Do you know what outstanding performances of A2B technology have in emerging automotive applications?
In the automotive cockpit electronics market, as automakers strive to differentiate their vehicles from their competitors, an increasingly obvious trend is that audio, voice and acoustic related applications are expanding rapidly. In addition, as the average consumer becomes more and more technology-savvy, their expectations for the driving experience and the level of personal interaction with the vehicle are also increasing significantly. Home theater quality audio systems have become commonplace in vehicles of all price points, and now sophisticated voice hands-free (HF) and in-car communication (ICC) systems are also emerging.
Active noise cancellation and road noise cancellation (ANC/RNC) systems, which were traditionally only deployed in top-tier premium vehicles, are now entering the mainstream market and are affordable to the general public. Looking ahead, audio or acoustic-based technologies will become a key component of the engine control unit (ECU) of L4/L5 autonomous vehicles, as the ECU needs to detect the presence of emergency vehicles.
What all these traditional and emerging applications have in common is their reliance on high-performance acoustic sensing technologies, such as microphones and accelerometers. Almost all emerging applications require multiple acoustic sensors, such as microphones or microphone arrays, to achieve optimal system-level performance, and therefore require a simple but cost-effective interconnect technology to ensure that the total system cost is minimized. The lack of microphone-optimized interconnect technology has long been a major pain point for automakers, requiring each microphone to be connected directly to the processing unit using expensive and bulky shielded analog cables. These added costs - primarily in terms of actual wiring, but also in terms of increased weight and reduced fuel efficiency - have in many cases prevented the widespread adoption of these applications, or at least limited them to the high-end market. Recent advances in digital microphone and connectivity technologies are expected to drive the rapid adoption of transformative applications in the next generation of vehicle infotainment systems. A2B technology will have great potential.
With the use of handheld phones while driving prohibited in most countries, hands-free devices with Bluetooth® have become standard equipment in almost all vehicles. A wide range of hands-free solutions are available on the market – from simple stand-alone units containing a speaker and a microphone to advanced solutions that are fully integrated into the vehicle's infotainment system. Until recently, most hands-free systems were implemented in a very similar way. Such systems only contained one (a few had two) microphones, and the microphone technology involved was of the 50-year-old electret condenser microphone (ECM) type. The speech quality of the transmitted audio was often unsatisfactory, especially with simple stand-alone units, where the distance between the microphone and the speaker's mouth could be quite large. Communication quality can be improved if the microphone is mounted as close to the mouth as possible, for example in the roof panel of the vehicle. In this case, however, microphones would need to be present in both front seats if the driver and passenger were to be supported equally.
A typical automotive ECM microphone is a device that combines an ECM unit with a small amplifier circuit in a single housing. The amplifier provides an analog signal with a voltage level that allows the signal to be transmitted over several meters of wire, which is also a requirement for typical automotive applications. Without amplification, the raw ECM signal is too low for such long wires, and the signal-to-noise ratio (SNR) will be too much reduced due to electromagnetic interference on the wires. Even if the signal is amplified, a shielded cable is required - usually a two-wire cable, with a bias voltage (8V) to power the microphone device. Given this wiring requirement, it is obvious that the number of ECM devices used in mainstream vehicles is limited due to weight and system cost constraints.
One of the few advantages of an ECM is its built-in acoustic directivity, which is usually tuned to a supercardioid polar pattern (MEMS microphones can also be made directional, but this usually requires more complex acoustic design). It is common to achieve 10 dB or more of back-attenuation, where "backward" means in the direction toward the windshield, from which only noise (i.e., no desired signals, such as the talker's voice) will be generated. Having higher sensitivity in the incoming direction of the desired signal is very beneficial for improving SNR. However, directional ECM units introduce unwanted side effects, such as a high-pass characteristic - sensitivity is reduced at lower frequencies. The 3 dB cutoff frequency of this high-pass response is usually in the 300 Hz to 350 Hz range. In the early days of HF technology, this high-pass characteristic was an advantage because engine noise was mainly present at lower frequencies and the engine sound itself was attenuated by the microphone. However, since the advent of wideband or HD telephony, this high-pass characteristic has become a problem. In wideband telephony, the effective bandwidth increases from 300 Hz to 3400 Hz to 100 Hz to 7000 Hz. The inherent high-pass filtering characteristics of the microphone make it necessary to amplify the signal between 100 Hz and 300 Hz in the post-processing unit, while if the microphone itself can provide better audio bandwidth, there is no need to amplify the signal in this range. Another disadvantage of ECM technology is that the sensitivity and frequency response of different devices vary greatly. The manufacturing tolerance of ECM is relatively large, which may not be a problem for single microphone applications. However, if multiple microphone signals are deployed in a microphone array application with small spacing, strict matching between microphones is essential to achieve optimal array performance. In this case, ECM is difficult to use. In addition, from a physical size perspective, traditional ECM units are generally not suitable for small microphone arrays.
