"I've had this song stuck in my head for days, and I just can't get rid of it."
I suspect that Bluetooth speakers will launch several commercials from wire interconnect companies:
"Monster Cables Protect Your Brain"
P.S. The Wi-Fi card on our home server is frequently disturbed from Bluetooth transfers from laptops. Bluetooth hops between frequencies in an attempt to avoid interference, but Wi-Fi stays on the same frequency continuously. Interference may appear as a slowdown in data speeds on the Wi-Fi network, or trouble maintaining a connection on a Bluetooth device.
I suspect that Bluetooth speakers will launch several commercials from wire interconnect companies:
"Monster Cables Protect Your Brain"
P.S. The Wi-Fi card on our home server is frequently disturbed from Bluetooth transfers from laptops. Bluetooth hops between frequencies in an attempt to avoid interference, but Wi-Fi stays on the same frequency continuously. Interference may appear as a slowdown in data speeds on the Wi-Fi network, or trouble maintaining a connection on a Bluetooth device.
Attachments
Thanks,
What spec is your pc? Is it a noiseless one with a high spec output board, does it have to run all the time to manage the room correction?
What would be the advantages of using an external DSP like the Nadja, MiniDSP or the upcoming one from Tranquility?
What the easiest software for room correction and what calibration mike do you use?
Sorry for all the questions but I've been looking for a long time and it's a bit like a candy store, too many choices or ways to go. I would like something nearing top sound quality, easy to set up and reasonably priced......too much to ask?
A cheap DEQX would be nearly ideal, but out of my price range and takes the fun (or frustration!) out of it.....
Lewinski,
I would assume your test mike would be similar to my Earthworks mic. Mine is a 1/4" mic and also has a calibration curve. I do have the mic base and able for a B&K microphone but need the capsule and power supply still. Then I will have a real professional level test mic that is good from just about dc to 100khz flat and stable. One of the big differences is that the B&K mic uses a nickel diaphragm on the mic vs a plastic film mic for the Earthworks. The plastic diaphragms are affected by humidity and temperature and the metal diaphragms are not, they are much more accurate.
I would assume your test mike would be similar to my Earthworks mic. Mine is a 1/4" mic and also has a calibration curve. I do have the mic base and able for a B&K microphone but need the capsule and power supply still. Then I will have a real professional level test mic that is good from just about dc to 100khz flat and stable. One of the big differences is that the B&K mic uses a nickel diaphragm on the mic vs a plastic film mic for the Earthworks. The plastic diaphragms are affected by humidity and temperature and the metal diaphragms are not, they are much more accurate.
REW is good package; note loopback timing reference.
Cool Edit; great wave editor. Early editions of Audition are essentially Cool Edit. Kirkeby is Cool Edit plugin for inverse transfer function from Farina. He also has Kirkeby for Audacity 2.0.5. Kirkeby transform is at heart of Sourcefordge DRC. In Cool Edit I also Farina's convolution plugin; Also available in his Aurora Plugins for Audacity package. With Kirkeby inverse I make my own swept sine waves for doing IR measurements. Cool Edit provides full functionality for creating and manipulation of both IIR and FIR type filters.
What do you do with your Earthworks microphones? I've got a matched pair of OM-1 microphones. With such microphones it is typically not necessary to apply correction curves of microphone to measurements in order to get great results.
JRiver Media is worth every penny. It readily does clean sample rate conversions to correction filters on the fly dependent on source sample rate. I've got a couple of 8ch analog out sound cards, and all eight may be used to drive an amplifier. Mostly stick with 2-way plus mono sub. I've done this on PIII 500MHz laptop with a Sound Blaster Extigy relic. Computational power is not a problem with virtually any of today's computers.
Barleywater,
I use my Earthworks mic and Earthworks pre-amp with a Clio analysis system. I do my speaker development with that when I am not working with my friend who has a complete B&K laboratory. It is just that the Earthworks mic has an upper limit of 20khz and that isn't close to what you can get on the top end as you can measure with a 1/4 B&K mic. At the same time the difference in price is worlds apart for the two different mics.
I use my Earthworks mic and Earthworks pre-amp with a Clio analysis system. I do my speaker development with that when I am not working with my friend who has a complete B&K laboratory. It is just that the Earthworks mic has an upper limit of 20khz and that isn't close to what you can get on the top end as you can measure with a 1/4 B&K mic. At the same time the difference in price is worlds apart for the two different mics.
