Merge pull request #3697 from pipecat-ai/mb/soniox-rt-4
Update SonioxSTTService default model to stt-rt-v4
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changelog/3697.changed.2.md
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- Update `SonioxSTTService` to set `vad_force_turn_endpoint` to `True`. This setting disabled the turn detection logic available natively in Soniox. Instead, Soniox relies on a local VAD to finalize the transcript. This configuration meaningfully reduces the time to final segment for Soniox. With this setting enabled, Soniox outputs a transcript in ~250ms (median). Pipecat enables smart-turn detection by default using the `LocalSmartTurnAnalyzerV3`. To use the native turn detection logic in Soniox, just set `vad_force_turn_endpoint` to `False`.
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changelog/3697.changed.md
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changelog/3697.changed.md
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- Update `SonioxSTTService` default model to `stt-rt-v4`.
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@@ -24,7 +24,8 @@ from pipecat.runner.types import RunnerArguments
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from pipecat.runner.utils import create_transport
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from pipecat.services.cartesia.tts import CartesiaTTSService
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from pipecat.services.openai.llm import OpenAILLMService
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from pipecat.services.soniox.stt import SonioxSTTService
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from pipecat.services.soniox.stt import SonioxInputParams, SonioxSTTService
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from pipecat.transcriptions.language import Language
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from pipecat.transports.base_transport import BaseTransport, TransportParams
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from pipecat.transports.daily.transport import DailyParams
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from pipecat.transports.websocket.fastapi import FastAPIWebsocketParams
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@@ -52,6 +53,10 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
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stt = SonioxSTTService(
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api_key=os.getenv("SONIOX_API_KEY"),
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params=SonioxInputParams(
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language_hints=[Language.EN],
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language_hints_strict=True,
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),
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)
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tts = CartesiaTTSService(
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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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