Migrate from Google Cloud TTS

Google Cloud bidirectional streaming sends one configuration request followed by text input requests over gRPC. Bland uses the same high-level flow over a standard WebSocket. Start with the Realtime TTS Quickstart.

Request mapping

Google Cloud streaming TTS	Bland `/v2/tts/ws`
Google Cloud credentials and project	Bland API key
First `StreamingSynthesisConfig` request	First `init` message
`VoiceSelectionParams.name`	Bland voice UUID
`StreamingSynthesisInput.text`	`speak.text` with a turn `context_id`
End the request iterator	Send `end_of_turn`, then keep the socket for later turns
`StreamingSynthesizeResponse.audio_content`	Binary WebSocket audio frame

for await (const token of llmTextStream) {
  ws.send(
    JSON.stringify({
      type: "speak",
      context_id: turnId,
      text: token,
    }),
  );
}

ws.send(JSON.stringify({ type: "end_of_turn", context_id: turnId }));

Google’s stream ends with the RPC. Bland is designed to stay open across conversational turns. Wait for utterance_end, then reuse the same socket with a new context_id. The Google voice name and language code do not carry over. Select a Bland voice UUID that supports your target language. Configure raw PCM or 8 kHz mu-law during init and update your playback pipeline for raw binary WebSocket frames. If your application uses Google’s synchronous SynthesizeSpeech with complete text, Synthesize Speech (HTTP) is the closer migration target. See the Google bidirectional streaming quickstart and Bland WebSocket reference.

Docs for agents: llms.txt

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Realtime Speech

HTTP Speech

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Voice Cloning

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Discovery

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Tooling

Migrate from Google Cloud TTS

Request mapping

​Request mapping

Request mapping