They can work even without board connection which makes them very useful for studying provided processing system configuration-it's more convenient than trying to decipher the contents of config files manually.
Convolver audio hijack software#
MiniDSP units are configured using specialized software called "plugins". This fact makes me think that miniDSP is optimized for Audio-Video applications where low latency is required, and the quality of the filters can be sacrificed, because when watching movies we normally pay more attention to the picture than to the sound. The FIR filter section has limited applicability due to short filter length, which gives relatively low resolution in frequency domain. That means, the processing in this miniDSP has low latency (due to low number of taps), but minimum phase and thus non-constant group delay. It also allows the total of 4096 taps for FIR filter to be arbitrarily distributed over all 4 output channels (with a limitation that a single channel can't have more than 2048 taps). The DSP implements 10 biquad IIR filters per both input channel, then 18 biquads for EQ and crossovers per each output channel. If digital input arrives at different rate, it automatically gets resampled. The DSP in "HD" products operates at 96 kHz sampling rate. This is in fact a very useful feature for our task. When connected to USB, besides 2 output channels the unit also offers 4 input channels that allow capturing processed audio data. Here is how the processing and routing chain is organized: It has a stereo ( 2 channel) input (switchable between analog, TOSLink, and USB), and 4 channels of analog output. Let's briefly describe the capabilities and structure of the miniDSP unit. Thus my Oppo BD unit would be only left with the tasks of decoding Dolby and DTS streams, and upmixing stereo into multichannel. The end result that I want to achieve is doing all the necessary speaker processing: crossovers, time alignment, speaker linearization, and room correction in one unit-the software DSP. For starters, I decided to follow the original design of the filters as close as possible (which means replicating their phase in addition to amplitude). Thus, my first task was to re-create LXmini's DSP crossovers and filters using Acourate. Another reason for choosing Acourate over miniDSP is that the former offers practically unlimited abilities to build filters, because it's all software.
Convolver audio hijack Pc#
So I decided to go another way-build a dedicated mini PC to run Acourate Convolver via my MOTU UltraLite AVB card. There are some balanced miniDSP units: 2x4 Bal, 4x10 HD, and 10x10 HD, but their form factors do not fit into my half-rack stack. The reason is that 2x4 HD has unbalanced line out connections, but for the rear speakers I would like to put the amplifier further away and would prefer to use balanced lines between the DSP and the power amplifier. I use a miniDSP 2x4 HD for the first pair of LXminis, but I decided I don't want to buy a second one. The original design of LXminis uses miniDSP processors for implementing the crossover and speaker linearization. I'm getting ready to build a second pair of Linkwitz LXmini-this time for rear channels.