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DM320011 is the Audio Development Board for PIC32 MCUs available from Microchip, priced $149.99 excluding shipment.
The board connects through the MFi dock edge connector to the accessory development platform for iPod and iPhone. Call this an iPod dockstation.
So now Microchip wants to place PIC32 chips in digital audio applications.
From a diyAudio perspective, that’s an interesting starting point.
Quite intriguing is the MFi dock edge connector showing at the right side. Looks overkill for connecting an iPod. Really, honestly, how can a PIC32MX7 featuring no I2S interfaces, be involved into digital audio? There must be a trick. Time to investigate. Let’s zoom in. (continue reading…)

Never heard about the C-Media CM6206 7.1 USB-audio chip ? It comes out as the chip used in most 5.1 and 7.1 USB-audio adapters.
Back in July 2010, Adrian Pardini tried to get it understood by ALSA – the Advanced Linux Sound Architecture.
There is a Linux patch about CM6206. Unfortunately there are reported quirks. Eric Lammerts, Clemens Ladisch and Dan Allongo continued working on ALSA CM6206 support.
In April 2011 Wolfgang Breyha and Takashi  Iwai continued on the Android Source Tree.

Go googling “revision 8129e79ed7932bd11d60518d62434a0b687e5771″

ALSA: usb-audio – Terratec Aureon 7.1 USB ID as C-Media cm6206 quirks
This patch adds support for the Terratec Aureon 7.1 USB which uses a C-Media cm6206 and needs all the quirks already found in the past.  (continue reading…)

LabVIEW from National Instruments (NI) is a renowned PC-based instrumentation system, specialized in signal analysis and measurement. LabVIEW became available on the Windows PC platform in 1992.
Since then it pervades the academic world, generating plenty R&D applications as spin-offs.
Many universities and tech schools impose LabVIEW tutorials to their students.

Back in 1992, an average PC was clocked at 33 MHz or so. One year later, thanks to the first i486 cores clocked at 3 times the external bus speed, the Windows PC performance got a big boost – when running on cache. (continue reading…)

Digital Audio Compilers are getting user-friendly nowadays. No more programming. Creating an audio application only consists of dragging and dropping blocks, interconnecting them, for defining digital audio signal flows. Let’s read the descriptions of three of them:

- ANALOG DEVICES – SigmaStudio
- FREESCALE – Symphony Studio
- DSP CONCEPTS – Audio Weaver
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Most soundtracks you are listening to, actually depend from Steinberg VST. Such technology is at the heart of digital sound creation (the VST instruments) and sound processing (the VST effects). From a diyAudio perspective, it is worth knowing a few VST basics, especially when experimenting with Synthmaker or Flowstone.

Here follows a neutral compilation of covering most important VST aspects. Worth visiting, is the Steinberg website, where it shows how VST has deeply penetrated all compartments of the sound industry.
Steinberg is currently promoting the The Art & Science Of Sound Recording video tutorial series on music production, presented by multi-platinum producer Alan Parsons. In more than 10 hours of footage, more than 40 fellow professionals join Alan Parsons in this guide to modern recording, including Erykah Badu, Jimmy Douglass, Jack Joseph Puig and Simon Phillips. This 10-hour DVD set applies classic, old-school recording experience to the modern recording scene and will be a standard work on the subject for years to come. (continue reading…)

About thirty years after the advent of consumer Digital Audio, the diyAudio communauty still needs a simple inexpensive way to graphically sketch an audio signal flow using a PC, automatically generate some corresponding audio DSP code, send it to an inexpensive target board, flash the microcontroller, and run it.

As diyAudio enthusiast, you know that the only way for guaranteeing reassuring listening sessions at home, without needing hiring a Geek and a Sound Engineer, is to get your precious audio delivered by a tiny target running a bulletproof Digital Audio OS instead of MS Windows, MS WOA, Google Android, or Apple iOS. You want to run a purely deterministic realtime software, tailored for high-end Digital Audio natively supporting audio sampling frequencies like 96 kHz or possibly 192 kHz.  You want to avoid any Sample Rate Conversion in the system. You want your DACs to be the sole and only clock masters. You know that inexpensive microcontrollers like PIC32MX1/MX2 and NXP LPC4330 featuring a 32 x 32 multiplier delivering a 64-bit result, deliver more precision than a 24-bit or a 28-bit DSP chip. You know all this, and you want to benefit from it for extensively mastering the audio reproduction chain. You, as DIY Audio enthusiast, you just realized you can’t live without that.  (continue reading…)

Although Graphical DSP Compilers are progressively becoming popular in most universities and tech school labs, they remain out of reach for many diyAudio enthusiasts. That’s not true anymore with Synthmaker and Flowstone.

