Add DS3 to I2S input

- I2S downsampling is disabled by default.

- I2S rate is always scaled up:
  The host may not issue a SET_SAMPLE_FREQ command so always scale up,
  and back down where the value may be a another command.

module_usb_audio/audio_io/audio_io.xc

@@ -31,11 +31,31 @@
 #include "commands.h"
 #include "xc_ptr.h"
 
+#ifndef I2S_DOWNSAMPLE_FACTOR
+#define I2S_DOWNSAMPLE_FACTOR (1)
+#endif
+#define I2S_RATE_SCALING (I2S_DOWNSAMPLE_FACTOR)
+
+#if I2S_RATE_SCALING != I2S_DOWNSAMPLE_FACTOR
+#error "Unsupported I2S rate scaling and downsampling configuration"
+#endif
+
+// I2S channels for stereo AEC reference input
+#define I2S_CHANS_DIGI_IN (2)
+// The PDM mic channels start after the I2S AEC reference channels
+#define PDM_RX_INDEX (I2S_CHANS_DIGI_IN)
+
+#if (NUM_USB_CHAN_IN < (I2S_CHANS_DIGI_IN + NUM_PDM_MICS))
+#error "Not enough USB input channels to support number of I2S and PDM inputs"
+#endif
+
 /* TODO 32 is max expected channels */
 static unsigned samplesOut[32];
 
 /* Two buffers for ADC data to allow for DAC and ADC ports being offset */
-static unsigned samplesIn[2][32];
+static unsigned samplesIn[2][I2S_DOWNSAMPLE_FACTOR][32];
+
+static int downsamplingCounter = 0;
 
 #if (DSD_CHANS_DAC != 0)
 extern buffered out port:32 p_dsd_dac[DSD_CHANS_DAC];
@@ -266,16 +286,24 @@ static inline unsigned DoSampleTransfer(chanend c_out, const int readBuffNo, con
 #else
         inuint(c_out);
 #endif
-        UserBufferManagement(samplesOut, samplesIn[readBuffNo], i_audMan);
+    }
+    unsigned downsampledI2S[2];
+    // TODO: run ds3 here
+    // Pass samplesIn[readBuffNo][0,I2S_DOWNSAMPLE_FACTOR][i]) to ds3()
+    UserBufferManagement(samplesOut, samplesIn[readBuffNo][downsamplingCounter], i_audMan);
 
 #if NUM_USB_CHAN_IN > 0
 #pragma loop unroll
-        for(int i = 0; i < NUM_USB_CHAN_IN; i++)
+    for(int i = 0; i < I2S_CHANS_DIGI_IN; i++)
     {
-            outuint(c_out, samplesIn[readBuffNo][i]);
+        outuint(c_out, downsampledI2S[i]);
+    }
+#pragma loop unroll
+    for (int i = PDM_RX_INDEX; i < (NUM_PDM_MICS + PDM_RX_INDEX); i++)
+    {
+        outuint(c_out, samplesIn[readBuffNo][downsamplingCounter][i]);
     }
 #endif
-    }
 
     return 0;
 }
@@ -479,6 +507,8 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
         return command;
     }
 
+    downsamplingCounter = 0;
+
     InitPorts(divide);
 
     /* TODO In master mode, the i/o loop assumes L/RCLK = 32bit clocks.  We should check this every interation
@@ -684,20 +714,22 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
 #endif
 
 #if (NUM_PDM_MICS > 0)
+            if ((I2S_DOWNSAMPLE_FACTOR - 1) == downsamplingCounter)
+            {
                 /* Get samples from PDM->PCM comverter */
                 c_pdm_pcm <: 1;
                 master
                 {
 #pragma loop unroll
-            for(int i = 0; i < NUM_PDM_MICS; i++)
+                    for(int i = PDM_RX_INDEX; i < (NUM_PDM_MICS + PDM_RX_INDEX); i++)
                     {
-                c_pdm_pcm :> samplesIn[readBuffNo][i];
+                        c_pdm_pcm :> samplesIn[readBuffNo][downsamplingCounter][i];
+                    }
                 }
             }
 #endif
         }
 
-
 #if (I2S_CHANS_ADC != 0)
             index = 0;
             /* Channels 0, 2, 4.. on each line */
@@ -793,6 +825,8 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
         frameCount+=2;
         if(frameCount == I2S_CHANS_PER_FRAME)
 #endif
+        {
+            if ((I2S_DOWNSAMPLE_FACTOR - 1) == downsamplingCounter)
             {
                 /* Do samples transfer */
                 /* The below looks a bit odd but forces the compiler to inline twice */
@@ -808,9 +842,15 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
                 }
 
                 /* Reset frame counter and flip the ADC buffer */
+                downsamplingCounter = 0;
                 frameCount = 0;
                 readBuffNo = !readBuffNo;
             }
+            else
+            {
+                ++downsamplingCounter;
+            }
+        }
     }
 
 #pragma xta endpoint "deliver_return"
@@ -924,7 +964,7 @@ chanend ?c_config, chanend ?c
     unsigned adatMultiple = 0;
 #endif
 
-    unsigned curSamFreq = DEFAULT_FREQ;
+    unsigned curSamFreq = DEFAULT_FREQ * I2S_RATE_SCALING;
     unsigned curSamRes_DAC = STREAM_FORMAT_OUTPUT_1_RESOLUTION_BITS; /* Default to something reasonable */
     unsigned curSamRes_ADC = STREAM_FORMAT_INPUT_1_RESOLUTION_BITS; /* Default to something reasonable - note, currently this never changes*/
     unsigned command;
@@ -1079,7 +1119,7 @@ chanend ?c_config, chanend ?c
         {
             /* TODO wait for good mclk instead of delay */
             /* No delay for DFU modes */
-            if ((curSamFreq != AUDIO_REBOOT_FROM_DFU) && (curSamFreq != AUDIO_STOP_FOR_DFU) && command)
+            if (((curSamFreq / I2S_RATE_SCALING) != AUDIO_REBOOT_FROM_DFU) && ((curSamFreq / I2S_RATE_SCALING) != AUDIO_STOP_FOR_DFU) && command)
             {
 #if 0
                 /* User should ensure MCLK is stable in AudioHwConfig */
@@ -1122,7 +1162,7 @@ chanend ?c_config, chanend ?c
 
 #if NUM_PDM_MICS > 0
                 /* Send decimation factor to PDM task(s) */
-                c_pdm_in <: curSamFreq;
+                c_pdm_in <: curSamFreq / I2S_DOWNSAMPLE_FACTOR;
 #endif
 
 #ifdef ADAT_TX
@@ -1163,7 +1203,7 @@ chanend ?c_config, chanend ?c
 
                 if(command == SET_SAMPLE_FREQ)
                 {
-                    curSamFreq = inuint(c_mix_out);
+                    curSamFreq = inuint(c_mix_out) * I2S_RATE_SCALING;
                 }
                 else if(command == SET_STREAM_FORMAT_OUT)
                 {
@@ -1176,7 +1216,7 @@ chanend ?c_config, chanend ?c
                 }
 
                 /* Currently no more audio will happen after this point */
-                if (curSamFreq == AUDIO_STOP_FOR_DFU)
+                if ((curSamFreq / I2S_RATE_SCALING) == AUDIO_STOP_FOR_DFU)
 				{
                   	outct(c_mix_out, XS1_CT_END);