liminfoAudioconvert
Free web tool: Audioconvert
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Choose fileMP3, OGG, FLAC, AAC, M4A, WAV, etc. (max 100MB)
About Audioconvert
The Audio Converter is a fully client-side audio format conversion tool that processes audio files directly in your browser using the Web Audio API and lamejs. It supports conversion between four output formats: MP3, WAV, OGG, and WebM. The conversion pipeline first decodes the input file using the browser's built-in AudioContext.decodeAudioData(), which supports a wide range of input formats including MP3, OGG, FLAC, AAC, M4A, WMA, WAV, WebM, and Opus (up to 100MB). After decoding, the tool re-encodes the raw PCM audio data into the selected output format.
This tool is used by podcasters, musicians, sound designers, game developers, and content creators who need to quickly convert audio files between formats without installing desktop software like Audacity or FFmpeg. Web developers who need to prepare audio assets in the right format for HTML5 audio elements (WAV for maximum compatibility, OGG/WebM for open formats, MP3 for broad support) also find it very useful. The MP3 bitrate selector (128, 192, 256, 320 kbps) is particularly useful for controlling the trade-off between file size and audio quality.
The technical encoding pipeline uses three distinct strategies depending on the target format. For WAV, a custom encoder writes the RIFF/WAVE header manually using a DataView with little-endian byte ordering, then interleaves all audio channels into 16-bit PCM samples. For MP3, the tool dynamically loads the lamejs library (/lame.min.js) and uses its Mp3Encoder class to encode in 1152-sample blocks, supporting both mono and stereo. For OGG and WebM, the browser's MediaRecorder API is used with real-time encoding — because MediaRecorder works in real time, these conversions take as long as the audio duration itself. After encoding, a download URL is created via URL.createObjectURL() and the audio is playable in the browser before downloading.
Key Features
- Converts audio to MP3, WAV, OGG, and WebM formats entirely in the browser
- MP3 bitrate selection: 128, 192, 256, and 320 kbps for quality vs. size control
- Supports a wide range of input formats: MP3, OGG, FLAC, AAC, M4A, WMA, WAV, WebM, Opus
- Audio info display after decoding: duration, channel count (mono/stereo), and sample rate
- In-browser preview: play the converted audio before downloading
- WAV encoding with custom RIFF/WAVE header writer using DataView (no library required)
- MP3 encoding via lamejs with stereo and mono channel support
- File size up to 100MB supported with drag-and-drop and click-to-select upload interface
Frequently Asked Questions
What audio formats can I convert to?
You can convert to four output formats: MP3, WAV, OGG, and WebM. The availability of OGG and WebM depends on whether your browser's MediaRecorder supports those MIME types. Chrome supports both; Firefox supports OGG. If your browser does not support a format, an error message will explain this.
What audio formats can I convert from (input)?
The tool accepts any audio file that your browser's AudioContext can decode, which includes MP3, OGG, FLAC, AAC, M4A, WMA, WAV, WebM, and Opus. The maximum file size is 100MB. The actual supported formats may vary by browser.
What bitrate should I choose for MP3 conversion?
128 kbps is suitable for speech, podcasts, and casual listening where file size matters most. 192 kbps offers a good balance for music. 256 kbps is high quality suitable for critical listening. 320 kbps is the maximum standard MP3 quality, indistinguishable from CD audio for most listeners. Higher bitrates produce larger files.
Why does OGG or WebM conversion take a long time?
OGG and WebM use the browser's MediaRecorder API, which encodes audio in real time. This means if your audio file is 3 minutes long, the encoding process will also take approximately 3 minutes. WAV and MP3 encoding are much faster because they process all audio data at once rather than in real time.
Is my audio file uploaded to a server?
No. All audio processing happens entirely within your browser. The file is read using the FileReader/ArrayBuffer API, decoded by the Web Audio API, and encoded using either client-side JavaScript (WAV, MP3) or the browser's MediaRecorder (OGG, WebM). No file data ever leaves your device.
What is the difference between WAV and MP3?
WAV is an uncompressed format that stores raw PCM audio data. It has perfect quality but large file sizes (roughly 10MB per minute for CD-quality stereo). MP3 uses lossy compression to reduce file size by discarding audio data that humans are less likely to notice. At 320 kbps, MP3 files are about 10x smaller than WAV with near-identical perceived quality.
Can I convert a stereo file to mono?
The tool preserves the channel configuration of the original file. It does not currently offer an option to change stereo to mono or vice versa. The audio info section after decoding will show you whether your source file is mono or stereo.
What happens if I try to convert an unsupported format?
If the file is not recognized as an audio file (based on MIME type and extension), you will see an error message immediately. If the file is recognized but cannot be decoded by AudioContext (for example, a corrupted file or an unsupported codec), an error will be shown after the decoding step.