r8brain PRO is a professional sample rate converter designed to deliver an unprecedented sample rate conversion (SRC) quality.  Unlike many existing SRC algorithms available on the market, r8brain PRO implements sample rate conversion processing in its full: interpolation and decimation steps without exploiting any kind of simplifications; the signal is first resampled to a least common multiple sample rate which makes conversion perfect.  In the core of the SRC algorithm, we use the convolution methods of our Pristine Space convolution processor which is known for its highly precise convolution processing.  This gives us high sample rate conversion quality in combination with comparably small processing times: sample rate conversion without compromises!

Like many existing SRC programs, r8brain PRO offers you a linear-phase conversion mode.  But more importantly, you also have an option of using the minimum-phase conversion mode, which finally brings SRC with true analog qualities to affordable digital audio workstations: in this mode, r8brain PRO works like an ideal digital-to-analog converter followed by an analog-to-digital converter to resample the audio.  This eliminates pre-ringing associated with linear-phase designs, while introducing only a minimal amount of phase coloration.

r8brain PRO can read mono, stereo and multi-channel files in both WAV and AIFF file formats, creating 16-, 24- and 32-bit mono, stereo and multi-channel WAV files in fixed and floating point formats.  EBU BWF (broadcasting) extensions, extensible wave format, sample loops and textual data residing inside the file are also supported.  For the sake of convenience, r8brain PRO allows you to perform batch conversions in a convenient manner.

r8brain PRO's bit-depth conversion is limited to flat dithering.  We have decided not to implement noise-shaping dithering because pro audio production software available on the market usually offers the user noise-shaping dithering of some kind already.  We also based our decision on the fact that the sample rate conversion process often adjusts peak structure of the original program material, thus, in many cases, making a subsequent peak-limiting a necessity.  To prevent output audio from clipping we have implemented a level normalization feature.