Professional Documents
Culture Documents
com
Follow Us
Table of Contents
Introduction
1.
2.
3.
4.
5.
6.
7.
References
About the Author
www.ti.com/audio
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
June 2014
Introduction
Soundbar Design from Start to Finish
The product development process is a long and winding road, with more than a few potholes along
the way.
Through numerous customer meetings over the years, I've worked through many product
developments alongside product managers, product designers, embedded programmers and
industrial engineers. I've been there through the happy times "Oh, it's awesome we do all the
processing in your amplifier chip!" through to "Why doesn't my I2C bus work?" I've seen products
from the initial marketing specification, through the design process, and over to the ramp production
phase.
When I was going through University over a decade ago, there were plenty of guides and university
textbooks that taught us how to set the value of resistors and capacitors for analog filters, and how
to bias a power transistor. What was missing was the process of developing a product in the real
world. The "hitchhikers guide to product design," if you will, with big bold words on the back that
read something soothing like "Don't panic, you probably just mis-wired your speaker."
In 2009, in the middle of the hype for soundbars, we developed a soundbar reference design
(Value-Soundbar-RDK). It was a great opportunity to pull together an entire TI-based solution from
inputs, control, processing, and amplification. These articles were published in Electronic Design.
Actually, they were really the memoirs of a product development, with all the ups and downs, with all the decisions made, and the hindsight
to say if it was a good idea or not!
Once we went to market, we realized that the same design could be repurposed for virtually any system that had loudspeakers, whether it be
PC or Bluetooth speakers, guitar amplifiers, and so on. Looking back, this guide should have been titled "Memoirs of an Audio Accessory
Design!"
I hope this guide gives some value to those on the beginning of the journey in product design. I wish I had something like this 15 years ago
when I started down this path.
Keep the soldering irons hot and the signal paths clean!
Dafydd
Back to top
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 1. Now that flat-panel televisions are only millimeters thick, many manufacturers are
pushing the amplifiers and speakers out to a soundbar.
Let's jump in feet first. Here's a specification straight off the plate of a marketing team at a company I recently visited:
USB, S/PDIF, and three times stereo analog inputs
Stereo analog inputs should support 2 VRMS inputs
S/PDIF sources should be coax
Stereo output (two at 20 W each)
Analog subwoofer output
Wireless subwoofer option
Infrared (IR) remote control that supports the NEC or RC5 protocol
Audio processing features need to include speaker equalization (EQ), volume, dynamic range control, and SRS WOW HD
The inspiration for these sorts of projects often happens instantly, leaving the designer to record it on whatever's closest. The picture in
Figure 2 may or may not be the actual first draft of the specs. I'll let you be the judge of that.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Back to top
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 1. In a typical audio signal chain, analog inputs must first pass through an ADC
before heading to the processor, whereas digital inputs can bypass the converter and be
sent straight to processing.
Sources: Analog inputs
In the real world, with analog sources like televisions, Blu-ray players, set-top boxes, and gaming consoles, most if not all manufacturers
stick to a maximum input level of 2 VRMS, or about 5.6 Vp-p. Anyone with a lot of design experience can tell you this standard is a pain to
deal with, as most classic audio converters run from a +5-V analog supply.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Back to top
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 1. A simple GPIO expander can be used to increase the amount of ports without the
need for interrupts.
The unexpected
One of the best pieces of advice I received in my years of working on customer systems was to always design for the unexpected. It is one
thing to design defensively, where you assume your design will have to be changed if something doesn't work. It's another thing to try and
stay one step ahead of marketing.
Marketing folks are constantly looking for ways to get the right product to the right customer. They are always searching for ways to get more
products out the door in a shorter time to a greater variety of end customers. They often talk in SKUs, which essentially means a separate
product.
Two products can have the same circuit board and components, but different firmware for different functionalities. To the end customer,
they're two different products that may be sold next to each other on the same shelf, but for different prices.
Each of those products is a separate SKU and typically will appeal to different parts of the market. By designing with flexibility in mind, a good
design should be able to be expanded upon quickly and easily to create different SKUs.
Consider adding stuffing options to your project that will allow add-on cards, if needed, with access to the control and data buses directly from
your host processor. This will immediately enable you to add an extra pair of channels, if needed, or to use a different amplifier in some
SKUs but not others simply by adding a card and changing the firmware. In our value soundbar reference design, we added headers to take
a wireless module and communicate with a wireless subwoofer (Figure 2).