Microphone arrays have a wide range of applicability, including in-car, as they offer similar (and often superior) directional performance compared to traditional ECMs. Spatial information about the direction of sound impact can be extracted from the microphone signals using two or more microphones appropriately grouped in an array. Such algorithms are often referred to as beamforming (BF). The term "beamforming" comes from the analogy with phased array antenna technology, where a radio "beam" emitted by an antenna array can be focused in a certain direction using simple purely linear filters and summing algorithms. Although there are no such beams in microphone arrays, the term beamforming is also common in the field of microphone signal processing, covering a wider range of linear and nonlinear algorithms than simple linear beamforming processing, allowing for higher performance and greater flexibility.
In addition to beamforming processing, the raw microphone signals almost always require post-processing because each HF microphone captures both the desired speech signal and interference from the environment (e.g., the cabin). Wind, road, and engine noise degrade the SNR, and the signal played through the speakers—often referred to as speaker echo—is also a source of unwanted signals. To reduce this interference and improve speech quality, sophisticated digital signal processing techniques are required, often referred to as acoustic echo cancellation and noise reduction (AEC/NR). AEC removes the speaker sound from the microphone, which would otherwise be transmitted as an echo of the person speaking at the other end of the line. NR improves the SNR of the transmitted signal while reducing the ever-present driving noise. While the International Telecommunication Union (ITU) has published detailed specifications (e.g., ITU-T P.1100 and P.1110) to define many of the performance details of HF systems, when talking in a moving vehicle, if the AEC/NR processing is not up to par, the subjective impression of the communication quality may be unsatisfactory. Together with the BF algorithms mentioned earlier, the combination of AEC/NR/BF enables a wide range of new applications, all of which involve some degree of digital audio signal processing. To support these applications, a new generation of microphone technology is needed that eliminates the shortcomings of traditional ECMs.
Micro-electromechanical systems (MEMS) technology is quickly becoming the new industry standard for microphones because it offers many advantages over traditional ECMs. First, MEMS allow sound sensors to be much smaller than existing ECM units. In addition, integrating a MEMS sensor with an analog-to-digital converter (ADC) in a single IC results in a digital microphone that can provide a signal that is ready for AEC/NR/BF processing.
Analog interface MEMS microphones also exist, but they have many of the same disadvantages as analog ECMs, and even require more complex amplifier circuits than ECMs if they work with traditional two-wire analog interfaces. Only with fully digital interface technology can the interference and SNR problems inherent in analog lines be significantly alleviated. In addition, MEMS is also preferred from a production perspective, because the production specification deviations of MEMS microphones are much smaller than those of ECM units, which is important for BF algorithms. Finally, the manufacturing process of MEMS IC microphones is greatly simplified, because automated mounting techniques can be used, and the overall production cost can be reduced. From an application perspective, smaller size is the biggest advantage, and because the sound inlet is very small, MEMS microphone arrays can be made virtually invisible. The sensor inlet and sound channel require special care in design and production quality. If the acoustic seal is not strong, noise from the internal structure can reach the sensor, and leakage between the two sensors can degrade the performance of the BF algorithm. Unlike typical ECM units, which can be designed and manufactured to be omnidirectional or directional, MEMS microphone elements are almost always manufactured to be omnidirectional (i.e., there is no inherent directionality in sound reception). Therefore, MEMS microphones are phase-faithful omnidirectional sound pressure sensors that provide ideal signals for advanced BF algorithms, and the attenuation direction and beamwidth can be configured by the user through software.
