Merge smart_mic_plus -> master

CHANGELOG.rst

@@ -1,6 +1,23 @@
 sc_usb_audio Change Log
 =======================
 
+7.0.0
+------
+
+6.30.0
+------
+    - RESOLVED:   Number of PDM microphone channels configured now based on
+    NUM_PDM_MICS define (previously hard-coded)
+    - RESOLVED:   PDM microphone clock divide now based MCLK defines (previously hard-coded)
+    - CHANGE:     Second microphone decimation core only run if NUM_PDM_MICS > 4
+
+6.20.0
+------
+    - RESOLVED:   Intra-frame sample delays of 1/2 samples on input streaming in TDM mode 
+    - RESOLVED:   Build issue with NUM_USB_CHAN_OUT set to 0 and MIXER enabled
+    - RESOLVED:   SPDIF_TX_INDEX not defined build warning only emitted when SPDIF_TX defined
+    - RESOLVED:   Failure to enter DFU mode when configured without input volume control
+
 6.19.0
 ------
     - RESOLVED:   SPDIF_TX_INDEX not defined build warning only emitted when SPDIF_TX defined

README.rst

@@ -1,7 +1,7 @@
 USB Audio Shared
 ................
 
-:Latest release: 6.18.1alpha0
+:Latest release: 6.19.0alpha0
 :Maintainer: xross
 :Description: USB Audio Shared Components. For use in the XMOS USB Audio Refererence Designs.
 

module_dfu/src/dfu.xc

@@ -203,7 +203,7 @@ static int DFU_Dnload(unsigned int request_len, unsigned int block_num, const un
 
 static int DFU_Upload(unsigned int request_len, unsigned int block_num, unsigned data_out[16], unsigned &DFU_state)
 {
-    unsigned int cmd_data[16];
+    unsigned int cmd_data[1];
     unsigned int firstRead = 0;
 
     // Start at flash address 0
@@ -236,8 +236,8 @@ static int DFU_Upload(unsigned int request_len, unsigned int block_num, unsigned
     {
         cmd_data[0] = !firstRead;
 
-        // Read whole (256bytes) page from the image on the flash
+        // Read whole (256bytes) page from the image on the flash into a memory buffer
-        flash_cmd_read_page((cmd_data, unsigned char[64]));
+        flash_cmd_read_page((cmd_data, unsigned char[1]));
         subPagesLeft = 4;
 
         // If address out of range, terminate!

module_usb_audio/audio_io/audio_io.xc

@@ -12,10 +12,11 @@
 #include <xs1.h>
 #include <xclib.h>
 #include <xs1_su.h>
+#include <string.h>
 
 #include "devicedefines.h"
 
-#include "dfu_interface.h"
+#include "xua_audio.h"
 #include "audioports.h"
 #include "audiohw.h"
 #ifdef SPDIF_TX
@@ -23,22 +24,50 @@
 #endif
 #ifdef ADAT_TX
 #include "adat_tx.h"
-
 #ifndef ADAT_TX_USE_SHARED_BUFF
 #error Designed for ADAT tx shared buffer mode ONLY
 #endif
 #endif
 
-#include "xua_audio.h"
-
 #include "commands.h"
 #include "xc_ptr.h"
 
-/* TODO 32 is max expected channels */
+#define MAX(x,y) ((x)>(y) ? (x) : (y))
-static unsigned samplesOut[32];
 
-/* Two buffers for ADC data to allow for DAC and ADC ports being offset */
+static unsigned samplesOut[MAX(NUM_USB_CHAN_OUT, I2S_CHANS_DAC)];
-static unsigned samplesIn[2][32];
+
+#ifndef ADAT_RX
+#define ADAT_RX 0
+#endif
+
+#ifndef SPDIF_RX
+#define SPDIF_RX 0
+#endif
+
+/* Two buffers for ADC data to allow for DAC and ADC I2S ports being offset */
+#define IN_CHAN_COUNT (I2S_CHANS_ADC + NUM_PDM_MICS + (8*ADAT_RX) + (2*SPDIF_RX)) 
+
+static unsigned samplesIn[2][MAX(NUM_USB_CHAN_IN, IN_CHAN_COUNT)];
+
+#if defined(ADAT_RX) && (ADAT_RX ==0)
+#undef ADAT_RX
+#endif
+
+#if defined(SPDIF_RX) && (SPDIF_RX ==0)
+#undef SPDIF_RX
+#endif
+
+static int downsamplingCounter = 0;
+#if (I2S_DOWNSAMPLE_FACTOR > 1)
+#include "src.h"
+static union ds3Data
+{
+    long long doubleWordAlignmentEnsured;
+    /* [Number of I2S channels][Number of samples/phases][Taps per phase] */
+    int32_t delayLine[I2S_DOWNSAMPLE_CHANS][I2S_DOWNSAMPLE_FACTOR][24];
+} ds3Data;
+static int64_t ds3Sum[I2S_DOWNSAMPLE_CHANS];
+#endif
 
 #if (DSD_CHANS_DAC != 0)
 extern buffered out port:32 p_dsd_dac[DSD_CHANS_DAC];
@@ -222,7 +251,7 @@ static inline void TransferAdatTxSamples(chanend c_adat_out, const unsigned samp
 }
 #endif
 
-/* sampsFromUsbToAudio: The sample frame the device has recived from the host and is going to play to the output audio interfaces */
+/* sampsFromUsbToAudio: The sample frame the device has received from the host and is going to play to the output audio interfaces */
 /* sampsFromAudioToUsb: The sample frame that was received from the audio interfaces and that the device is going to send to the host */
 void UserBufferManagement(unsigned sampsFromUsbToAudio[], unsigned sampsFromAudioToUsb[], client audManage_if i_audMan);
 
@@ -468,6 +497,10 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
     }
 #endif
 
+#if (I2S_DOWNSAMPLE_FACTOR > 1)
+    memset(&ds3Data.delayLine, 0, sizeof ds3Data);
+#endif
+
     unsigned command = DoSampleTransfer(c_out, readBuffNo, underflowWord, i_audMan);
 
 #ifdef ADAT_TX
@@ -482,6 +515,8 @@ 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
@@ -597,11 +632,17 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
         else
 #endif
         {
+#if (I2S_DOWNSAMPLE_FACTOR > 1)
+            if (0 == downsamplingCounter)
+            {
+                memset(&ds3Sum, 0, sizeof ds3Sum);
+            }
+#endif
 #if (I2S_CHANS_ADC != 0)
             /* Input previous L sample into L in buffer */
             index = 0;
             /* First input (i.e. frameCount == 0) we read last ADC channel of previous frame.. */
-            unsigned buffIndex = (frameCount < 3) ? !readBuffNo : readBuffNo;
+            unsigned buffIndex = (frameCount > 1) ? !readBuffNo : readBuffNo;
 
 #pragma loop unroll
             /* First time around we get channel 7 of TDM8 */
@@ -614,6 +655,26 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
 
                 /* Note the use of readBuffNo changes based on frameCount */
                 samplesIn[buffIndex][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i] = bitrev(sample); // channels 0, 2, 4.. on each line.
+#if (I2S_DOWNSAMPLE_FACTOR > 1)
+                if ((I2S_DOWNSAMPLE_FACTOR - 1) == downsamplingCounter)
+                {
+                    samplesIn[readBuffNo][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))] =
+                        src_ds3_voice_add_final_sample(
+                            ds3Sum[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i],
+                            ds3Data.delayLine[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i][downsamplingCounter],
+                            src_ff3v_ds3_voice_coefs[downsamplingCounter],
+                            samplesIn[readBuffNo][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))]);
+                }
+                else
+                {
+                    ds3Sum[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i] =
+                        src_ds3_voice_add_sample(
+                        ds3Sum[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i],
+                        ds3Data.delayLine[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i][downsamplingCounter],
+                        src_ff3v_ds3_voice_coefs[downsamplingCounter],
+                        samplesIn[readBuffNo][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))]);
+                }
+#endif // (I2S_DOWNSAMPLE_FACTOR > 1)
             }
 #endif
 
@@ -687,20 +748,22 @@ 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_MIC_INDEX; i < (NUM_PDM_MICS + PDM_MIC_INDEX); i++)
                     {
                         c_pdm_pcm :> samplesIn[readBuffNo][i];
                     }
                 }
+            }
 #endif
         }
 
-
 #if (I2S_CHANS_ADC != 0)
             index = 0;
             /* Channels 0, 2, 4.. on each line */
@@ -712,6 +775,26 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
                 asm volatile("in %0, res[%1]" : "=r"(sample)  : "r"(p_i2s_adc[index++]));
 
                 samplesIn[buffIndex][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i] = bitrev(sample); // channels 1, 3, 5.. on each line.
+#if ((I2S_DOWNSAMPLE_FACTOR > 1) && !I2S_DOWNSAMPLE_MONO)
+                if ((I2S_DOWNSAMPLE_FACTOR - 1) == downsamplingCounter)
+                {
+                    samplesIn[buffIndex][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i] =
+                        src_ds3_voice_add_final_sample(
+                            ds3Sum[((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i],
+                            ds3Data.delayLine[((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i][downsamplingCounter],
+                            src_ff3v_ds3_voice_coefs[downsamplingCounter],
+                            samplesIn[readBuffNo][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i]);
+                }
+                else
+                {
+                    ds3Sum[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i] =
+                        src_ds3_voice_add_sample(
+                        ds3Sum[((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i],
+                        ds3Data.delayLine[((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i][downsamplingCounter],
+                        src_ff3v_ds3_voice_coefs[downsamplingCounter],
+                        samplesIn[readBuffNo][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i]);
+                }
+#endif // ((I2S_DOWNSAMPLE_FACTOR > 1) && !I2S_DOWNSAMPLE_MONO)
 
