ST / X_NUCLEO_CCA02M1

Library to handle the X-NUCLEO-CCA02M1 MEMS Microphones Expansion Board.

Dependencies:   ST_I2S ST_FREQUENCY_DIVIDER USBDEVICE

Dependents:   HelloWorld_CCA02M1 HelloWorld_CCA02M1_mbedOS HelloWorld_CCA02M1 Karaoke_CCA01M1_CCA02M1_mbedOS

Fork of X_NUCLEO_CCA02M1 by ST Expansion SW Team

MEMS Microphones Library

Library to handle the X-NUCLEO-CCA02M1 MEMS Microphones Expansion Board. A single board allows to record a standard 2-channel stereo signal as an array of PCM samples (16 bit/sample); in principle, it could make use of six additional MEMS microphones to realize a 8-channel audio system.


Microphones configuration

Currently the configurations supported are the following:

  • Stereo@48KHz
  • Stereo@44.1KHz (CD audio quality)
  • Stereo@32KHz
  • Stereo@16KHz
  • Stereo@8KHz
  • Mono@48KHz
  • Mono@44.1KHz
  • Mono@32KHz
  • Mono@16KHz
  • Mono@8KHz

Mono configurations need a Jumper connecting PB_5 and PB_13 on the Morpho connector to properly work.


Platform compatibility

  • This board can be currently used with the Nucleo F4 Family only, please see the ST_I2S library compatibility for further information.
  • The library is compatible both with mbed OS 5.x and mbed classic 2.x (to work with mbed classic, the main application has to import the "events" library, which is not included into the "mbed" library).


I2S Peripheral Usage

By default this board makes use of the I2S peripheral available on Nucleo boards.


Acquiring through the USB

In order to acquire the recorded PCM audio channel with an audio SW on a PC, please connect the expansion board to a USB port of the PC, and the Nucleo board to a USB power supply.

Diff: BSP/PDM2PCMAudio.cpp

Revision:
20:9952bef19da1
Parent:
11:b2f7f79026e4
--- a/BSP/PDM2PCMAudio.cpp	Thu May 04 10:39:39 2017 +0000
+++ b/BSP/PDM2PCMAudio.cpp	Fri May 05 11:34:10 2017 +0000
@@ -68,55 +68,58 @@
 /* Methods -------------------------------------------------------------------*/
 
 /**
-* @brief  Converting audio data from PDM to PCM.
-* @param  output_buffer     Pointer to output PCM buffer data.
-* @param  input_buffer      Pointer to input PDM buffer data.
-* @param  volume            Volume level (it must be in the range [0..64]).
-* @param  decimation_factor Decimation factor (it must be either 64 or 128).
-* @retval COMPONENT_OK in case of success, COMPONENT_ERROR otherwise.
-*/
-status_t PDM2PCMAudio::convert(int16_t *output_buffer, uint16_t *input_buffer, uint32_t volume, uint32_t decimation_factor)
+ * @brief  Getting number of PCM samples from nummber of PDM samples.
+ * @param  PCM_samples Number of PCM samples.
+ * @retval Number of equivalent PDM samples.
+ */
+uint32_t PDM2PCMAudio::pcm2pdm_samples(uint32_t PCM_samples)
+{
+    return PCM_samples * (_decimation_factor >> 4);
+}
+
+/**
+ * @brief  Converting audio data from PDM to PCM.
+ * @param  output_buffer     Pointer to output PCM buffer data.
+ * @param  input_buffer      Pointer to input PDM buffer data.
+ * @param  volume            Volume level (it must be in the range [0..PDM2PCM_MAX_VOLUME]).
+ * @retval COMPONENT_OK in case of success, COMPONENT_ERROR otherwise.
+ */
+status_t PDM2PCMAudio::convert(int16_t *output_buffer, uint16_t *input_buffer, uint32_t volume)
 {
     if (!(volume <= PDM2PCM_MAX_VOLUME))
-        error("Volume level not supported: it must be in the range [0..64].\r\n");
+        error("Volume level not supported: it must be in the range [0..%d].\r\n", PDM2PCM_MAX_VOLUME);
 
-    switch (decimation_factor)
+#ifdef PDM2PCM_AUDIO_DEBUG
+                _pdm2pcm_audio_signal = 1;
+#endif
+    switch (_decimation_factor)
     {
         case 64:
             for (uint32_t index = 0; index < _channels; index++) {
-#ifdef PDM2PCM_AUDIO_DEBUG
-                _pdm2pcm_audio_signal = 1;
-#endif
 #ifdef USE_OPEN_PDM2PCM_LIBRARY
-                Open_PDM_Filter(&((uint8_t *) input_buffer)[index], (uint16_t *) &(output_buffer[index]), volume, (TPDMFilter_InitStruct *) &_PDM2PCM_filter[index]);
+                Open_PDM_Filter_64(&((uint8_t *) input_buffer)[index], (uint16_t *) &(output_buffer[index]), volume, (TPDMFilter_InitStruct *) &_PDM2PCM_filter[index]);
 #else                
                 PDM_Filter_64_LSB(&((uint8_t *) input_buffer)[index], (uint16_t *) &(output_buffer[index]), volume, (PDMFilter_InitStruct *) &_PDM2PCM_filter[index]);
 #endif
-#ifdef PDM2PCM_AUDIO_DEBUG
-                _pdm2pcm_audio_signal = 0;
-#endif
             }
             break;
 
         case 128:
             for (uint32_t index = 0; index < _channels; index++) {
-#ifdef PDM2PCM_AUDIO_DEBUG
-                _pdm2pcm_audio_signal = 1;
-#endif
 #ifdef USE_OPEN_PDM2PCM_LIBRARY
-                Open_PDM_Filter(&((uint8_t *) input_buffer)[index], (uint16_t *) &(output_buffer[index]), volume, (TPDMFilter_InitStruct *) &_PDM2PCM_filter[index]);
+                Open_PDM_Filter_128(&((uint8_t *) input_buffer)[index], (uint16_t *) &(output_buffer[index]), volume, (TPDMFilter_InitStruct *) &_PDM2PCM_filter[index]);
 #else                
                 PDM_Filter_128_LSB(&((uint8_t *) input_buffer)[index], (uint16_t *) &(output_buffer[index]), volume, (PDMFilter_InitStruct *) &_PDM2PCM_filter[index]);
 #endif
-#ifdef PDM2PCM_AUDIO_DEBUG
-                _pdm2pcm_audio_signal = 0;
-#endif
             }
             break;
 
         default:
             error("Decimation factor not supported: it must be either 64 or 128.\r\n");
     }
+#ifdef PDM2PCM_AUDIO_DEBUG
+                _pdm2pcm_audio_signal = 0;
+#endif
 
     return COMPONENT_OK;
 }