Audio formats

All output is mono at 24 000 Hz (sample_rate must be 24000 if given). The format field picks the container:

format	Content-Type	What you get	Use it for
wav (default)	audio/wav	RIFF/WAVE, 16-bit PCM	files, quick tests, players
pcm16	audio/L16; rate=24000 style raw	signed 16-bit little-endian samples, no header	your own decoder/mixer, telephony bridges, browsers via Web Audio
opus	audio/ogg; codecs=opus	Ogg/Opus, 48 kbit/s, 20 ms packets, one Ogg page per packet	web playback, storage, WebRTC-adjacent pipelines

Streaming vs buffered

POST /v1/tts streams by default ("stream": true): chunked transfer, the first bytes after the first decoded 80 ms frame.

• WAV while streaming carries a placeholder length in its header. Most players and ffmpeg handle it; strict parsers may not. Send "stream": false to receive a complete file with a correct length (and Content-Length).
• Opus while streaming is a valid, growing Ogg stream; with "stream": false you get a complete, seekable file.
• pcm16 has no header either way.

On the WebSocket there is no container for a message stream, so format: wav is downgraded to pcm16 (the start event tells you which format is in effect); opus sends Ogg pages as binary messages.

Converting

# WAV → MP3 / M4A
ffmpeg -i out.wav -c:a libmp3lame -q:a 3 out.mp3
ffmpeg -i out.wav -c:a aac -b:a 96k out.m4a
# raw PCM → 8 kHz μ-law for telephony
ffmpeg -f s16le -ar 24000 -ac 1 -i out.pcm -ar 8000 -acodec pcm_mulaw out.ul

Frame timing

The engine produces audio in 80 ms frames (1 920 samples). The first frame is decoded alone, then frames are decoded four at a time, so after the first chunk audio arrives in ~320 ms steps that are each ahead of real time (real-time factor ≈ 0.14–0.19). Buffer ~100–200 ms before starting playback for jitter-free output over the public internet; see the streaming guide.