Merge pull request #3697 from pipecat-ai/mb/soniox-rt-4
Update SonioxSTTService default model to stt-rt-v4
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@@ -93,7 +93,7 @@ class SonioxInputParams(BaseModel):
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client_reference_id: Client reference ID to use for transcription.
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"""
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model: str = "stt-rt-preview"
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model: str = "stt-rt-v4"
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audio_format: Optional[str] = "pcm_s16le"
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num_channels: Optional[int] = 1
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@@ -152,7 +152,7 @@ class SonioxSTTService(WebsocketSTTService):
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url: str = "wss://stt-rt.soniox.com/transcribe-websocket",
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sample_rate: Optional[int] = None,
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params: Optional[SonioxInputParams] = None,
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vad_force_turn_endpoint: bool = False,
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vad_force_turn_endpoint: bool = True,
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ttfs_p99_latency: Optional[float] = SONIOX_TTFS_P99,
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**kwargs,
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):
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@@ -164,7 +164,8 @@ class SonioxSTTService(WebsocketSTTService):
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sample_rate: Audio sample rate.
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params: Additional configuration parameters, such as language hints, context and
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speaker diarization.
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vad_force_turn_endpoint: Listen to `VADUserStoppedSpeakingFrame` to send finalize message to Soniox. If disabled, Soniox will detect the end of the speech.
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vad_force_turn_endpoint: Listen to `VADUserStoppedSpeakingFrame` to send finalize message to Soniox.
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If disabled, Soniox will detect the end of the speech. Defaults to True.
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ttfs_p99_latency: P99 latency from speech end to final transcript in seconds.
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Override for your deployment. See https://github.com/pipecat-ai/stt-benchmark
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**kwargs: Additional arguments passed to the STTService.
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