Free & Private Online Browser Utility
Audio Speed & Pitch Changer
Modify audio playback speed and pitch independently or synchronously inside your browser using the Web Audio API. You can speed up lectures, slow down music rehearsals, and export processed audio to standard WAV format. Audio buffers are decoded and processed in local browser memory without uploading sensitive voice or musical recordings to remote servers.
How to use
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Step 1: Load Audio File
Select an MP3, WAV, or OGG audio file from your device to decode into memory.
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Step 2: Adjust Speed and Pitch
Use the sliders to modify playback rate (0.25x to 3x) or transpose pitch by semitones.
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Step 3: Export Processed Audio
Audition the modified sound in real time and export the resulting audio stream as a WAV file.
Worked example
Input: 1-minute spoken MP3 recording at 1.0x playback rate
Output: 40-second audio stream at 1.5x speed with preserved pitch in WAV format
Processed using AudioContext.decodeAudioData and buffer playback manipulation.
Tips & limits
- PCM WAV export generates uncompressed audio that preserves acoustic fidelity without lossy artifacts.
- Slowing playback to 0.75x is especially useful for analyzing complex musical solos or fast foreign language dialogue.
- Large audio files may take a few seconds to decode into memory on mobile devices.
Common problems
- Audio file fails to decode
Verify your audio is in a standard format such as MP3, WAV, or OGG; proprietary DRM-protected files cannot be decoded.
- Exported WAV file has large byte size
WAV uses uncompressed 16-bit PCM samples; convert the exported WAV to MP3 or Opus if file size is a priority.
Frequently Asked Questions (FAQ)
Does changing speed also change the pitch?
By default, simple resampling changes both speed and pitch together. Our pitch adjustment slider allows you to compensate for tempo changes.
Are audio files uploaded to a cloud server?
No. The Web Audio API decodes audio samples into local browser RAM, keeping your files on your device.
What formats can I upload?
The tool supports audio formats supported natively by your browser's AudioContext, including MP3, WAV, OGG, and AAC.
How finely can the speed control be set?
The speed slider runs from 0.25x to 3.0x and stops every 0.05, so tempos like 1.15x or 2.45x are selectable rather than rounded to the nearest quarter. Pitch is a separate control that covers -12 to +12 semitones.
How do the two sliders combine into one playback rate?
The component multiplies them: your semitone offset becomes 2^(semitones / 12) and that factor is applied to the speed value. Setting +12 semitones therefore doubles the rate, exactly like setting speed to 2.0x.
The Web Audio API Playback Engine: Speed vs Pitch Independence
The HTML5 Web Audio API manages digital sound reproduction through an audio node graph anchored by an AudioContext. Within this pipeline, playback rate adjustments are executed via the playbackRate AudioParam of an AudioBufferSourceNode. By adjusting the playback rate, the browser resamples the audio buffer in real time, shifting the time interval between successive discrete audio samples.
In analog playback and native hardware resampling, playback speed and fundamental acoustic pitch are inextricably linked: increasing the playback velocity shortens the waveform period, which proportionally multiplies the fundamental frequency and raises perceived pitch. To achieve independent control over tempo and pitch, modern digital audio workstations (DAWs) decouple the two parameters through phase vocoder algorithms or pitch-synchronous overlap-and-add (PSOLA). In the mathematical model used here, pitch transposition is expressed in semitones using the equal-temperament formula f = f₀ × 2^(semitones / 12), allowing precise harmonic adjustments for musicians and vocalists.
| Playback Rate | Effect on Tempo | Effect on Pitch |
|---|---|---|
| 0.5x | Half speed (doubled playback duration) | Half pitch (-12 semitones / 1 octave down) |
| 0.75x | 75% slower (extended phrasing) | -4 semitones (major third down) |
| 1.0x | Normal (original duration) | Normal (unmodified pitch and key) |
| 1.25x | 25% faster (accelerated tempo) | +3.8 semitones (~minor third up) |
| 1.5x | 50% faster (brisk delivery) | +7 semitones (perfect fifth up) |
| 2.0x | Double speed (halved playback duration) | +12 semitones (1 octave up) |
Audio Export: WAV PCM Encoding Without External Libraries
Exporting high-fidelity audio from client-side memory requires serializing decoded floating-point buffers into the standard Resource Interchange File Format (RIFF) WAV container. A standard WAV file consists of three sequential byte chunks: the 12-byte RIFF header (declaring container bounds and the 'WAVE' format signature), the 24-byte fmt subchunk (specifying audio format category, channel count, sample rate, and byte alignment), and the variable-length data subchunk containing the raw pulse-code modulation (PCM) audio samples.
Web Audio internally processes audio data as 32-bit floating-point numbers normalized between -1.0 and +1.0. To generate standard CD-quality 16-bit uncompressed WAV output, each floating-point sample is clamped to prevent clipping artifacts and converted into a signed 16-bit two's-complement integer ranging from -32,768 to +32,767. This tool achieves zero external runtime dependencies by extracting channel sample arrays via AudioBuffer.getChannelData() and writing binary bytes sequentially with a JavaScript DataView into an ArrayBuffer before packaging into a downloadable Blob.
| Byte Offset | Chunk / Field | Type / Size | Purpose / Technical Specification |
|---|---|---|---|
| 00 - 03 | ChunkID ("RIFF") | 4-byte ASCII | Identifies the file as a RIFF binary container (0x46464952) |
| 04 - 07 | ChunkSize | 32-bit uint (LE) | Total file size in bytes minus 8 (excludes RIFF identifier and size) |
| 08 - 11 | Format ("WAVE") | 4-byte ASCII | Specifies the WAV audio stream format payload (0x45564157) |
| 12 - 15 | Subchunk1ID ("fmt ") | 4-byte ASCII | Format chunk descriptor header (0x20746D66) |
| 16 - 19 | Subchunk1Size | 32-bit uint (LE) | Fixed at 16 bytes for standard uncompressed Linear PCM |
| 20 - 35 | Audio Format & Rates | 16 bytes PCM | Specifies 1 (PCM), channels (1/2), sampleRate, byteRate, and 16-bit depth |
| 36 - 43 | Subchunk2ID ("data") | 8 bytes + data | Data chunk header (0x61746164) followed by interleaved 16-bit PCM samples |
sector Use Cases & Professional Applications
Musician Instrument Transcription & Ear Training Practice
Guitarists, pianists, and vocalists slow down blistering solos, complex jazz chords, and rapid vocal riffs to 50% or 75% speed without audio clipping, enabling note-by-note transcription and deliberate practice.
Podcast, Audiobook & Lecture Rapid Research Auditing
Students, journalists, and audio researchers accelerate recorded university lectures, interviews, and long-form podcasts to 1.5x–2.0x speed, cutting listening time in half while maintaining sharp voice intelligibility.
Video Editing Sound Design & Tempo Synchronization
Content creators and video editors adjust the duration and tempo of background music tracks and voiceover stems to align accurately with fast-paced visual transitions in TikTok, Reels, and YouTube shorts.