# Gradium

> Text-to-speech with GradiumTTSService, over Gradium's WebSocket API.

Source: https://polvallverdu.github.io/pipedog/docs/services/tts/gradium/



## Overview [#overview]

* **`GradiumTTSService`**: streams text over a WebSocket to Gradium and pushes 48 kHz audio as it arrives.

<Cards>
  <Card title="Upstream page" href="https://docs.pipecat.ai/api-reference/server/services/tts/gradium">
    Pipecat's documentation for the same service.
  </Card>

  <Card title="Source" href="https://github.com/polvallverdu/pipedog/tree/main/packages/gradium/src">
    `packages/gradium/src/tts.ts`.
  </Card>

  <Card title="Gradium docs" href="https://docs.gradium.ai/api-reference/endpoint/tts-websocket">
    The text-to-speech WebSocket reference.
  </Card>
</Cards>

## Installation [#installation]

```bash
bun add @pipedog/gradium
```

```ts
import { GradiumTTSService } from "@pipedog/gradium/tts";
```

## Prerequisites [#prerequisites]

* A Gradium account and an API key from [Gradium](https://gradium.ai/).
* Pass the key as `apiKey`. The examples read it from `GRADIUM_API_KEY`.

## GradiumTTSService [#gradiumttsservice]

Connects to `wss://api.gradium.ai/api/speech/tts`, with the key in the `x-api-key` header.

### Configuration [#configuration]

| Option | Type | Required | Description |
|---|---|---|---|
| `apiKey` | `string` | yes | Gradium API key for authentication. |
| `url` | `string \| undefined` | no | Gradium websocket API endpoint. Defaults to "wss://api.gradium.ai/api/speech/tts". |
| `jsonConfig` | `string \| null \| undefined` | no | Optional JSON configuration string for additional model settings. |
| `settings` | `GradiumTTSSettings \| null \| undefined` | no | Runtime-updatable settings. |
| `name` | `string \| null \| undefined` | no | Custom name; defaults to `ClassName#count`. |
| `taskManager` | `BaseTaskManager \| null \| undefined` | no | Task manager for handling tasks. |
| `enableDirectMode` | `boolean \| undefined` | no | Process frames immediately instead of using internal queues. |
| `metrics` | `FrameProcessorMetrics \| null \| undefined` | no | Optional metrics collector for this processor. |
| `reconnectBackoffMinWait` | `number \| undefined` | no | Minimum time, in seconds, to wait between reconnection attempts. |
| `reconnectBackoffMaxWait` | `number \| undefined` | no | Maximum time, in seconds, to wait between reconnection attempts. |
| `reconnectOnError` | `boolean \| undefined` | no | Whether to automatically reconnect on connection errors. |
| `wsCloseTimeout` | `number \| undefined` | no | Maximum time, in seconds, to wait for the peer to acknowledge the websocket closing handshake before dropping the connection. Applied to connections opened through `_websocketConnect()`. Increase it for peers that need longer to complete a graceful close. |
| `wsSendTimeout` | `number \| null \| undefined` | no | Maximum time, in seconds, a single send may wait on a peer that has stopped reading before the connection is dropped and the send throws `ConnectionClosedError`. Applied to connections opened through `_websocketConnect()`. Pass `null` to wait indefinitely. Defaults to `WS_SEND_TIMEOUT`. |
| `textAggregationMode` | `TextAggregationMode \| null \| undefined` | no | How to aggregate incoming text before synthesis. `TextAggregationMode.SENTENCE` (default) buffers until sentence boundaries, `TextAggregationMode.TOKEN` streams tokens directly for lower latency. |
| `pushTextFrames` | `boolean \| undefined` | no | Whether to push TextFrames and LLMFullResponseEndFrames. |
| `pushStopFrames` | `boolean \| undefined` | no | Whether to automatically push TTSStoppedFrames. |
| `pushStartFrame` | `boolean \| undefined` | no | Whether to automatically create audio contexts and push TTSStartedFrames. When true, the base class handles `createAudioContext` and yields `TTSStartedFrame` before each synthesis call, so `runTts` implementations do not need to. |
| `stopFrameTimeoutS` | `number \| undefined` | no | Idle time before pushing TTSStoppedFrame when pushStopFrames is true. |
| `pushSilenceAfterStop` | `boolean \| undefined` | no | Whether to push silence audio after TTSStoppedFrame. |
| `silenceTimeS` | `number \| undefined` | no | Duration of silence to push when pushSilenceAfterStop is true. |
| `pauseFrameProcessing` | `boolean \| undefined` | no | Whether to hold incoming frames after a turn until its audio contexts end. Needed by services whose audio doesn't carry its request's context ID. |
| `maxConsecutiveZeroAudioContexts` | `number \| undefined` | no | How many consecutive TTS contexts may complete without producing any audio before the service is reported unable to do its job. Catches a provider that accepts requests and stays silent — an unknown voice ID, say — which no error ever surfaces. Every silent context reports an error the service can carry on from, whatever the limit is; on reaching the limit the service reports a permanent error instead, stops being given work, and the pipeline worker applies its `ProcessorUnusablePolicy`. Set to 0 to report silent contexts without ever writing the service off. |
| `appendTrailingSpace` | `boolean \| undefined` | no | Whether to append a trailing space to text before sending to TTS. This helps prevent some TTS services from vocalizing trailing punctuation (e.g., "dot"). Only applied in sentence aggregation mode; when streaming tokens, the incoming text's own whitespace is preserved. |
| `skipTextTypes` | `string[] \| null \| undefined` | no | List of text types that should not be spoken. |
| `textTransforms` | `[string, TextTransform][] \| null \| undefined` | no | A list of TextTransforms to transform text just before sending it to TTS. Each TextTransform takes the aggregated text and its type, and returns the transformed text. To register, provide a list of tuples of `[textType \| "*", transformFunction]`. |
| `transportDestination` | `string \| null \| undefined` | no | Destination for generated audio frames. |
| `reuseContextIdWithinTurn` | `boolean \| undefined` | no | Whether the service should reuse context IDs within the same turn. |
| `audioCache` | `TTSAudioCacheStore \| null \| undefined` | no | Where to keep synthesized audio for reuse (e.g. a shared `TTSAudioCache`). Text whose audio is in the cache is served from it, with no request to the provider. Lines get there through `warmAudioCache()`, and also from live synthesis when `cacheSynthesizedAudio` allows it. Only services that yield their audio from `runTts` (HTTP and streaming-RPC services, see `supportsAudioCache`) use it, and only in sentence aggregation mode. Defaults to null (no cache). pipedog only. |
| `cacheSynthesizedAudio` | `boolean \| ((text: string) => boolean) \| undefined` | no | Whether audio synthesized live is stored in `audioCache`, or a function that decides per text. Defaults to false: only warmed lines are stored, since a generated reply can repeat what the user said. pipedog only. |
| `audioCacheNamespace` | `string \| undefined` | no | Extra part of the cache key, for constructor options that change the audio but are not in `settings`. pipedog only. |

