> ## Documentation Index > Fetch the complete documentation index at: https://smallestai-ff1e543d.mintlify.site/llms.txt > Use this file to discover all available pages before exploring further. # Audio Specifications > Supported audio encoding formats and requirements for real-time WebSocket transcription ## Supported Encoding Formats The Pulse STT WebSocket API supports the following audio encoding formats for real-time streaming:

Encoding	Description	Use Case
`linear16`	16-bit linear PCM	Recommended for best quality
`linear32`	32-bit linear PCM	High-fidelity audio
`alaw`	A-law encoding	Telephony systems
`mulaw`	μ-law encoding	Telephony systems (North America)
`opus`	Opus compressed audio	Low bandwidth, high quality
`ogg\_opus`	Ogg Opus container	Ogg container with Opus codec

## Supported Sample Rates Sample rate is the number of times the audio signal is measured per second. A higher sample rate naturally implies audio of better detail and higher quality. However it increases the size of the audio file. The WebSocket API supports the following sample rates: * **8000 Hz** * **16000 Hz** * **22050 Hz** * **24000 Hz** * **44100 Hz** * **48000 Hz** ## Audio Requirements ### Chunk Size The recommended size is `4096 bytes` per chunk. Sending audio in consistent 4096-byte chunks helps maintain optimal latency and processing efficiency. It minimizes the tradeoff between processing latency and network latency, finding the right fit between number of requests and the size of each request. ### Channels Currently, we support only single-channel (mono) transcription. Multi-channel support is coming soon. ### Streaming Rate For optimal real-time performance: * Stream chunks at regular intervals (e.g., every 50-100ms) * Maintain consistent chunk sizes when possible * Avoid sending chunks too rapidly or too slowly ## Format Recommendations ### Best Quality (Default) Use 16 kHz mono Linear PCM (`linear16`) for the optimal mix of accuracy and processing speed: ``` Encoding: linear16 Sample Rate: 16000 Hz Channels: Mono Chunk Size: 4096 bytes ``` ### Telephony Quality Use 8 kHz μ-law or A-law encoding for low bandwidth usage: ``` Encoding: mulaw or alaw Sample Rate: 8000 Hz Channels: Mono Chunk Size: 4096 bytes ``` ### High Fidelity For broadcast or high-quality scenarios, use higher sample rates: ``` Encoding: linear16 or linear32 Sample Rate: 44100 or 48000 Hz Channels: Mono Chunk Size: 4096 bytes ``` ## Audio Preprocessing Before streaming audio to the WebSocket API, ensure your audio is: 1. **Converted to the correct format**: Use the specified encoding (linear16, linear32, alaw, mulaw, opus, or ogg\_opus) 2. **Set to the correct sample rate**: Match the `sample_rate` parameter in your WebSocket URL 3. **Mono channel**: Downmix stereo or multi-channel audio to mono 4. **Properly chunked**: Split audio into 4096-byte chunks for streaming ### Example: Converting Audio for Streaming ```python theme={null} import numpy as np import soundfile as sf # Read audio file audio, sample_rate = sf.read('input.wav') # Convert to mono if stereo if len(audio.shape) > 1: audio = np.mean(audio, axis=1) # Resample to 16 kHz if needed if sample_rate != 16000: from scipy import signal audio = signal.resample(audio, int(len(audio) * 16000 / sample_rate)) # Convert to 16-bit PCM audio_int16 = (audio * 32767).astype(np.int16) # Split into 4096-byte chunks chunk_size = 4096 chunks = [audio_int16[i:i+chunk_size//2] for i in range(0, len(audio_int16), chunk_size//2)] ``` ## Query Parameters Specify encoding and sample rate in the WebSocket connection URL: ```javascript theme={null} const url = new URL("wss://api.smallest.ai/waves/v1/pulse/get_text"); url.searchParams.append("encoding", "linear16"); url.searchParams.append("sample_rate", "16000"); ```