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R3710 Datasheet(PDF) 8 Page - ON Semiconductor |
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R3710 Datasheet(HTML) 8 Page - ON Semiconductor |
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8 / 18 page  RHYTHM R3710 www.onsemi.com 8 SIGNAL PATH There are two inputs into the audio signal path. The first input is the front microphone and the second input can be a second microphone or telecoil input as selected by a programmable MUX. The front microphone input is intended as the main microphone audio input. Analog input signals should be ground referenced to MGND (microphones, telecoils, DAI). MGND is internally connected to GND to minimize noise, and should not be connected to any external ground point. The audio input is buffered, sampled and converted into digital form using an A/D converter. The digital output is converted into a selectable 32 kHz or 16 kHz, 20−bit digital audio signal. Further IIR filter blocks process the microphone signal. These are followed by four cascaded biquad filters: pre1, pre2, pre3 and pre4. These filters can be used for frequency response shaping before the signal goes through channel and adaptive processing. The channel and adaptive processing consists of the following: • Frequency band analysis • 1, 2, 4, 6 or 8 channel WDRC • 16 frequency shaping bands (spaced linearly at 500 Hz intervals, except for first and last bands) • 128 frequency band adaptive noise reduction • Frequency band synthesis After the processing the signal goes through two more biquad filters, post1 and post2, which are followed by the AGC−O block. The AGC−O block incorporates the Wideband Gain and the Volume Control. There are also two more biquad filters, post3 and post4, and the Peak Clipper. The last stage in the signal path is the D/A H−bridge. White noise can be shaped, attenuated and then added into the signal path at two possible locations: before the Volume Control (between the Wideband Gain and the Volume Control) or after the Volume Control (between post 4 and the Peak Clipper) as shown in Figure 1. Functional Block Description iSceneDetect 1.0 Environment Classification The iSceneDetect feature, when enabled, will sense the environment and automatically control the enhancement algorithms without any user involvement. It will detect speech in quiet, speech in noise, music, quiet and noise environments and make the necessary adjustments to the parameters in the audio path, such as ANR, WDRC and FBC, in order to optimize the hearing aid settings for the specific environment. iSceneDetect will gradually make the adjustments so the change in settings based on the environment is smooth and virtually unnoticeable. This feature will enable the hearing aid wearer to have an instrument which will work in any environment with a single memory. Evoke Advanced Acoustic Indicators Advanced acoustic indicators provide alerting sounds that are more complex, more pleasing and potentially more meaningful to the end user than the simple tones used on previous products. The feature is capable of providing pulsed, multi−frequency pure tones with smooth on and off transitions and also damped, multi−frequency tones that can simulate musical notes or chords. A unique indicator sound can be assigned to each of the ten system events: memory select (A, B, C, or D), low battery warning, digital VC movement and digital VC minimum/maximum. Each sound can consist of a number of either pure tones or damped tones but not both. A pure tone sound can consist of up to four tones, each with a separate frequency, amplitude, duration and start time. Each frequency component is smoothly faded in and out with a fade time of 64 ms. The start time indicates the beginning of the fade in. The duration includes the initial fade−in period. By manipulating the frequencies, start times, durations and amplitudes various types of sounds can be obtained (e.g., various signalling tones in the public switched telephone network). A damped tone sound can consist of up to six tones, each with a separate frequency, amplitude, duration, start time and decay time. Each frequency component starts with a sudden onset and then decays according to the specified time constant. This gives the audible impression of a chime or ring. By manipulating the frequencies, start times, durations, decays and amplitudes, various musical melodies can be obtained. Acoustic indication can be used without the need to completely fade out the audio path. For example, the low−battery indicator can be played out and the user can still hear an attenuated version of the conversation. Adaptive Feedback Canceller The Adaptive Feedback Canceller (AFC) reduces acoustic feedback by forming an estimate of the hearing aid feedback signal and then subtracting this estimate from the hearing aid input. The forward path of the hearing aid is not affected. Unlike adaptive notch filter approaches, the AFC algorithm does not reduce the hearing aid’s gain. The AFC is based on a time−domain model of the feedback path. The third−generation AFC (see Figure 5) allows for an increase in the stable gain (see Note) of the hearing instrument while minimizing artefacts for music and tonal input signals. As with previous products, the feedback canceller provides completely automatic operation. NOTE: Added stable gain will vary based on hearing aid style and acoustic setup. Please refer to the Adaptive Feedback Cancellation Information note for more details. |
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