Set up LAME and FFmpeg
The app is a front end for two free encoders. You add them once, next to the exe.
- Get LAME. Download the static
lame.exe from free-codecs.com. It is all you need for WAV to MP3.
- Get FFmpeg (optional). Download
ffmpeg.exe from free-codecs.com for MP3, FLAC, M4A and other input, and for AAC, FLAC, OGG, OPUS and WAV output.
- Put them together. Copy both files into the folder with
X-Audio-Converter.exe, start the app and open Settings - both should show OK.
Tip: use the static lame.exe build. Other builds may need a Visual C++ runtime DLL and stop with an error code.
Convert WAV to MP3
The classic job, done with the LAME encoder.
- Open the file. Click Browse or drag the WAV onto the window. The output name is filled in for you.
- Pick the quality. Keep MP3 (.mp3) and choose a bitrate - 192 kbps for music, 320 kbps for the best quality - or tick Use VBR with quality 2.
- Convert. Click Convert to MP3, then Open output folder to find the file.
Speech or podcasts? Tick Voice mode for smaller files tuned for voice.
Convert a whole folder
Turn an album or a folder of recordings into MP3 or any other format in one go.
- Select the files. Click Folder to take every audio file in a folder, or Ctrl+click several files in Browse. Dropping files or folders on the window works too.
- Include subfolders if needed. Tick Include subfolders (keep folder structure) first to convert a whole music library tree.
- Choose the output folder. The output box switches to a folder. Every file keeps its name and sub folder, with the new extension.
- Convert. The progress bar shows which file is running, and a summary lists any file that could not be read.
Tip: in batch mode each file keeps its own title tag, while typed Artist, Album and Year apply to all of them.
Convert to AAC, FLAC, OGG or OPUS
With FFmpeg in place, the Output format list unlocks five more formats.
- AAC / M4A - for iPhone, iTunes and video editors. 192 to 256 kbps sounds transparent.
- FLAC - lossless. The compression level only changes file size and speed, never quality.
- OPUS - the most efficient codec. 96 to 128 kbps is enough for music, 32 to 64 kbps for voice.
- OGG Vorbis - open format for games and players that support it. Quality 5 is a good default.
WAV PCM is also available, for example to feed audio into an editor that does not read FLAC.
Make every file the same loudness
Tracks from different albums or recordings often play at very different volumes. Normalizing fixes that in the same pass.
- Load the batch. Pick the files or the folder as usual.
- Tick Normalize loudness. It sits next to Sample rate and works for every output format.
- Convert. Each file is measured and brought to -16 LUFS, with a -1.5 dB true-peak limit so nothing clips.
Good to know: -16 LUFS suits podcasts, playlists and mobile listening. Leave it off for albums you want to keep exactly as mastered.
Tags and sample rate
Your music library stays organised, and the output can match a video or CD project.
- Tags are copied. Title, artist, album, year and track come over from the source file automatically.
- Untagged files? Keep Fill empty tags from the file name ticked in the Tags tab.
03. Artist - Title.wav becomes track 3, the artist and the title.
- Override them. Type a value in the Tags tab to replace it - it works for MP3, M4A, FLAC, OGG and OPUS.
- Match the rate. Set Sample rate to 48000 for video projects or 44100 for CD, or keep the original.
Studio work? For soxr-quality resampling, use the free X Sample Rate Converter.