Following on from this. I've got a unit preordered and am intending using it for an LX521 build. I would like to be able to process upto 24/192 for 8 channel output at the same rate. Looking at the spec (p36) maximum instructions at 192kHz sample rate is 1536 per sample.
Am I right to be concerned that there might not be enough taps per channel?
Would quantization be an issue? I am concerned that with 1536 per channel that this solution might not be optimal for my application, and I might be better either building the ASP (zero quantization error), or going down the route of BruteFIR on a raspberry pi, Acourate or pick up a more powerful evaluation board.
Anyone have information on what tap lengths are needed, without downsampling?
I'm new to DSP, thanks.
Am I right to be concerned that there might not be enough taps per channel?
Would quantization be an issue? I am concerned that with 1536 per channel that this solution might not be optimal for my application, and I might be better either building the ASP (zero quantization error), or going down the route of BruteFIR on a raspberry pi, Acourate or pick up a more powerful evaluation board.
Anyone have information on what tap lengths are needed, without downsampling?
I'm new to DSP, thanks.
Is stripped down version for PA
2 in 2 out / 2 in 4 out or 4 in 4 out
with Display and Rotary Encoder coming soon or already available ?
regards
Yes please! Would be handy to have them published
Sorry for the delay, here it is attached. I tried to rename the project files but ran into a bunch of issues with library dependencies - for anyone reading this, despite the file names these are NOT the source files for Addon A - go to the main Aurora github for these.
I have tested the optical input and USB input on this board. The optical output is UNTESTED. Please use at your own risk!
There are also some earlier versions with XLR connections.
I’ve just been informed that my order has shipped - is anyone able to answer my question regarding if this board is able to perform at 192kHz if the sample rate is 1536 per sample if I’m using 8 channels?
I am a bit confused about your question. It seems that you mix things up.
Quantization/quantization error is not related to sample rate. It depends on the bit depth. Thus, there is no difference of a system with 48kHz and 192kHz if both run at 24bit resolution and with the same quantization algorithm.
The maximum number of taps per FIR is constant. Therefore, samplerate defines maximum length of the filter impulse response in milli seconds. E.g. a filter with 8192 taps is 171ms long @ 48kHz. Now doubling the sample rate half the IR length: 8192taps @96kHz are 86ms and @192kHz 43ms.
The taps number and the sample rate determine the frequency resolution. With 96kHz and 192kHz you get much more frequency taps but many of them fall outside the audible frequency band and the resolution at lower frequencies comes less if the tap number is held constant. Therefore, FIRs at higher sample rates tend to have higher tap count to improve the low frequency resolution. Yet you still have a lot of taps outside the audible range. That is the price you pay.
Finally, the 1536 are the instructions per sample. This is a bit misleading. The ADAU1452 is a SIMD device. One instruction executes multiple operations in one clock cycle.
Raphael
Quantization/quantization error is not related to sample rate. It depends on the bit depth. Thus, there is no difference of a system with 48kHz and 192kHz if both run at 24bit resolution and with the same quantization algorithm.
The maximum number of taps per FIR is constant. Therefore, samplerate defines maximum length of the filter impulse response in milli seconds. E.g. a filter with 8192 taps is 171ms long @ 48kHz. Now doubling the sample rate half the IR length: 8192taps @96kHz are 86ms and @192kHz 43ms.
The taps number and the sample rate determine the frequency resolution. With 96kHz and 192kHz you get much more frequency taps but many of them fall outside the audible frequency band and the resolution at lower frequencies comes less if the tap number is held constant. Therefore, FIRs at higher sample rates tend to have higher tap count to improve the low frequency resolution. Yet you still have a lot of taps outside the audible range. That is the price you pay.
Finally, the 1536 are the instructions per sample. This is a bit misleading. The ADAU1452 is a SIMD device. One instruction executes multiple operations in one clock cycle.
Raphael
Thanks for taking your time to answer this - my experience with DSP is very low, so appologies that I need some education.
