About

This tool addresses a critical gap in digital media workflows: the secure, high-fidelity extraction of audio from video containers. Unlike server-side converters that compress dynamic range or introduce artifacting, this application utilizes the Web Audio API to access the raw Pulse-Code Modulation (PCM) data directly from the source file. This is essential for sound engineers, podcasters, and forensic analysts who require bit-perfect accuracy.

The architecture is strictly Client-Side. When you load a video, the browser decodes the binary stream into an IEEE 754 floating-point buffer. This allows for immediate visualization and manipulation without network latency. The integrated Non-Destructive Editor features a precise waveform render, allowing users to visually identify transient peaks (drum hits, speech starts) and silence. You can audition the selection before export, ensuring that the cut points t_start and t_end are mathematically exact.

Formulas

The fidelity of the extracted audio is determined by the quantization error and the sampling theorem. To avoid aliasing, the sampling rate f_s must satisfy the Nyquist-Shannon theorem relative to the maximum frequency B in the signal.

f_s ≥ 2B

When trimming audio, we calculate the byte offset in the linear memory buffer. For a stereo file (C = 2) with bit depth D (in bytes), the memory pointer offset P for a given time t is:

Pbyte = floort × f_s × C × D

Reference Data

Format	Compression	Bitrate (kbps)	Algorithm	Best For
WAV (PCM)	Lossless	1411 +	Linear Quantization	Mastering, Archiving, Editing
WebM (Opus)	Lossy High-Eff	64 − 510	MDCT / LPC	Web Streaming, Voice, Storage
MP3	Lossy	128 − 320	Perceptual Coding	Legacy Compatibility
AAC	Lossy	256	MDCT	Video Containers (MP4)
FLAC	Lossless	Var	Linear Prediction	Compressed Archiving
Ogg (Vorbis)	Lossy	192	MDCT	Open Source Gaming

Frequently Asked Questions

Visual feedback allows you to "see" the sound. In a long video file, finding a specific quote or sound effect by ear is tedious. The waveform displays amplitude peaks, making it easy to spot where someone starts talking or where a loud event occurs. This tool provides a zoomed overview of the entire dynamic range for precise cutting.

No. You cannot add information that isn't there. If the source video uses low-bitrate AAC audio (e.g., 96kbps), converting it to WAV simply creates a larger file containing that same low-quality audio. However, extracting to WAV prevents *further* quality loss that would occur if you re-encoded it to MP3 again (generation loss).

WAV is raw, uncompressed audio. It is huge in file size but contains exact data (perfect for editing in software like Audacity or Logic Pro). WebM (audio-only) uses the Opus codec, which is modern and highly efficient. It produces very small files with quality that is often indistinguishable from the original to the human ear. Use WAV for editing, WebM for sharing.

Technically, no, but practically, yes. The browser must decode the entire audio stream into RAM (Random Access Memory). A 3-hour movie might require several gigabytes of RAM to decode as raw PCM data. If your computer runs out of memory, the browser tab may crash. For 4K movies or very long files, ensure you have a powerful machine.

Before playback or visualization can occur, the compressed audio stream (AAC, Vorbis, etc.) inside the video container must be fully decoded into raw linear PCM data. This is a CPU-intensive task. Once decoded, operations like trimming and playing are instantaneous.