feat(cartesia): align WebSocket TTS with latest API and buffering guidance
- Bump default cartesia_version to 2026-03-01. - Replace deprecated use_original_timestamps with use_normalized_timestamps so word timestamps match what was actually spoken. - Add max_buffer_delay_ms init arg; auto-derive 0 in SENTENCE mode to avoid the doc-warned "middle ground" of client + server buffering, leave unset in TOKEN mode for managed buffering. - Silently consume flush_done messages now emitted per transcript when server-side buffering is disabled.
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@@ -232,12 +232,13 @@ class CartesiaTTSService(WebsocketTTSService):
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*,
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api_key: str,
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voice_id: str | None = None,
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cartesia_version: str = "2025-04-16",
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cartesia_version: str = "2026-03-01",
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url: str = "wss://api.cartesia.ai/tts/websocket",
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model: str | None = None,
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sample_rate: int | None = None,
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encoding: str = "pcm_s16le",
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container: str = "raw",
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max_buffer_delay_ms: int | None = None,
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params: InputParams | None = None,
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settings: Settings | None = None,
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text_aggregation_mode: TextAggregationMode | None = None,
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@@ -263,6 +264,12 @@ class CartesiaTTSService(WebsocketTTSService):
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sample_rate: Audio sample rate. If None, uses default.
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encoding: Audio encoding format.
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container: Audio container format.
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max_buffer_delay_ms: Server-side buffering window before generation
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starts. ``0`` disables server buffering (custom buffering); any
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value in (0, 5000] enables managed buffering. If ``None``,
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derived from ``text_aggregation_mode``: ``0`` for ``SENTENCE``
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(avoids stacking client and server buffering), unset for
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``TOKEN`` (uses Cartesia's 3000ms default).
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params: Additional input parameters for voice customization.
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.. deprecated:: 0.0.105
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@@ -353,6 +360,15 @@ class CartesiaTTSService(WebsocketTTSService):
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self._output_encoding = encoding
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self._output_sample_rate = 0 # Set in start() from self.sample_rate
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# Cartesia warns against the "middle ground" of client-side sentence
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# aggregation plus the server's default 3000ms buffer. When the user
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# doesn't pick a value, send 0 in SENTENCE mode (custom buffering) and
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# leave it unset in TOKEN mode so the server default applies (managed
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# buffering).
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if max_buffer_delay_ms is None and not self._is_streaming_tokens:
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max_buffer_delay_ms = 0
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self._max_buffer_delay_ms = max_buffer_delay_ms
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self._receive_task = None
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def can_generate_metrics(self) -> bool:
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@@ -466,9 +482,12 @@ class CartesiaTTSService(WebsocketTTSService):
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"sample_rate": self._output_sample_rate,
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},
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"add_timestamps": add_timestamps,
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"use_original_timestamps": False if self._settings.model == "sonic" else True,
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"use_normalized_timestamps": True,
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}
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if self._max_buffer_delay_ms is not None:
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msg["max_buffer_delay_ms"] = self._max_buffer_delay_ms
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if self._settings.language:
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msg["language"] = self._settings.language
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@@ -647,6 +666,13 @@ class CartesiaTTSService(WebsocketTTSService):
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await self.stop_all_metrics()
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await self.push_error(error_msg=f"Error: {msg}")
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self.reset_active_audio_context()
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elif msg["type"] == "flush_done":
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# Cartesia emits flush_done as a per-transcript boundary marker
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# within a context (e.g. when max_buffer_delay_ms=0 causes the
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# server to flush each submission). We don't need it: each turn
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# already has its own context_id and audio chunks are tagged
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# with it. Acknowledge silently.
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pass
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else:
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await self.push_error(error_msg=f"Error, unknown message type: {msg}")
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