What I basically want to know is if there is sufficient processing power available in the Aurora DSP to implement the attached acourate example configuration, at 192/24 bit in and out for a pair of four way open baffle speakers whilst satisfying the Nyquist sampling theorem? I do not know what to look for or how to determine if the the specifications of Aurora DSP are sufficient for this and that the functions that I need are present... that's the crux of my question.
p.s. I decided to download the mac version of Audverion control. Nice and easy to use and at my skill level
. I notice that gain for a partiqular channel cannot be set to +10dB - is there reason why gain can only be reduced?Attachments
@stretchneck I don't really understand the output from acourate but to me it looks like that it basically configures IIRs only. That you can do with Aurora. For FIRs it would be important to know the filter length (taps).
Gain can only be reduced because 0dB is full scale output for master volume and I wanted to keep it the same on all gain/volume settings.
@jcga Yes, there is currently much progress on the new web app. You can track it here: Commits * freeDSP/freeDSP-aurora * GitHub
Currently some friends are testing the alpha builds. Those of you how are curious and want to get an idea of it can test it, too. Here you can find a document how to install such a preview version:
freeDSP-aurora/AN001 Firmware Update EN.pdf at feature/wepapp * freeDSP/freeDSP-aurora * GitHub
I think in the next few days the first beta release will be published followed by a final release soon.
Raphael
Gain can only be reduced because 0dB is full scale output for master volume and I wanted to keep it the same on all gain/volume settings.
@jcga Yes, there is currently much progress on the new web app. You can track it here: Commits * freeDSP/freeDSP-aurora * GitHub
Currently some friends are testing the alpha builds. Those of you how are curious and want to get an idea of it can test it, too. Here you can find a document how to install such a preview version:
freeDSP-aurora/AN001 Firmware Update EN.pdf at feature/wepapp * freeDSP/freeDSP-aurora * GitHub
I think in the next few days the first beta release will be published followed by a final release soon.
Raphael
The first official beta version for the new web app is pubislhed. You can find it here:
Release v2.0.0-beta.1 * freeDSP/freeDSP-aurora * GitHub
Please read the README.md and the Application Note AN001 (freeDSP-aurora/DOCUMENTATION at develop * freeDSP/freeDSP-aurora * GitHub) for how to install the new web app on your board.
The web app should not have any trouble with connection caused by Windows, anti virus software, firewalls etc. pp. as long as your web browser can access your WIFI. You need to enable JavaScript of course.
Please note, so far only Safari and Firefox have been tested. Chrome was reported working, too.
Raphael
Release v2.0.0-beta.1 * freeDSP/freeDSP-aurora * GitHub
Please read the README.md and the Application Note AN001 (freeDSP-aurora/DOCUMENTATION at develop * freeDSP/freeDSP-aurora * GitHub) for how to install the new web app on your board.
The web app should not have any trouble with connection caused by Windows, anti virus software, firewalls etc. pp. as long as your web browser can access your WIFI. You need to enable JavaScript of course.
Please note, so far only Safari and Firefox have been tested. Chrome was reported working, too.
Raphael
Hi,
I received my Aurora DSP, but i have no idea how to use this!
Only the board, Wi-Fi antenna, and power supply cables are included. Nothing else.
I read the manual, and there is nothing on the type of connectors that's needed to hook up. If we don't want to use the add ons, how can we use this board?
Please consider copying what ICEpower does for their manuals. They spell out every connector type for every pin out on the board, and they're for commercial use where you'd think the engineers would already know this. DIYers on average are far less knowledgeable, and will greatly benefit from more instructions
Please help on getting started. Thanks
I received my Aurora DSP, but i have no idea how to use this!
Only the board, Wi-Fi antenna, and power supply cables are included. Nothing else.
I read the manual, and there is nothing on the type of connectors that's needed to hook up. If we don't want to use the add ons, how can we use this board?
Please consider copying what ICEpower does for their manuals. They spell out every connector type for every pin out on the board, and they're for commercial use where you'd think the engineers would already know this. DIYers on average are far less knowledgeable, and will greatly benefit from more instructions
Please help on getting started. Thanks
Last edited:
I'm not official support for this, but are you sure you read the User Manual and not just the Getting Started Guide?