### Settings [#settings]

Runtime-updatable settings, passed as `settings: new GradiumTTSService.Settings({...})`
(`GradiumTTSSettings`). Any setting you leave out keeps the default below.

| Setting    | Type               | Default              | Description                                  |
| ---------- | ------------------ | -------------------- | -------------------------------------------- |
| `model`    | `string`           | `"default"`          | Speech model.                                |
| `voice`    | `string`           | `"_6Aslh2DxfmnRLmP"` | Voice id. Changing it reconnects the socket. |
| `language` | `Language \| null` | `null`               | Language.                                    |

### Usage [#usage]

```ts title="tts.ts"
import { GradiumTTSService } from "@pipedog/gradium/tts";

const tts = new GradiumTTSService({
  apiKey: process.env.GRADIUM_API_KEY ?? "",
  settings: new GradiumTTSService.Settings({ voice: "_6Aslh2DxfmnRLmP" }),
});
```

### Notes [#notes]

* The sample rate is fixed at 48000 Hz.
* `jsonConfig` takes a JSON string of extra model settings, sent with the setup message.

| Runtime            | Works | What it needs |
| ------------------ | ----- | ------------- |
| Bun                | ✅ yes | Nothing.      |
| Node.js            | ✅ yes | Nothing.      |
| Deno               | ✅ yes | Nothing.      |
| Cloudflare Workers | ✅ yes | Nothing.      |

Generated from `pipecat.runtimes` and `pipecat.compat` in `packages/gradium/package.json`: ✅ means the compat smoke test passes on that runtime.

## Runtime notes [#runtime-notes]

* The WebSocket goes through the core `websocketConnect`, with headers on the handshake.
* The deprecated `InputParams`, `params`, `voice_id` and `model` constructor options are not ported.
* Tested against a local WebSocket server (upstream has no Gradium TTS test), not against Gradium itself.