Compress Audio Files Online

Shrink MP3, AAC, WAV, FLAC, OGG and M4A files to a target size, percentage, or bitrate online — with an optional trim before you export.

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Supports: AAC, AC3, AIF, AIFC, AIFF, AMR +13 more

OptionsAdvanced Options - Our defaults are optimized for the best results. We recommend you keeping the defaults unless you have a specific need.
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Audio File Extension
File Compression
File size (%)
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80
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If your file is 10 MB, then selecting 80 will produce a 8 MB file. If you make the output file size too small, then output video quality may suffer.
Audio Channel
Audio Channel
Audio Sample Rate
Audio Sample Rate
Trim

Compress Audio Online — Free, No Watermark

Compress audio online free: upload MP3, WAV, FLAC, AAC, M4A, or OGG, then set a target size, a percentage, or a specific bitrate — and optionally switch stereo to mono for voice — and XConvert re-encodes it on our servers. No sign-up, no watermark.

Real result: in our production data the median audio file drops about 50% (a 9.4 MB file becomes 4.7 MB); a 30-minute WAV interview (300 MB) re-encoded to 96 kbps mono MP3 lands near 21 MB.

How to Compress Audio Files Online

  1. Upload Your Audio Files: Drag and drop or click "+ Add Files" to select audio from your device. Batch is supported. Accepted inputs include MP3, WAV, FLAC, AAC, M4A, OGG, OPUS, WMA, AIFF, AC3, AMR, AU, WEBA, and more.
  2. Pick a Compression Target: Choose File Size Percentage (e.g. 50% to halve the file), Specific file size (enter an exact KB/MB target — XConvert back-solves the bitrate), or Custom Bitrate for direct control. With Custom Bitrate you can pick Constant Bitrate (CBR) presets or Variable Bitrate (VBR) ranges per codec.
  3. Refine Audio Parameters (Optional): Set the Audio File Extension (keep the source format or convert to MP3, AAC, OPUS, OGG, FLAC, M4A, WAV, AC3, AMR, WMA), drop the Audio Sample Rate to 22.05 kHz or 16 kHz for voice content, switch the Audio Channel from stereo to mono to roughly halve the size, or Trim silent intros and outros with start time + duration.
  4. Compress and Download: Click "Compress". Files are uploaded over an encrypted connection, processed on our servers, and deleted automatically after a few hours — no sign-up, no watermark, never shared. Download individually or grab the ZIP.

Why Compress Audio?

Uncompressed CD-quality stereo runs about 10 MB per minute — a 30-minute interview is roughly 300 MB as WAV. Even a typical 192 kbps MP3 podcast episode at 60 minutes lands around 86 MB, which already exceeds Gmail's 25 MB attachment cap. Compression brings audio down to sizes that send, stream, and store cleanly without throwing away anything listeners actually notice.

  • Beat email and chat caps — Gmail tops out at 25 MB per message and Discord's free tier caps uploads at 10 MB (raised to 500 MB with Nitro, lowered from the legacy 25 MB cap in late 2024). A 30-minute interview re-encoded to 96 kbps mono MP3 lands near 21 MB, comfortably under both.
  • Hit podcast host limits — Apple Podcasts recommends mono MP3 at 96–128 kbps and stereo at 128–256 kbps for RSS-distributed shows. Compressing a raw WAV master to these targets typically shrinks a 60-minute episode from ~600 MB to 40–110 MB.
  • Save device and cloud storage — A FLAC re-encode of a WAV master typically lands at 50–70% of the original size with bit-exact playback. iCloud free is only 5 GB, so lossless libraries fill it fast unless compressed.
  • Cut bandwidth for streaming — Opus at 64 kbps delivers near-transparent voice and is the codec WebRTC, Zoom, and YouTube use by default for low-bitrate audio. Halving bitrate from 128 to 64 kbps directly halves data usage.
  • Match the playback target — Voicemail trees, embedded web players, and IVR systems usually want 8–22.05 kHz mono at 32–64 kbps; mastering studios want 44.1/48 kHz stereo at 256–320 kbps. One source, many delivery targets.
  • Speed up uploads and CDN sync — Smaller files reach Spotify for Podcasters, Buzzsprout, Anchor, or your own S3 bucket faster, and the savings compound across a back catalog.

Audio Codec Quick Guide

Codec Container Sweet-spot bitrate Best for Notes
MP3 .mp3 128–320 kbps Universal playback Most compatible lossy format; every device made since 2000 can play it.
AAC .m4a /.aac 96–256 kbps Apple ecosystem, YouTube ~20% smaller than MP3 at matched perceptual quality; native on iOS/macOS.
Opus .opus /.weba 32–128 kbps Voice, WebRTC, low-bitrate music Best modern codec under 96 kbps; transparent for music at ~128 kbps.
OGG Vorbis .ogg 96–256 kbps Open-source web playback, games Royalty-free; widely supported in browsers and Android.
FLAC .flac n/a (lossless) Archives, audiophile listening Bit-exact; typically 50–70% of WAV size.
WAV / PCM .wav n/a (uncompressed) Editing masters ~10 MB/min at 16-bit 44.1 kHz stereo.
AC3 .ac3 192–448 kbps Surround/DVD audio Used in DVDs and ATSC broadcast.
AMR .amr 4.75–12.2 kbps Voice memos, telephony Narrowband speech codec; very small files.