If you visit their website, there's a very prominent link to their GitHub page, which has all the documentation for this project. There's a User Manual with all the pin-outs and exact part numbers for all the connectors used so you can order them as needed. In addition, there are full CAD models, chassis panel layouts, full schematics, Kicad source files, gerber files, and even full source code available that should be able to answer any questions you have.
Do note that this board is mainly designed for people already familiar with DIY Digital Audio, so if your primary experience has been with kit amplifiers like ICEpower then something like a DSP might be a bit overwhelming with the number of connections and functionality. If you can describe your application I and other users of this forum might be able to help point you in the right direction.
If you visit their website, there's a very prominent link to their GitHub page, which has all the documentation for this project. There's a User Manual with all the pin-outs and exact part numbers for all the connectors used so you can order them as needed. In addition, there are full CAD models, chassis panel layouts, full schematics, Kicad source files, gerber files, and even full source code available that should be able to answer any questions you have.
Do note that this board is mainly designed for people already familiar with DIY Digital Audio, so if your primary experience has been with kit amplifiers like ICEpower then something like a DSP might be a bit overwhelming with the number of connections and functionality. If you can describe your application I and other users of this forum might be able to help point you in the right direction.
I'm not official support for this, but are you sure you read the User Manual and not just the Getting Started Guide?
If you visit their website, there's a very prominent link to their GitHub page, which has all the documentation for this project. There's a User Manual with all the pin-outs and exact part numbers for all the connectors used so you can order them as needed. In addition, there are full CAD models, chassis panel layouts, full schematics, Kicad source files, gerber files, and even full source code available that should be able to answer any questions you have.
Do note that this board is mainly designed for people already familiar with DIY Digital Audio, so if your primary experience has been with kit amplifiers like ICEpower then something like a DSP might be a bit overwhelming with the number of connections and functionality. If you can describe your application I and other users of this forum might be able to help point you in the right direction.
Thank you. Now I see the connector types and stuff.
I'm planning on using this for a small active speaker, one Aurora inside each speaker. It will be 4 way, so it'll take an input, and output 4 balanced channels to the 4 drivers to balanced input amps.
Hello, I am currently looking for a 4-channel crossover product. I am glad to see your product here. The mp-dsp is currently consistent with your hardware solution. HYPEX in software may not go too far. So have you done some optimizations in terms of JITTER and master clock? DAC can be made modular, or updated to a slightly better output chip such as PCM1798, or you found that the signal quality of the DSP output does not need a better DAC? Hope to get a piece for testing. Of course, I have to pay the bill.
HYPEX is comfortable in three-way projects.
But in the quarter-way project, I wanted to smash it
The new firmware just released is slightly better，At least it can MUTE a channal.
Hi dspverden
While making some connection changes, I was careless & I believe I shorted something with a stray +12v wire. I see no physical sign of damage but I did see a small spark & now when I apply 12v I hear a small whining sound coming from the board itself (with no connections made to the board apart from the 12v power.
Is there any way for me to determine the extent of the damage I've done to this board?
While making some connection changes, I was careless & I believe I shorted something with a stray +12v wire. I see no physical sign of damage but I did see a small spark & now when I apply 12v I hear a small whining sound coming from the board itself (with no connections made to the board apart from the 12v power.
Is there any way for me to determine the extent of the damage I've done to this board?
Hi,
I did not make any optimizations for Jitter and master clock except routing it carefully on the PCB keeping in mind the best practice advises (avoiding stubs etc. pp.).
The DAC from AKM I have selected because it offers very good performance. It even outperforms most of the stereo chips and this thing is 8 channels!
You are free to upgrade to another DAC. You can use the freeDSP-Expansion header for this.
Raphael
Ouch. That's not good to hear.
You can do the following: Measure the voltages on each LDO in SOT case. They should all be either +5V or +3V3. This way you will find out the voltage branch that you damaged.
Reminder for all: As stated in the manual: Always do your wiring with all power off and double check every connection!
Thanks for your quick reply dspverden.
IC106 measures 1.4v. The rest are correct (either 5v or 3.3v)
Thanks