As handheld devices become more compact, the demand for higher sound output from smaller speakers continues to rise. At the same time, managing background noise and echoes—both linear and nonlinear during hands-free calls—has become a major challenge in achieving clear voice communication in noisy environments. ForteMedia’s FM2010 chip addresses these issues through its patented Mini Array Microphone (SAM) technology. This advanced solution uses spatial filtering to capture sound from both near and far distances while suppressing acoustic noise and reducing low-frequency interference, all at a single-chip cost. In this article, we’ll explore the design principles of mini-array microphone technology in mobile devices, the key features of the FM2010 chip, and its practical application in GSM phones.
Mini Array Microphone Technology Design Essentials SAM technology leverages two microphones—a Uni-MIC (main microphone) and an Omni-MIC (reference microphone)—to create a compact array. These are positioned either back-to-back or side-by-side, taking advantage of their physical differences. The FM2010 processes these signals to form a directional beam pattern, effectively acting as a spatial filter that suppresses unwanted noise. The performance of this system depends on the microphone characteristics, structural design, and parameter settings of the FM2010, which influence the beam’s direction, angle, and noise suppression efficiency.
1. Microphone Selection A 4mm Uni-MIC and Omni-MIC pair is 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 more than 10dB between 0° and 180°. The Omni-MIC offers a sensitivity of -40dB ±1.5dB with a flat frequency response across 300Hz to 3.4kHz. It is recommended to use the Uni-MIC model B4015UL403 and the Omni-MIC model B4015AL-398 from Shandong Weifang Yilida (IEA).
2. Structural Design The structural design must preserve the directional characteristics of the Uni-MIC and ensure the cone-shaped pickup beam remains effective. For hands-free calling, additional attention should be given to speaker placement, microphone damping, and airtightness. The direction of the pickup beam determines where noise is suppressed, so it's crucial that the desired signal lies within the beam’s coverage. During product design, the orientation of the Uni-MIC should be carefully considered. The structure must maintain a sensitivity difference of over 6dB between 0° and 180° after the microphone is installed in the housing, ensuring minimal impact on frequency response. Proper damping can reduce nonlinear echo and improve the system’s echo return loss.
3. Signals Processed by FM2010 The signals captured by the mini-array microphone and those processed by the FM2010 are compared in Figure 4. A sound source located 0.3 meters away produces an SPL of 83dB. The test signals include outputs from the Uni-MIC and Omni-MIC at 0° and 180°, as well as the line output (Lout) of the mini-array microphone at the same angles. As shown, the signal within the pickup beam (0°) differs from the one outside (180°) by up to 20dB. This means that any unsteady noise outside the beam is suppressed by 20dB relative to the useful signal. The SAM beam has an effective range of 2 meters, with the beam angle determined by the Uni-MIC’s directivity and FM2010 parameter settings.
The signals from the Uni-MIC and Omni-MIC are first amplified via a programmable gain amplifier, then converted to digital format, and high-pass filtered before being sent to the speech processor. This includes linear and nonlinear echo cancellation, VAD detection, noise suppression, and volume control. After digital conversion, the microphone signal is output through the line output (Lout) to the TWL3014/16 analog baseband processor, which sends the signal to the OMAP733/750 digital baseband processor. Once demodulated and decoded, the audio signal is sent back through the TWL3014/16 to the receiver and earpiece, then amplified and played through the hands-free speaker.
The FM2010 is controlled via the SHI interface, along with PWD, RESET, and ANA_IRQ control pins. Upon power-up, PWD is set high, ANA_IRQ is low, and after a reset, parameters such as clock source, frequency, and DSP speed are configured. The chip is then put into power-saving mode. Depending on whether the device is in call, outgoing, or recording mode, the FM2010 is woken up, and corresponding mode parameters are loaded. In handheld noise cancellation mode, parameters like microphone count, gain, volume, and VAD settings are adjusted. For hands-free conference mode, additional parameters such as microphone inversion and echo cancellation thresholds are required. The debugging process for hands-free personal mode is similar to the handheld mode, but echo cancellation settings require more fine-tuning.
After the call ends, the FM2010’s CODEC is turned off, and the chip returns to power-saving mode. In the GSM phone audio test mode, the FM2010 operates in pass-through mode, allowing the internal DSP to bypass processing. The Uni-MIC signal is amplified and directly output through the LOUT, and can be connected to the microphone amplifier via the SHI interface. The LOUT gain parameter can be adjusted accordingly.
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