Open Source DSP XOs - Page 37 - diyAudio
Go Back   Home > Forums > Source & Line > Digital Line Level

Digital Line Level DACs, Digital Crossovers, Equalizers, etc.

Please consider donating to help us continue to serve you.

Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving
Reply
 
Thread Tools Search this Thread
Old 20th September 2012, 09:15 AM   #361
diyAudio Member
 
abraxalito's Avatar
 
Join Date: Sep 2007
Location: Hangzhou - Marco Polo's 'most beautiful city'. 700yrs is a long time though...
Blog Entries: 109
Send a message via MSN to abraxalito Send a message via Yahoo to abraxalito Send a message via Skype™ to abraxalito
I doubt very much they're unable - more unwilling to. I guess they see peripherals as the way that customers differentiate their ARM offerings. Yet I think it would be a good thing if ARM was more proactive in peripheral design - at least for the more 'core' stuff like serial, because then we might see more innovation in other areas - like NXP's SPIFI and SGPIO for instance. NXP's SSP is ARM's design but few other vendors use that (TI does I think, STM doesn't). An ARM designed I2S would be a boon.
__________________
Seek not the favour of the multitude...rather the testimony of few. And number not voices, but weigh them. - Kant
The capacity for impartial observation is commonly called 'cynicism' by those who lack it.
  Reply With Quote
Old 13th October 2012, 05:57 PM   #362
diyAudio Member
 
Join Date: Jun 2009
Minor update: the LPC4310, 4320, 4330, and 4350 parts' status has transitioned to production and the datasheet was revised 10/11. Nothing really of note unless one happens to be targeting the 180 BGA where some of pins moved but there's a bit more detail on power consumption that's interesting for regulator sizing and such (the SGPIO peripheral's draw is still <tbd> but it's probably a few hundred microamps). Looks like I can run my XO+EQ with about 20mA. Versus 17mA for the SPDIF receiver and 16mA for the DAC's digital side. And 120mA typ for DAC analog and the output buffers. Darn op amps.

LPCXpresso 4.3.0 is out too. But I've been busy with layout verification and have to get code for Q31 filter bank synthesis together so haven't updated---there'll probably be another update by the time I get back to firmware.
  Reply With Quote
Old 14th October 2012, 01:12 AM   #363
diyAudio Member
 
Join Date: Jun 2009
Quote:
Originally Posted by twest820 View Post
Minor update: the LPC4310, 4320, 4330, and 4350 parts' status has transitioned to production and the datasheet was revised 10/11. Nothing really of note unless one happens to be targeting the 180 BGA where some of pins moved but there's a bit more detail on power consumption that's interesting for regulator sizing and such (the SGPIO peripheral's draw is still <tbd> but it's probably a few hundred microamps). Looks like I can run my XO+EQ with about 20mA. Versus 17mA for the SPDIF receiver and 16mA for the DAC's digital side. And 120mA typ for DAC analog and the output buffers. Darn op amps.

LPCXpresso 4.3.0 is out too. But I've been busy with layout verification and have to get code for Q31 filter bank synthesis together so haven't updated---there'll probably be another update by the time I get back to firmware.
what are you making ?
  Reply With Quote
Old 14th October 2012, 03:13 PM   #364
diyAudio Member
 
Join Date: Jun 2009
An embedded replacement for my PC XO and EQ. At the moment it's a WM8805+LPC4300 in LQFP144 (the 4310, 4320, or 4330 are all available in this package and are pin compatible)+CS4565 SPDIF->DSP->DAC board. The BOM's been through several iterations and might change again but it looks fairly well converged at this point.
  Reply With Quote
Old 24th October 2012, 05:36 AM   #365
diyAudio Member
 
Join Date: Jun 2009
Thought Iíd mention Iíve implemented equivalents of arm_biquad_cascade_df1_q31 (32 bit fixed point samples with 64 bit accumulate), arm_biquad_cas_df1_32x64_q31 (32 bit fixed point samples with 64 bit accumulate and feedback), and a higher precision version that would be probably be called arm_biquad_cas_df1_64x64_q31 (32 bit fixed point samples with 64 bit accumulate, coefficients, and feedback). Iíve been using the code on a PC to do some of the numerical characterization Abraxalito, Steph, and I were discussing over on the IIR Lab thread and have found that, chosen appropriately, the filters hit the quantization noise floor of 24 bit audio. This is comfortably below the noise floor of an ES9012 DAC in mono.

As a rule of thumb, for redbook audio Q31 is sufficient for filters centered at 1kHz and above, Q31_32x64 is sufficient to around 200Hz, and Q31_64x64 is a good choice below that. Back of the envelope Iím estimating my relatively complex 38 biquad three way XO+EQ would require a Cortex M4 core running at 60MHz. Scale accordingly for higher sample rates.

