INTERACT FORUM

With MC now equipped with its own convolver (before with a plugin), DRC can be used also to equalize in real time external analog sources like: turntable, FM tuner, etc.

With its arsenal of DSPs, MC is not not just THE multimedia player, it becomes an important part of the processing chain for any audio source.

Also, having the possibility to re-route the analog input directly to the FFT analyzer (while MC plays from disk) would allow to check in real time how the DRC equalization performs:
this would require a bit more sophistication on the FFT analyzer though (i.e.: selectable vertical scale, averaging, and extended linear frequency response (16-32K FFT points)) or
the use of a VST plugin like Voxengo SPAN (free). The user would then play a pink noise track from disk and check in real time the frequency response from the microphone. 

An input plugin that plays from the line-in (probably with ASIO) is something we've considered doing, but it's not available currently.

Could you explain why this feature would be useful?  Is latency (time from input at microphone to output) important?

Thanks.

Matt,

we have talked about this in the past. Now with the Convolution engine running this feature is more relevant than ever. Many users of Acourate and Audiolense have custom build a dedicated Linux PC running BruteFIR just for that task - convolution of ANY source stream. This would allow to plug-in any external source to the line ins of the sound interface (like SACD players, TV receivers, gaming consoles,...) and use all filters of the MC DSP Studio.

But this solution would still leave alot of streams "in the dark" - all audio streams being played on the PC with apps other than MC - like games, video/audio editing applications, internet streams,...

So the "all out" solution would be implementing the MC audio engine (incl. all DSP functions) as "virtual soundcard" - with the possibility to route any external source from the ADC inputs through it as well. We have discussed this idea in the past and i know you are quite fond of the concept. It would open up JRiver MC to a whole new customer group - especially with convolution running now!

Thank you!

I think there is also interest in routing applications like spotify through mc's audio engine by either the software loopback/mixer of a soundcard or software like virtual audio cable.

+1 for usable (ASIO) audioinputs to connect beloved turntables CD players...

With that feature JR MC has the potential to become the heart of every stereo setup with the ability to use the superior dsp chain.
The digital and analog world could finally grow together - itīs a dream!

best regards
Markus

with great respect for your work and maybe the best audiophile music-player avaible today  

An input plugin that plays from the line-in (probably with ASIO) is something we've considered doing, but it's not available currently.

Could you explain why this feature would be useful?  Is latency (time from input at microphone to output) important?

Thanks.

Matt,

Thank you, and all the rest for replying to this question. 

I will try to make this brief (not possible  ) and yet include enough detail.

The purpose for the request is to use MC17 as an audiophile line preamp in my "audiophile" 2 channel system.

It can be viewed here;
http://forum.audiogon.com/cgi-bin/fr.pl?vevol&1317958032
and
http://cgim.audiogon.com/cgi-bin/vs.pl?vevol&1317958032&viewitem&1&showall

In my never ending quest for audio Nirvana, I have recently made a big improvement in the playback quality of my system by doing away with a pair of interconnects and my active line section.

I have experimented with a few passive solutions through the years, but continued to return to an active piece.  While researching an extremely well thought of passive line section from Australia, I ran across a comment from the "manufacture/researcher" that stated a single set of interconnects from source to amps was better than even his volume control.

I have always stayed away from controlling volume in the digital domain for fear of throwing away bits and reducing sound quality.

I gave it a try and was blown away.  Theory be damned (ie. FLAC and WAV have to sound the same because the "math" says so), "the proof is in the pudding". 
But..... that left me with a problem.  No way to get vinyl/phono to the amps.  A switch box did not seem like a good idea, extra contacts, and an extra set of interconnects between my digital source and the amps.

I pressed my older M-Audio Audiophile Firewire 24/96 into service.  TT to phono preamp, phono preamp to M-Audio, and M-Audio by 4 wire Firewire to computer.  The computer is an HP touch screen Q506 (dual core, 4 gig mem, Windows 7 64) dedicate to audio playback.

Turns out this works very well, but not directly into MC17.  I have not figured out how to get it to pass through.  And in fact I don't want it to just "pass through".  I would like to be able to use some plugs in for MC17, like a high pass filter for rumble, real-time upsampling to 192, and possible dynamics expansion, a la the old Phase Linear 1000 peak unlimiter, downward expander.

So, there is my "brief" reply to your question.

Thanks for any help.

Bruce

Yes, I would very much like to use REW to get a better understanding of what I am doing with with digital room correction. It is much easier to find the best sounding solution if you can also measure to better understand what you are hearing. Way too many variables now to find an optimum with ears only.

If I recall you have about 16 channels with active crossovers - I can't fathom setting all that up without fully being able to test what is going on!

When I finally get all my amps built, hopefully in the next couple of months, I will have active crossovers on the 3 fronts of our 5.1 system. I sure do hope by then that I will be able to somehow utilize all of REW's capabilities with MC.

Please, please please!

Rod

Just a crazy thought - what about Rewire support?  It's used to shuttle audio and midi between many studio apps today.  I think everyone's "live" input scenario may be a bit different, and rewire would be a good way to allow users to set up their own "input" app.  For example, I could set up a Reaper session to handle multiple inputs (turntable, mic, ...), EQ them as needed, and then route the master bus over to MC. 

Conceptually, it's quite similar to having Reaper run a VST plugin, but offers more than just simple inline one after another routing.

Here's a short description:
http://wiki.cockos.com/wiki/index.php/ReWire

An input plugin that plays from the line-in (probably with ASIO) is something we've considered doing, but it's not available currently. Could you explain why this feature would be useful?  Is latency (time from input at microphone to output) important?

See my post here:
http://yabb.jriver.com/interact/index.php?topic=69312.msg469867#msg469867

Low latency is good but most measurement software is designed to deal with it.