|
|||||||
| Home | Forums | Rules | Articles | Store | Gallery | Blogs | Register | Donations | FAQ | Calendar | Search | Today's Posts | Mark Forums Read | Search |
| 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 |
|
|
|
Thread Tools | Search this Thread |
|
|
#21 | |
|
diyAudio Member
Join Date: Sep 2004
Location: Montevideo
|
Quote:
It works at 8 bit resolution, what means roughly 48 dB of dynamic range and about the same (roughly) in terms of S/N ratio. This certainly fits well many applications such as practice amplifiers and similar, but falls rather short of the S/N acceptable for live performance or recording. There is no attempt to process signals in the sense of scaling / mixing. It is limited to time delay, time reversal, pitch shifting and so on. Again this may be more than enough for many people. This is quite OK at 8 bits, for any further attempt to make calculations not only will probably exceed the speed capabilites of the processor, but also introduce round off errors, being each lost bit about a 6 dB loss in S/N. As reference, consumer audio works at 16 bits, and DSP processing mandates a headroom of several extra bits to accomodate truncation. This is why usually 32 bit processors are called for this, better yet if they can handle floating point math. Yet, this is truly a DSP example in the sense it works along the same basic philosophy. On another line, I received today the TI samples (FIFOs, ADCs and DACs) and must now figure out how to handle 0.5 mm pin pitch devices Most probably I will start with a delay building block comprising A/D, FIFO delay and DA. Once I have this in line I will combine them along the idea posted in the previous note. Will keep informed on progress. Rodolfo |
|
|
|
|
|
#22 |
|
diyAudio Member
|
Ingrast, try this to handle the 0.5mm pin pitch: http://members.lycos.nl/anthonyvh/index.php?page=smd
I hope that helps! I'm eager to see results. I wish I could do things like this. Well, maybe in 3-5 years
__________________
Website: http://members.lycos.nl/anthonyvh |
|
|
|
|
#23 |
|
diyAudio Member
Join Date: Sep 2004
Location: Montevideo
|
Hey Devil_h@ck!
You've got excellent material there!! The soldering part I was not really worried about. The PCB manufacturing was, and your page I found excellent. The process I use is slightly different, with bare copper clad and dry film hot pressed. I use soda ash developer and NaOH for stripping. Problem is it gets really hard to have good repetitive control of all parameters so as to have consistent quality fine resolution. The idea of using 2 piggybacked masks is cute! My Epson 1520 makes good artwork on transparency but much better on paper as you noted. That you can get good SMD quality with basically the same tools is encouraging, will keep you informed. Rodolfo |
|
|
|
|
#24 |
|
diyAudio Member
|
Ingrad, I'm glad that I can help a tiny bit
__________________
Website: http://members.lycos.nl/anthonyvh |
|
|
|
|
#25 |
|
diyAudio Member
Join Date: Sep 2004
Location: Montevideo
|
I devoted some hours last weekend to work on the digital delay project.
Here is a draft design for the basic delay building block. NOTE: This design is untested, being only the glue logic digitally simulated to make sure no big blunder was lurking before committing to actual hardware construction. As such there may - most probably there are - undetected bugs to be found and corrected. Brief description. Process starts with conversion by U1, PCM1802 24 bit stereo A/D chip. This device generates signals: BCK: Clock signal for downstream devices to synchronize with data. FSYNC:Frame synchronization signal, marks the beginning of a serial frame LRCK: Left/right channel flag to indicate which one is being outputted. DOUT: Actual data stream. The output stream is 24 bit I2S format selected by pins FMT0 and FMT1. Timing is derived from a master clock running at 256x the sampling frequency. Varying the master clock between 8.19 and 24.57 MHz modifies the actual sampling rate from 32 ksamples/s to 96 ksamples/s, which in combination with a fixed 1024 or 4096 sample delay introduced by FIFO U2, yields the variable time delay. DOUT is sent to the input of IC5/6 forming a 16 bit shift register. We settle to 16 bit actual word length to simplify hardware though U1 provides 24 bits and U2 can handle 18 bits. LRCK and BCK are combined by IC1A to enable (EN_SHFT) the shift clock. Any one Low resets 4 bit counter IC2A to all zeros. IC3A detects the arrival of the 15th pulse (BIT15) and IC7A delays this signal a further period (BIT16) to enable shifting in the 16th most significant data bit. The delayed signal inhibits further shift clock pulses, freezing IC5/6 to present a parallel 16 bit word corresponding to the A/D most significant 16 output bits. IC7B delays BIT16 a further clock period to accommodate a gating signal for writing the FIFO. FIFO writing is accomplished by pulling down input WEN to which end the inverted combination of FSYNC and LRCK (IC11A) comes handy. Output of gate