# Audio Formats

Supported audio formats for file upload and WebSocket streaming.

Lesan AI supports a wide range of audio formats for file-based transcription and a subset of formats optimized for real-time WebSocket streaming.

## File Upload Formats

The following formats are supported for file uploads via the `/transcribe` and `/transcribe/batch` endpoints:

- **MP3** (.mp3) — Most common format. Good compression with acceptable quality.
- **WAV** (.wav) — Uncompressed audio. Best quality but largest file size.
- **M4A** (.m4a) — AAC-encoded audio in MP4 container. Common on Apple devices.
- **FLAC** (.flac) — Lossless compression. Best quality-to-size ratio.
- **OGG** (.ogg) — Vorbis or Opus encoded. Common in web applications.
- **WebM** (.webm) — Opus encoded in WebM container. Native browser recording format.
- **AAC** (.aac) — Raw AAC audio. Common in mobile applications.

### File Size Limits

- **Maximum file size** — 500 MB per file
- **Maximum duration** — 4 hours per file
- **Minimum duration** — 0.1 seconds

For files exceeding these limits, consider splitting the audio into smaller segments before uploading.

### Recommendations

- **Best quality** — FLAC or WAV at 16kHz or higher sample rate
- **Best compression** — MP3 at 128kbps or OGG at 96kbps
- **Browser recordings** — WebM with Opus codec (native MediaRecorder output)
- **Mobile recordings** — M4A (iOS) or OGG (Android)

## WebSocket Streaming Formats

The [streaming API](/guides/streaming) supports these formats, optimized for low-latency real-time audio:

- **pcm\_s16le** (default) — Raw PCM, 16-bit signed, little-endian. No encoding overhead, lowest latency. Requires setting the correct `sample_rate` parameter.
- **wav** — WAV container format. Send the WAV header with the first chunk; subsequent chunks are raw PCM.
- **webm\_opus** — WebM container with Opus codec. Best for browser-based streaming using `MediaRecorder`.
- **opus\_raw\_16k** — Raw Opus frames at 16kHz. Low bandwidth, good for mobile applications.

### Format Selection

Choose the right streaming format for your use case:

- **Server-side processing** — Use `pcm_s16le` for lowest latency and simplest implementation
- **Browser applications** — Use `webm_opus` for native MediaRecorder compatibility
- **Mobile / low bandwidth** — Use `opus_raw_16k` for compressed audio with minimal overhead

## Sample Rate Guidelines

The sample rate affects transcription quality and bandwidth:

- **16000 Hz** (recommended) — Optimal for speech recognition. Best accuracy-to-bandwidth ratio.
- **8000 Hz** — Telephony quality. Acceptable for voice calls but lower accuracy.
- **44100 Hz / 48000 Hz** — High-quality audio. Automatically downsampled to 16kHz internally. Uses more bandwidth with no accuracy benefit.

## Converting Audio Formats

Use ffmpeg to convert audio to a supported format:

```bash
# Convert any format to 16kHz WAV (best for upload)
ffmpeg -i input.mp4 -ar 16000 -ac 1 output.wav

# Convert to MP3 (good compression)
ffmpeg -i input.wav -ar 16000 -ac 1 -b:a 128k output.mp3

# Convert to FLAC (lossless compression)
ffmpeg -i input.wav -ar 16000 -ac 1 output.flac

# Extract raw PCM for streaming
ffmpeg -i input.wav -f s16le -ar 16000 -ac 1 output.pcm
```

```python
# Using pydub to convert audio files
from pydub import AudioSegment

# Load any supported format
audio = AudioSegment.from_file("input.mp4")

# Convert to 16kHz mono WAV
audio = audio.set_frame_rate(16000).set_channels(1)
audio.export("output.wav", format="wav")

# Convert to MP3
audio.export("output.mp3", format="mp3", bitrate="128k")

# Get raw PCM bytes for streaming
pcm_data = audio.raw_data
```

See the [ASR guide](/guides/asr) for file upload examples, or the [Streaming guide](/guides/streaming) for real-time audio streaming.