Design of handheld communication products based on mini array microphone technology

Publisher:清新自然Latest update time:2011-05-27 Source: 今日电子Keywords:FM2010 Reading articles on mobile phones Scan QR code
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The space restrictions on handheld product structures are increasing, speakers are getting smaller, and the required sound is getting louder. Suppressing various noises and echoes (linear and nonlinear echoes during video hands-free calls) and achieving clear voice communication in a noisy environment is a problem that needs to be solved.

The FM2010 chip launched by ForteMedia of the United States is based on the patented technology of mini array microphone (SAM). It adopts spatial filtering technology, long/near distance directional pickup of sound signals, suppression of acoustic noise, and elimination of acoustic echo. It is a low-power and low-cost single chip. This article will introduce the design points of mini array microphone technology in handheld communication products, the main functions of the FM2010 chip, and its typical application in GSM mobile phones. Figure 1 is a schematic diagram of anti-noise in handheld mode; Figure 2 is a schematic diagram of anti-noise in hands-free mode.

SAM handheld mode noise immunity diagram

Figure 1 Schematic diagram of SAM handheld mode noise immunity

SAM hands-free mode anti-noise schematic diagram

Figure 2 Schematic diagram of SAM hands-free mode noise immunity

Key points of mini array microphone technology design

SAM technology can use Uni-MIC (main microphone) and Omni-MIC (reference microphone) to form a mini array microphone, which can be placed back to back or side to side, as shown in Figure 3. By using the difference in the physical characteristics of the two microphones, after being processed by the FM2010 chip, a cone-shaped pickup beam is formed, forming a spatial filter to suppress non-steady-state noise. The characteristics, structural design of the microphone and the parameter adjustment of FM2010 determine the direction, angle and effect of suppressing non-steady-state noise of the cone pickup beam.

Microphone placement

Figure 3 Microphone placement [page]

1. Microphone selection

It is recommended to use 4mm Uni and Omni microphones. The sensitivity of Uni-MIC is -40dB ±3dB; the spectrum drops less than 8.5dB at 300Hz, and the rise is less than 3.5dB at 3.4kHz; the directivity is cardioid, the sensitivity difference between 0° and 90° is greater than 4dB, and the sensitivity difference between 0° and 180° is greater than 10dB. The sensitivity of Omni-MIC is -40dB±1.5dB; the spectrum is flat from 300Hz to 3.4kHz. It is recommended to use Uni microphone B4015UL403 and Omni microphone B4015AL-398 from Shandong Weifang Yilida (IEA).

2 Structural design

The key issues in structural design are the characteristics of the Uni microphone and the direction of the conical pickup beam. If there is a hands-free call function, more consideration needs to be given to the speaker, microphone vibration reduction and microphone airtightness. The direction of the conical pickup beam determines the direction of suppressing non-steady-state noise, so the useful signal should be kept within the pickup beam, otherwise it will be treated as noise and suppressed. Therefore, when designing the product appearance, the direction of the Uni microphone should be fully considered. The structural design should ensure that after the Uni microphone is installed in the microphone sleeve and the entire casing, the sensitivity difference between 0° and 180° is greater than 6dB, and the sensitivity and frequency characteristics remain basically unchanged. Microphone vibration reduction can reduce nonlinear echo, and airtightness can reduce linear echo, thereby improving the signal-to-echo ratio of the system.

3 Signal processed by FM2010

The comparison of the signal picked up by the mini array microphone and the signal output after being processed by FM2010 is shown in Figure 4. The sound source is 0.3m away from the mini array microphone, and the sound intensity is 83dB (SPL). The test signals are the signals picked up by the Uni microphone 0° and 180°, the Omni microphone 0° and 180°, and the line output (Lout) mini array microphone 0° and 180°, and the signals output after being processed by FM2010. As can be seen from Figure 4, for the same signal size, the final output signal inside the pickup beam (0°) and outside the pickup beam (180°) can differ by 20dB. In other words, as long as the non-steady-state noise is outside the pickup beam, it will be suppressed by 20dB relative to the useful signal. The effective range of the cone pickup beam is 2m. The angle of the SAM cone pickup beam depends on the directional characteristics of the Uni microphone and the parameter adjustment of FM2010. Figure 5 is the cone pickup beam direction diagram of the actual test SAM.

SAM's conical pickup beam suppresses noise

Figure 4 SAM's conical pickup beam suppresses noise

SAM's conical pickup beam pattern

Figure 5 SAM's conical pickup beam pattern and acoustic effect [page]

Application of FM2010 in GSM mobile phone

FM2010 adopts low-power and small-size design, consuming less than 25mW. It integrates DSP , CODEC, ROM and RAM on the chip , making it suitable for mobile handheld device applications.