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Back to top
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 1. Note the clock relationship in this simple block diagram of a S/PDIF receiver
driving a processor and digital amplifier. The clocks for the entire system can be derived
from the incoming S/PDIF stream.
In an analog inputs only system, you the designer must generate an internal master clock within your product. This master clock is used to
drive the ADC, processor and digital amplifier, if the digital amplifier requires I2S. Most designers in the industry generate these clocks using
one of four methods:
1. Direct digital synthesis (DDS) device: Some designs use a DDS device where a relatively inexpensive IC generates a very
high-speed clock. PC motherboards use this type of clock generation. A DDS device can generate different rates on the fly. For
example, it can generate the master clock for 44.1 kHz for one setting, then switch to 96-kHz mode in another setting.
2. PLL circuit: Depending on the type of PLL (fixed multiple, or multiply and divide structure), different master clocks can be
generated in a manner that is similar to DDS. PLLs require a known clock rate to multiply/divide from or to use as a reference.
The reference can be a fixed-rate CMOS output oscillator or a Pierce oscillator.
3. Fixed-rate CMOS output oscillators: Many designers buy an off-the-shelf CMOS oscillator for systems where the sampling rate
is fixed. These simple devices tend to be very reliable. Just add a power supply (3.3 V and GND), pull up the "enable" pin, and
you have a very clean master clock output at a fixed frequency. Semiconductor manufacturers often use these oscillators on
evaluation boards and in systems that can afford the extra dollar or so.
4. Pierce oscillator: Used more in consumer audio systems, a Pierce oscillator can be created using a simple crystal and a $0.10
piece of logic. The Pierce oscillator is by far the cheapest way to generate a master clock (Figure 2). Again, these systems
usually have a fixed clock rate. Some customers may use two different Pierce oscillator circuits because they are low cost, if the
system needs to support 44.1 kHz (and multiples) and 48 kHz (and multiples: 96 kHz, 192 kHz).
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 2. Pierce oscillators are easy to design. But if you expect to support multiple sample
frequencies, the solution size can increase.
Of course, nothing comes for free. While cheap in chip price, Pierce oscillators involve other costs such as board space and a higher number
of components than CMOS oscillators. Also, the output clock from a Pierce oscillator does not always have a perfect 50/50 duty cycle. It is
actually closer to 52/48 percent.
This doesn't cause a problem in many systems because the digital circuitry most likely will be single-edge clocking, but it is something to
consider. A Pierce oscillator generates a single clock source. From there, it needs to be divided down to BCK and LRCK, a task typically
performed by the clock master in your product.
Many experienced designers may have had different experiences and may have additional advice with this next part, but this is what I have
learned.
The ADC is the most critical part of any digital audio circuit. It is also the part that is most sensitive to jitter. If you mess up the analog-todigital conversion, there is little you can do to compensate for it in the digital domain. This is why the master clock is generated next to the
ADC in many professional audio systems. The ADC is used in master-mode, which causes the ADC to divide down and distribute the SCK,
BCK, and LRCK to the rest of the audio signal chain (Figure 3).
Figure 3. In the master and slave relationship, many designers allow the ADC to perform
the clock division from MCK to get the best ADC performance.
In our example where you have a digital input feeding a SRC, the SRC behaves as a clock domain isolation barrier (Figure 4). The DSP and
amplification system needs to run from its own clock source, which is shared with the SRC output side.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 4. Designers can convert different clock rates to the system clock rate by
using an SRC.
This type of system allows simple switching from analog to digital inputs, as the data rate is synchronized from an ADC or SRC (S/PDIF
source). This method is nice and easy. Without the SRC, the system must mute, clear the processing pipeline, switch the sample rate (bank
switching), start running data through the new pipeline, and unmute.
One final note: never, ever try to transfer data from one system to another, such as going from a CD player S/PDIF output to a DAC input,
without the slave side locked in to the transmitter (Figure 5). In most systems, the DAC input slaves to the S/PDIF. If they do not, users will
have lots of noise issues with pops and clicks.
Figure 5. The difference in clock rates as device A transmits to device B (both with their
own crystals) generate overruns and underruns in buffered audio memory due to their
asynchronous behavior. Eventually this will cause pops and clicks.
You may have two crystals that say they are both 48 kHz, but that does not mean they are exactly the same. Any drift in specification causes
the transmitter to generate data faster or slower than the receiver. This immediately causes buffer underruns or overruns, which cause lots
of pops and clicks (bad for speakers), and potentially can crash your system.