SynthMaker is an audio programming tool that allows anyone to create their own sounds, virtual instruments and effects without having to write any code. These instruments and effects can then be used for playing live via MIDI or as VST plugins for use in making music using any of the popular recording software.

SynthMaker allows to program without writing a single line of code. This is achieved by linking together components. Components are simply building blocks that perform some action. All changes take effect instantly with no build time. Synthmaker has a highly efficient sound engine. This allows you to process one sample at a time without compromising on performance. Feedback paths are no problem which is essential if you want to design your own filters. Exporting an instrument or an effect is a simple one-click process. Having identified  the module you want to export just click Create on the resulting dialog box. You can export to VST plugin or as a standalone instrument that can be played via MIDI. (continue reading…)

Active crossovers are a popular diyAudio topic, however only a few hobbyists managed to build a home-brew digital crossover on a kitchen table, for less than $99 all inclusive, software included.

This is going to change with the recent arrival of the PIC32MX1/MX2 family (twin I2S), and the NXP LPC4330 (twin I2S and possibly more through the SGPIO). I2S stands for Inter-IC Sound, Integrated Interchip Sound, or IIS. This is different from SPI and I2C. I2S is a 20 year-old standard, still widely in use, for connecting a µC on an audio ADC, DAC or Codec. On top of this, the PIC32 and ARM Cortex chips feature a fast 32-bit x 32-bit multiplier delivering a 64-bit result. This enables computing audio with a 192 dB dynamic range, about 50 dB more than the best available ADCs and DACs. Quite a headroom would you say.

The PIC32MX1/MX2 has Flash memory and runs at 40 MHz. This is a nice single chip µC, easy to use. Like all PIC32 chips, there is a MIPS M4K core inside, 32-bit of course. The PIC32MX1/MX2 is fully supported by Microchip MPLAB ICD3 programmer and debugger. Call it a one-stop-shop solution.  The PIC32MX1/MX2 is a nice recent addition (early 2012) to a mature µC architecture. We need experimenting with the PIC32MX1/MX2 in first place. Back in 2008 most µC programmers got intimidated by 32-bit µC, thinking they were expensive, complex, requiring C for programming, and always short of RAM with Linux as background. Lucio Di Jasio from Microchip did a great job in optimally introducing the PIC32 to people having started their carrier more than 25 years ago, writing firmware for 8-bit µC like the Intel MCS51 one-chips or Motorola 68K CPUs back in those times.

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State Variable Filters went popular inside modular electronic music instruments, a technique pioneered by Moog in the sixties. The flexibility delivered by a State Variable Filter remains unbeatable, making its success. In 1987, the Musical Applications of Microprocessors book written by Hal Chamberlin triggered a revolution in exposing many different sound generation and processing techniques, aiming at synthesizing natural sounding percussive sounds.

Both analog and digital State Variable Filters got described in Hal Chamberlin book. All State Variable Filters rely on two cascaded integrators with adjustable time constants, surrounded by two independent feedback loops. Worth noting is that the State Variable Filters has not found its way in HiFi and High-End Audio. The State Variable Filter requires two or three opamps, quite a lot compared to a  Sallen-Key or a Rauch filter only needing one opamp. In HiFi, you don’t require the flexibility provided by the State Variable Filter. In Pro-Audio, there can be a State Variable Filter used per parametric equalizer (independent Fc and Q adjusts, variable boost or cut), however as this is a bandpass-only structure, one should regard it as a reduced set  State Variable Filter.

Flexibility is the keyword. With a State Variable Filter, any 2nd-order transfer function can be synthesized with the aid of an external mixer combining the three canonic signals (lowpass, bandpass, highpass) in any proportion. (continue reading…)

The IIR overflow syndrome is a popular theme in DSP literature.  As illustration, let’s design a 2nd-order Butterworth highpass cutting at 100 Hz. The coefficients get calculated using the RBJ Cookbook. They also get calculated using IIR_Lab, displaying the corresponding Bode Plot:

Clearly, the RBJ Cookbook coefficients are the same as IIR_Lab. The Bode Plot confirms the filter behavior. Apparently we are dealing with a gentle, trouble-free Butterworth highpass cutting at 100 Hz.

A rapid tour on Wikipedia makes us learn about two main implementations.
The Merged Sections implementation official name is Direct-Form II.
The Split Sections implementation official name is Direct-Form I.

Let us evaluate both, from the overflow syndrome perspective. Do they materialize gentle, trouble-free, universal filters as you would expect from digital technology? Can the results differ? Isn’t digital supposed to be immaculate ?

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