             }
 #endif
@@ -796,6 +879,8 @@ 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 */
@@ -811,9 +896,15 @@ 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"
@@ -927,7 +1018,7 @@ chanend ?c_config, chanend ?c
     unsigned adatMultiple = 0;
 #endif
 
-    unsigned curSamFreq = DEFAULT_FREQ;
+    unsigned curSamFreq = DEFAULT_FREQ * I2S_DOWNSAMPLE_FACTOR;
     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;
@@ -1082,7 +1173,7 @@ 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_DOWNSAMPLE_FACTOR) != AUDIO_REBOOT_FROM_DFU) && ((curSamFreq / I2S_DOWNSAMPLE_FACTOR) != AUDIO_STOP_FOR_DFU) && command)
             {
 #if 0
                 /* User should ensure MCLK is stable in AudioHwConfig */
@@ -1125,7 +1216,7 @@ 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
@@ -1158,15 +1249,12 @@ chanend ?c_config, chanend ?c
 #if (NUM_PDM_MICS > 0)
                    c_pdm_in,
 #endif
-                   null
+                   null, i_audMan
-//#ifdef RUN_DSP_TASK
-                   , i_audMan
-//#endif
                    );
 
                 if(command == SET_SAMPLE_FREQ)
                 {
-                    curSamFreq = inuint(c_mix_out);
+                    curSamFreq = inuint(c_mix_out) * I2S_DOWNSAMPLE_FACTOR;
                 }
                 else if(command == SET_STREAM_FORMAT_OUT)
                 {
@@ -1179,7 +1267,7 @@ chanend ?c_config, chanend ?c
                 }
 
                 /* Currently no more audio will happen after this point */
-                if (curSamFreq == AUDIO_STOP_FOR_DFU)
+                if ((curSamFreq / I2S_DOWNSAMPLE_FACTOR) == AUDIO_STOP_FOR_DFU)
 				{
                   	outct(c_mix_out, XS1_CT_END);
 

module_usb_audio/audio_io/userbuffermanagement.c

@@ -4,10 +4,10 @@
 
 /* Default implentation for UserBufferManagement() */
 void UserBufferManagement(unsigned sampsFromUsbToAudio[], unsigned sampsFromAudioToUsb[]
-        , unsigned i_dsp
+        , unsigned i_audMan
         ) __attribute__ ((weak));
 void UserBufferManagement(unsigned sampsFromUsbToAudio[], unsigned sampsFromAudioToUsb[]
-        , unsigned i_dsp
+        , unsigned i_audMan
 )
 {
     /* Do nothing */

module_usb_audio/audio_io/xua_audio.h

@@ -1,10 +1,7 @@
 #ifndef __audio_h__
 #define __audio_h__
 
-//#include "devicedefines.h"
 #include "dfu_interface.h"
-//#include "xua_dsp.h"
-
 
 typedef interface audManage_if
 {
@@ -18,8 +15,6 @@ typedef interface audManage_if
 } audManage_if;
 
 
-
-
 /** The audio driver thread.
  *
  *  This function drives I2S ports and handles samples to/from other digital
@@ -46,9 +41,7 @@ void audio(chanend c_in,
 #if (NUM_PDM_MICS > 0)
     , chanend c_pdm_in
 #endif
-//#ifdef RUN_DSP_TASK
     , client audManage_if i_audMan
-//#endif
 );
 
 void SpdifTxWrapper(chanend c_spdif_tx);

module_usb_audio/devicedefines.h

@@ -121,6 +121,41 @@
 #define I2S_WIRES_ADC            (I2S_CHANS_ADC / I2S_CHANS_PER_FRAME)
 #endif
 
+/**
+ * @brief Incoming I2S (device to host) channels can be downsampled by a factor of 3.
+ *
+ * Default: 1 i.e. downsampling is disabled.
+ */
+#ifndef I2S_DOWNSAMPLE_FACTOR
+#define I2S_DOWNSAMPLE_FACTOR (1)
+#else
+    #if (I2S_DOWNSAMPLE_FACTOR != 3) && (I2S_DOWNSAMPLE_FACTOR != 1)
+        #error Unsupported I2S downsampling configuration
+    #endif
+#endif
+
+/**
+ * @brief Only downsample one channel per I2S frame.
+ *
+ * Default: 0 i.e. mono mode is disabled, all channels will be downsampled.
+ */
+#ifndef I2S_DOWNSAMPLE_MONO
+#define I2S_DOWNSAMPLE_MONO (0)
+#endif
+
+/**
+ * @brief Number of incoming (device to host) I2S channels to downsample.
+ *
+ * Default: The number of I2S incoming channels, or half this if mono downsampling is enabled.
+ */
+#if (I2S_DOWNSAMPLE_MONO == 1)
+    #define I2S_DOWNSAMPLE_CHANS (I2S_CHANS_ADC / 2)
+    #if ((I2S_DOWNSAMPLE_FACTOR > 1) && (I2S_MODE_TDM == 1))
+        #error Mono I2S downsampling is not avaliable in TDM mode
+    #endif
+#else
+#define I2S_DOWNSAMPLE_CHANS I2S_CHANS_ADC
+#endif
 
 /**
  * @brief Max supported sample frequency for device (Hz). Default: 192000
@@ -468,14 +503,14 @@
  * @brief Device firmware version number in Binary Coded Decimal format: 0xJJMN where JJ: major, M: minor, N: sub-minor version number.
  */
 #ifndef BCD_DEVICE_J
-#define BCD_DEVICE_J             6
+#define BCD_DEVICE_J             7
 #endif
 
 /**
  * @brief Device firmware version number in Binary Coded Decimal format: 0xJJMN where JJ: major, M: minor, N: sub-minor version number.
  */
 #ifndef BCD_DEVICE_M
-#define BCD_DEVICE_M             19
+#define BCD_DEVICE_M             0
 #endif
 
 /**
@@ -1046,7 +1081,7 @@
 #undef OUT_VOLUME_IN_MIXER
 #else
 #if defined(MIXER)
-// Enabled by default
+// Disabled by default
 //#define OUT_VOLUME_IN_MIXER
 #endif
 #endif
@@ -1066,7 +1101,7 @@
 #undef IN_VOLUME_IN_MIXER
 #else
 #if defined(MIXER)
-/* Enabled by default */
+/* Disabled by default */
 //#define IN_VOLUME_IN_MIXER
 #endif
 #endif
@@ -1218,7 +1253,11 @@ enum USBEndpointNumber_Out
 #define DFU_VENDOR_ID               VENDOR_ID
 #define DFU_BCD_DEVICE              BCD_DEVICE
 #define DFU_MANUFACTURER_STR_INDEX  offsetof(StringDescTable_t, vendorStr)/sizeof(char *)
+#if (AUDIO_CLASS == 2)
 #define DFU_PRODUCT_STR_INDEX       offsetof(StringDescTable_t, productStr_Audio2)/sizeof(char *)
+#else
+#define DFU_PRODUCT_STR_INDEX       offsetof(StringDescTable_t, productStr_Audio1)/sizeof(char *)
+#endif
 #endif
 
 /* USB test mode support enabled by default (Required for compliance testing) */

module_usb_audio/endpoint0/audiorequests.xc

@@ -45,9 +45,9 @@ extern unsigned char mixSel[MAX_MIX_COUNT][MIX_INPUTS];
 
 /* Global var for current frequency, set to default freq */
 unsigned int g_curSamFreq = DEFAULT_FREQ;
-//unsigned int g_curSamFreq48000Family = DEFAULT_FREQ % 48000 == 0;
-
 #if 0
+unsigned int g_curSamFreq48000Family = DEFAULT_FREQ % 48000 == 0;
+
 /* Original feedback implementation */
 long long g_curSamFreqMultiplier = (DEFAULT_FREQ * 512 * 4) / (DEFAULT_MCLK_FREQ);
 #endif
@@ -131,6 +131,7 @@ static void setG_curSamFreqMultiplier(unsigned x)
 }
 #endif
 
+#if (OUTPUT_VOLUME_CONTROL == 1) || (INPUT_VOLUME_CONTROL == 1)
 /* Update master volume i.e. i.e update weights for all channels */
 static void updateMasterVol( int unitID, chanend ?c_mix_ctl)
 {
@@ -266,6 +267,7 @@ static void updateVol(int unitID, int channel, chanend ?c_mix_ctl)
         }
     }
 }
+#endif
 
 /* Handles the audio class specific requests
  * returns:     XUD_RES_OKAY if request dealt with successfully without error,
@@ -275,7 +277,7 @@ static void updateVol(int unitID, int channel, chanend ?c_mix_ctl)
 int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, chanend c_audioControl, chanend ?c_mix_ctl, chanend ?c_clk_ctl
 )
 {
-    unsigned char buffer[512];
+    unsigned char buffer[128];
     int unitID;
     XUD_Result_t result;
     unsigned datalength;
@@ -312,6 +314,7 @@ int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
                                     return result;
                                 }
 
+#if MAX_FREQ != MIN_FREQ
                                 if(datalength == 4)
                                 {
                                     /* Re-construct Sample Freq */
@@ -369,7 +372,7 @@ int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
                                     /* Allow time for our feedback to stabilise*/
                                     FeedbackStabilityDelay();
                                 }
-
+#endif /* MAX_FREQ != MIN_FREQ */
                                 /* Send 0 Length as status stage */
                                 XUD_DoSetRequestStatus(ep0_in);
                             }
@@ -514,6 +517,7 @@ int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
                     break;
                 }
 
