Skip to navigation
APIsLegacySpeech to Text Translate

WebSocket

View as Markdown

WebSocket channel for real-time speech to text streaming with English translation.

Note: This API Reference page is provided for informational purposes only. The Try It playground may not provide the best experience for streaming audio. For optimal streaming performance, please use the SDK or implement your own WebSocket client.

Handshake

WSS
wss://api.sarvam.ai/speech-to-text-translate/ws

Headers

Api-Subscription-KeystringRequired
API subscription key for authentication

Query parameters

modelenumOptionalDefaults to saaras:v2.5

Model to be used for speech to text translation.

  • saaras:v2.5 (default): Translation model that translates audio from any spoken Indic language to English.
    • Example: Hindi audio → English text output

For the latest model (saaras:v3), use the /speech-to-text endpoint with mode="translate".

Allowed values:
sample_rateenumOptionalDefaults to 16000
Audio sample rate for the WebSocket connection. When specified as a connection parameter, only 16kHz and 8kHz are supported. 8kHz is only available via this connection parameter. If not specified, defaults to 16kHz.
Allowed values:
high_vad_sensitivityenumOptional

Enable high VAD (Voice Activity Detection) sensitivity

Allowed values:
positive_speech_thresholdstringOptionalDefaults to 0.7

VAD probability threshold (0.0–1.0) above which a frame is considered speech. Overrides the server default when provided.

negative_speech_thresholdstringOptionalDefaults to 0.45

VAD probability threshold (0.0–1.0) below which a frame is considered silence. Overrides the server default (or the high_vad_sensitivity preset) when provided.

min_speech_framesstringOptionalDefaults to 2
Minimum number of consecutive speech frames required to start a speech segment. Overrides the server default when provided.
first_turn_min_speech_framesstringOptionalDefaults to 8
Minimum speech frames required specifically for the first user turn. Overrides the server default when provided.
negative_frames_countstringOptionalDefaults to 18

Number of negative (silence) frames needed within the window to end a speech segment. Overrides the server default (or the high_vad_sensitivity preset) when provided.

negative_frames_windowstringOptionalDefaults to 24

Sliding window size (in frames) over which negative frames are counted. Overrides the server default (or the high_vad_sensitivity preset) when provided.

start_speech_volume_thresholdstringOptional

Volume level (dB) below which audio is considered too quiet to be speech. When not provided, no volume-based filtering is applied.

interrupt_min_speech_framesstringOptionalDefaults to 2

Minimum speech frames required to register a barge-in / interruption. Overrides the server default when provided.

pre_speech_pad_framesstringOptionalDefaults to 9
Number of audio frames to prepend before the detected speech onset, ensuring the beginning of speech is not clipped. Overrides the server default when provided.
num_initial_ignored_framesstringOptionalDefaults to 0
Number of leading audio frames to skip entirely at connection start. Useful for discarding connection setup noise. Overrides the server default when provided.
vad_signalsenumOptional
Enable VAD signals in response
Allowed values:
flush_signalenumOptional
Signal to flush the audio buffer and finalize transcription and translation
Allowed values:
input_audio_codecenumOptional

Audio codec/format of the input stream. Use this when sending raw PCM audio. Supported values: wav, pcm_s16le, pcm_l16, pcm_raw.

Allowed values:

Send

Audio Translation MessageobjectRequired
Send audio data for real-time speech to text streaming with translation
OR
Translation Config MessageobjectRequired
Send configuration for speech to text streaming with translation
OR
Speech Translate Flush SignalobjectRequired
Send signal to flush audio buffer and finalize transcription and translation

Receive

TranslationobjectRequired
Receive real-time transcription and translation results from the WebSocket