In general, it is important to organize all signal processing blocks in an integrated algorithm suite. If functional blocks are implemented in isolation from each other, processing delays will be unnecessarily increased and overall system performance will be degraded. For example, the BF algorithm should always be implemented together with the AEC, preferably by the same provider. If the BF algorithm introduces any nonlinear effects in the signal, the AEC will certainly produce unsatisfactory results. The ideal result of digital signal processing is best achieved by an integrated algorithm package that receives undegraded microphone signals.
Below is a detailed comparison of the standard linear BF and ADI's proprietary algorithms to fully understand the performance potential of advanced BF algorithms. The curves in Figure 1 show the polar characteristics and frequency response of three different BF algorithms in the in-beam and out-of-beam directions. The standard linear hypercardioid algorithm based on a two-microphone array is used as a benchmark (black curve). The benchmark curve shows the maximum attenuation in the typical zero-angle direction (i.e., the maximum out-of-beam attenuation), as well as a "back lobe" at 180°, where the out-of-beam attenuation is lower. The resulting back lobe is the result of a trade-off with beamwidth in the linear algorithm. The cardioid beam (not shown) has the maximum attenuation exactly at 180°. However, its acceptance area is wider than that of the hypercardioid configuration. Beams with less pronounced back lobes and higher out-of-beam attenuation can be achieved with nonlinear algorithms, and the red curve shows this type of ADI's proprietary two-microphone algorithm (microphone spacing: 20 mm).
Figure 1. Polarity attenuation characteristics of different BF algorithms.
There are two omnidirectional microphones in the array, so there is always rotational symmetry in the beam shape. In other words, the attenuation at X° in the polar diagram is the same as the attenuation at 360° - x°. This assumes that the 0° to 180° line of the polar diagram is equivalent to an imaginary line connecting the two microphones. A three-dimensional beam shape can be imagined by rotating the two-dimensional polar curve about the axis of the microphone. Asymmetric beam shapes or narrower beams without rotational symmetry require at least three microphones arranged in a triangle. For example, in a typical overhead console installation, a two-microphone array can attenuate the sound coming from the windshield. However, when so oriented, the two-microphone array cannot distinguish between the driver and the passenger. Rotating the array 90° can distinguish between the driver and the passenger, but the noise generated by the windshield will be indistinguishable from the sound in the cabin. Only by using three or more omnidirectional microphones configured in an array can the windshield noise be attenuated and the driver and passenger be distinguished. The green curve in Figure 1 shows an exemplary polar characteristic of the corresponding ADI proprietary three-microphone algorithm, where the microphones are arranged in an equilateral triangle with a spacing of 20 mm.
Polar plots are calculated using band-limited white noise arriving at the microphone array from different angles. The audio bandwidth is limited to 100 Hz to 7000 Hz, which is the wideband (or HD Voice) bandwidth of advanced cellular phone networks. Figure 2 compares the frequency response curves of the different algorithm types. In the in-beam direction, the frequency response of all algorithms is flat within the desired audio bandwidth, as expected. The out-of-beam frequency response is calculated for the out-of-beam half-space (90° to 270°), confirming that the out-of-beam attenuation is high over a wide frequency range.
Figure 2. In-beam (dashed line) and out-of-beam (bold line) frequency responses of different BF algorithms.
The relationship between array microphone spacing and audio bandwidth and sampling rate deserves further discussion. Wideband HD voice uses a sampling rate of 16 kHz, which is a good choice for speech transmission. The current 16 kHz wideband sampling rate makes a huge difference in speech quality and speech intelligibility compared to the 8 kHz sampling rate used by earlier narrowband systems. Driven by speech recognition providers, there is a growing demand for higher sampling rates such as 24 kHz or 32 kHz. Voice-band applications may require sampling rates as high as 48 kHz, which is typically the main system audio sampling rate. The underlying motivation is to avoid sample rate conversion internally. However, the additional computational resources required to support these high sampling rates are not commensurate with the actual results they produce, so 16 kHz or 24 kHz are now widely accepted as the recommended sampling rates for most voice-band applications.
For beamforming applications, high sampling rates are problematic because spatial aliasing occurs at a frequency equal to the speed of sound divided by twice the microphone spacing. Beamforming cannot be performed at this aliasing frequency, so spatial aliasing is undesirable. Spatial aliasing can be avoided in wideband systems (16 kHz sampling rate) if the microphone spacing is limited to 21 mm or less. At higher sampling rates, the spacing needs to be smaller to avoid spatial aliasing. However, too small a microphone spacing is not desirable because microphone tolerances, especially the intrinsic (non-acoustic) noise of the microphone sensor, become a problem. If the spacing is too small, interference (such as intrinsic noise) and sensitivity deviations between the microphones of an array may overwhelm the signal differences between the microphones, causing the signal differences to become insignificant. In practice, the microphone spacing should not be less than 10 mm.