+#if (OUTPUT_VOLUME_CONTROL == 1) || (INPUT_VOLUME_CONTROL == 1)
                 /* Feature Units */
                 case FU_USBOUT:
                 case FU_USBIN:
@@ -629,6 +633,7 @@ int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
                     }
 
                     break; /* FU_USBIN */
+#endif
 
 #if defined(MIXER) && (MAX_MIX_COUNT > 0)
                 case ID_XU_OUT:
@@ -1076,7 +1081,7 @@ int AudioEndpointRequests_1(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp
      */
 
     XUD_Result_t result;
-    unsigned char buffer[1024];
+    unsigned char buffer[128];
     unsigned length;
 
     /* Host to Device */
@@ -1094,7 +1099,7 @@ int AudioEndpointRequests_1(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp
                 {
                     return result;
                 }
-
+#if (MAX_FREQ != MIN_FREQ)
                 if(controlSelector == SAMPLING_FREQ_CONTROL)
                 {
                     /* Expect length 3 for sample rate */
@@ -1116,22 +1121,6 @@ int AudioEndpointRequests_1(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp
                             if(curSamFreq48000Family || curSamFreq44100Family)
                             {
                                 g_curSamFreq = newSampleRate;
-#if 0
-                                /* Original feedback implementation */
-
-                                int newMasterClock;
-
-                                if(g_curSamFreq48000Family)
-                                {
-                                    newMasterClock = MCLK_48;
-                                }
-                                else
-                                {
-                                    newMasterClock = MCLK_441;
-                                }
-
-                                setG_curSamFreqMultiplier((g_curSamFreq*512*4)/newMasterClock);
-#endif
 
                                 /* Instruct audio thread to change sample freq */
                                 outuint(c_audioControl, SET_SAMPLE_FREQ);
@@ -1147,6 +1136,9 @@ int AudioEndpointRequests_1(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp
                         return XUD_SetBuffer(ep0_in, buffer, 0);
                     }
                 }
+#else
+                return XUD_SetBuffer(ep0_in, buffer, 0);
+#endif     
             }
             break;
         }
@@ -1171,12 +1163,13 @@ int AudioEndpointRequests_1(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp
 int AudioClassRequests_1(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, chanend c_audioControl, chanend ?c_mix_ctl, chanend ?c_clk_ctl
 )
 {
-    unsigned char buffer[1024];
+    unsigned char buffer[128];
     unsigned unitID;
     XUD_Result_t result;
 
     /* Inspect request */
     /* Note we could check sp.bmRequestType.Direction if we wanted to be really careful */
+#if (OUTPUT_VOLUME_CONTROL == 1) || (INPUT_VOLUME_CONTROL == 1)
     switch(sp.bRequest)
     {
         case UAC_B_REQ_SET_CUR:
@@ -1288,7 +1281,7 @@ int AudioClassRequests_1(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
             buffer[1] = (VOLUME_RES_MIXER >> 8);
             return XUD_DoGetRequest(ep0_out, ep0_in, buffer, 2, sp.wLength);
     }
-
+#endif
     return 1;
 }
 #endif

module_usb_audio/endpoint0/descriptors.h

@@ -35,13 +35,14 @@ typedef struct
     STR_TABLE_ENTRY(vendorStr);
     STR_TABLE_ENTRY(serialStr);
 
+#if (AUDIO_CLASS == 2)
     /* Audio 2.0 Strings */
     STR_TABLE_ENTRY(productStr_Audio2);           /* Product string for Audio 2 */
     STR_TABLE_ENTRY(outputInterfaceStr_Audio2);   /* iInterface for streaming intefaces */
     STR_TABLE_ENTRY(inputInterfaceStr_Audio2);    /* iInterface for streaming intefaces */
     STR_TABLE_ENTRY(usbInputTermStr_Audio2);      /* Users sees as output from host */
     STR_TABLE_ENTRY(usbOutputTermStr_Audio2);     /* User sees as input to host */
-
+#endif
 #if defined (AUDIO_CLASS_FALLBACK) || (AUDIO_CLASS == 1)
     /* Audio 1.0 Strings */
     STR_TABLE_ENTRY(productStr_Audio1);           /* Product string for Audio 1 */
@@ -50,6 +51,7 @@ typedef struct
     STR_TABLE_ENTRY(usbInputTermStr_Audio1);      /* Users sees as output from host */
     STR_TABLE_ENTRY(usbOutputTermStr_Audio1);     /* User sees as input to host */
 #endif
+#if (AUDIO_CLASS == 2)
     STR_TABLE_ENTRY(clockSelectorStr);            /* iClockSel */
     STR_TABLE_ENTRY(internalClockSourceStr);      /* iClockSource for internal clock */
 #ifdef SPDIF_RX
@@ -58,6 +60,7 @@ typedef struct
 #ifdef ADAT_RX
     STR_TABLE_ENTRY(adatClockSourceStr);          /* iClockSource for external S/PDIF clock */
 #endif
+#endif
 #ifdef DFU
     STR_TABLE_ENTRY(dfuStr);                      /* iInterface for DFU interface */
 #endif
@@ -291,7 +294,9 @@ typedef struct
 #if defined(MIXER) && (MAX_MIX_COUNT > 7)
     STR_TABLE_ENTRY(mixOutStr_8);
 #endif
+#ifdef IAP
     STR_TABLE_ENTRY(iAPInterfaceStr);
+#endif
 #ifdef IAP_EA_NATIVE_TRANS
     STR_TABLE_ENTRY(iAP_EANativeTransport_InterfaceStr);
 #endif
@@ -302,12 +307,13 @@ StringDescTable_t g_strTable =
     .langID                      = "\x09\x04", /* US English */
     .vendorStr                   = VENDOR_STR,
     .serialStr                   = "",
+#if (AUDIO_CLASS == 2)
     .productStr_Audio2           = PRODUCT_STR_A2,
     .outputInterfaceStr_Audio2   = APPEND_PRODUCT_STR_A2(),
     .inputInterfaceStr_Audio2    = APPEND_PRODUCT_STR_A2(),
     .usbInputTermStr_Audio2      = APPEND_PRODUCT_STR_A2(),
     .usbOutputTermStr_Audio2     = APPEND_PRODUCT_STR_A2(),
-
+#endif
 #if defined (AUDIO_CLASS_FALLBACK) || (AUDIO_CLASS == 1)
     .productStr_Audio1           = PRODUCT_STR_A1,
     .outputInterfaceStr_Audio1   = APPEND_PRODUCT_STR_A1(),
@@ -315,6 +321,7 @@ StringDescTable_t g_strTable =
     .usbInputTermStr_Audio1      = APPEND_PRODUCT_STR_A1(),
     .usbOutputTermStr_Audio1     = APPEND_PRODUCT_STR_A1(),
 #endif
+#if (AUDIO_CLASS == 2)
     .clockSelectorStr            = APPEND_VENDOR_STR(Clock Selector),
     .internalClockSourceStr      = APPEND_VENDOR_STR(Internal Clock),
 #ifdef SPDIF_RX
@@ -323,6 +330,7 @@ StringDescTable_t g_strTable =
 #ifdef ADAT_RX
     .adatClockSourceStr          = APPEND_VENDOR_STR(ADAT Clock),
 #endif
+#endif
 #ifdef DFU
     .dfuStr                      = APPEND_VENDOR_STR(DFU),
 #endif
@@ -368,7 +376,9 @@ StringDescTable_t g_strTable =
 #if defined(MIXER) && (MAX_MIX_COUNT > 8)
 #error
 #endif
+#ifdef IAP
     .iAPInterfaceStr             = "iAP Interface",
+#endif
 #ifdef IAP_EA_NATIVE_TRANS
     .iAP_EANativeTransport_InterfaceStr = IAP2_EA_NATIVE_TRANS_PROTOCOL_NAME,
 #endif
@@ -397,6 +407,7 @@ USB_Descriptor_Device_t devDesc_Audio1 =
 };
 #endif
 
+#if (AUDIO_CLASS == 2)
 /* Device Descriptor for Audio Class 2.0 (Assumes High-Speed ) */
 USB_Descriptor_Device_t devDesc_Audio2 =
 {
@@ -438,7 +449,7 @@ unsigned char devDesc_Null[] =
     0,                              /* 16 iSerialNumber : Index of serial number decriptor */
     0x01                            /* 17 bNumConfigurations : Number of possible configs */
 };
-
+#endif
 
 /****** Device Qualifier Descriptors *****/
 
@@ -751,7 +762,7 @@ typedef struct
 
 }__attribute__((packed)) USB_Config_Descriptor_Audio2_t;
 
-#if 1
+#if (AUDIO_CLASS == 2)
 USB_Config_Descriptor_Audio2_t cfgDesc_Audio2=
 {
     .Config =
@@ -2252,7 +2263,10 @@ const unsigned num_freqs_a1 = MAX(3, (0
 #define NUM_CONTROL_INTERFACES 0
 #endif
 
-#define AC_TOTAL_LENGTH             (AC_LENGTH + (INPUT_INTERFACES_A1 * (17 + NUM_USB_CHAN_IN_FS + num_freqs_a1 * 3)) + (OUTPUT_INTERFACES_A1 * (17 + NUM_USB_CHAN_OUT_FS + (num_freqs_a1 *3))))
+#define AC_TOTAL_LENGTH             (AC_LENGTH + \
+                                    (INPUT_INTERFACES_A1 *  (9 + (7* INPUT_VOLUME_CONTROL) + (NUM_USB_CHAN_IN_FS * INPUT_VOLUME_CONTROL)  + num_freqs_a1 * 3)) +\
+                                    (OUTPUT_INTERFACES_A1 * (9 + (7 * OUTPUT_VOLUME_CONTROL) + (NUM_USB_CHAN_OUT_FS * OUTPUT_VOLUME_CONTROL) + (num_freqs_a1 *3))))
+
 #define STREAMING_INTERFACES        (INPUT_INTERFACES_A1 + OUTPUT_INTERFACES_A1)
 
