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Sigma DSP for Active Speaker Systems ‐ Application Note Brewster Lamacchia 19‐Aug‐16 Momentum Data Systems

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Sigma DSP for Active Speaker Systems ‐ Application Note Brewster Lamacchia 19‐Aug‐16 Momentum Data Systems Sigma DSP for active speaker systems ‐ application note Brewster LaMacchia 19‐Aug‐16 Momentum Data Systems http://www.mds.com Introduction The availability of low cost, efficient, high performance class D amplifier devices in the past few years has enabled the use of in‐speaker electronics for better performance in not just subwoofers but also in primary channels of consumer oriented audio components. The availability of A/D and D/A devices with enough dynamic range for high performance home audio has also been a key component in an active (powered) speaker design that uses DSP. While powered PA speakers for “Pro” audio have been available for a number of years, those systems typically lacked sophisticated signal processing and had limited performance parameters. Passive crossovers can be replaced with DSP based filters that allow optimal matching of woofer and tweeters (2‐ way systems) or more complex multi‐band drivers. With increased DSP processing not only can filters be implemented, but extra equalization to flatten the system response (or tune it for a specific sound characteristic) can be performed. That same DSP device can also be used to implement features that protect the drivers from damage due to thermal (overheating) effects of high power levels, as well as prevent hard clipping of the amplifier and/or excessive driver excursion, which have very audible side effects (increased distortion and heating for the former, and popping noise and/or mechanical failure for the latter). MDS has developed hardware platforms for in speaker electronics based on Analog Device’s Sigma DSP. The Sigma DSP integrates a number of useful peripherals and has a very simple and low cost power supply design. The Sigma Studio tool allows non‐DSP experts to create the DSP applications needed to achieve maximum sound performance and high sound quality from lower cost transducers.1 The software programmable nature of the DSP based design also means that a common hardware platform can be used across multiple systems designs with the obvious savings in cost and development time for multiple end user products. Lastly the software based approach allows customer selection of processing features based on their personal preferences as well as handling equalization for differing placement in a room. This paper only looks at the considerations for in speaker DSP processing; the creation of multi‐speaker (surround) sources and bass management for systems that use a sub‐woofer is outside of the scope of this paper. Other aspects of speaker design are also only briefly summarized, but references to more in depth materials are included. From the development engineer’s perspective, in addition to an easy to use tool like Sigma Studio, is the availability of low cost desktop audio systems put together with soundscards and low cost and/or shareware 1 For more information about Sigma Studio see https://wiki.analog.com/resources/tools‐software/sigmastudio 1 software for signal generation and analysis. While final design will require a good test room and calibrated measurement equipment, a lot can be done outside of the test room to get the system initially prototyped and debugged. Even with these admissions of brevity and skipping of important detail, this paper may appear long to the casual reader. Despite being about sound, the topic is not one that lends itself to sound bites. Summary In an active (powered) speaker there are typically 3 major PCB assemblies as illustrated in Figure 1: Power supply – provides the higher voltage needed by the amplifier (typically 24 – 48V) as well as digital (3.3) and analog (+/‐ 15 or +5) power Amplifiers – typically in the 25W to 100W/driver range for multiway speakers, and 300W or more for subwoofers. DSP processor board – includes A/D and D/A FIGURE 1 MAJOR COMPONENTS OF AN ACTIVE SPEAKER SYSTEM In a soundbar application there would typically be two (a simple stereo soundbar) sets of amplifiers and transducers or three sets (left, center, right). In a soundbar, which tend to be compact and thus have a low enclosure volume, an external subwoofer is used for bass frequencies, typically the crossover is 120 Hz (small sound bars) to 80 Hz (larger soundbars). So called bookshelf sized speakers also typically require a subwoofer for good sound. A typical speaker DSP processor board is described by the block diagram in Figure 2. A small microprocessor is usually used to manage the system operation. The amplifier and/or power supply may have temperature 2 measurement capability and the microprocessor can monitor those and signal the user should the system overheat. Systems that allow user adjustment of the speaker’s operation would use the microprocessor to manage the user interface. By