A2B technology is specifically designed to simplify the connectivity challenges of emerging automotive microphone and sensor-intensive applications. From an implementation perspective, A2B is a serial topology with a single master device and multiple sub-nodes (up to 10). The third-generation A2B transceiver family, currently in full production, has five members, all available in automotive, industrial, and consumer temperature ranges. The full-featured AD2428W and four reduced-featured, lower-cost derivatives—AD2429W, AD2427W, AD2426W, and AD2420W—form the latest pin-compatible enhanced A2B transceiver family from Analog Devices.
The AD2427W and AD2426W have reduced functionality (only for sub-nodes) and are targeted at microphone connectivity applications such as hands-free, ANC/RNC, or ICC. The AD2429W and AD2420W are entry-level A2B derivatives that offer significant cost advantages over full-featured devices, particularly for cost-sensitive applications such as automotive eCall and multi-element microphone arrays. Table 1 compares the features of various third-generation A2B transceivers .
Table 1. Comparison of A 2 B transceiver features
The AD242x family supports daisy-chaining a single master device and up to 10 child nodes, with a total bus distance of up to 40 meters and a maximum distance of up to 15 meters between nodes. A2B 's daisy-chain topology is a major advantage over existing ring/parallel topologies and is beneficial to the integrity and robustness of the overall system. If one connection in an A2B daisy chain is affected, the entire network will not collapse. Only nodes downstream of the faulty connection will be affected. A2B 's embedded diagnostics can determine the cause of the fault, issue an interrupt signal, and initiate corrective action.
The master-slave node topology of A2B is inherently more efficient than existing digital bus architectures . After a simple bus initialization process, the bus is operational without further processor intervention. An added benefit of A2B's unique architecture is that system latency is completely deterministic (less than 50 µs) and is independent of the location of the audio node on the A2B bus. This feature is extremely important for voice and audio applications such as ANC/RNC and ICC, where audio samples from multiple remote sensors must be processed in a time-coherent manner.
All A2B transceivers can transmit audio, control, clock, and power signals on a single unshielded twisted pair. This reduces overall system cost for the following reasons.
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The number of physical cables is reduced compared to traditional implementations.
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The actual cabling used can be lower cost and lighter weight unshielded twisted pair rather than more expensive shielded cable.
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Most importantly, for specific application scenarios, A 2 B technology provides bus power capability, delivering up to 300 mA of current to the audio nodes on the A 2 B daisy chain. With this bus power capability, there is no need to use a local power supply on the audio ECU, further reducing system cost.
The total 50 Mbps bus bandwidth provided by A2B technology can support up to 51 upstream and downstream audio channels using standard audio sampling rates (44.1 kHz, 48 kHz, etc.) and bit widths (16, 24 bits). This provides considerable flexibility and connectivity for a wide range of audio I/O devices. Maintaining a fully digital audio signal chain between audio ECUs ensures the highest quality audio without audio performance degradation due to ADC/DAC conversion.
Open circuit, wire short, wire reverse, wire short to power or ground. This feature is very important from a system integrity perspective because the A2B node upstream of the fault point can still operate normally in the event of an open circuit, wire short or wire reverse fault. The diagnostic function also provides the ability to effectively isolate system-level faults, which is critical from the perspective of the automotive dealer/installer.
The recently announced fourth generation A2B transceiver AD243x is an evolution of existing technology, improving key functional parameters (node count increased to 17, bus power increased to 50 W), while adding an additional SPI control channel (10 Mbps), providing efficient software over the air (SOTA) capability for remote programming of smart A2B nodes . The new features of the AD243x family make it ideal for new applications, such as microphone nodes equipped with LEDs in ultra-advanced microphone architectures.
From a single voice microphone to a multi-element BF microphone array for HF communication, from ANC to RNC, from ICC to alarm sound detection, microphones are increasingly used in the automotive industry. In line with technology and market trends, almost every new car on the road today is equipped with at least one microphone module for HF communication. Premium and luxury cars may have six or more microphone modules, which are necessary to realize the full potential of BF, AEC, ANC, RNC, ICC, etc. Digital MEMS microphones have obvious advantages in these applications.
