To answer my own question for anyone else doing this with JRiver, the link below is a tremendous help.
Guide to Speaker/Room Correction Using Freeware and JRiver
Guenter, nice cabinet. Mine is from a local thrift store and the delivery cost more than the cabinet did 
After ordering the wrong thing from MiniDSP (a MiniStreamer); and finding out "no refunds", oh well, I ordered the correct thing (a USBStreamer) so I can do ADAT. After much fiddling, and about $350 invested (not counting idiot purchase of wrong items !) I am pleased to report I have JRiver playing and a basic crossover set up. There is a learning curve here, and I have much to learn. I may have finally found my very low noise Nirvana because of my Balanced outs on the "DAC" (a Behringer ADA8200).
Now, how to import REW data into there...
JRiver looks like a huge app, I just wanted to be able to play music
Sent from my NV570P using Tapatalk
After ordering the wrong thing from MiniDSP (a MiniStreamer); and finding out "no refunds", oh well, I ordered the correct thing (a USBStreamer) so I can do ADAT. After much fiddling, and about $350 invested (not counting idiot purchase of wrong items !) I am pleased to report I have JRiver playing and a basic crossover set up. There is a learning curve here, and I have much to learn. I may have finally found my very low noise Nirvana because of my Balanced outs on the "DAC" (a Behringer ADA8200).
Now, how to import REW data into there...
JRiver looks like a huge app, I just wanted to be able to play music
Sent from my NV570P using Tapatalk
Now I have a nearly all-Behringer system
Perhaps this is not to brag about, but hey it works and it gets loud if I want.
But no, I want to comment on how cheaply (or not quite so) the entire signal chain (PC -> speakers) can be done "stupid cheap" nowadays. Right now I'm working my bugs out of my new JRMC + ADA8200 DAC. The laptop running things is a <$200 buy. My "main" PC (Pipo X9) lists for about $150. I've not tried a Raspberry PI but they are what, $50? Point: the PC platform is super inexpensive; consider my laptop could even be my media library if I wanted; even a small hard disk is 100's of GB today; mine is mostly empty.
The software (player + DSP + ?) can be free ( = "Linux" ) or reasonable. I am still evaluating JRMC. I'll probably buy it. It costs a whopping $50.
No, the biggest expense in the system is going to be the DAC(s) and amplifiers. Even my disreputable Behringer system (DAC and amps) probably total ~ $800 when new. But a cheap HTR could bring that cost down to maybe $300 or even less.
In review, that means a multi-channel, actively-EQed system is within reach from as cheap as $450 (Windows based) up to say $1200.
OK, at the really low end my $30 Android phone, $20 class-D amp and ... but that won't rattle the rafters
We now conclude this pontification and return you to your regularly scheduled "let's add .01 msec to the midrange and see how it sounds" program
Perhaps this is not to brag about, but hey it works and it gets loud if I want.But no, I want to comment on how cheaply (or not quite so) the entire signal chain (PC -> speakers) can be done "stupid cheap" nowadays. Right now I'm working my bugs out of my new JRMC + ADA8200 DAC. The laptop running things is a <$200 buy. My "main" PC (Pipo X9) lists for about $150. I've not tried a Raspberry PI but they are what, $50? Point: the PC platform is super inexpensive; consider my laptop could even be my media library if I wanted; even a small hard disk is 100's of GB today; mine is mostly empty.
The software (player + DSP + ?) can be free ( = "Linux" ) or reasonable. I am still evaluating JRMC. I'll probably buy it. It costs a whopping $50.
No, the biggest expense in the system is going to be the DAC(s) and amplifiers. Even my disreputable Behringer system (DAC and amps) probably total ~ $800 when new. But a cheap HTR could bring that cost down to maybe $300 or even less.
In review, that means a multi-channel, actively-EQed system is within reach from as cheap as $450 (Windows based) up to say $1200.
OK, at the really low end my $30 Android phone, $20 class-D amp and ... but that won't rattle the rafters
We now conclude this pontification and return you to your regularly scheduled "let's add .01 msec to the midrange and see how it sounds" program
There is a very old plugin Foo_dsp_xover to Foobar. It works and is very easy to use, but it's not supported anymore. You can also use several VST plugins with a VST wrapper for foobar. I have tried this the last few days. But, the VST plugins are made for Synthezisers etc, and not specifically for crossover. It works and it's free, but compared to MiniDSP it's not very userfriendly.
What about soundquality from HDMI - is it lossless or converted to DTS or some other surround format?
Great.
