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