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NRF9160-SICA-R7 Datasheet(PDF) 150 Page - Nordic Semiconductor |
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NRF9160-SICA-R7 Datasheet(HTML) 150 Page - Nordic Semiconductor |
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150 / 400 page ![]() Peripherals 6.9.2 Module operation By default, bits from the left PDM microphone are sampled on PDM_CLK falling edge, and bits for the right are sampled on the rising edge of PDM_CLK, resulting in two bitstreams. Each bitstream is fed into a digital filter which converts the PDM stream into 16-bit PCM samples, then filters and down-samples them to reach the appropriate sample rate. The EDGE field in the MODE register allows swapping left and right, so that left will be sampled on rising edge, and right on falling. The PDM module uses EasyDMA to store the samples coming out from the filters into one buffer in RAM. Depending on the mode chosen in the OPERATION field in the MODE register, memory either contains alternating left and right 16-bit samples (Stereo), or only left 16-bit samples (Mono). To ensure continuous PDM sampling, it is up to the application to update the EasyDMA destination address pointer as the previous buffer is filled. The continuous transfer can be started or stopped by sending the START and STOP tasks. STOP becomes effective after the current frame has finished transferring, which will generate the STOPPED event. The STOPPED event indicates that all activity in the module is finished, and that the data is available in RAM (EasyDMA has finished transferring as well). Attempting to restart before receiving the STOPPED event may result in unpredictable behavior. 6.9.3 Decimation filter In order to convert the incoming data stream into PCM audio samples, a decimation filter is included in the PDM interface module. The input of the filter is the two-channel PDM serial stream (with left channel on clock high, right channel on clock low). Depending on the RATIO selected, its output is 2 × 16-bit PCM samples at a sample rate either 64 times or 80 times (depending on the RATIO register) lower than the PDM clock rate. The filter stage of each channel is followed by a digital volume control, to attenuate or amplify the output samples in a range of -20 dB to +20 dB around the default (reset) setting, defined by GPDM,default. The gain is controlled by the GAINL and GAINR registers. As an example, if the goal is to achieve 2500 RMS output samples (16-bit) with a 1 kHz 90 dBA signal into a -26 dBFS sensitivity PDM microphone, do the following: • Sum the PDM module's default gain ( GPDM,default ) and the gain introduced by the microphone and acoustic path of his implementation (an attenuation would translate into a negative gain) • Adjust GAINL and GAINR by the above summed amount. Assuming that only the PDM module influences the gain, GAINL and GAINR must be set to -GPDM,default dB to achieve the requirement. With GPDM,default=3.2 dB, and as GAINL and GAINR are expressed in 0.5 dB steps, the closest value to program would be 3.0 dB, which can be calculated as: GAINL = GAINR = (DefaultGain - (2 * 3)) Remember to check that the resulting values programmed into GAINL and GAINR fall within MinGain and MaxGain. 6.9.4 EasyDMA Samples will be written directly to RAM, and EasyDMA must be configured accordingly. The address pointer for the EasyDMA channel is set in SAMPLE.PTR register. If the destination address set in SAMPLE.PTR is not pointing to the Data RAM region, an EasyDMA transfer may result in a HardFault or RAM corruption. See Memory on page 21 for more information about the different memory regions. DMA supports Stereo (Left+Right 16-bit samples) and Mono (Left only) data transfer, depending on the setting in the OPERATION field in the MODE register. The samples are stored little endian. 4418_1315 v2.0 150 |
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