IC8B (WCLK) is a negative going half cycle of BCK, in whose rising edge data is actually written to U2. For the first 1024 or 4096 frames, the FIFO fills up with data until it becomes full. Output FF goes low once the last available storage position is filled, and we gate an inverted BCK half cycle with FF (IC11B/C) to generate RCLK. On the positive transition of RCLK, the word written 1024 or 4096 frames back is outputted by U2, while this very read operation frees one place thus rising the FF signal back to idle (High). From here on, FF cycles once each audio frame. Signal LRCK comes handy to gate the FIFO output word into IC8/9 forming a parallel load shift register, we use the same BCK for shifting. When LRCK goes up, IC8/9 latch the 16 bit word and begin shifting it out. The shift register cascade serial input is grounded, so after the first 16 bits just read from the FIFO are shifted out, zeroes follow for the D/A to read, emulating zero value bits up to the 24th it expects. The clock may be left running, for the D/A takes care for itself once it has read the needed bits. A multiplexer might have been used here instead of the shift register with the same effect for serializing with a little different logic, probably a bit simpler but I had already ordered the S/R's so this is it for the time being. The DSD1791 24 bit stereo D/A converter is a complex chip programmable through a serial control interface. Fortunately it is designed to operate stand alone with reasonable defaults, one of which is the I2S 24 bit data format. It is driven by the same BCK, LRCK and FSYNC signals generated by U1, only it uses delayed input data for conversion. Master clock generator details have not yet been worked out, but I plan to use a TLC2932 PLL which provides a control range adequate for varying the frequency from 8.19 to 24.57 MHz as required. Superimposing a low frequency signal on the control voltage will provide sort of tremolo effect. Since both the A/D and D/A are stereo chips, I plan to use one half of each for a long 4096 stage delay and for a shorter 1024 stage one. Right now I will focus on manufacture of carrier boards for the smallest SMD devices. This carrier boards will have DIP style rows of .1 pitch holes to make is easier for the motherboard to design and construct. All devices are available as free samples from Texas Instruments. I highly recommend to register there if you are interested in building prototype systems like this. Will keep posted (hopefully with photographs) of further advances. Rodolfo |
|
|
|
|
#26 |
|
diyAudio Member
|
I'm ancious to see/hear results
__________________
Website: http://members.lycos.nl/anthonyvh |
|
|
|
|
#27 |
|
diyAudio Member
Join Date: Sep 2004
Location: Montevideo
|
I managed to devote some hours last weekend to the project, here are the reults so far.
I tried with encouraging results fabrication of fine line carrier boards for the smallest SMD's. I am confident with some time I will have the little beasts tamed enough to build the prototype. I finished the basic module design, here is the schematic for one delay block and A/D - D/A conversion. the next post is the PCB, which I designed as single sided with lots of vertical jumpers and that can be latter converted to double sided. This module includes conversion, FIFO and clocking. Provision for a second delay module is included in expansion connector SV3 to export and import the second audio channel generated / received from the stereo A/D - D/A chips. Connector SV1 provides access to plug a pot (10k is fine) for clock speed adjustment. SV2 is the audio input and SV4 the output. Separate +3.3 VCC, +5 and -5V must be supplied as well as separate digital and analog grounds. This module should be integrated with an analog one where actual combination of signals is performed. Hope to build it next week or the other, will keep posted. Rodolfo |
|
|
|
|
#28 |
|
diyAudio Member
Join Date: Sep 2004
Location: Montevideo
|
Somehow I did not upload the file, here it is, compressed so the server does not complain
|
|
|
|
|
#29 |
|
diyAudio Member
Join Date: Sep 2004
Location: Montevideo
|
And here is the board.
|
|
|
|
|
#30 |
|
diyAudio Member
|
Keep up the good work
__________________
Website: http://members.lycos.nl/anthonyvh |
|
|
| Currently Active Users Viewing This Thread: 1 (0 members and 1 guests) | |
| Thread Tools | Search this Thread |
|
|
|
|
||||
| Thread | Thread Starter | Forum | Replies | Last Post |
| echo, reverb using bucket brigade delay lines from panasonic | Workhorse | Solid State | 20 | 6th April 2011 01:37 AM |
| Digital reverb help | spirod | Digital Source | 0 | 11th October 2006 07:12 AM |
| Digital Reverb ! | -_nando-_ | Digital Source | 5 | 11th April 2006 06:29 PM |
| digital delay | Jack Thomas | Digital Source | 1 | 1st August 2002 03:02 AM |
| digital reverb | JBL | Digital Source | 1 | 8th October 2001 11:14 PM |
| New To Site? | Need Help? |
| Page generated in 0.14218 seconds (81.86% PHP - 18.14% MySQL) with 11 queries |