 /* Number of interfaces for Audio  1.0 (+1 for control ) */
@@ -2268,7 +2282,7 @@ const unsigned num_freqs_a1 = MAX(3, (0
 #define CHARIFY_SR(x) (x & 0xff),((x & 0xff00)>> 8),((x & 0xff0000)>> 16)
 
 #if (MIN_FREQ_FS < 12000) && (MAX_FREQ_FS > 48000)
-#error SAMPLE RATE RANGE TO GREAT FOR UAC1 ON WINDOWS
+#error SAMPLE RATE RANGE TOO GREAT FOR UAC1 ON WINDOWS
 #endif
 
 unsigned char cfgDesc_Audio1[] =
@@ -2297,7 +2311,7 @@ unsigned char cfgDesc_Audio1[] =
     USB_CLASS_AUDIO,
     UAC_INT_SUBCLASS_AUDIOCONTROL,
     0x00,                                 /* Unused */
-    8,                                    /* iInterface - re-use iProduct */
+    offsetof(StringDescTable_t, productStr_Audio1)/sizeof(char *), /* iInterface - re-use iProduct */
 
     /* CS (Class Specific) AudioControl interface header descriptor (4.3.2) */
     AC_LENGTH,
@@ -2325,8 +2339,9 @@ unsigned char cfgDesc_Audio1[] =
     NUM_USB_CHAN_OUT_FS,                  /* bNrChannels */
     0x03, 0x00,                           /* wChannelConfig */
     offsetof(StringDescTable_t, outputChanStr_1)/sizeof(char *), /* iChannelNames */
-    11,                                   /* iTerminal */
+    offsetof(StringDescTable_t, usbInputTermStr_Audio1)/sizeof(char *), /* iTerminal */
 
+#if (OUTPUT_VOLUME_CONTROL == 1)
     /* CS_Interface class specific AC interface feature unit descriptor - mute & volume for dac */
     (8 + NUM_USB_CHAN_OUT_FS),
     UAC_CS_DESCTYPE_INTERFACE,
@@ -2364,6 +2379,7 @@ unsigned char cfgDesc_Audio1[] =
 #error NUM_USB_CHAN_OUT_FS > 8 currently supported
 #endif
     0x00,                                 /* String table index */
+#endif
 
     /* CS_Interface Output Terminal Descriptor - Analogue out to speaker */
     0x09,
@@ -2372,7 +2388,11 @@ unsigned char cfgDesc_Audio1[] =
     0x06,                                 /* Terminal ID */
     0x01, 0x03,                           /* Type - streaming out, speaker */
     0x00,                                 /* Associated terminal - unused */
-    0x0A,                                 /* sourceID  */
+#if (OUTPUT_VOLUME_CONTROL == 1)
+    0x0A,                                 /* sourceID  - FU */
+#else
+    0x01,                                 /* sourceID - IT */
+#endif
     0x00,                                 /* Unused */
 #endif
 
@@ -2387,7 +2407,7 @@ unsigned char cfgDesc_Audio1[] =
     NUM_USB_CHAN_IN_FS,                   /* bNrChannels */
     0x03, 0x00,                           /* wChannelConfigs */
     offsetof(StringDescTable_t, inputChanStr_1)/sizeof(char *), /* iChannelNames */
-    12,                                   /* iTerminal */
+    offsetof(StringDescTable_t, usbOutputTermStr_Audio1)/sizeof(char *), /* iTerminal */
 
     /* CS_Interface Output Terminal Descriptor - USB Streaming Device to Host*/
     0x09,
@@ -2396,9 +2416,14 @@ unsigned char cfgDesc_Audio1[] =
     0x07,                                 /* Terminal ID */
     0x01, 0x01,                           /* Type - streaming */
     0x01,                                 /* Associated terminal - unused */
-    0x0B,                                 /* sourceID - from selector unit ?? */
+#if INPUT_VOLUME_CONTROL
+    0x0B,                                 /* sourceID - FU */
+#else
+    0x02,                                 /* sourceID - IT */
+#endif
     0x00,                                 /* Unused */
 
+#if (INPUT_VOLUME_CONTROL == 1)
     /* CS_Interface class specific AC interface feature unit descriptor - mute & volume for adc */
     (8 + NUM_USB_CHAN_IN_FS),
     UAC_CS_DESCTYPE_INTERFACE,
@@ -2436,6 +2461,7 @@ unsigned char cfgDesc_Audio1[] =
 #endif
     0x00,                                 /* String table index */
 #endif
+#endif
 
 #if (NUM_USB_CHAN_OUT > 0)
     /* Standard AS Interface Descriptor (4.5.1) */
@@ -2447,7 +2473,7 @@ unsigned char cfgDesc_Audio1[] =
     0x01,                                 /* bInterfaceClass - AUDIO */
     0x02,                                 /* bInterfaceSubclass - AUDIO_STREAMING */
     0x00,                                 /* bInterfaceProtocol - Not used */
-    0x09,                                 /* iInterface */
+    offsetof(StringDescTable_t, outputInterfaceStr_Audio1)/sizeof(char *), /* iInterface */
 
     /* Standard As Interface Descriptor (4.5.1) */
     0x09,
@@ -2462,7 +2488,7 @@ unsigned char cfgDesc_Audio1[] =
     0x01,                                 /* Interface class - AUDIO */
     0x02,                                 /* subclass - AUDIO_STREAMING */
     0x00,                                 /* Unused */
-    0x09,                                 /* String table index  */
+    offsetof(StringDescTable_t, outputInterfaceStr_Audio1)/sizeof(char *), /* iInterface */ 
 
     /* Class-Specific AS Interface Descriptor (4.5.2) */
     0x07,
@@ -2580,7 +2606,7 @@ unsigned char cfgDesc_Audio1[] =
     0x01,                                 /* Interface class - AUDIO */
     0x02,                                 /* subclass - AUDIO_STREAMING */
     0x00,                                 /* Unused */
-    0x0A,                                 /* iInterface */
+    offsetof(StringDescTable_t, inputInterfaceStr_Audio1)/sizeof(char *), 
 
     /* Standard Interface Descriptor - Audio streaming IN */
     0x09,
@@ -2591,7 +2617,7 @@ unsigned char cfgDesc_Audio1[] =
     0x01,                                 /* Interface class - AUDIO */
     0x02,                                 /* Subclass - AUDIO_STREAMING */
     0x00,                                 /* Unused */
-    0x0A,                                /* iInterface*/
+    offsetof(StringDescTable_t, inputInterfaceStr_Audio1)/sizeof(char *), 
 
     /* CS_Interface AC interface header descriptor */
     0x07,

module_usb_audio/endpoint0/endpoint0.c

@@ -206,6 +206,8 @@ void Endpoint0(chanend c_ep0_out, chanend c_ep0_in, chanend c_audioControl,
     XUD_ep ep0_out = XUD_InitEp(c_ep0_out);
     XUD_ep ep0_in  = XUD_InitEp(c_ep0_in);
 
+#if 0
+    /* Dont need to init globals.. */
     /* Init tables for volumes (+ 1 for master) */
     for(int i = 0; i < NUM_USB_CHAN_OUT + 1; i++)
     {
@@ -218,7 +220,7 @@ void Endpoint0(chanend c_ep0_out, chanend c_ep0_in, chanend c_audioControl,
         volsIn[i] = 0;
         mutesIn[i] = 0;
     }
-
+#endif
     VendorRequests_Init(VENDOR_REQUESTS_PARAMS);
 
 #ifdef MIXER
@@ -311,7 +313,7 @@ void Endpoint0(chanend c_ep0_out, chanend c_ep0_in, chanend c_audioControl,
                         switch (sp.wIndex)
                         {
                             /* Check for audio stream from host start/stop */
-#if (NUM_USB_CHAN_OUT > 0)
+#if (NUM_USB_CHAN_OUT > 0) && (AUDIO_CLASS == 2)
                             case INTERFACE_NUMBER_AUDIO_OUTPUT:
                                 /* Check the alt is in range */
                                 if(sp.wValue <= OUTPUT_FORMAT_COUNT)
@@ -346,7 +348,7 @@ void Endpoint0(chanend c_ep0_out, chanend c_ep0_in, chanend c_audioControl,
                                 break;
 #endif
 
-#if (NUM_USB_CHAN_IN > 0)
+#if (NUM_USB_CHAN_IN > 0) && (AUDIO_CLASS == 2)
                             case INTERFACE_NUMBER_AUDIO_INPUT:
                                 /* Check the alt is in range */
                                 if(sp.wValue <= INPUT_FORMAT_COUNT)

module_usb_audio/flashlib_user.c

@@ -19,9 +19,15 @@
 #define portout(a,b) {__asm__  __volatile__("out res[%0], %1": : "r" (a) , "r" (b));}
 
 #ifdef DFU_FLASH_DEVICE
+
+#ifdef QUAD_SPI_FLASH
+/* Using specified flash device rather than all supported in tools */
+fl_QuadDeviceSpec flash_devices[] = {DFU_FLASH_DEVICE};
+#else
 /* Using specified flash device rather than all supported in tools */
 fl_DeviceSpec flash_devices[] = {DFU_FLASH_DEVICE};
 #endif
+#endif
 