keeping the system management tasks off of the DSP the DSP operation stays very deterministic, which is very desirable for real time audio applications as even getting a single sample delayed/wrong out of millions2 will be instantly perceived by the listener as a defective product. FIGURE 2 SIGMA DSP CONTROLLER BOARD FOR AN ACTIVE SPEAKER The block diagram also includes an analog activity monitor that uses the microprocessor’s ADC input to detect when analog signal is present and the speaker should come out of low power standby3. While the diagram shows +/‐ analog supplies (for the op‐amps) it is possible to design a system (with slightly reduced performance) that operates with a single rail +5V analog supply. During customer development and system voicing ADI’s Sigma Studio USBi tool can be used to develop and test the DSP programs. Please see http://www.analog.com/en/design‐center/evaluation‐hardware‐and‐ 2 Assuming a 48 kHz sample rate, one error out of 3 million samples would be a possible “pop” from the system once per minute. 3 A device like this active speaker must draw less than 0.5W in standby to be sold in many places in the world. The device must detect when no input is present and after some preset time period enter standby state. It is likewise desirable to exit the state when signal returns; the microprocessor has low power modes and can also power off the amplifiers and other DSP controller board components. 3 software/evaluation‐boards‐kits/eval‐adusb2ebz.html#eb‐overview for more information on USBi and the Sigma Studio tools. Once development is complete the output of the Sigma Studio tool is to a (header) .h file that is then compiled in to the microprocessor application software. The microprocessor software consists of two main parts; the speaker controller program (which includes the code to download to the Sigma DSP) and the bootloader/flash updater. It is also possible to have the Sigma DSP load code directly from an EEPROM at boot, but that architecture isn’t used here as: Some have a digital interface with multiple input sample rates, and due to processing limits the application (and not just coefficients) needed to change at different sample rates. Storage is already available in the microprocessor and an external SPI flash connected to it. Code download over I2C is reasonably fast Consistency with other MDS developed designs that combine a host plus DSP Other applications, particularly where the DSP program can not be field updated, may make more sense to be implemented as boot locally. Obviously a system that doesn’t need a microcontroller would also boot the DSP locally. The flash updater supports updating the speaker software from a USB port with the corresponding control software for this. The bootloader ensures that a failed firmware update can not result in a bricked speaker. Sampling rate is typically 48 kHz but with the higher performance Sigma DSP parts operation can be at 96 kHz. Internally the Sigma DSP is a 24 bit precision device for typical operations, with a block floating point scheme used to avoid digital overflow in intermediate calculations. Other things to consider While this paper focuses on the DSP and signal processing, this is a small subset of the issues faced by designers of active speakers. We assume the reader already has a general understanding of audio, see for example The Audio Expert (1). Unlike many electronics applications, speakers can not be miniaturized in to arbitrary form factors. The science of speaker design has to take in room acoustics (2) and mechanical consideration of the cabinet and the transducers (3). Speaker’s acoustic properties are a function of the type of and location in the room they are placed (4) and this also affects the design goals. There is also a visual aspect to the product – some speakers are hidden in walls, others are sculpted to look like avant‐garde art pieces. Besides performance, key considerations are the desired size and cost of the system as well as manufacturability. Speaker design was already a complex task before adding power hungry electronics inside of a sealed enclosure with exposure to high vibration levels. For example, what might be a normally simple task of selecting the audio input connectors and other items on the rear panel has to take in to account that the enclosure must be sealed (other than the port tube/opening in non‐ sealed designs) as small holes will produce whistling noises. Many designs will place the electronics inside of a basket that is mounted to the rear and provides an air seal. Doing this however can significantly change the volume of air inside of the speaker, which in turn affects many aspects of the speaker design. 4 In larger speakers or subwoofer designs the higher acoustical power creates significant pressure variations on circuit boards. This can expose two potential problems. First is mechanical resonances in the PCB which can either create buzzing noises or result in stress failure of soldered connections.
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