convert stereo WAV to mono
Converting WAV to mono never runs through any lossy codec at any step: it decodes PCM, mixes the samples, writes PCM again. The L+R sum used in Average mode is divided by 2, an exact -6.02dB attenuation measured directly in the tool, enough to keep content that is identical on both channels at its original level, with no risk of clipping.
The sample rate does not change, only the bit depth drops to 16, this tool's WAV encoder's fixed ceiling.
The 0.50 ratio is exact for WAV, unlike MP3, whose size follows the chosen bitrate, not the channel count.
The divide-by-2 exactly cancels out the duplication of content that is identical on both channels, the -6dB only costs level on content that differs between the two sides.
Averaging sums L and R and halves the result, which preserves centred content (vocals, kick) but can cancel anything recorded in opposite phase. The L and R modes simply discard one channel: you lose whatever only existed on the other side, but you gain a guarantee that nothing will cancel. When the correlation strip shows negative stretches, that is exactly the trade-off to weigh.
Because adding without dividing would let two channels peaking simultaneously (+1.0 each) exceed the PCM's amplitude ceiling, reaching 2.0. Dividing by 2 is an exact -6.02dB attenuation that brings that theoretical peak back to 1.0, at the cost of halving the level of content that is not identical on both channels.
It runs through no lossy compression step, it is PCM decoding, sample mixing and PCM writing again. The one limitation is the fixed 16-bit export ceiling: a 24-bit or 32-bit float source file comes out requantized, even though the original sample rate is preserved.
Yes, for WAV. In a 2-second test at 44100Hz, the stereo file was 352,844 bytes and the mono one 176,444, a ratio of 0.50. That is different from MP3, where size follows the configured bitrate, not the number of recorded channels.
Your audio is decoded and processed in your browser. No file is ever uploaded to the server.