As handheld devices become more compact, the size of speakers continues to shrink, while the demand for louder sound increases. At the same time, various noises and echoes—both linear and nonlinear during video hands-free calls—must be effectively suppressed. Achieving clear voice communication in noisy environments remains a significant challenge.
ForteMedia’s FM2010 chip leverages its patented Mini Array Microphone (SAM) technology, which uses spatial filtering to capture sound from both near and far distances, suppress acoustic noise, and reduce low-frequency distortions—all at a low cost per chip. This article explores the key design aspects of mini-array microphone technology in portable communication devices, the main features of the FM2010 chip, and its typical applications in GSM mobile phones.
Mini Array Microphone Technology: Key Design Considerations SAM technology utilizes two microphones—an Uni-MIC (main microphone) and an Omni-MIC (reference microphone)—to create a mini-array configuration. These can be placed back-to-back or side-by-side, leveraging their physical differences. The signals are then processed by the FM2010 chip to form a directional pickup beam, acting as a spatial filter that suppresses background noise. The performance of this beam is influenced by the microphone characteristics, structural design, and parameter settings on the FM2010.
1. Microphone Selection A 4mm Uni-MIC and Omni-MIC are recommended for optimal performance. The Uni-MIC has a sensitivity of -40dB ±3dB, with a flat frequency response below 8.5dB at 300Hz and less than 3.5dB at 3.4kHz. Its directional pattern is heart-shaped, with a sensitivity difference of over 4dB between 0° and 90°, and over 10dB between 0° and 180°. The Omni-MIC has a sensitivity of -40dB ±1.5dB and a flat response across 300Hz to 3.4kHz. It is advisable to use the Uni-MIC B4015UL403 and Omni-MIC B4015AL-398 from Shandong Weifang Yilida (IEA).
2. Structural Design The primary concern in structural design is maintaining the directional characteristics of the Uni-MIC and ensuring the pickup beam direction aligns with the desired signal source. For hands-free calling, additional attention should be given to speaker-microphone damping and microphone airtightness. The direction of the cone-shaped pickup beam determines where noise is suppressed, so it's essential that the useful signal lies within the beam. Otherwise, it may be mistakenly treated as noise. During product design, the orientation of the Uni-MIC must be carefully considered. The structure should ensure a sensitivity difference of more than 6dB between 0° and 180° after the Uni-MIC is installed in the housing, with minimal changes to frequency response and sensitivity. Proper microphone damping helps reduce nonlinear echo and improves the system's echo ratio.
3. Signals Processed by FM2010 The signals captured by the mini-array microphone and those processed by the FM2010 are compared in Figure 4. The sound source is located 0.3 meters away with a sound level of 83dB (SPL). The test signals include the outputs from the Uni-MIC and Omni-MIC at 0° and 180°, as well as the line output (Lout) from the mini-array microphone at 0° and 180°. As shown in the figure, the signal levels differ by up to 20dB between the pickup beam (0°) and outside the beam (180°). This means that any unsteady noise outside the beam is suppressed by 20dB relative to the useful signal. The effective range of the cone-shaped beam is approximately 2 meters, and its angle depends on the directional characteristics of the Uni-MIC and the parameter settings on the FM2010.
The signals from the Uni-MIC and Omni-MIC are first amplified by a programmable gain amplifier, converted to digital via ADC, and high-pass filtered before being sent to the speech processor for tasks such as linear and nonlinear echo cancellation, VAD detection, noise suppression, and microphone volume adjustment. After digital conversion, the microphone signal is output through the line output and sent to the TWL3014/16 analog baseband processor’s MICIP/MIC1N input. The signal is then processed through uplink processing and sent to the OMAP733/750 digital baseband processor. Once received, the signal is demodulated and decoded. After confidential processing, the audio data is sent back to the TWL3014/16 for downlink processing via the receiver (HSO) and earpiece (EARP/EARN), eventually reaching the external power amplifier to drive the hands-free speaker. Two signals are also sent to the FM2010’s line input, where they undergo analog-to-digital conversion and high-pass filtering before being processed as an echo reference signal.
The control flow of the FM2010 is managed via the SHI interface, PWD, RESET, and ANA_IRQ pins. Upon power-on, PWD is set high, ANA_IRQ is set low, and after a reset, the chip receives parameters such as clock source, frequency, and DSP speed. It then enters power-saving mode. As shown in Figure 8, the FM2010 is activated based on whether a call is in progress, outgoing, or recording. Parameters are sent accordingly depending on the device mode—handheld or hands-free. In handheld mode, parameters like microphone count, gain, volume, and cancellation settings are adjusted. In hands-free conference mode, additional parameters such as microphone inversion and echo cancellation settings (including different voice levels) are configured. The debugging process for the hands-free personal-only mode is similar to the handheld noise-cancelling mode, but requires more precise echo cancellation adjustments. After the call ends, the FM2010’s CODEC is turned off, and the chip returns to power-saving mode.
As shown in Figure 9, the GSM phone audio test mode is used for testing purposes. When entering this mode, the FM2010 operates in pass-through mode, meaning no internal DSP processing occurs. The Uni-MIC signal is amplified by the programmable gain amplifier and directly output through the LOUT amplifier, which can be connected to the microphone amplifier via the SHI interface. The LOUT amplifier gain parameter can also be adjusted.
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