There are ways around this using buffers and generating interrupts once the receiving buffer is mostly empty or mostly full. However, that is
another article waiting to be written. Designers involved with USB/FireWire and other non-time-guaranteed protocols typically have lots of
experience with this.
Our multi-SKU (end product from same design) strategy drives the need for multiple clock generation strategies. In the value soundbar
reference design, we have two stuffing options for clock mastering. We use the DIR9001 S/PDIF receiver that, when locked, generates all
clocks for us. When unlocked, it simply uses an onboard crystal with some dividers to generate an "analog mode" clock source (Figure 6).
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 6. The DIR9001 circuit uses the CKSEL pin to select between the S/PDIF clock
recovery outputs, or the reference crystal to generate the clocks for the system. CKSEL
can be connected to the S/PDIF lock pin on the device for auto switching.
For systems with digital inputs, the S/PDI receiver uses the crystal as a reference to calculate the sampling rate. When the S/PDIF is
unlocked, the S/PDIF receiver (DIR9001) then generates the audio master clock for the system.
For analog-only systems, use the same crystal footprint to save PCB space. However, use an additional buffer in place of the DIR9001 to
generate a fixed 24.576-MHz clock. This is divided down by the PCM3070 for use in its codecs and multiplied to a higher frequency (using a
PLL) for use in the miniDSP. Doing so saves the cost between using the DIR9001 and using a $0.10 crystal buffer in the systems.
Back to top
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 1. This is the value soundbar reference design amplifier section. The speaker
connections are not shown.
Some class D amplifier chips omit the PWM modulator, "banishing" it to a separate chip. This minimizes interaction between high- and lowlevel circuitry. More importantly, it also allows the amplifier to be built with high-voltage geometry. Outputs up to 600 W from a single IC
become possible.
The newest type of switching amplifier accepts an I2S input, converting the I2S data into a PWM drive signal. These amps are easy to work
with, as many DSPs and ADCs provide an I2S data stream at little or no cost. Some even include EQ and dynamic-range control.
Figure 2 shows such a system. The PCM1808 ADC is the clock master, driving the TAS57xx. The MSP430 MCU loads the signal-processing
coefficients into the TAS57xx amplifier and monitors the analog inputs, using its internal ADC to turn the system on when an input signal is
present.
Figure 2. A simple design using an ADC and switching amplifier. The amplifier directly
accepts digital data in I2S format.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 3. The value soundbar reference design uses diodes specifically designed for ESD
protection.
Power supply solutions
Switching amplifiers usually require two or three dc voltages. A higher voltage rail typically is required for the output amplifier either a single
24 V+ rail, or a split high rail supply. Usually there is a power rail for digital (and low-level analog, if any) circuitry at 3.3 V or 5 V. The rail
voltages are usually obtained from a power supply connected to the ac power line. The digital voltage is commonly derived from the positive
rail voltage, using a switched-mode power supply (SMPS) or linear regulator (Figure 4).
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 4. The left-hand side of the chassis holds the switching supply for the switching
amplifier on the right side.
The rail supplies can be linear or switching. Linear supplies typically require large, heavy transformers, but their rectifiers and capacitors are
inexpensive. Switched-mode supplies use small, light transformers, but more and more expensive electronic components are required.
Switching supplies are compact and cool-running necessities if the supplies are mounted within the product. Voltage regulation is an
inherent part of their architecture and does not require additional power-wasting components. They can be designed to operate on line
voltages from 100 to 250 V, without having to rewire the transformer. This universality makes it easy for a single product to serve domestic
and foreign markets.
Switching supplies generate high-frequency noise. Close attention to board layout and bypassing is needed to keep noise out of analog
circuitry.
Texas Instruments offers reference power-supply designs for AV use. One is a 720-W class G supply that can power many of TI's class D
amplifiers. A class G power supply uses power rail switching to minimize idle power losses in the power stage.
In this specific case, a logic input pin on the power supply can be used to tell the power supply that either a 50-V or 25-V supply is required.
Halving the power supply voltage from 50 V to 25 V divides the idle power consumption by four (P=V2/R)
"If we electrocute the user, he won't buy any more of our stuff."
The ac line (100 to 250 V) often directly powers consumer electronics. Line voltage is potentially lethal, so there are stringent Conformit
Europenne (CE) and Underwriters Laboratory (UL) safety standards for line-powered products.