I purchased the EMM-6 mic three years ago when I was first exploring room measuremnts with REW and room acoustics. I asked around experienced users and it was a good balance between price and quality. It doesn't go flat to 100 kHz, though. But does a good job for where I am at now. At the time I also purchased a Tascam US122 external sound card to plug the mic to and take measurements, which I now not use anymore. Taking incremental steps and learning along the way works for me.
Yes, but at 'phase coherence' in the specsheet they mention 'minimum', not linear. So probably IIR filtering (on a DSP that handles FIR in the MiniShark, so they presumably could have done that here as well. Why not?)
My PC was built optimizing performance as an audio player, but with higher processing power as I was already thinking of DSP.
Motherboard is Intel S1200KPR: a server board without wi-fi, Bluetooth, and other electrical noise generating bells and whistles that I won't use.
Processor: Intel Xeon E3 1265lv2
The operating system, JRiver and Acourate are stored on a small SSD. The music files are stored on another (large) SSD.
8 GB RAM
Paul Pang Audio USB card (audiophile USB card). Exploring with battery power to this card.
Linear power supply to the PC, with a Paul Pang Audio ATX power supply inside the PC. PPA SATA cables.
Streacom FC8 fanless case.
Sounds like I'm a computer geek - I'm not! This was the first computer I ever built. But I did do my due diligence alright.
Oh, and internet access is through an external wi-fi brigde, connected to the PC thru LAN to avoid any electricall noise inside the PC.
And I also use a software called Audiophile Optimizer, which made a huge impact on sound.
The computer runs all the time I listen to music. It's an alll-in-one solution in that it stores and plays the music, does room correction and will do crossovers (I'm waiting for the multichannel DAC to arrive).
Don't know. I'm not familiar with those. Well, some with MiniDSP. Are you referring to their hardware? Then it allows for the crossovers and you can connect to several stereo DACs. Some MiniDSP units come with Dirac inside, which is another software package similar to Acourate. In my view, though, Dirac is more user friendly and easier to set up, but less powerfull than Acourate. And multichannel in Dirac gets expensive too.
No problem. I ask plenty of questions too. It's the way to learn!
I don't know of no option that is not expensive, easy to use, and powerful. Tradeoffs are in order. I enjoy the journey, not just the end destination, so DIY and learning is my way.
These are good primers on Acourate. Worth a read!
Computer Audiophile - Acourate Digital Room and Loudspeaker Correction Software Walkthrough
Computer Audiophile - Advanced Acourate Digital XO Time Alignment Driver Linearization Walkthrough
20kHz is more than sufficient for doing speaker alignments.
Just to make sure we are on same page, please outline your development process once you have selected drivers and have them mounted.
I am of speaker design philosophy that for XO between two drivers that the driver acoustic centers are not more than 1/4 wavelength of XO frequency apart. At greater separation lobing happens and consideration for aiming primary lobe is warranted.
When 1/4 wave conditions are met (and often even if they aren't) a single microphone location along design's listening axis works well for measuring IR of each driver.
In REW, use of loopback timing reference preserves relative timing between multiple IR measurements. Alternately, periodic measurements of drivers may be made by placing copies of stimulus in multiple tracks routed to individual drivers. Know period between stimulus blocks readily allows alignment of the responses for filter development.
Filters applied to measurements in editor predict response. Sum of responses predicts system response. This eliminates need for remeasuring speaker for each filter that is added or tweaked.
Shifting of IR in time frame is delay adjustment. Course adjustment is accomplished by whole sample shifts of waveform. This works well at lower frequencies but leaves much to be desired at higher frequencies. Example: 1kHz has about 1 degree phase shift for each 3 microseconds of delay; a 96kHz sample rate based digital delay line provided about 10 microseconds delay adjustment per sample shift. Sub sample delays are readily achieved in software. REW is capable of sub microsecond shifts at 44.1kHz sample rate.
Where to?
Just to make sure we are on same page, please outline your development process once you have selected drivers and have them mounted.
I am of speaker design philosophy that for XO between two drivers that the driver acoustic centers are not more than 1/4 wavelength of XO frequency apart. At greater separation lobing happens and consideration for aiming primary lobe is warranted.
When 1/4 wave conditions are met (and often even if they aren't) a single microphone location along design's listening axis works well for measuring IR of each driver.
In REW, use of loopback timing reference preserves relative timing between multiple IR measurements. Alternately, periodic measurements of drivers may be made by placing copies of stimulus in multiple tracks routed to individual drivers. Know period between stimulus blocks readily allows alignment of the responses for filter development.
Filters applied to measurements in editor predict response. Sum of responses predicts system response. This eliminates need for remeasuring speaker for each filter that is added or tweaked.