The increasing number of microphones presents a major challenge to vehicle infotainment system engineers – how to simplify the connection harness and make it lightest. This is not a simple task for traditional analog systems. Analog microphones require at least one pair of double-shielded wires (ground and signal/power), pins and connector cavities for interconnection. The amount of wires is always twice the number of microphone modules in the system. At the same time, the length of wire required to connect each microphone module causes the total weight of the harness to increase faster. A simple way to alleviate this problem is to share the microphone signal between multiple applications, thereby reducing the number of microphones used in the system. For example, the same microphone signal can be used for HF communication and as the Error input in the ANC system. However, different applications may require different microphone characteristics. In the example mentioned above, the HF microphone signal is often more desirable to have a rising frequency response shape (i.e., the sensitivity decreases as the frequency decreases) to eliminate the low-frequency noise content in the cabin. This is a useful and very effective technique to improve the speech intelligibility delivered by the speech microphone. In contrast, the ANC microphone requires a sufficiently high sensitivity level at low frequencies, because the main purpose of the ANC algorithm is to reduce low-frequency noise. Therefore, in order for two applications in an analog system to share the same microphone, the signal from the microphone needs to be fed into different circuits for appropriate frequency filtering. In this case, one or more ground loops may form, which may cause serious noise problems.
As a digital bus with daisy-chain capability, A2B technology , together with digital MEMS microphones, provides a multi-microphone signal interconnection and/or sharing solution that is ideally suited to meet the needs of the rapidly expanding audio, voice, noise cancellation, and other acoustic applications in vehicles. Consider a fictitious but illustrative scenario: an automotive application requires an HF microphone module, an ANC microphone module, and a simple array microphone module consisting of two microphone elements for BF, all three of which are integrated around a dome light module. Figures 3a and 3b show how this design can be implemented using a traditional analog system and a digital A2B system , respectively .
Figure 3. (a) Analog system design using analog microphone components (shielded wire). (b) Digital system design using digital microphone components (A2B
technology
and UTP wire).
Because analog systems cannot easily support microphone sharing, each application module (HF, ANC, and BF) requires a dedicated microphone and a separate wiring harness to connect the corresponding functional circuits. This results in the need for four separate microphone elements and three sets of wiring harnesses (a total of seven wires plus shielding). On the other hand, digital A2B systems can easily support shared signals, so the number of microphone elements can be reduced from four to two. In this specific example, a single microphone module consisting of two wideband omnidirectional microphone elements can be used to provide two acoustic signal channels to meet the needs of all application modules. Once the signals of these two channels reach the central processing unit (such as the head unit or independent power amplifier) through a simple UTP line, they can be shared and digitally processed to support HF, ANC, and BF applications.
While the example shown in Figure 3 may not represent a real-world situation, it clearly demonstrates the advantages of A2B technology over traditional analog technology. Digital audio bus systems such as A2B technology address the challenges of automakers, allowing them to come up with new audio and acoustic-related concepts to enhance the user experience and enable faster time to market for these concepts.
In fact, the commercialization of A2B technology has enabled many applications in the automotive market, some of which are new and some of which were previously difficult to achieve. For example, Harman International, a leading provider of automotive audio solutions, has developed a series of digital microphone and sensor modules that use A2B systems to enable a variety of automotive applications. Figure 4 shows some common automotive A2B microphones and sensors and how they are used in automobiles. These sensors include: single A2B microphones , multi-element microphone arrays for ANC and voice communication, A2B accelerometers for RNC , externally mounted bumper A2B microphones , and roof A2B microphone arrays for emergency alarm detection and acoustic environment monitoring . With the empowerment of these A2B microphones and accelerometers, more and more application solutions that require multi-sensor inputs are being developed to further enhance the user experience in the automotive industry.
Future vehicle architectures will increasingly rely on high-performance acoustic sensing technologies such as microphones and accelerometers. A fully digital approach including sensors, interconnects, and processors can deliver significant performance and system cost advantages. Analog Devices is collaborating with Harman International to deliver cost-effective solutions that create value and differentiation for end customers.
Figure 4. Common A
2
B microphones and sensors.
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