Taking TI's mobile phone platform as an example, as shown in Figure 6, there are four main interfaces between FM2010 and the analog baseband processor: reference signal for echo cancellation (HS REF) for receiver, reference signal for echo cancellation for hands-free speaker (HF REF), array microphone power supply and control (MIC PWR), and output signal (LOUT) of array microphone signal after being processed by FM2010. There are six main interfaces between FM2010 and the digital baseband processor: clock signal (13MHz), SHI interface selection control (SHI_S), SHI interface (SHI), pass-through mode selection control (IRQ_ANA), reset control (RESET), and power saving control (PWD).

Application diagram of FM2010 on TI mobile phone platform

Figure 6 FM2010 application block diagram on TI mobile phone platform

The signals picked up by Uni microphone and Omni microphone are amplified by programmable gain amplifier , converted by analog to digital and high-pass filtered, and then sent to voice processor for processing (linear echo cancellation, nonlinear echo cancellation, VAD detection, noise suppression processing, microphone volume setting). After digital-to-analog conversion, the output voice data is sent from the line output end to the microphone input end (MICIP/MIC1N) of TWL3014/16 analog baseband processor, and then sent to the digital baseband processor OMAP733/750 for processing after uplink processing. After the GSM mobile phone receives the signal and demodulates it, it is decoded by the signal digital baseband processor OMAP733/750, and after confidentiality processing, the digital audio signal is sent to the TWL3014/16 analog baseband processor, and then output from the receiver (HSO) and earphone (EARP/EARN) through downlink processing. One channel is sent to the receiver, and the other channel is sent to the external power amplifier for amplification and driving the hands-free speaker. At the same time, the two channels of signals are sent to the line input of FM2010 respectively, converted into voice data through analog-to-digital conversion and high-pass filtering, and sent to the voice processor for processing as an echo reference signal. The typical application is shown in Figure 7.

FM2010 Typical Application Schematic Diagram

Figure 7 FM2010 typical application schematic

The working mode of FM2010 in GSM mobile phone can be set as needed, mainly including test mode, handheld noise cancellation mode, hands-free conference mode, and hands-free personal mode. Test mode is mainly used for mobile phone audio testing (Acoustics). At this time, FM2010 is set to enter the direct mode (IRQ_ANA). FM2010 only amplifies the Uni microphone signal without any other processing, and outputs it from the line output pin; in the handheld noise cancellation mode, FM2010 works in the Uni and Omni microphones two microphone SAM mode to eliminate steady-state and non-steady-state noise in the call environment; the hands-free conference mode is used in a small multi-person conference mode. At this time, FM2010 works in the microphone inversion mode. FM2010 amplifies and processes the signal of the Omni microphone, eliminates the steady-state echo in the call environment, and outputs it from the line output pin. The microphone can pick up omni-directional signals; the hands-free personal mode can be used for conference occasions or video calls for one person. [page]

The control flow of OMAP733/750 to FM2010: the operation of FM2010 is realized through the SHI interface, PWD, RESET, and ANA_IRQ control pins. After the whole machine is powered on, PWD is first set high and ANA_IRQ is set low. After reset, parameters are sent to FM2010, mainly clock source, clock frequency, DSP working rate, and then the chip enters power saving mode. As shown in Figure 8, the call mode wakes up FM2010 according to whether there is an incoming call, outgoing call or recording operation. After reset, the corresponding mode parameters of FM2010 are sent according to the handheld/hands-free mode. For the handheld noise cancellation mode, the number of microphones, microphone gain, microphone volume, echo cancellation parameters and VAD parameters need to be sent. For the hands-free conference mode, the number of microphones, microphone gain, microphone volume, microphone inversion, and echo cancellation parameters (mainly including when the other party speaks quietly, speaks normally, and speaks loudly) need to be sent. The debugging method of the hands-free personal dedicated mode is basically the same as the handheld noise cancellation mode, and the echo cancellation parameters need more adjustment. After the call is over, turn off the CODEC of FM2010 and set FM2010 to power saving mode. As shown in Figure 9, the GSM mobile phone audio test mode is mainly used for mobile phone testing purposes. After entering the test mode, FM2010 works in the pass-through mode. At this time, the internal DSP does not process the microphone input signal. The Uni microphone input signal is amplified by the programmable gain amplifier and directly output from the LOUT amplifier. The online microphone amplifier and LOUT amplifier gain parameters can be controlled through the SHI interface.

Call mode control process

Figure 8 Call mode control flow

Test mode control flow

Figure 9 Test mode control flow

FM2010 SHI interface control timing: SHI interface works in slave mode, with clock signal provided externally, and maximum clock frequency of 400kHz. When setting the mode and parameters of FM2010, first send a transmission start status, then send 1-byte SHI device address C0, 2-byte synchronization word FCF3, then control command word 3B, 2-byte parameter address, 2-byte parameter data, repeat then control command word 3B, 2-byte parameter address, 2-byte parameter data, and finally start command control command word 3B, 2-byte parameter address 1E 3A, 2-byte parameter data 00 00 to end, so that FM2010 can run normally.

Keywords:FM2010 Reference address:Design of handheld communication products based on mini array microphone technology

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