a best effort version of the bot's output
- The `RTVIObserver` now emits `bot-output` messages based off
the new `AggregatedTextFrame`s (`bot-tts-text` and
`bot-llm-text` are still supported and generated, but
`bot-transcript` is now deprecated in lieu of this new, more
thorough, message).
- The new `RTVIBotOutputMessage` includes the fields:
- `spoken`: A boolean indicating whether the text was spoken by TTS
- `aggregated_by`: A string representing how the text was aggregated
("sentence", "word", "my custom aggregation")
- Introduced new fields to `RTVIObserver` to support the new
`bot-output` messaging:
- `bot_output_enabled`: Defaults to True. Set to false to disable
bot-output messages.
- `skip_aggregator_types`: Defaults to `None`. Set to a list of
strings that match aggregation types that should not be included
in bot-output messages. (Ex. `credit_card`)
`CartesiaTTSService`:
- Modified use of custom default text_aggregator to avoid deprecation warnings and push users
towards use of transformers or the `LLMTextProcessor`
- Added convenience methods for taking advantage of Cartesia's SSML tags: spell, emotion,
pauses, volume, and speed.
`RimeTTSService`:
- Modified use of custom default text_aggregator to avoid deprecation warnings and push users
towards use of transformers or the `LLMTextProcessor`
- Added convenience methods for taking advantage of Rime's customization options: spell,
pauses, pronunciations, and inline speed control.
Introduced `LLMTextProcessor`: A new processor meant to allow customization for how
LLMTextFrames should be aggregated and considered. It's purpose is to turn
`LLMTextFrame`s into `AggregatedTextFrame`s. By default, a TTSService will still
aggregate `LLMTextFrame`s by sentence for the service to consume. However, if you
wish to override how the llm text is aggregated, you should no longer override the
TTS's internal text_aggregator, but instead, insert this processor between your LLM
and TTS in the pipeline.
This frame introduces an `aggregated_by` field to describe the type of text included
in the frame and allows unspoken groupings of text to be pushed through the pipeline
and treated similar to TTSTextFrames.
Adding support for setting whether or not the text in the TextFrame
should be added to the LLM context (by the LLM assistant aggregator).
Defaults to `True`.
Modified the BaseTextAggregator type so that when text gets aggregated, metadata can
be associated with it. Currently, that just means a `type`, so that the aggregation
can be classified or described. Changes made to support this:
- **IMPORTANT**: Aggregators are now expected to strip leading/trailing white space
characters before returning their aggregation from `aggregation()` or `.text`. This
way all aggregators have a consistent contract allowing downstream use to know how
to stitch aggregations back together
- Introduced a new `Aggregation` dataclass to represent both the aggregated `text` and
a string identifying the `type` of aggregation (ex. "sentence", "word", "my custom
aggregation")
- **BREAKING**: `BaseTextAggregator.text` now returns an `Aggregation` (instead of `str`).
To update: `aggregated_text = myAggregator.text` -> `aggregated_text = myAggregator.text.text`
- **BREAKING**: `BaseTextAggregator.aggregate()` now returns `Optional[Aggregation]`
(instead of `Optional[str]`). To update:
```
aggregation = myAggregator.aggregate(text)
if (aggregation):
print(f"successfully aggregated text: {aggregation.text}") // instead of {aggregation}
```
- `SimpleTextAggregator`, `SkipTagsAggregator`, `PatternPairAggregator` updated to
produce/consume `Aggregation` objects.
- All uses of the above Aggregators have been updated accordingly.
This update ensures that audio is flushed immediately after sending bare text to the WebSocket, improving the responsiveness of the Text-to-Speech service.
This commit introduces the RimeNonJsonTTSService class, enabling Text-to-Speech synthesis over WebSocket endpoints that require plain text messages. The service includes configuration parameters for language, segmentation, and audio settings, and handles WebSocket connections for raw audio byte transmission. Limitations include the lack of support for word-level timestamps and context IDs.