 #ifdef QUAD_SPI_FLASH
 /*
@@ -83,7 +89,11 @@ int flash_cmd_enable_ports()
 #endif
 
 #ifdef DFU_FLASH_DEVICE
+#ifdef QUAD_SPI_FLASH
+    result = fl_connectToDevice(&p_qflash, flash_devices, 1);
+#else
     result = fl_connectToDevice(&p_flash, flash_devices, 1);
+#endif
 #else
     /* Use default flash list */
 #ifdef QUAD_SPI_FLASH

module_usb_audio/main.xc

@@ -317,9 +317,12 @@ VENDOR_REQUESTS_PARAMS_DEC_
             c_sof, epTypeTableOut, epTypeTableIn, p_usb_rst,
             clk, 1, XUD_SPEED_HS, XUD_PWR_CFG);
 #else
+        {
+        set_core_high_priority_on();
         XUD_Manager(c_xud_out, ENDPOINT_COUNT_OUT, c_xud_in, ENDPOINT_COUNT_IN,
             c_sof, epTypeTableOut, epTypeTableIn, p_usb_rst,
             clk, 1, XUD_SPEED_FS, XUD_PWR_CFG);
+        }
 #endif
 
         /* USB Packet buffering Core */
@@ -485,6 +488,8 @@ void usb_audio_io(chanend c_aud_in, chanend ?c_adc,
 #define USER_MAIN_CORES
 #endif
 
+#ifndef EXCLUDE_USB_AUDIO_MAIN
+
 /* Main for USB Audio Applications */
 int main()
 {
@@ -558,6 +563,8 @@ int main()
     USER_MAIN_DECLARATIONS
     par
     {
+        USER_MAIN_CORES
+        
         on tile[XUD_TILE]:
         par
         {
@@ -668,17 +675,17 @@ int main()
 #ifdef MIC_PROCESSING_USE_INTERFACE
         on stdcore[PDM_TILE].core[0]: pdm_buffer(c_ds_output, c_pdm_pcm, i_mic_process);
 #else
-        on stdcore[PDM_TILE]: pdm_buffer(c_ds_output, c_pdm_pcm);
+        on stdcore[PDM_TILE].core[0]: pdm_buffer(c_ds_output, c_pdm_pcm);
 #endif
 #endif
 
-        USER_MAIN_CORES
-    }
 
 #ifdef SU1_ADC_ENABLE
         xs1_su_adc_service(c_adc);
 #endif
+    }
 
     return 0;
 }
 
+#endif

module_usb_audio/mixer/mixer.xc

@@ -242,10 +242,10 @@ static inline void GetSamplesFromHost(chanend c)
 #else
             ptr_samples[i] = sample;
 #endif
+        }
+    }
 #endif
 }
-    }
-}
 
 #pragma unsafe arrays
 static inline void GiveSamplesToDevice(chanend c, xc_ptr ptr, xc_ptr multOut)
@@ -366,7 +366,7 @@ static void mixer1(chanend c_host, chanend c_mix_ctl, chanend c_mixer2)
         /* Forward on Request for data to decouple thread */
         outuint(c_host, request);
 
-        /* Between request to decouple and respose ~ 400nS latency for interrupt to fire */
+        /* Between request to decouple and response ~ 400nS latency for interrupt to fire */
         select
         {
             case inuint_byref(c_mix_ctl, cmd):

module_usb_audio/module_build_info

@@ -29,4 +29,7 @@
 #
 EXCLUDE_FILES += descriptors_2.rst
 
-MODULE_XCC_FLAGS += $(XCC_FLAGS) -falways-inline
+#MODULE_XCC_FLAGS += $(XCC_FLAGS) -falways-inline
+
+XCC_FLAGS_endpoint0.c = -Os -mno-dual-issue $(XCC_FLAGS)
+XCC_FLAGS_dbcalc.xc = -Os -mno-dual-issue $(XCC_FLAGS)

module_usb_audio/pdm_mics/mic_array_conf.h

@@ -1,7 +1,9 @@
 #ifndef MIC_ARRAY_CONF_H_
 #define MIC_ARRAY_CONF_H_
 
+#include "customdefines.h"
+
 #define MIC_ARRAY_MAX_FRAME_SIZE_LOG2 0
-#define MIC_ARRAY_NUM_MICS 8
+#define MIC_ARRAY_NUM_MICS (NUM_PDM_MICS)
 
 #endif /* MIC_ARRAY_CONF_H_ */

module_usb_audio/pdm_mics/pdm_mic.h

@@ -1,37 +0,0 @@
-
-#include "mic_array.h"
-
-#ifdef MIC_PROCESSING_USE_INTERFACE
-/* Interface based user processing */
-typedef interface mic_process_if
-{
-    void transfer_buffers(mic_array_frame_time_domain * unsafe audio, int output[]);
-    void init();
-} mic_process_if;
-
-
-[[combinable]]
-void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio
-#ifdef MIC_PROCESSING_USE_INTERFACE
-   , client mic_process_if i_mic_process
-#endif
-);
-
-[[combinable]]
-void user_pdm_process(server mic_process_if i_mic_data);
-
-/* PDM interface and decimation cores */
-void pdm_mic(streaming chanend c_ds_output[2]);
-
-#else
-
-/* Simple user hooks/call-backs */
-unsafe void user_pdm_process(mic_array_frame_time_domain * unsafe audio, int output[]);
-
-void user_pdm_init();
-
-/* PDM interface and decimation cores */
-void pdm_mic(streaming chanend c_ds_output[2]);
-
-#endif
-

module_usb_audio/pdm_mics/pdm_mic.xc

@@ -12,11 +12,12 @@
 #include <string.h>
 #include <xclib.h>
 #include <stdint.h>
+#include <assert.h>
 
 #include "mic_array.h"
 #include "xua_pdm_mic.h"
 
-#define MAX_DECIMATION_FACTOR 12
+#define MAX_DECIMATION_FACTOR (96000/MIN_FREQ)
 
 /* Hardware resources */
 in port p_pdm_clk                = PORT_PDM_CLK;
@@ -35,11 +36,11 @@ mic_array_frame_time_domain mic_audio[2];
 void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio, client mic_process_if i_mic_process)
 #else
 #pragma unsafe arrays
+[[combinable]]
 void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio)
 #endif
 {
     unsigned buffer;
-    int output[NUM_PDM_MICS];
     unsigned samplerate;
 
 #ifdef MIC_PROCESSING_USE_INTERFACE
@@ -48,6 +49,12 @@ void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio)
     user_pdm_init();
 #endif
 
+#if NUM_PDM_MICS > 4
+    unsigned decimatorCount = 2;
+#else
+    unsigned decimatorCount = 1;
+#endif
+
     mic_array_decimator_conf_common_t dcc;
     const int * unsafe fir_coefs[7];
     mic_array_frame_time_domain * unsafe current;
@@ -105,20 +112,21 @@ void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio)
         dc[1].mic_gain_compensation[3]=0;
         dc[1].channel_count = 4;
 
-        mic_array_decimator_configure(c_ds_output, 2, dc);
+        mic_array_decimator_configure(c_ds_output, decimatorCount, dc);
 
-        mic_array_init_time_domain_frame(c_ds_output, 2, buffer, mic_audio, dc);
+        mic_array_init_time_domain_frame(c_ds_output, decimatorCount, buffer, mic_audio, dc);
 
         /* Grab a first frame of mic data */
         /* Note, loop is unrolled once - allows for while(1) select {} and thus combinable */
-        current = mic_array_get_next_time_domain_frame(c_ds_output, 2, buffer, mic_audio, dc);
+        current = mic_array_get_next_time_domain_frame(c_ds_output, decimatorCount, buffer, mic_audio, dc);
     }
 
     /* Run user code */
+    /* TODO ideally processing done inplace - it then doesn't matter if it is run or not */
 #ifdef MIC_PROCESSING_USE_INTERFACE
-    i_mic_process.transfer_buffers(current, output);
+    i_mic_process.transfer_buffers(current);
 #else
-    user_pdm_process(current, output);
+    user_pdm_process(current);
 #endif
     int req;
     while(1)
@@ -135,18 +143,18 @@ void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio)
 #pragma loop unroll
                         for(int i = 0; i < NUM_PDM_MICS; i++)
                         {
-                            c_audio <: output[i];
+                            c_audio <: current->data[i][0];
                         }
                     }
 
                     /* Get a new frame of mic data */
-                    mic_array_frame_time_domain * unsafe current = mic_array_get_next_time_domain_frame(c_ds_output, 2, buffer, mic_audio, dc);
+                    mic_array_frame_time_domain * unsafe current = mic_array_get_next_time_domain_frame(c_ds_output, decimatorCount, buffer, mic_audio, dc);
 
                     /* Run user code */
 #ifdef MIC_PROCESSING_USE_INTERFACE
-                    i_mic_process.transfer_buffers(current, output);
+                    i_mic_process.transfer_buffers(current);
 #else
-                    user_pdm_process(current, output);
+                    user_pdm_process(current);
 #endif
                 }
                 else
@@ -159,17 +167,17 @@ void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio)
                     dcc.output_decimation_factor = decimationfactor;
                     dcc.coefs=fir_coefs[decimationfactor/2];
                     dcc.fir_gain_compensation = fir_gain_compen[decimationfactor/2];
-                    mic_array_decimator_configure(c_ds_output, 2, dc);
+                    mic_array_decimator_configure(c_ds_output, decimatorCount, dc);
-                    mic_array_init_time_domain_frame(c_ds_output, 2, buffer, mic_audio, dc);
+                    mic_array_init_time_domain_frame(c_ds_output, decimatorCount, buffer, mic_audio, dc);
 