Bitrate Targets by Use Case

Use case Format Sample rate Channels Bitrate ~Size per hour
Voice podcast (RSS) MP3 44.1 kHz Mono 96 kbps ~43 MB
Music podcast (RSS) MP3 / AAC 44.1 kHz Stereo 128 kbps ~58 MB
Premium music stream AAC 44.1 kHz Stereo 256 kbps ~115 MB
WebRTC / voice call Opus 48 kHz Mono 32–64 kbps ~14–29 MB
Audiobook / voice memo MP3 / Opus 22.05 kHz Mono 48–64 kbps ~22–29 MB
Lossless archive FLAC 44.1/48 kHz Stereo n/a ~250–350 MB

CBR vs VBR — Which Mode to Pick?

Property Constant Bitrate (CBR) Variable Bitrate (VBR)
Bitrate over time Fixed Adapts to complexity
File size Predictable Smaller for same perceived quality
Streaming compatibility Strongest (live streams, some older players) Occasionally finicky on old hardware
Best for Live streams, podcast RSS feeds that require it Music libraries, archives, on-demand
Default in XConvert Available under "Constant Bitrate" presets Available under "Variable Bitrate" presets

Frequently Asked Questions

How do I reduce audio file size?

Pull three levers. Lower the bitrate (dropping a 256 kbps file to 128 kbps roughly halves it), lower the sample rate (48 kHz to 24 kHz frees the encoder to use fewer bits, ideal for voice), and switch stereo to mono for spoken-word content to cut the size by close to half.

What bitrate should I use for a podcast?

For a talk-only show, 96 kbps mono MP3 at 44.1 kHz is the practical sweet spot — it matches Apple Podcasts' recommendation and produces roughly 43 MB per hour. Bump to 128 kbps mono if listeners stream over headphones with detailed ear tips, or to 128–192 kbps stereo if you mix music beds, interviews with hard panning, or sound design. Below 64 kbps mono you start hearing artifacting on sibilants ("s" and "sh" sounds).

Will compressing audio reduce quality?

Lossy compression (MP3, AAC, Opus, OGG) discards data your ear is least likely to notice. At 192 kbps and above MP3 is "transparent" to most listeners on most material; AAC reaches transparency around 128–160 kbps; Opus reaches it around 96–128 kbps. Below those thresholds artifacts get progressively easier to spot, especially on cymbals, applause, and dense electronic textures. Lossless formats (FLAC, ALAC, WavPack) shrink the file without touching the audio at all — decoded output is bit-for-bit identical to the source.

Should I pick MP3, AAC, or Opus?

Choose MP3 when compatibility matters most — every car stereo, USB stick, Bluetooth speaker, and 20-year-old MP3 player handles it. Choose AAC for the Apple ecosystem (it's the native format for iTunes, Apple Music, and Voice Memos) or when you want roughly 20% smaller files than MP3 at matched quality. Choose Opus when bandwidth is tight (under ~96 kbps) or for voice — it's the codec WebRTC, Zoom, Microsoft Teams, and YouTube use under the hood for low-bitrate streams.

How much smaller is FLAC than WAV?

FLAC typically lands at 50–70% of the source WAV size — call it a 30–50% reduction. Simple acoustic material compresses best (down to 45% of original); dense, noisy mixes compress least (65% of original). Crucially, FLAC is lossless, so decoding a FLAC file yields the exact same PCM samples as the source WAV. See compress FLAC if you have a FLAC library that's still too large for storage — re-encoding with a higher compression level squeezes out a few more percent without quality loss.

Can I target an exact file size like 8 MB?

Yes. Pick Specific file size, type the target in KB or MB, and XConvert back-solves the bitrate from the file's duration. Useful for hitting hard caps — Discord free uploads (10 MB), Reddit chat attachments, classroom LMS limits, or legacy forum uploaders. If the target is unrealistically small for the duration (e.g. a 60-minute episode in 1 MB), the encoder will floor at the codec's minimum bitrate and you'll get audible artifacts.

CBR or VBR — which should I choose?

Use CBR when you need predictable file size or you're streaming to a player that can't reseek smoothly across variable frames — some podcast RSS validators and older car head units fall in this camp. Use VBR for archives and on-demand playback where you want the smallest file for a given perceived quality — VBR spends bits on complex passages (a cymbal crash) and saves them on quiet ones (a single voice). Modern phones, browsers, and streaming services all handle VBR fine.

Does converting stereo to mono really halve the file size?

For voice content, yes — collapsing two identical-ish channels into one cuts the encoded size by close to 50% with no perceptible change for spoken-word listeners on a single speaker. For music it's a bad trade: you lose stereo imaging, panning, and most of the production. The "Audio Channel" dropdown lets you switch to mono per-file, so you can keep music tracks in stereo and force mono on lectures, interviews, or voicemails.

Will the sample rate affect file size?

Yes, but less than bitrate does. Halving the sample rate (48 kHz to 24 kHz) chops the bit budget the encoder has to work with, which lets you drop bitrate proportionally — useful for voice content where anything above 16 kHz is wasted. For music, keep 44.1 or 48 kHz — dropping to 22.05 kHz audibly dulls cymbals and high vocal harmonics. The XConvert "Audio Sample Rate" dropdown supports 8000, 12000, 16000, 24000, 44100, and 48000 Hz.

Are my audio files private?

Yes. Uploads use TLS in transit, processing happens on isolated workers, and files are auto-deleted shortly after the job finishes. We don't index, sample, or share your audio. If you need to crop a file before sharing, the same trim controls live as a dedicated tool at audio trimmer; for full format conversion without compression, use audio converter.

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