On the other hand, if an amplifier is powered by an outboard supply or "wall wart," only the supply has to meet tight safety standards. As 30 V
(ac or dc) is considered "safe" (it isn't high enough to produce a strong shock sensation, and it definitely is not lethal), an amplifier powered
by a 30-V external supply has safety standards that are more relaxed and easier to meet. You can start selling the product sooner, and there
is less trouble getting certification for international markets.
If you use an external power supply, you probably will want a commodity model from a reputable manufacturer. Keep an eye on the following:
Output: If the amplifier has a continuous average output of 50 W (100 W peak), a 24-V power supply has to deliver at
least 4.5 A.
Voltage regulation: Voltage regulation can be important in systems that use open-loop class D amplifiers, which typically switch
the power rail directly to the speaker load, so any power supply unit (PSU) noise couples to the speaker load. However, all of our
analog input power amplifiers are internally closed-loop amplifiers, and have good power supply rejection ratio (PSRR).
Tolerance for abuse: How well does the supply tolerate shorts (one or many, brief or sustained)? "Accidents happen." Good
supplies bounce back from abuse. Bad supplies die.
For the value soundbar reference design, we used an off-the-shelf 24 V supply. Surprisingly, we had to go through three vendors to find a
reliable product. My office looked like the elephants' graveyard of power supplies.
Selecting a regulator for the low-voltage supply
Once you have found a reliable supply of the right capacity, you need to decide on how to generate the low voltages. (Depending on the
design, these might power analog as well as digital stages.) You can use inexpensive commodity linear regulators (such as the UA78M33)
or an SMPS. A linear regulator is usually the least expensive solution, but pay attention to the regulator's thermal and current limits.
If the circuit needs 5 V, and the supply is 24 V, the regulator drops the difference, 19 V. The current through the regulator is the current
through the load, so a 1-A load would require a linear regulator to dissipate a blistering 19 W. For heavy loads, an SMPS is therefore the
best choice.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Back to top
6. Layout
Design reuse is a big part of modern product design, including home audio systems. Reducing systems to sections that can be copied and
pasted over and over again is key to getting to market quickly. If the marketing team you are working with suddenly decides to add a
function, be it software or hardware, a library of working circuits is always advantageous.
System partitioning for simple upgrades
Considerations regarding form factor are always more than just what circuit designers have in their backpack. In consumer products, the
product's industrial design, including the product/end-user interface, will influence your layout considerations significantly.
Are all of the connectors on one end of the product? Are the inputs right next to the outputs, or are they on completely separate ends of the
product? Such things will drive your solution significantly. For example, in a product where input connectors are right next to output
connectors (such as a laptop PC with its headphone and input jacks), then it makes sense to use a codec with the input and output pins near
each other.
On the other extreme, in a system with multiple inputs on one side of the unit and multiple inputs on the other side, it may make more sense
to keep the ADCs and DACs separate. This keeps the analog signal traces as short as possible, making them less susceptible to
interference and crosstalk.
Now that we are starting to worry about design reuse and how to apply it to a circuit that may have a completely different form factor, let us
throw another consideration into the mix-system partitioning, which could be considered the layout equivalent of a block diagram.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 1. By designing the PCB in such a way, rather than a "whatever fits and makes an
easy layout," various sections can be replaced without destroying the entire layout effort. It
also allows each section to be tested and debugged individually.
When it comes to the value soundbar reference design kit, a number of things need to be considered.
Form factor
The form factor is essentially a long cabinet that is 32 inches wide, 4 inches tall, and about 4 inches deep. With speakers at both ends of the
speaker cabinet, the only empty space available is a gap in the middle.
Let us assume we want a reasonable amount of cabinet space at both ends of the unit, along with cabinet porting and other features. On
both ends of the soundbar we allow 10 inches of room for speakers and acoustic space. That leaves 12 inches of PCB width and about 3.5
inches of front-to-back depth.
All of the connectors in our soundbar are on the same plane as we do not have that many connectors. Also, doing so helps users connect
wires from awkward angles. Imagine trying to connect various wires with no real idea of where along the back panel they are!
In this case, the soundbar has multiple inputs and a single subwoofer line out. Other outputs such as the amplifier outputs are inside the
case.
We discussed mounting the PCB vertically or horizontally. Some may disagree, but unless the PCB can be strongly supported, don't mount
external connectors vertically on the PCB because the PCB will flex. On the value soundbar reference design kit, we mounted the
connectors horizontally to minimize this flexing.
