--- license: bsd-3-clause language: - en library_name: onnx tags: - pipecat - turn-detection - end-of-turn - voice-agents - streaming - onnx --- # Streaming Turn Streaming, audio-only turn-taking model for [Pipecat](https://github.com/pipecat-ai/pipecat) voice agents. On every 40 ms of user audio it gives the probability that the user's turn is over, and whether speech over the bot is a backchannel or a floor-take. It never sees a transcript. - 4.0M parameters: log-mel, causal conv encoder, GRU. 16 kHz mono in, 25 frames a second out. - 0.37 ms per 40 ms chunk on one Apple M-series core (ONNX Runtime). - English only. Experimental: one training run per version, and little live testing. ## Versions | Folder | End the turn when | |---|---| | `streaming-turn-v0.4` (latest) | `eot` ≥ 0.5, or `eot_late` ≥ 0.4 | | `streaming-turn-v0.2` | `eot` ≥ 0.5, or 0.8 s after the last sound if `eot` reached 0.3 | v0.4 adds `eot_late`, a second end-of-turn output for a user who has gone quiet. Each folder holds `streaming-turn-fp32.onnx` (the streaming model), `model.json` (tensor names, chunk size, class lists) and a `.pt` file (PyTorch config and weights). ## Use with Pipecat `agent.turn_analyzers` is in the code repository, [jptaylor/streaming-turn](https://github.com/jptaylor/streaming-turn). ```python from pathlib import Path from huggingface_hub import snapshot_download from pipecat.audio.vad.silero import SileroVADAnalyzer from pipecat.frames.frames import BotStartedSpeakingFrame, BotStoppedSpeakingFrame from pipecat.processors.aggregators.llm_response_universal import ( LLMContext, LLMContextAggregatorPair, LLMUserAggregatorParams, ) from pipecat.turns.user_stop import TurnAnalyzerUserTurnStopStrategy from pipecat.turns.user_turn_strategies import UserTurnStrategies from agent.turn_analyzers import StreamingTurnAnalyzer, StreamingTurnParams VERSION = "streaming-turn-v0.4" repo = snapshot_download("jptaylor/streaming-turn", allow_patterns=f"{VERSION}/*") analyzer = StreamingTurnAnalyzer( Path(repo) / VERSION, params=StreamingTurnParams(threshold=0.5, late_threshold=0.4), # v0.2: StreamingTurnParams(threshold=0.5, likely_threshold=0.3) ) class StreamingTurnStop(TurnAnalyzerUserTurnStopStrategy): """Tells the model when the bot is speaking: it was trained with that input.""" async def process_frame(self, frame): if isinstance(frame, (BotStartedSpeakingFrame, BotStoppedSpeakingFrame)): analyzer.set_bot_speaking(isinstance(frame, BotStartedSpeakingFrame)) return await super().process_frame(frame) user_aggregator, assistant_aggregator = LLMContextAggregatorPair( LLMContext(), user_params=LLMUserAggregatorParams( vad_analyzer=SileroVADAnalyzer(), user_turn_strategies=UserTurnStrategies(stop=[StreamingTurnStop(turn_analyzer=analyzer)]), ), ) ``` ## Outputs One frame per 40 ms chunk. | Output | | |---|---| | `eot` | probability that the user's turn is over | | `eot_late` | the same, for a user who has gone quiet (v0.4 only) | | `intent` | continuer, assessment, negative, floor_take, non_speech. Backchannel = continuer + assessment; floor-take = negative + floor_take | | `leadin` | none, acknowledgement, hesitation, marker, reaction (barely beats guessing) | | `vad`, `vap` | auxiliary: voice activity now, and both speakers' activity over the next 2 s | Inputs: `chunk` (640 samples, float32) and `bot_state` (is the bot speaking, and for how long), plus the streaming state tensors named in `model.json`. ## Results Held-out Open Yap 1K conversations (1,259 turn ends, 8,191 pauses), replayed through Pipecat's Silero VAD with nobody answering, at the settings above: | | Cuts in on a pause | Answers within 500 ms | Within 1 s | Waits for the 3 s timeout | |---|---|---|---|---| | v0.4 | 4.9% | 56.4% | 79.7% | 2.9% | | v0.2 | 4.6% | 51.4% | 82.2% | 15.5% | On Smart Turn v3.2's English test clips (7,820, read 200 ms after the speech ends): v0.4 93.3%, v0.2 93.8%, Smart Turn v3.2 93.5%. ## Limitations - The held-out conversations are from a corpus the model trained on (different conversations). - Keep `bot_state` up to date. v0.4 holds back until the bot has spoken in a session. - A finished sentence it takes for an unfinished one waits for the 3 s timeout. - Half of the end-of-turn targets on Smart Turn clips are Smart Turn v3.2's own predictions, so it is not independent of v3.2. - Trained on human-to-human conversation and single-speaker clips, with simulated echo. Phone codecs are untested. ## Training data [Open Yap 1K](https://theagenticdatacompany.com/open-yap-1k) v1.0, [otoSpeech-full-duplex-280h](https://huggingface.co/datasets/otoearth/otoSpeech-full-duplex-280h) (CC BY 4.0), the English part of the [Smart Turn v3.2 training data](https://huggingface.co/datasets/pipecat-ai/smart-turn-data-v3.2-train) (CC BY 4.0), and 600 synthetic hard negatives. A frozen [Whisper base.en](https://huggingface.co/openai/whisper-base.en) encoder was the distillation target, and Smart Turn v3.2 gave soft end-of-turn labels. No training data is included here.