type of the output. Currently only audio is supported.
target	object	Destination of the output. See below.

target

Field	Type	Description
type	string	Output audio transport: webrtc or ws. Must match the source transport in input_stream.
format	string	WebSocket transport only. Output audio format: pcm_s16le or zlib_pcm_s16le (zlib-compressed PCM).
sample_rate	integer	WebSocket transport only. Output sample rate. Fixed at 24000 Hz.
channels	integer	WebSocket transport only. Output channels. Fixed at 1 (mono).

---

pipeline

Holds the configuration for all processing steps: preprocessing, transcription, and translation.

transcription

Settings for Automatic Speech Recognition (ASR).

source_language

• Type: string
• Description: Language code of the input audio (e.g., "en", "es", "fr"). Set to "auto" to enable automatic language detection (optionally restricted by detectable_languages). See Supported languages.

detectable_languages

• Type: array of strings
• Description: When source_language is "auto", only languages from this list will be detected. Leave empty to allow any supported language.

segment_confirmation_silence_threshold

• Type: float
• Description: Seconds of silence needed to confirm the end of a segment (0.3–2.0, default 0.7). Recommended range: 0.5–0.9 s, depending on the speaker's tempo and pauses. Increase it if the speaker frequently pauses between words; setting it too low can cause unwanted sentence splitting.

speakers_total

• Type: integer or null
• Description: Expected number of speakers (1–1000). Helps speaker handling when known in advance.

only_confirm_by_silence

• Type: bool
• Description: When true, segments are confirmed only by silence detection.

sentence_splitter

• Type: object
• Description: Controls how longer sentences are split into smaller parts (sometimes with slight rephrasing, without losing the meaning) to speed up processing.

Field	Type	Description
enabled	bool	Whether to enable automatic sentence splitting.

verification

• Type: object
• Description: Transcription verification settings.

Field	Type	Description
auto_transcription_correction	bool	WIP. Enables automatic transcription verification using an LLM.
transcription_correction_style	string or null	Style of the LLM correction.

translations

• Type: array of objects
• Description: An array of translation targets. Each object defines translation settings for one target language; add one object per target language.

target_language

• Type: string
• Description: Language to translate into (e.g., "en-us", "es", "fr"). See Supported languages.

allowed_source_languages

• Type: array of strings
• Description: Restricts this translation target to specific source languages. Used for conditional multi-language translation together with "source_language": "auto".

translate_partial_transcriptions

• Type: bool
• Description: Enables translation of partial (unconfirmed) transcriptions.

speech_generation

• Type: object
• Description: Text-to-speech (TTS) settings for this target language.

Field	Type	Description
voice_cloning	bool	Experimental. Mimics the original speaker's voice. It usually takes 10–20 seconds of speech before the voice changes are applied.
voice_id	string or null	A specific voice ID (voice cloning must be disabled). "default_low" or "default_high" automatically picks the best default voice for the language. Manage voices in the Palabra web portal.
voice_timbre_detection	object	Automatically detects voice timbre and assigns voice IDs accordingly. See below.

voice_timbre_detection

Field	Type	Description
enabled	bool	Enables voice timbre detection (voice cloning must be disabled).
high_timbre_voices	array of strings	Voice ID to use for high-timbre voices. Currently only one ID is supported; "default_high" can be used.
low_timbre_voices	array of strings	Voice ID to use for low-timbre voices. Currently only one ID is supported; "default_low" can be used.

---

translation_queue_configs

Configures the behavior of unspoken TTS buffers.

The global key holds the default settings. You can add language-specific overrides by using the language code as a key (for example, "es" for Spanish).

Field	Type	Description
desired_queue_level_ms	integer	Desired average TTS buffer size in milliseconds (2000–163840). With auto_tempo enabled, the system tries to keep the buffer at this level. Recommended: 5000–10000 ms.
max_queue_level_ms	integer	Maximum TTS queue size in milliseconds (3000–163840). If the queue grows beyond this limit, it is reduced to desired_queue_level_ms by dropping older queued audio. Must be greater than desired_queue_level_ms; should be at least 2–3× larger.
auto_tempo	bool	Automatically corrects speech tempo based on the queue state. Recommended to keep on.
auto_tempo_max_delay_ms	integer	Maximum buffer delay in milliseconds for auto tempo (60–10000, default 250).
min_tempo	float	Minimum allowed speech speed (1.0–2.0, default 1.0).
max_tempo	float	Maximum allowed speech speed (1.0–2.0, default 1.35). Must be ≥ min_tempo.

If you don't provide translation_queue_configs, the server applies a default global config: desired_queue_level_ms: 5000, max_queue_level_ms: 20000, min_tempo: 1.15, max_tempo: 1.45, auto_tempo: true.

---

allowed_message_types

• Type: array of strings
• Default: ["translated_transcription", "partial_transcription", "validated_transcription"]
• Description: Specifies which message types you will receive over the WebSocket. The same messages are also sent in the WebRTC data channel.
  • "partial_transcription" — emitted for partial transcription segments as they are recognized.
  • "partial_translated_transcription" — emitted for partial translated transcriptions if translate_partial_transcriptions is enabled.
  • "validated_transcription" — emitted when a transcription segment is fully confirmed.
  • "translated_transcription" — emitted when a transcription segment has been translated.

---

See also: Translation management API · Publishing and receiving audio