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.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"

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

Capability matrix

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.