Shifting of IR in time frame is delay adjustment. Course adjustment is accomplished by whole sample shifts of waveform. This works well at lower frequencies but leaves much to be desired at higher frequencies. Example: 1kHz has about 1 degree phase shift for each 3 microseconds of delay; a 96kHz sample rate based digital delay line provided about 10 microseconds delay adjustment per sample shift. Sub sample delays are readily achieved in software. REW is capable of sub microsecond shifts at 44.1kHz sample rate.
Where to?
Because Bruno is usually very cautious to use exact wording I assume that this speaker behaves exactly like those that are called linear phase by the the people who are often using sloppy sales language instead of exact engineering terminology.
In other words: When you apply linear phase crossovers to multiway speakers then you end up with a minimum phase speaker. I.e. one that does only show phase-distorion at its lower and upper cutoff due to the natural rolloff behaviour - but no excess-phase inbetween.
There are some possibilities to apply techniques allowing a complete linear phase behaviour over the whole frequency response of a speaker but these come at the cost of increased total delay.
Regards
Charles
Charles, you may be right, but why then would Kii not advertise it? I suspect linear phase is not worth the bother, certainly not if complications such as pre-ringing are taken into account.
Off for the day to help a friend with his sailboat. Keep it going everybody and I will pick it up tonight. Great information so far, so much to learn and implement.
Overkill,
Thanks for your contributions, I have a better focus now on what I need to place inside the speaker and what should be external.
A "Blue Sky" standard and interoperability would be a very good thing in my eyes. Enough with proprietary systems that only work with themselves, not what the consumer needs or wants.
Now if only a Wisa standard was embraced by the major producers of consumer products this would really move wireless audio and even audio video integration to a workable level.
Overkill,
Thanks for your contributions, I have a better focus now on what I need to place inside the speaker and what should be external.
A "Blue Sky" standard and interoperability would be a very good thing in my eyes. Enough with proprietary systems that only work with themselves, not what the consumer needs or wants.
Now if only a Wisa standard was embraced by the major producers of consumer products this would really move wireless audio and even audio video integration to a workable level.
Blue sky it is. Assume this needs to include physical dimensions for form factor development otherwise you have bolt on electronics that interfere with speaker rear panel layouts.
Ummm.. folks. I tried to share this earlier. Considering this stuff will run on a 30 dollar raspbian linux board... and include access to shares... and has HDMI out to a tv already.. and a USB port... and libraries that are extendable to bluetooth....
The wheel exists.
Kodi Community – May 2015 | Kodi
The wheel exists.
Kodi Community – May 2015 | Kodi
JRiver clone
Nice idea but as you say that wheel already exists...JRiver
I see software that will do a Sonus / blu tooth / normal WiFi / streaming job...
Where is the high end DSP / Eq, DAC's / A to D's / amplification and the ability to maintain 12S from source to loudspeakers in 24 bit 96KHz with sub 5ms latency....?
Have I missed a sister web site or link?
Nice idea but as you say that wheel already exists...JRiver
I see software that will do a Sonus / blu tooth / normal WiFi / streaming job...
Where is the high end DSP / Eq, DAC's / A to D's / amplification and the ability to maintain 12S from source to loudspeakers in 24 bit 96KHz with sub 5ms latency....?
Have I missed a sister web site or link?
Looks like they are "building" it. Who knows what that means. It's an open source group so that can mean literally anything. They aren't highly motivated by the normal means and there's nobody to go after if there is blatant prior art involved.
And, it could be terrible. Hideous. But most motivated DIY people don't have high expectations either.
And, it could be terrible. Hideous. But most motivated DIY people don't have high expectations either.
Meh, I recently bought in (literally) to the raspberry pi thing. I assumed I could use that awesome 8-channel HDMI audio output via an audio extractor. But all I ran into were problems since my monitor was DVI and audio is not supported for it, or everything was mixed down to 2 ch. I finally found a USB solution for spdif out and am using that as digital audio input to a DSP crossover from miniDSP. In that role the R-Pi is working great for me. Good luck if you want to use multichannel audio (unless your monitor enables it)!
That's the MCM audio extractor?
It says it has the EDID audio switching. If configured properly, the source will interrogate the audio extractor rather than the monitor, and read the EDID data from an EEPROM inside the audio extractor. The 7.1 switch setting should force the source to use 8-channel LPCM. The monitor type shouldn't matter.
Are you sure you have everything configured correctly? I was thinking of getting one of those, but if the EDID audio switching doesn't work, it's not an option.
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