About 30 years ago, before every computer had CD quality audio built in, audio cards and chips were technological marvels. MIDI chips, FM synthesis, and synths on a chip reigned supreme but one little device – just a handful of resistors – sounded fantastic. it was the Covox Speech Thing, a simple resistor ladder wired up to the parallel port of a computer that would output 8-bit audio to an external amplifier. [FK] recently built his own Covox (Czech, Google translatrix) with just 18 resistors, and the results sound fantastic.
Instead of fancy chips, the original Covox Speech Thing used the 8 bit parallel port on a PC. Back in the olden days, this was the fastest way to get digital data out of a computer, but since it was digital only, a DAC was required to turn this into audio. A simple resistor ladder was sufficient, and this hardware was eventually supported by the old DOS games from Sierra and Id.
[FK] has a demo of this LPT DAC available here, but we’re not thinking that link will last long. If anyone has a better link, leave a note in the comments and we’ll update this post. Thanks [beavel] for sending this in.
Audiophiles tend to put analog systems on a pedestal. Analog systems can provide great audio performance, but they tend to be quite costly. They’re also hard to tinker with, since modifying parameters involves replacing components. To address this, [tshen2] designed the DSP 01.
The DSP 01 is based around the Analog Devices ADAU1701. This DSP chip includes two ADCs for audio input, and four DACs for audio output. These can be controlled by the built in DSP processor core, which has I/O for switches, buttons, and knobs.
[tshen2]’s main goal with the DSP 01 was to implement an audio crossover. This device takes an input audio signal and splits it up based on frequency so that subwoofers get the low frequency components and tweeters get the higher frequency components. This is critical for good audio performance since drivers can only perform well in a certain part of the audio spectrum.
Analog Devices provides SigmaStudio, a free tool that lets you program the DSP using a drag-and-drop interface. By dropping a few components in and programming to EEPROM, the DSP can be easily reconfigured for a variety of applications.
While FPGAs get all the credit for being the hip new thing, they are inherently digital devices. Without a proper ADC and DAC, you won’t be delving into the analog domain with your programmable logic. Maxim has just put out a chip that does just that: an analog swiss army knife with 20 pins that are configurable as analog to digital converter, digital to analog converters, GPIO, or any mix of the above.
The MAX11300 includes twenty IO ports, each capable of becoming an ADC, DAC, or GPIO, with pairs of ports capable of being configured as a logic level translator or an analog switch. The ADCs and DACs are 12-bit, with input and output ranges from -10V to +10V.
As a nice little bonus, the chip is controlled over SPI, making this an interesting device for a small “do anything analog” tool we’re sure will hit Tindie or Seeed Studio before the year is out. Luckily for whoever would create such a device, Maxim has a nice GUI for configuring each of the 20 pins on their chip, Of course Maxim already offers an evaluation kit for the MAX11300. It’s $100 USD and is Windows only.
The MAX11300 is available in either 40-pin TQFN or 48-pin TQFP packages (with the larger, easier to solder TQFP shipping later) for about $5.80 USD in quantity 1000, or $11.37 in quantity one.Video below showing off the MAX11300 reading and writing analog values to a few pins, and a good look at the configuration software.
Continue reading “The Analog Swiss Army Knife”
Some projects are both educational and useful. We believe that [Jasper’s] Arduino based electronic load is one of those project.
[Jasper’s] electronic load can not only act as a constant current load, but also as a constant power and constant resistive load as well. The versatile device has been designed for up to 30V, 5A, and 15W. It was based on a constant current source that is controlled by a DAC hooked up to the Arduino. By measuring both the resulting voltage and current of the load, the system can dynamically adapt to achieve constancy. While we have seen other Arduino based constant loads before, [Jasper’s] is very simple and straight forward compartively. [Jasper] also includes both the schematic and Arduino code, making it very easy to reproduce.
There are tons of uses for a voltage controlled current source, and this project is a great way to get started with building one. It is an especially great project for putting together your knowledge of MOSFET theory and opamp theory!
Before the days of iPod docks in every conceivable piece of audio equipment, most devices were actually built very well. Most shelf top equipment usually came with well designed circuits using quality components, and late 90s CD players were no exception. [Mariosis] heard of some very nice DACs found in some of these units and decided to take one out for a spin. He’s using a Raspberry Pi to play audio with the DAC found in a late 90s Kenwood CD player.
After fortune favored a CD player with a dead drive on [Mariosis]’ workbench, he dug up the service manual and found some interesting chips – a PCM56 DAC, a little bit of logic, and an SM5807 oversampling chip that does all the conversion for the DAC.
This oversampling chip uses an I2S – not I2C – bus to carry the data from the CD to the DAC. There is, of course, an I2S driver for the Raspi, but the first attempts at playing audio didn’t result in anything. It turned out there was a problem with what the oversampler expected – the ‘standard’ I2S signal delays the data one tick behind the LRCLK signal.
There are two ways to fix this problem: programming a kernel driver, or building some custom logic to fix the problem. Obviously breaking out some flip-flops and NOR gates was the cooler option, giving [Mariosis] a great sounding stereo with a vintage DAC.
As [Jan-Erik] had already built a simple USB connected Digital-to-Analog Converter (DAC), he decided to make the high-end version of it.
The prototype you see in the picture above is based on:
- the PCM2707C from Texas Instruments which takes care of the USB communication and outputs I2S audio data
- the PCM1794A, a 132dB SNR 24-bit 192kHz DAC which receives I2S protocol
- the OPA4134, a high performance audio operational amplifier
The on-board +3.3V and -5V voltages are generated by inductor-less power supplies. As [Jan-Erik] mentions in his write-up, the ‘high-end’ was put between single quotes because the PCB is single sided and uses through hole passive components. The board was designed using Kicad, etched by himself and put in a machined enclosure. All the production files can be downloaded from his website so you may produce it within a day.
If you look around a few electronic music forums, you’ll see a lot of confusion over the difference between a bitcrusher – a filter that reduces the bit depth of an audio signal – and a sample rate reducer – a filter that does exactly what its name implies. With the popularization of 8-bit and retro synth music, this difference is obviously of grave import of concern to saints and kings alike. [Michael] is more than happy to walk us through the difference with real-time sample and bit rate adjustment with his audio hacker board.
The audio hacker board is an Arduino shield with a 12-bit DAC and a 12-bit ADC. With two 1/8″ jacks and a pair of pots, [Michael] was easily able to whip up a sketch that is able to adjust the sample rate and bit depth of an audio signal in real-time.
Contrary to nearly everyone’s opinion of what ‘8-bit’ music is, it’s actually the sample rate that makes music sound like a cassette deck jury rigged into a Nintendo Entertainment System. Reducing the bitrate just makes any audio source sound louder and worse.
Check out the excellent demo video of the effect of bitcrushers and sample rate reducers below.
Continue reading “The difference between bitcrushers and sample rate reducers”