标签：#audio

A new reading subscription from Everand offers access to both ebooks and audiobooks, and Fable's book club community.

Whisper is remarkably robust. But "robust" doesn't mean "immune to noise." If you've ever run a meeting recording through Whisper and gotten back garbage — or worse, confidently wrong text — the problem is usually the audio, not the model. Here's the thing: different noise types fail differently. Electrical hum causes Whisper to hallucinate syllables. Echo makes it drop words entirely. Static makes it confuse phonemes. Knowing which noise you have tells you exactly which fix to apply. This post covers: ✅ How each noise type (hum, hiss, echo, wind, static) degrades Whisper output ✅ A Python preprocessing pipeline that detects and removes noise before transcription ✅ How to call the StemSplit Denoise API for cloud GPU noise removal (no local setup) ✅ Measured WER improvements you can reproduce The Noise → Transcription Failure Map Noise Type What It Sounds Like How It Breaks Whisper Hum (50/60 Hz) Constant low-frequency "buzz" Inserts phantom syllables, lowers confidence Hiss High-frequency "shhh" Loses sibilants, confuses "s/sh/f" sounds Echo / Room reverb Words "bounce" and overlap Drops end-of-sentence words, merges phrases Wind Burst plosives, low-frequency rumble Transcribes as "[inaudible]", breaks sentence segmentation Static / crackling Random pops and snaps Breaks word boundaries, causes mid-word cuts These aren't hypothetical. They're reproducible failure modes. Let me show you how to handle each one. Prerequisites pip install openai-whisper requests python-dotenv soundfile numpy librosa You'll need: A StemSplit API key from stemsplit.io/developers (free 5-minute tier, no credit card) ffmpeg installed ( brew install ffmpeg / sudo apt install ffmpeg ) The Preprocessing Pipeline Here's the full pipeline before we break it down: Audio file → [Noise detection] → [Denoise via StemSplit API] → [Post-process: normalize, trim silence] → Whisper → Transcript Step 1: Detect What Kind of Noise You Have Before throwing everything at a denoiser, it helps to know what you

With true plug-and-play support, versatile connectivity, and excellent sound, Universal Audio’s Volt 876 makes recording simple.

The M-Track Duo HD Producer Pack removes the last remaining excuse for terrible audio and potentially terrible opinions.

This is a sanitized engineering note about server-side audio noise reduction for WebRTC calls. Source article: https://www.lodan.me/posts/server-side-webrtc-noise-reduction-pion-ffmpeg-rnn/ What the prototype tests The goal is not to replace WebRTC's built-in audio processing. The narrower test is: receive a WebRTC Opus track with Pion read RTP packets in OnTrack decode Opus payloads to PCM pipe raw PCM into FFmpeg apply the arnndn RNN noise reduction filter validate the output as a file before considering real-time forwarding Why this boundary matters RTP, Opus, PCM, and FFmpeg raw audio input are different boundaries. If the PCM format is wrong, FFmpeg may still produce a file, but the result should not be trusted. For example, if the Go side writes int16 PCM, the FFmpeg input format should be reviewed as s16le , not casually treated as s32le . Production concerns The prototype is useful because it isolates the audio path, but production use needs more work: buffering and latency CPU and memory isolation FFmpeg process lifecycle model choice packet loss and jitter RTP timestamps audio/video sync whether the processed audio is returned to WebRTC or only recorded The full article has diagrams and the longer explanation: https://www.lodan.me/posts/server-side-webrtc-noise-reduction-pion-ffmpeg-rnn/

Fosi Audio announced a new sound card today with a unique feature designed to give FPS players an advantage. The C3 Gaming Sound Card, which sits outside your PC or laptop and connects with a USB-C cable, includes the company's StepSense "audio enhancement technology" powered by a model that was "trained on extensive FPS audio […]