Streaming

Murf TTS API supports real-time streaming capabilities, allowing developers to generate and play text-to-speech (TTS) audio dynamically as it is being generated in real-time, reducing the time-to-first-byte. This ensures minimal latency, making it ideal for conversational AI, real-time applications, and voice-enabled assistants.

In addition to HTTP streaming endpoint, Murf TTS will soon support Websocket streaming which enables bidirectional streaming for real-time audio generation.

Quickstart

Streaming enables returning raw audio bytes (e.g., MP3 data) directly over HTTP using chunked transfer encoding. This allows clients to process or play audio incrementally as it is generated. This section focuses on how streaming works for requests made to the Text to Speech API.

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# pip install murf
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from murf import Murf
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client = Murf(
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    api_key="YOUR_API_KEY", # Not required if you have set the environment variable MURF_API_KEY
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)
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res = client.text_to_speech.stream(
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    text="Hi, how can I help you this fine day?",
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    voice_id="en-US-natalie"
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)
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for audio_chunk in res:
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  with open("stream_output.wav", "ab") as f:
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    f.write(audio_chunk)

FAQs