                     /* Get a new mic data frame */
-                    mic_array_frame_time_domain * unsafe current = mic_array_get_next_time_domain_frame(c_ds_output, 2, buffer, mic_audio, dc);
+                    mic_array_frame_time_domain * unsafe current = mic_array_get_next_time_domain_frame(c_ds_output, decimatorCount, buffer, mic_audio, dc);
 
                     /* Run user code */
 #ifdef MIC_PROCESSING_USE_INTERFACE
-                    i_mic_process.transfer_buffers(current, output);
+                    i_mic_process.transfer_buffers(current);
 #else
-                    user_pdm_process(current, output);
+                    user_pdm_process(current);
 #endif
                 }
                 break;
@@ -183,10 +191,23 @@ void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio)
 
 void pdm_mic(streaming chanend c_ds_output[2])
 {
-    streaming chan c_4x_pdm_mic_0, c_4x_pdm_mic_1;
+    streaming chan c_4x_pdm_mic_0;
+#if (NUM_PDM_MICS > 4)
+    streaming chan c_4x_pdm_mic_1;
+#else
+    #define c_4x_pdm_mic_1 null
+#endif
 
-    /* Note, this divide should be based on master clock freq */
+    /* Mics expect a clock in the 3Mhz range, calculate the divide based on mclk */
-    configure_clock_src_divide(pdmclk, p_mclk, 2);
+    /* e.g. For a 48kHz range mclk we expect a 3072000Hz mic clock */
+    /* e.g. For a 44.1kHz range mclk we expect a 2822400Hz mic clock */
+
+    /* Note, codebase currently does not handle a different divide for each clock */
+    assert((MCLK_48 / 3072000) == (MCLK_441 / 2822400));
+       
+    unsigned micDiv = MCLK_48/3072000;
+    
+    configure_clock_src_divide(pdmclk, p_mclk, micDiv/2);
     configure_port_clock_output(p_pdm_clk, pdmclk);
     configure_in_port(p_pdm_mics, pdmclk);
     start_clock(pdmclk);
@@ -195,7 +216,9 @@ void pdm_mic(streaming chanend c_ds_output[2])
     {
         mic_array_pdm_rx(p_pdm_mics, c_4x_pdm_mic_0, c_4x_pdm_mic_1);
         mic_array_decimate_to_pcm_4ch(c_4x_pdm_mic_0, c_ds_output[0], MIC_ARRAY_NO_INTERNAL_CHANS);
+#if (NUM_PDM_MICS > 4)
         mic_array_decimate_to_pcm_4ch(c_4x_pdm_mic_1, c_ds_output[1], MIC_ARRAY_NO_INTERNAL_CHANS);
+#endif
     }
 }
 #endif

module_usb_audio/pdm_mics/pdm_mic_user.c

@@ -0,0 +1,22 @@
+
+#include "devicedefines.h"
+
+#if (NUM_PDM_MICS > 0) && !defined(MIC_PROCESSING_USE_INTERFACE)
+
+#include "mic_array_frame.h" 
+
+/* Deafult implementations of user_pdm_init() and user_pdm_process().  Both can be over-ridden */
+void user_pdm_init() __attribute__ ((weak));
+void user_pdm_init()
+{
+    return;
+}
+
+
+void user_pdm_process() __attribute__ ((weak));
+void user_pdm_process(mic_array_frame_time_domain * audio)
+{
+    return;
+}
+
+#endif

module_usb_audio/pdm_mics/xua_pdm_mic.h

@@ -5,7 +5,7 @@
 /* Interface based user processing */
 typedef interface mic_process_if
 {
-    void transfer_buffers(mic_array_frame_time_domain * unsafe audio, int output[]);
+    void transfer_buffers(mic_array_frame_time_domain * unsafe audio);
     void init();
 } mic_process_if;
 
@@ -26,11 +26,12 @@ void pdm_mic(streaming chanend c_ds_output[2]);
 #else
 
 /* Simple user hooks/call-backs */
-void user_pdm_process(mic_array_frame_time_domain * unsafe audio, int output[]);
+void user_pdm_process(mic_array_frame_time_domain * unsafe audio);
 
 void user_pdm_init();
 
 /* PDM interface and decimation cores */
+[[combinable]]
 void pdm_buffer(streaming chanend c_ds_output[2], chanend c_audio);
 
 /* PDM interface and decimation cores */

module_usb_audio/usb_buffer/decouple.xc

@@ -14,25 +14,47 @@
 #endif
 #define MAX(x,y) ((x)>(y) ? (x) : (y))
 
-#define HS_PACKET_SIZE ((((MAX_FREQ+7999)/8000))+3)     // Samples per channel
-#define FS_PACKET_SIZE  ((((MAX_FREQ_FS+999)/1000))+3)  // Samples per channel
-
 /* TODO use SLOTSIZE to potentially save memory */
-#define BUFF_SIZE_OUT MAX(4 * HS_PACKET_SIZE * NUM_USB_CHAN_OUT, 4 * FS_PACKET_SIZE * NUM_USB_CHAN_OUT_FS)
+/* Note we could improve on this, for one subslot is set to 4 */
-#define BUFF_SIZE_IN  MAX(4 * HS_PACKET_SIZE * NUM_USB_CHAN_IN, 4 * FS_PACKET_SIZE * NUM_USB_CHAN_IN_FS)
+/* The *4 is conversion to bytes, note we're assuming a slotsize of 4 here whic is potentially as waste */
+#define MAX_DEVICE_AUD_PACKET_SIZE_MULT_HS  ((MAX_FREQ/8000+1)*4)
+#define MAX_DEVICE_AUD_PACKET_SIZE_MULT_FS  ((MAX_FREQ_FS/1000+1)*4)
 
-/* Maximum USB buffer size (1024 bytes + 1 word to store length) */
+/*** IN PACKET SIZES ***/
-#define MAX_USB_AUD_PACKET_SIZE 1028
+/* Max packet sizes in bytes. Note the +4 is because we store packet lengths in the buffer */
+#define MAX_DEVICE_AUD_PACKET_SIZE_IN_HS  (MAX_DEVICE_AUD_PACKET_SIZE_MULT_HS * NUM_USB_CHAN_IN + 4)
+#define MAX_DEVICE_AUD_PACKET_SIZE_IN_FS  (MAX_DEVICE_AUD_PACKET_SIZE_MULT_FS * NUM_USB_CHAN_IN_FS + 4)
 
-#define OUT_BUFFER_PREFILL (MAX(NUM_USB_CHAN_OUT_FS*FS_PACKET_SIZE*3+4,NUM_USB_CHAN_OUT*HS_PACKET_SIZE*4+4)*1)
+#define MAX_DEVICE_AUD_PACKET_SIZE_IN (MAX(MAX_DEVICE_AUD_PACKET_SIZE_IN_FS, MAX_DEVICE_AUD_PACKET_SIZE_IN_HS))
-#define IN_BUFFER_PREFILL (MAX(FS_PACKET_SIZE*2+4, NUM_USB_CHAN_IN*HS_PACKET_SIZE*4+4)*2)
+
+/*** OUT PACKET SIZES ***/
+#define MAX_DEVICE_AUD_PACKET_SIZE_OUT_HS  (MAX_DEVICE_AUD_PACKET_SIZE_MULT_HS * NUM_USB_CHAN_OUT + 4)
+#define MAX_DEVICE_AUD_PACKET_SIZE_OUT_FS  (MAX_DEVICE_AUD_PACKET_SIZE_MULT_FS * NUM_USB_CHAN_OUT_FS + 4)
+
+#define MAX_DEVICE_AUD_PACKET_SIZE_OUT (MAX(MAX_DEVICE_AUD_PACKET_SIZE_OUT_FS, MAX_DEVICE_AUD_PACKET_SIZE_OUT_HS))
+
+/*** BUFFER SIZES ***/
+
+#define BUFFER_PACKET_COUNT 3  /* How many packets too allow for in buffer - minimum is 3! */
+
+#define BUFF_SIZE_OUT_HS    MAX_DEVICE_AUD_PACKET_SIZE_OUT_HS * BUFFER_PACKET_COUNT
+#define BUFF_SIZE_OUT_FS    MAX_DEVICE_AUD_PACKET_SIZE_OUT_FS * BUFFER_PACKET_COUNT
+
+#define BUFF_SIZE_IN_HS     MAX_DEVICE_AUD_PACKET_SIZE_IN_HS * BUFFER_PACKET_COUNT
+#define BUFF_SIZE_IN_FS     MAX_DEVICE_AUD_PACKET_SIZE_IN_FS * BUFFER_PACKET_COUNT
+
+#define BUFF_SIZE_OUT       MAX(BUFF_SIZE_OUT_HS, BUFF_SIZE_OUT_FS)
+#define BUFF_SIZE_IN        MAX(BUFF_SIZE_IN_HS, BUFF_SIZE_IN_FS)
+
+#define OUT_BUFFER_PREFILL  (MAX(MAX_DEVICE_AUD_PACKET_SIZE_OUT_HS, MAX_DEVICE_AUD_PACKET_SIZE_OUT_FS))
+#define IN_BUFFER_PREFILL  (MAX(MAX_DEVICE_AUD_PACKET_SIZE_IN_HS, MAX_DEVICE_AUD_PACKET_SIZE_IN_FS)*2)
 