Multi-SKU design
The best way to maximize the return on investment (ROI) of a design is to spin it into as many products as possible. This may be as simple
as adding extra analog inputs and an alarm clock function to one product, or as complicated as upgrading the power amplifiers used and
changing those little 2-inch mid-tweeter loudspeakers to 4-inch full-range speakers.
By designing as many options as possible onto one PCB, you allow as much of the design as possible to pass various safety certifications as
possible, on the initial qualification. This allows designers to get multiple SKUs designed and qualified in as short a time possible and save
significant amounts of money on one qualification instead of three or four.
Options can be added to the PCBs that are enabled by stuffing options (Figure 2). The signal paths can be modified by changing which
resistors are added in series with the tracks being enabled. On a simpler level, specific devices can be stuffed one way or the other. Think of
designs used to create exclusive products.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 2. Stuffing options allow the same design and PCB layout to be used for an analogonly SKU versus a mixed analog and digital input SKU. In this case, an analog-only SKU
does not require the DIR9001. By using the stuffing option, we remove the DIR9001, but
reuse the footprint for the external crystal that the DIR9001 to generate the clocks for the
analog SKU (along with the crystal oscillator).
In the value soundbar reference design kit, we propose two different clock sources, depending on the setup. In an analog-only build SKU, the
DIR9001 (S/PDIF receiver) is not necessary. However, we still need an audio clock source to drive the audio codec. The same reference
crystal is used (Y2) in the diagram. A crystal buffer driver (OSC) is used instead to create a nice 3.3-V CMOS clock source that drives the
converters and DSP in the PCM3070. If both devices are inserted, we have contention on the MCLK bus. But in the case where it is one or
(XOR) the other, we should have no issue.
Figure 3 depicts another example. The S/PDIF receiver (DIR9001) has an external two-input multiplexer that we use to select from two
different S/PDIF sources. But in our design, we have three sources (USB to S/PDIF converter, coax, and S/PDIF).
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Back to top
7. Software's role
Soundbar design encompasses many divergent yet essential steps. For example, the marketing team's specification must be decoded into
component selections. Various converter and amplifier topologies must be examined, and there is in-circuit and production line
programming.
In terms of hardware, the soundbar reference design we have been developing uses a 16-bit ultra-low-power MSP430 MCU to handle the
control, user interface, and housekeeping duties. The PCM3070's dual miniDSPs manage the audio processing (Figure 1).
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 1. The final soundbar reference design hardware includes the miniDSP codec,
microcontroller, USB, S/PDIF receiver, audio power amplifier, and power supply.
The code for the miniDSPs is kept externally on an additional EEPROM, as we had some concerns regarding the number of different process
flows required for the miniDSP. By process flow, I refer to a different audio processing flow for, say, S/PDIF content versus analog input, or
different audio processing for 44.1-kHz content versus 48 kHz. Each process flow requires its own code and coefficients, increasing the
amount of flash space required on the microcontroller, if we decide to keep everything in the microcontroller.
In practice (and hindsight), we found that customers typically have two process flows (44.1 and 48 kHz). In fact, most have one process flow,
with different coefficients for 44.1 kHz and 48 kHz.
In the rest of this article, we discuss the software that was developed to boot the code for the miniDSP codec from an external source. Mainly
because that is what was done, but also because it provides a good example of such software management, should you have multiple
process flows in your product.
The software in an audio system, such as a soundbar or PC speaker, can be split into a few different sections: control and user interface
(decoding IR, receiving button presses and de-bouncing them), housekeeping (boot up and maintenance of multiple devices), and audio
signal chain processing (audio processing that sits between the input and speaker amplifiers).
Control and user interface
A separate daughter card with physical button interfaces was created to fit in with the typical industrial design of soundbars. The physical
user interface is simply a bunch of switches connected to general-purpose I/O (GPIO) pins on the microcontroller, along with a bunch of ESD
diodes.
Software-wise, it is painfully simple. An interrupt detects one of the buttons pushed (with a signal pulled to ground), a small timer is run (for
debounce), and the I/O port is checked again to see which button was pulled to ground. Based on which button is pulled to ground, a
function is run (such as switch to USB input).
The IR remote control is a tougher challenge.1 There are two different consumer IR protocols, NEC and RC-5. Both work using a similar
physical layer. The protocol really has more to do with timing and word depths. Credit where it is due, the NEC format (or Japanese format)
is mainly attributed to the team at NEC, while RC-5 was developed by Phillips.