Then something like this could come in handy, if you have a multichannel poweramplifier.
New HDMI to HDMI + Audio Decoder Extractor Splitter HDMI to SPDIF Digital Optical/ Analog 5.1 Support Dolby Digital AC3,DTS,LPCM-in HDMI from Consumer Electronics on Aliexpress.com | Alibaba Group
Then something like this could come in handy, if you have a multichannel poweramplifier.
New HDMI to HDMI + Audio Decoder Extractor Splitter HDMI to SPDIF Digital Optical/ Analog 5.1 Support Dolby Digital AC3,DTS,LPCM-in HDMI from Consumer Electronics on Aliexpress.com | Alibaba Group
From my point of view it makes no sense to take a splitter.
You don't know the quality of this low budged splitter and the dac inside for the analog output.
And to put 8 Channel or 6 Channel PCM (up to 192 khz and 24 bit if wanted) in 5,1 spdif stream makes also no sense.
Better is to go directly in HDMI AVR and use its high quality dac.
Regards
Guenter
You don't know the quality of this low budged splitter and the dac inside for the analog output.
And to put 8 Channel or 6 Channel PCM (up to 192 khz and 24 bit if wanted) in 5,1 spdif stream makes also no sense.
Better is to go directly in HDMI AVR and use its high quality dac.
Regards
Guenter
I have monoblocks and don't intend to get a AVR, so I need either a good DAC/soundcard - or a HDMI splitter.
Thanks for the inspiration, by the way. I have a spare laptop, and have now setup the software and it works. I use windows XP, VSThost, VG asio bridge hifi cable and some different VST plug-ins from reaper and other VST stuf for testing. It's rather straightforward, and can be set-up in an hour the first time (with a good guide).
Thanks for the inspiration, by the way. I have a spare laptop, and have now setup the software and it works. I use windows XP, VSThost, VG asio bridge hifi cable and some different VST plug-ins from reaper and other VST stuf for testing. It's rather straightforward, and can be set-up in an hour the first time (with a good guide).
Bump for a great thread and winner recipe!
However, I am wondering about delay and such if you have for instance a console or other device that you want to use occasionally.
Do you mean different delay for each channel?
This is possible to adjust with Calf x-over for room correction.
Or do you mean the delay in general between starting a playback and hearing it.
in this case there maybe a delay of some ms but I cannot recognize it.
If you use the limiter in calf you have for sure to adjust a delay.
But thats the same for all (also Hardware) limiters.
Regards
Guenter
This is possible to adjust with Calf x-over for room correction.
Or do you mean the delay in general between starting a playback and hearing it.
in this case there maybe a delay of some ms but I cannot recognize it.
If you use the limiter in calf you have for sure to adjust a delay.
But thats the same for all (also Hardware) limiters.
Regards
Guenter
No.
I meant delay when taking audio from analog output of a source to the input of the computer to process the xo and such, when using a gaming console or similar. I tried this on my windows setup and the delay seems to vary a bit and I could not get it right, tried settings up to 174ms but with no success. It was very noticeable. Automatic setup suggested everything from 114 to 174, it was just "wrong". Most auto things on the tv are off, such as image enhancing and so on.
So long story short, try playing guitar hero or similar on a console with audio through the pc for processing, also very irritating on racing games like Forza.
Best regards
Kent
In KXStudio (and so JACK) latency can be adjusted.
First you have to use the lowlatency kernel (is set by default)
Then it is a balance between buffer size and getting xruns.
This will vary for each computer system.
For a fast system buffer can be smaller without getting xruns and so latency shorter.
For my system with a buffer size of 512 samples I don't get any xruns and have a latency of 12 ms. That is okay for my needs.
But I am sure I can take a smaller buffer size without xruns and then shorter latency. I have not tried yet.
regards
Guenter
First you have to use the lowlatency kernel (is set by default)
Then it is a balance between buffer size and getting xruns.
This will vary for each computer system.
For a fast system buffer can be smaller without getting xruns and so latency shorter.
For my system with a buffer size of 512 samples I don't get any xruns and have a latency of 12 ms. That is okay for my needs.
But I am sure I can take a smaller buffer size without xruns and then shorter latency. I have not tried yet.
regards
Guenter
Last edited:
Guenter:
This does not really answer my question, as the measured delay will most likely differ from the numbers you present. Inputs, processing and outputs induce delay. For instance, if I run my Steinberg (Yamaha) UR28M in headless mode (no need for computer to use inputs/outputs) the delay between input/output is 12ms which is very good, when connected to a computer for sw recording and processing purposes this value will increase by a significant amount. The UR28M can re-route audio very fast, but when sw is involved for filtering etc it is a different story. The standard audio jacks on the motherboard seem to have much worse delay for some reason unknown to me, and for example when routing from the motherboard jack input to the UR28M output the delay is so unpredictable I am having troubloe finding a specific value to set permanently.