 /* Volume and mute tables */
-#ifndef OUT_VOLUME_IN_MIXER
+#if !defined(OUT_VOLUME_IN_MIXER) && (OUTPUT_VOLUME_CONTROL == 1)
 unsigned int multOut[NUM_USB_CHAN_OUT + 1];
 static xc_ptr p_multOut;
 #endif
-#ifndef IN_VOLUME_IN_MIXER
+#if !defined(IN_VOLUME_IN_MIXER) && (INPUT_VOLUME_CONTROL == 1)
 unsigned int multIn[NUM_USB_CHAN_IN + 1];
 static xc_ptr p_multIn;
 #endif
@@ -41,20 +63,12 @@ static xc_ptr p_multIn;
 unsigned g_numUsbChan_Out = NUM_USB_CHAN_OUT;
 unsigned g_numUsbChan_In = NUM_USB_CHAN_IN;
 
-/* Note we could improve on this, for one subslot is set to 4 */
-#define MAX_DEVICE_AUD_PACKET_SIZE_MULT_HS  ((MAX_FREQ/8000+1)*4)
-#define MAX_DEVICE_AUD_PACKET_SIZE_MULT_FS  ((MAX_FREQ_FS/1000+1)*4)
-
-#define MAX_DEVICE_AUD_PACKET_SIZE_HS ((MAX_FREQ/8000+1)*NUM_USB_CHAN_IN*4)
-#define MAX_DEVICE_AUD_PACKET_SIZE_FS ((MAX_FREQ_FS/1000+1)*NUM_USB_CHAN_IN_FS*4)
-
-#define MAX_DEVICE_AUD_PACKET_SIZE (MAX(MAX_DEVICE_AUD_PACKET_SIZE_FS, MAX_DEVICE_AUD_PACKET_SIZE_HS))
-
 /* Circular audio buffers */
-unsigned outAudioBuff[BUFF_SIZE_OUT + (MAX_USB_AUD_PACKET_SIZE>>2) + 4];
+unsigned outAudioBuff[(BUFF_SIZE_OUT >> 2)+ (MAX_DEVICE_AUD_PACKET_SIZE_OUT >> 2)];
-unsigned audioBuffIn[BUFF_SIZE_IN + (MAX_DEVICE_AUD_PACKET_SIZE>>2) + 4];
+unsigned audioBuffIn[(BUFF_SIZE_IN >> 2)+ (MAX_DEVICE_AUD_PACKET_SIZE_IN >> 2)];
 
-unsigned inZeroBuff[(MAX_DEVICE_AUD_PACKET_SIZE>>2)+4];
+/* Shift down accounts for bytes -> words */
+unsigned inZeroBuff[(MAX_DEVICE_AUD_PACKET_SIZE_IN >> 2)];
 
 void GetADCCounts(unsigned samFreq, int &min, int &mid, int &max);
 
@@ -91,8 +105,13 @@ xc_ptr g_aud_to_host_wrptr;
 xc_ptr g_aud_to_host_dptr;
 xc_ptr g_aud_to_host_rdptr;
 xc_ptr g_aud_to_host_zeros;
+#if (AUDIO_CLASS == 2)
 int sampsToWrite = DEFAULT_FREQ/8000;  /* HS assumed here. Expect to be junked during a overflow before stream start */
 int totalSampsToWrite = DEFAULT_FREQ/8000;
+#else
+int sampsToWrite = DEFAULT_FREQ/1000;  /* HS assumed here. Expect to be junked during a overflow before stream start */
+int totalSampsToWrite = DEFAULT_FREQ/1000;
+#endif
 int aud_data_remaining_to_device = 0;
 
 /* Audio over/under flow flags */
@@ -109,15 +128,21 @@ unsigned unpackData = 0;
 unsigned packState = 0;
 unsigned packData = 0;
 
-/* Default to something sensible but the following are setup at stream start: */
+/* Default to something sensible but the following are setup at stream start (unless UAC1 only..) */
+#if (AUDIO_CLASS == 2)
 unsigned g_curSubSlot_Out = HS_STREAM_FORMAT_OUTPUT_1_SUBSLOT_BYTES;
 unsigned g_curSubSlot_In  = HS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES;
-
-/* Init to something sensible, but expect to be re-set before stream start */
-#if (AUDIO_CLASS==2)
-int g_maxPacketSize = MAX_DEVICE_AUD_PACKET_SIZE_HS;
 #else
-int g_maxPacketSize = MAX_DEVICE_AUD_PACKET_SIZE_FS;
+unsigned g_curSubSlot_Out = FS_STREAM_FORMAT_OUTPUT_1_SUBSLOT_BYTES;
+unsigned g_curSubSlot_In  = FS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES;
+#endif
+
+
+/* IN packet size. Init to something sensible, but expect to be re-set before stream start */
+#if (AUDIO_CLASS==2)
+int g_maxPacketSize = MAX_DEVICE_AUD_PACKET_SIZE_IN_HS;
+#else
+int g_maxPacketSize = MAX_DEVICE_AUD_PACKET_SIZE_IN_FS;
 #endif
 
 #pragma select handler
@@ -146,7 +171,7 @@ void handle_audio_request(chanend c_mix_out)
         outSamps = g_aud_from_host_wrptr - g_aud_from_host_rdptr;
         if (outSamps < 0)
         {
-            outSamps += BUFF_SIZE_OUT*4;
+            outSamps += BUFF_SIZE_OUT;
         }
 
         /* If we have a decent number of samples, come out of underflow cond */
@@ -505,7 +530,7 @@ __builtin_unreachable();
                     space_left += datalength;
                     SET_SHARED_GLOBAL(g_aud_to_host_rdptr, rdPtr);
 
-                } while(space_left < (BUFF_SIZE_IN*4/2));
+                } while(space_left < (BUFF_SIZE_IN/2));
             }
 
             sampsToWrite = totalSampsToWrite;
@@ -594,22 +619,22 @@ void decouple(chanend c_mix_out
 
     int t = array_to_xc_ptr(outAudioBuff);
 
-#ifndef OUT_VOLUME_IN_MIXER
+#if !defined(OUT_VOLUME_IN_MIXER) && (OUTPUT_VOLUME_CONTROL == 1)
     p_multOut = array_to_xc_ptr(multOut);
 #endif
-#ifndef IN_VOLUME_IN_MIXER
+#if !defined(IN_VOLUME_IN_MIXER) && (INPUT_VOLUME_CONTROL == 1)
     p_multIn = array_to_xc_ptr(multIn);
 #endif
 
     aud_from_host_fifo_start = t;
-    aud_from_host_fifo_end = aud_from_host_fifo_start + BUFF_SIZE_OUT*4;
+    aud_from_host_fifo_end = aud_from_host_fifo_start + BUFF_SIZE_OUT;
     g_aud_from_host_wrptr = aud_from_host_fifo_start;
     g_aud_from_host_rdptr = aud_from_host_fifo_start;
 
     t = array_to_xc_ptr(audioBuffIn);
 
     aud_to_host_fifo_start = t;
-    aud_to_host_fifo_end = aud_to_host_fifo_start + BUFF_SIZE_IN*4;
+    aud_to_host_fifo_end = aud_to_host_fifo_start + BUFF_SIZE_IN;
     g_aud_to_host_wrptr = aud_to_host_fifo_start;
     g_aud_to_host_rdptr = aud_to_host_fifo_start;
     g_aud_to_host_dptr = aud_to_host_fifo_start + 4;
@@ -622,14 +647,14 @@ void decouple(chanend c_mix_out
     g_aud_to_host_zeros = t;
 
     /* Init vol mult tables */
-#ifndef OUT_VOLUME_IN_MIXER
+#if !defined(OUT_VOLUME_IN_MIXER) && (OUTPUT_VOLUME_CONTROL == 1)
     for (int i = 0; i < NUM_USB_CHAN_OUT + 1; i++)
     {
         asm volatile("stw %0, %1[%2]"::"r"(MAX_VOL),"r"(p_multOut),"r"(i));
     }
 #endif
 
-#ifndef IN_VOLUME_IN_MIXER
+#if !defined(IN_VOLUME_IN_MIXER) && (INPUT_VOLUME_CONTROL == 1)
     for (int i = 0; i < NUM_USB_CHAN_IN + 1; i++)
     {
         asm volatile("stw %0, %1[%2]"::"r"(MAX_VOL),"r"(p_multIn),"r"(i));
@@ -664,10 +689,16 @@ void decouple(chanend c_mix_out
     aud_to_host_flag = 0;
     SET_SHARED_GLOBAL(g_aud_to_host_flag, aud_to_host_flag);
 
-    /* NOTE: IN EP not marked ready at this point - Initial size of zero buffer not known
+    /* NOTE: For UAC2 IN EP not marked ready at this point - Initial size of zero buffer not known
      * since we don't know the USB bus-speed yet.
      * The host will send a SetAltInterface before streaming which will lead to this core
      * getting a SET_CHANNEL_COUNT_IN. This will setup the EP for the first packet */
+#if (AUDIO_CLASS == 1)
+    /* For UAC1 we know we only run at FS */
+    /* Set buffer back to zeros buffer */
+    SET_SHARED_GLOBAL(g_aud_to_host_buffer, g_aud_to_host_zeros);
+    SetupZerosSendBuffer(aud_to_host_usb_ep, sampFreq, g_curSubSlot_In);
+#endif
 #endif
 
     while(1)
@@ -687,12 +718,12 @@ void decouple(chanend c_mix_out
             /* Check for freq change or other update */
 
             GET_SHARED_GLOBAL(tmp, g_freqChange_flag);
+#if (MIN_FREQ != MAX_FREQ)
             if (tmp == SET_SAMPLE_FREQ)
             {
                 SET_SHARED_GLOBAL(g_freqChange_flag, 0);
                 GET_SHARED_GLOBAL(sampFreq, g_freqChange_sampFreq);
 
