Convert RAW Audio to AAC — Free Online Converter
Convert Raw PCM Audio (.raw-audio) to Advanced Audio Coding (.aac) online for free. Fast, secure audio conversion with no watermarks or registration....
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How to Convert
Upload your .raw file by dragging it into the upload area or clicking to browse.
Choose your output settings. The default settings work great for most files.
Click Convert and download your .aac file when it's ready.
About RAW Audio to AAC Conversion
Raw PCM audio is pure uncompressed sample data written directly to disk with zero framing — no file header, no metadata, no codec information. The binary stream must be interpreted with externally specified parameters: sample rate (e.g., 44100 Hz), sample format (e.g., signed 16-bit little-endian), and channel count (e.g., 2 for stereo). This format appears in DSP development, embedded firmware audio buffers, real-time audio capture from ALSA/WASAPI/CoreAudio raw interfaces, and scientific instrumentation.
AAC (Advanced Audio Coding) is the dominant lossy audio codec for the modern era, used as the default format by Apple Music, YouTube, and virtually every mobile operating system. It delivers noticeably better sound quality than MP3 at equivalent bitrates. Converting raw PCM to AAC transforms unstructured binary samples into a universally playable, efficiently compressed audio file.
Why Convert RAW Audio to AAC?
Raw PCM files occupy enormous disk space — a one-minute stereo recording at 44.1 kHz / 16-bit consumes 10.5 MB, with no metadata to describe what the data even is. No media player, streaming service, or operating system can open a raw audio file without manual parameter specification. The data is effectively trapped in an unusable form for distribution or archival.
AAC encoding compresses raw PCM by roughly 10:1 at transparent quality (256 kbps) or 20:1 at near-transparent quality (128 kbps), while adding the container metadata that makes the file self-describing and universally playable. For audio engineers, DSP developers, and scientists who produce raw captures, AAC is the bridge to sharing and consuming that content on any device.
Common Use Cases
- Compressing raw audio buffers from ALSA or WASAPI capture into distributable AAC files
- Converting DSP algorithm output from binary PCM to a format suitable for A/B listening tests on headphones
- Packaging scientific acoustic measurements into AAC for inclusion in research presentations
- Creating playable audio files from embedded firmware test captures for quality assurance reports
- Reducing storage footprint of raw audio archives while maintaining high perceptual quality
How It Works
FFmpeg ingests the raw file using explicit input parameters: `-f s16le -ar 44100 -ac 2` for signed 16-bit little-endian stereo at 44.1 kHz. These parameters are mandatory — without them, FFmpeg cannot interpret the headerless byte stream. The raw samples are decoded to internal planar float format, then encoded using the AAC-LC encoder (native or libfdk_aac if available) at the target bitrate. Output is wrapped in an ADTS raw stream (.aac) or an M4A/MP4 container depending on the output extension.
Quality & Performance
Raw PCM is mathematically perfect uncompressed audio, making it the ideal source for AAC encoding — there is no pre-existing compression to compound with. At 256 kbps stereo, AAC-LC is generally considered transparent (indistinguishable from the source in double-blind tests) for most content. At 128 kbps, quality remains excellent for speech and most music. The only quality variable is ensuring the correct input parameters are specified; wrong values produce garbage regardless of AAC bitrate.
Device Compatibility
| Device | RAW Audio | AAC |
|---|---|---|
| Windows PC | Partial | Partial |
| macOS | Partial | Native |
| iPhone/iPad | Partial | Native |
| Android | Partial | Partial |
| Linux | Partial | Partial |
| Web Browser | No | No |
Recommended Settings by Platform
Spotify
Resolution: N/A
Bitrate: 320 kbps
OGG Vorbis preferred
Apple Music
Resolution: N/A
Bitrate: 256 kbps
AAC format required
SoundCloud
Resolution: N/A
Bitrate: 128 kbps
Lossless FLAC/WAV for best quality
Podcast
Resolution: N/A
Bitrate: 128 kbps
MP3 mono for spoken word
Tips for Best Results
- 1Use `-f f32le` for 32-bit float samples common in DAW bounce exports and real-time audio APIs
- 2Add metadata after conversion with `-metadata title=...` to make the AAC file self-documenting
- 3For maximum quality from 24-bit raw sources, encode at 320 kbps AAC or consider FLAC for lossless preservation
- 4Pipe raw audio directly from a capture tool into FFmpeg to avoid intermediate disk writes: `arecord -f S16_LE | ffmpeg -f s16le -ar 44100 -ac 2 -i pipe:0 output.aac`
- 5Always specify byte order explicitly — big-endian (`s16be`) vs little-endian (`s16le`) will produce completely different audio from the same data
Raw audio to AAC conversion applies modern perceptual coding to pristine uncompressed samples, producing compact, universally compatible files from the most primitive audio format with no generational quality penalty.