Anyway.
I just read and re-read manuals on the RX-Vx81 line of receivers from Yamaha. Just sent an email to them regarding a "loophole" in the RX-V681/781, where it seems to be possible to connect passive subs to speaker outputs assigned for "Zone 2", connect input configured to "Zone 2" to the subwoofer outputs, set 24db/oct lpf for sub out, and use the bi-amp function for front speakers with hpf and passive xo for tweeter to create a 3way setup fully compatible with the YPAO function of the receiver!
And not only that! But if this works (eagerly awaiting response from Yamaha), all the inputs of the receiver will work without concern of any re-routing of audio! With a remote! And it says 110db s/n ratio or greater in the manual.
One box to conquer them all!
I'm not really a Yamaha fan, but maybe I'm becoming one pending on their answer.
Sorry about derailing the thread a bit.
Best regards.
Kent
This does not really answer my question, as the measured delay will most likely differ from the numbers you present. Inputs, processing and outputs induce delay. For instance, if I run my Steinberg (Yamaha) UR28M in headless mode (no need for computer to use inputs/outputs) the delay between input/output is 12ms which is very good, when connected to a computer for sw recording and processing purposes this value will increase by a significant amount. The UR28M can re-route audio very fast, but when sw is involved for filtering etc it is a different story. The standard audio jacks on the motherboard seem to have much worse delay for some reason unknown to me, and for example when routing from the motherboard jack input to the UR28M output the delay is so unpredictable I am having troubloe finding a specific value to set permanently.
Anyway.
I just read and re-read manuals on the RX-Vx81 line of receivers from Yamaha. Just sent an email to them regarding a "loophole" in the RX-V681/781, where it seems to be possible to connect passive subs to speaker outputs assigned for "Zone 2", connect input configured to "Zone 2" to the subwoofer outputs, set 24db/oct lpf for sub out, and use the bi-amp function for front speakers with hpf and passive xo for tweeter to create a 3way setup fully compatible with the YPAO function of the receiver!
And not only that! But if this works (eagerly awaiting response from Yamaha), all the inputs of the receiver will work without concern of any re-routing of audio! With a remote! And it says 110db s/n ratio or greater in the manual.
One box to conquer them all!
I'm not really a Yamaha fan, but maybe I'm becoming one pending on their answer.
Sorry about derailing the thread a bit.
Best regards.
Kent
Kent:
I tried to simulate your situation to find a delay and lipsync value.
In my equipment also is a sound interface "Roland Edirol UA-25".
This I connected by USB to my KXStudio setup (PC1).
The UA-25 has a optical (TOS link) input port.
This port I connected to a second computers (PC2) soundcard.
I searched for a lipsync frame on youtube.
I played for example this stream on PC2:
https://www.youtube.com/watch?v=s_PbyRpKrRk
I realy could find no subjective delay by playing this or also other test frames or videos.
In this case the whole connection chain was digital up to the AV-Receiver.
Regards
Guenter
I tried to simulate your situation to find a delay and lipsync value.
In my equipment also is a sound interface "Roland Edirol UA-25".
This I connected by USB to my KXStudio setup (PC1).
The UA-25 has a optical (TOS link) input port.
This port I connected to a second computers (PC2) soundcard.
I searched for a lipsync frame on youtube.
I played for example this stream on PC2:
https://www.youtube.com/watch?v=s_PbyRpKrRk
I realy could find no subjective delay by playing this or also other test frames or videos.
In this case the whole connection chain was digital up to the AV-Receiver.
Regards
Guenter
Loafmeat said:
In my case it was an analog chain. So there must be several conversions along the line.
But in your example with a digital chain:
Good. I`ll take your word for it and assume that my problem is the analog chain, the source or m$, maybe all of the above.
Thank you for your time.
Reviving an old - but interesting - 3ad 'cause i'm approaching to multichannel amplification.
François Bourdon, the foo_dsp_xover author, replied that:
So I collected some interesting links that may inspire:
Well the easiest/cheapest way (for listening digital-only music) is to use an HT amplifier, driven by a PC trough HDMI...
François Bourdon, the foo_dsp_xover author, replied that:
So I collected some interesting links that may inspire:
- Status
- This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.