-
                 /* Pass on to mixer */
                 DISABLE_INTERRUPTS();
                 inuint(c_mix_out);
@@ -734,7 +765,12 @@ void decouple(chanend c_mix_out
                 speedRem = 0;
                 continue;
             }
-            else if(tmp == SET_STREAM_FORMAT_IN)
+#endif
+#if (AUDIO_CLASS == 2)
+#if (MIN_FREQ != MAX_FREQ)
+            else
+#endif
+            if(tmp == SET_STREAM_FORMAT_IN)
             {
                 unsigned dataFormat, usbSpeed;
 
@@ -790,6 +826,7 @@ void decouple(chanend c_mix_out
                 SET_SHARED_GLOBAL(g_aud_from_host_rdptr, aud_from_host_fifo_start);
                 SET_SHARED_GLOBAL(g_aud_from_host_wrptr, aud_from_host_fifo_start);
 
+                /* NOTE, this is potentially usefull for UAC1 */
                 unpackState = 0;
 
                 outUnderflow = 1;
@@ -819,6 +856,7 @@ void decouple(chanend c_mix_out
                 SET_SHARED_GLOBAL(g_freqChange, 0);
                 ENABLE_INTERRUPTS();
             }
+#endif
         }
 
 #if (NUM_USB_CHAN_OUT > 0)
@@ -865,7 +903,7 @@ void decouple(chanend c_mix_out
                 space_left = aud_from_host_fifo_end - g_aud_from_host_wrptr;
             }
 
-            if (space_left <= 0 || space_left >= MAX_USB_AUD_PACKET_SIZE)
+            if (space_left <= 0 || space_left >= MAX_DEVICE_AUD_PACKET_SIZE_OUT)
             {
                 SET_SHARED_GLOBAL(g_aud_from_host_buffer, aud_from_host_wrptr);
                 XUD_SetReady_OutPtr(aud_from_host_usb_ep, aud_from_host_wrptr+4);
@@ -890,8 +928,8 @@ void decouple(chanend c_mix_out
             GET_SHARED_GLOBAL(aud_from_host_rdptr, g_aud_from_host_rdptr);
             space_left = aud_from_host_rdptr - aud_from_host_wrptr;
             if (space_left <= 0)
-                space_left += BUFF_SIZE_OUT*4;
+                space_left += BUFF_SIZE_OUT;
-            if (space_left >= (BUFF_SIZE_OUT*4/2))
+            if (space_left >= (BUFF_SIZE_OUT/2))
             {
                 /* Come out of OUT overflow state */
                 outOverflow = 0;
@@ -907,10 +945,10 @@ void decouple(chanend c_mix_out
 #if (NUM_USB_CHAN_IN > 0)
         {
             /* Check if buffer() has sent a packet to host - uses shared mem flag to save chanends */
-            int tmp;
+            int sentPkt;
-            GET_SHARED_GLOBAL(tmp, g_aud_to_host_flag);
+            GET_SHARED_GLOBAL(sentPkt, g_aud_to_host_flag);
             //case inuint_byref(c_buf_in, tmp):
-            if (tmp)
+            if (sentPkt)
             {
                 /* Signals that the IN endpoint has sent data from the passed buffer */
                 /* Reset flag */
@@ -928,7 +966,7 @@ void decouple(chanend c_mix_out
                     fill_level = aud_to_host_wrptr - aud_to_host_rdptr;
 
                     if (fill_level < 0)
-                        fill_level += BUFF_SIZE_IN*4;
+                        fill_level += BUFF_SIZE_IN;
 
                     if (fill_level >= IN_BUFFER_PREFILL)
                     {

module_usb_audio/usb_buffer/interrupt.h

@@ -81,6 +81,7 @@
 #define do_interrupt_handler(f,args)    \
   asm(ISSUE_MODE_SINGLE\
       ".align 4\n" \
+      ".cc_top __"#f"_handler.function,__"#f"_handler\n" \
       "__" #f "_handler:\n"  \
       "ENTSP_lu6 0\n" \
       "kentsp " #args "/2*2 + 20\n" \
@@ -91,7 +92,8 @@
       restore_state(f,args)       \
       "krestsp " #args "/2*2 + 20 \n" \
       "__kret:\n" \
-      "kret\n");
+      "kret\n" \
+      ".cc_bottom __"#f"_handler.function");
 
 #define register_interrupt_handler(f, args, nstackwords) \
   asm (" .section .dp.data,       \"adw\", @progbits\n"  \

module_usb_audio/usb_buffer/usb_buffer.xc

@@ -258,6 +258,11 @@ void buffer(register chanend c_aud_out, register chanend c_aud_in,
     XUD_SetReady_In(ep_hid, g_hidData, 1);
 #endif
 
+#if (AUDIO_CLASS == 1)
+    /* In UAC1 we dont use a stream start event (and we are always FS) so mark FB EP ready now */
+    XUD_SetReady_In(ep_aud_fb, fb_clocks, 3);
+#endif
+
     while(1)
     {
         XUD_Result_t result;
@@ -305,6 +310,7 @@ void buffer(register chanend c_aud_out, register chanend c_aud_in,
                 {
                     unsigned cmd = inuint(c_aud_ctl);
 
+#if (MAX_FREQ != MIN_FREQ)
                     if(cmd == SET_SAMPLE_FREQ)
                     {
                         unsigned receivedSampleFreq = inuint(c_aud_ctl);
@@ -344,7 +350,10 @@ void buffer(register chanend c_aud_out, register chanend c_aud_in,
                         * handshake elsewhere */
                         SET_SHARED_GLOBAL(g_freqChange_sampFreq, receivedSampleFreq);
                     }
-                    else if(cmd == SET_STREAM_FORMAT_IN)
+                    else 
+#endif
+#if (AUDIO_CLASS == 2)
+                        if(cmd == SET_STREAM_FORMAT_IN)
                     {
                         unsigned formatChange_DataFormat = inuint(c_aud_ctl);
                         unsigned formatChange_NumChans = inuint(c_aud_ctl);
@@ -356,8 +365,10 @@ void buffer(register chanend c_aud_out, register chanend c_aud_in,
                         SET_SHARED_GLOBAL(g_formatChange_DataFormat, formatChange_DataFormat);
                         SET_SHARED_GLOBAL(g_formatChange_SampRes, formatChange_SampRes);
                     }
+                    /* FIXME when FB EP is enabled there is no inital XUD_SetReady */
                     else if (cmd == SET_STREAM_FORMAT_OUT)
                     {
+                        
                         XUD_BusSpeed_t busSpeed;
                         unsigned formatChange_DataFormat = inuint(c_aud_ctl);
                         unsigned formatChange_NumChans = inuint(c_aud_ctl);
@@ -385,6 +396,7 @@ void buffer(register chanend c_aud_out, register chanend c_aud_in,
                         }
 #endif
                     }
+#endif
                     /* Pass on sample freq change to decouple() via global flag (saves a chanend) */
                     /* Note: freqChange flags now used to communicate other commands also */
                     SET_SHARED_GLOBAL0(g_freqChange, cmd);                /* Set command */

module_usb_audio/warnings.xc

@@ -65,11 +65,10 @@ Warnings relating to configuration defines located in this XC source file rather
 
 /* Sanity check on FS channel counts */
 #if (NUM_USB_CHAN_OUT_FS > NUM_USB_CHAN_OUT)
-#error NUM_USB_CHAN_OUT expected to be less than or equal to NUM_USB_CHAN_OUT
+#error NUM_USB_CHAN_OUT_FS expected to be less than or equal to NUM_USB_CHAN_OUT
 #endif
 
 #if (NUM_USB_CHAN_IN_FS > NUM_USB_CHAN_IN)
-#error NUM_USB_CHAN_IN expected to be less than or equal to NUM_USB_CHAN_IN
+#error NUM_USB_CHAN_IN_FS expected to be less than or equal to NUM_USB_CHAN_IN
 #endif
 
-

xpd.xml

@@ -15,7 +15,7 @@
         </component>
         <component description = "MIDI" type = "demoCode" scope = "General Use" path = "module_usb_midi" local = "false" id = "module_usb_midi" name = "MIDI Module for USB Audio Framework">
             <board></board>
-            <componentDependency version = "6.18.0">module_queue</componentDependency>
+            <componentDependency version = "6.18.1">module_queue</componentDependency>
             <keyword>MIDI</keyword>
         </component>
     </components>
@@ -104,7 +104,8 @@
     <release parenthash = "136edf6bf7393d30e1f85197b72585c22611d94f" version = "6.15.2rc0" githash = "ef12c7afda6d448e22b7eb53c21d23e986db7b0d"></release>
     <release parenthash = "d0929ae0676896b94a119e4b7c44ffadec1471d8" version = "6.16.0alpha0" githash = "5f2fec1d2ab32999c69536635f21ca42312dc267"></release>
     <release parenthash = "60b6ca27825c4ab385fb945acf23089a4240a5ba" version = "6.18.0alpha0" githash = "37855df2f1b9a0c32d8c37e9c98853df4740e446"></release>
-    <release parenthash = "f770595930dac7e19fce394908a5817c34a79cda" version = "6.18.1alpha0"></release>
+    <release parenthash = "f770595930dac7e19fce394908a5817c34a79cda" version = "6.18.1alpha0" githash = "084863d720a4175b7248d68edb13f0d853321142"></release>
+    <release parenthash = "c3cae7fe525f1470ffae49e1f0955324924a1947" version = "6.19.0alpha0"></release>
     <subpartnumber>XM-004720-SM</subpartnumber>
     <vendor>XMOS</vendor>
     <version_defines>