IR transmitters and receivers typically have to be matched in frequency. For example, the ones used in the soundbar reference design run at
38 kHz. Data is amplitude modulated into the transmitter and filtered out by the receiver. At the receive end, the system will not see the
carrier frequency, only content that looks like UART serial stream data. For instance, with RC-5, we end up with content coming out of the
receiver at 1.778 Mspb (64/38 kHz).
From a code perspective on the microcontroller, an IR decoding function is started when the serial stream's starting bits trigger the
microcontroller's GPIO pin. Once a valid, relevant code has been received, the appropriate button-press function is run, essentially making
the IR command emulate a physical button press.
Housekeeping
The same MSP430 MCU is used for all housekeeping duties, such as booting up the PCM3070, ensuring the TPA31xx amplifier is muted,
and keeping an eye on the lock status of the digital audio receiver (DIR9001). The PCM3070 has pages of registers that look after the
hardware side of things such as clock configuration and input channel multiplexer selection. Then, another specific set of pages looks after
the instruction and coefficient data.
The microcontroller's code base maintains the hardware registers for the miniDSP codec, since they tend to be fixed for all inputs and usage
cases across Texas Instruments' customer base (analog versus digital input). Once initially booted, only a few register changes are needed.
For example, changing from analog input 1 to analog input 2 is a matter of a few register writes.
The miniDSP code for the miniDSP codec is much bigger and likely to change from customer to customer. In that case, we put each process
flow and related coefficients into an external EEPROM, which can be loaded by the microcontroller in a copy and paste fashion.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Figure 2. Designers can find the I2C address and register to control a multiplexer in their
process flow. This is essential for the microcontroller to perform the correct action on
button-press.
That register address will be used in your MSP430 MCU software, specifically in your button-action code. Other examples may be in
equalization settings, allowing users to change between pop, jazz, and classical mode EQs.2 Multiple images (or configuration files or CFGs)
can be stored on the EEPROM and can be loaded from the microcontroller with a simple call.
Getting into production
You now have an MSP430 MCU hex file, along with an external EEPROM hex file. How on earth are you going to get the code into their
respective places? In the soundbar reference design kit, we opted to have a separate board with pogo pins that the soundbar board could be
placed upon in the factory (or screwed onto in the development environment).
This programming board has headers for the standard MSP430 MCU 14-pin programming spi-bi-wire, as well as a TAS1020B USB
streaming device used to send I2C commands. Using this tool, the microcontroller and the external EEPROM can be programmed
separately, with separate tools.
The caveat to all this is that the MSP430 MCU wants to be the I2C master, once it has been flashed/programmed. However, the TAS1020B
also wants to be the I2C master. As such, the sequence of programming is very important. The EEPROM must be programmed first via the
TAS1020. Next, temporarily disconnect the TAS1020, then program the microcontroller.
In hindsight, putting all code in the microcontroller's onboard flash would have saved a lot of the heartburn associated with two I2C masters
fighting on the same bus. Additionally, I would have specified a simple UART interface on the microcontroller to be connected to a simple
USB serial port cable that could be used for debug.
The soundbar reference design kit has been quite successful in different form factors as soundbars, MP3 docks, and PC speakers. The
combination of easy-to-use tools for the MSP430 MCU, PurePath Studio, and SRS WOW HD royalty-free makes the package rather
compelling!
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Back to top
References
I found a great resource for IR knowledge that you might find interesting: www.sbprojects.com/knowledge/ir/nec.php.
To find out more about the platform head, check out this value soundbar reference design kit.
Download these datasheets: PCM3070, TAS1020 and TPA3116D2.
For audio support, visit the TI E2E Community forum for audio amplifiers.
Back to top
Dafydd Roche is one of TI's Audio Systems Engineers. When he's not busy trying to define smarter, easy-to-use products so customers can
stand out from the crowd, he's busy shaking his cubicle walls doing listening tests with TI's DACs and amplifiers. Dafydd enjoys Tex-Mex
food (a change from his native Wales), talking with customers about their development issues, and putting the world in order on TI's E2E
audio forum (e2e.ti.com). He can be reached at ti_dafyddroche@list.ti.com.
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]
Back to top
http://www.ti.com/ww/en/analog/Soundbar-Design-eBook/[02/08/2015 15:40:56]