If you want to do your own investigations grab the latest Cross Time DSP commit and have look at the unit tests to get started.
  Reply With Quote
Old 1st November 2012, 10:44 AM   #366
diyAudio Member
 
Join Date: Apr 2003
Location: Tampere Finland Europe
Quote:
Originally Posted by twest820 View Post
An embedded replacement for my PC XO and EQ. At the moment it's a WM8805+LPC4300 in LQFP144 (the 4310, 4320, or 4330 are all available in this package and are pin compatible)+CS4565 SPDIF->DSP->DAC board. The BOM's been through several iterations and might change again but it looks fairly well converged at this point.
You mean CS4365 DAC? Do you have all of them (SPDIF,DSP and DAC) on the same board or separate ones? I think modular design would suite best for people who like to use different components (SPDIF receiver, ASRC, clock, DAC's etc). How are you clocking the DAC? Is it interface using SGIOP? I'd like to use the Crystal DAC as well but also with XMOS and it's I2S lib need you have an external clock (24.576 MHZ for an example) feeding XMOS which is the master and generates other clocks. With XMOS it's also easy to use ADAT interface as output.
  Reply With Quote
Old 2nd November 2012, 03:38 AM   #367
diyAudio Member
 
Join Date: Jun 2009
Yes, 4365; 4565 is a typo. SPDIF, DSP, and DAC are all on the same board with one of the WM8805's clock outputs going to the CS4365 and the other to the LPC4300. I'm using a mix of the 4300's I2S and SGPIO peripherals and the preliminary schematic includes an expansion header with duplex clocking, I2S receive, and the seven SGPIO lines that aren't used for driving the 4365.

So it's not modular per se, but if one wants to patch in an ADC, different multichannel DAC, alternate clock, or pretty much any arbitrary I2S source or sink it's not particularly difficult.
  Reply With Quote
Old 2nd November 2012, 09:30 AM   #368
diyAudio Member
 
Join Date: Apr 2003
Location: Tampere Finland Europe
Quote:
Originally Posted by twest820 View Post
So it's not modular per se, but if one wants to patch in an ADC, different multichannel DAC, alternate clock, or pretty much any arbitrary I2S source or sink it's not particularly difficult.
So you haven't been considering adding a sample rate converter? Anyway, the LPC4300 looks interesting because of the SGPIO, using multiple I2S ports feels complicated because separate clocks are needed for both interfaces.

Btw. is the 4300 pinout compatible with 2294 for an example, there are these modular (empty) boards I'd like to use:

Empty ARM MCU Cards for NXP LPC devices - mikroElektronika

E: To answer myself they aren't compatible.

Last edited by mhelin; 2nd November 2012 at 09:54 AM.
  Reply With Quote
Old 2nd November 2012, 12:28 PM   #369
diyAudio Member
 
Join Date: Apr 2003
Location: Tampere Finland Europe
Looked at the LPC4330-Xplorer schematics. The board has onboard UDA1380 codec /A/D SNR 97 dB, DAC 100 dB) using I2S0 but the I2S1 pins are used by the flash. The SGPIO pins are supposed to be exported to the headers. So it can be used for developing the software and DAC at least.
  Reply With Quote
Old 2nd November 2012, 01:08 PM   #370
diyAudio Member
 
Join Date: Jun 2009
Quote:
Originally Posted by mhelin View Post
So you haven't been considering adding a sample rate converter?
Considered and rejected based on ABX results (and some maths). The time domain degradation from resampling is inversely proportional to the change in sample rate and proportional to the steepness of the antializasing filter. I found 4x resampling with a slow rolloff was OK, but preserving the original bit rate was subjectively preferable. It can also be computationally preferable; for biquads CPU scales rather like O(N log N) with sample rate due to the increased precision needed when the filters are operated at low normalzied frequency.
  Reply With Quote

Reply


Hide this!Advertise here!
Thread Tools Search this Thread
Search this Thread:

Advanced Search

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Trackbacks are Off
Pingbacks are Off
Refbacks are Off


Similar Threads
Thread Thread Starter Forum Replies Last Post
Volume / Source selector - open source project ? AuroraB Analog Line Level 22 22nd September 2012 03:21 PM
Violet DSP Evolution - an Open Baffle Project cuibono Multi-Way 211 18th May 2010 03:26 AM
Open call for suggestions on Open Source DIY Audio Design gfergy Everything Else 1 15th April 2007 08:33 AM
Open Source, Open Architecture! zenmasterbrian Digital Source 185 23rd February 2007 11:35 PM


New To Site? Need Help?

All times are GMT. The time now is 11:02 PM.


vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2014 DragonByte Technologies Ltd.
Copyright ©1999-2014 diyAudio

Content Relevant URLs by vBSEO 3.3.2