# Streaming TTS

Hermes can stream TTS audio as it arrives from the provider, instead of waiting
for the full audio before playing. This is used by voice mode (CLI/TUI live
conversation), the dashboard speak-stream WebSocket, and — via the gateway
`StreamingTTSConsumer` — any platform adapter that opts into streaming audio.
Voice replies start speaking after the first clause instead of after full
generation + synthesis.

## Architecture

The streaming pipeline has four parts:

1. **Producer** — the LLM emits text deltas as it generates a response
2. **Sentence chunker** — `tools.tts_streaming.SentenceChunker` accumulates
   deltas, strips `<think>` blocks (even split across deltas), and flushes
   complete sentences
3. **TTS provider** — a registered `StreamingTTSProvider` turns each sentence
   into raw PCM chunks (int16 mono at the provider's declared `sample_rate`)
4. **Audio sink** — `sounddevice.OutputStream` for local playback
   (`tools.tts_tool_speaker.stream_tts_to_speaker`), or a gateway platform adapter's
   `write_streaming_tts` seam (`gateway/streaming_tts_consumer.py`)

Providers with no chunked API still get per-*sentence* playback via the proven
sync `text_to_speech_tool` path, so edge (the default) is conversational too.
All spoken text is cleaned by `tools.tts_text_normalize.prepare_spoken_text`
(one cleaner, all paths).

## How to pick a provider

By default the dispatcher streams with the provider you already configured
(`tts.provider`) when that provider has a chunked API — it never silently
swaps your voice for a different provider just to get streaming.

To override, set `tts.streaming.provider` in your `config.yaml`:

- a provider name (`elevenlabs`, `gemini`, `openai`, `xai`) pins that streamer
- `auto` walks the priority list `elevenlabs → gemini → openai → xai` and uses
  the first one whose credentials resolve — an explicit opt-in to "best
  chunked voice available"

```yaml
tts:
  provider: gemini
  streaming:
    provider: gemini      # or "auto"
  gemini:
    model: gemini-2.5-flash-preview-tts
    voice: Kore
```

## Capability matrix

| Provider    | Transport                             | Chunked PCM | Credentials |
|-------------|---------------------------------------|-------------|-------------|
| elevenlabs  | chunked HTTP (`pcm_24000`)            | yes         | `ELEVENLABS_API_KEY` / `tts.elevenlabs` |
| openai      | chunked HTTP (`with_streaming_response`, `pcm`) | yes | `tts.openai.api_key` → env → managed gateway |
| gemini      | SSE (`streamGenerateContent?alt=sse`) | yes         | `GEMINI_API_KEY` / `GOOGLE_API_KEY` |
| xai         | WebSocket (`wss://api.x.ai/v1/tts`)   | yes         | xAI OAuth or `XAI_API_KEY` |
| edge, piper, kitten, neutts, mistral, minimax, deepinfra, … | — | no (per-sentence sync fallback) | as usual |

All credential lookups go through `resolve_provider_secret()`
(config > env/.env > credential pool) — never bare env reads. Streamed bodies
are capped at 16 MiB per sentence, mirroring the sync providers' bounded
upstream-body invariant.

## Adding a new streaming provider

1. Subclass `StreamingTTSProvider` in `tools/tts_streaming.py`
2. Set `sample_rate` (and `channels` / `sample_width` if not int16 mono)
3. Implement `available()` (a pure probe — never install anything) and
   `stream(self, text) -> Iterator[bytes]` yielding raw PCM chunks
4. Decorate with `@register("yourname")`
5. Add tests in `tests/tools/test_tts_streaming.py`

The ABC enforces the contract; the registry makes the provider discoverable;
the dispatcher (`stream_tts_to_speaker`) and the gateway consumer handle the
sentence buffer, stop events, and audio sink for free.

## Gateway streaming (platform adapters)

`gateway/streaming_tts_consumer.py` bridges agent deltas to an adapter's
streaming-audio seam. Adapters opt in by overriding, on
`BasePlatformAdapter`:

- `supports_streaming_tts(chat_id, audio_format) -> bool`
- `begin_streaming_tts / write_streaming_tts / finish_streaming_tts /
  abort_streaming_tts`

All default to unsupported/no-op, so existing adapters are untouched. When a
turn's streaming audio completes, the whole-file auto-TTS reply for that turn
is suppressed (no double playback); when streaming fails before any audio was
audible, the gateway falls back to the legacy whole-file voice reply.
