Audio Sample Rate Converter Convert audio files to different sample rates (44.1kHz, 48kHz, etc.).

Sample rate conversion uses the Web Audio API's OfflineAudioContext, which has built-in high-quality resampling. The source file is decoded using an AudioContext at its native sample rate. An OfflineAudioContext is created at the target sample rate with the appropriate number of channels and a duration calculated as: newLength = originalDuration * targetSampleRate. The decoded AudioBuffer is played into the OfflineAudioContext via an AudioBufferSourceNode, and the context's built-in resampler handles the rate conversion using sinc interpolation. The rendered output AudioBuffer contains the audio at the new sample rate. This is then encoded as a 16-bit PCM WAV file. Upsampling adds interpolated samples; downsampling applies an anti-aliasing filter before decimation to prevent aliasing artifacts.