新增：添加了对AK4438VN的支持

- 添加了两个宏，不使用时，不会影响编译

CHANGELOG.rst

@@ -1,16 +1,43 @@
 lib_xua Change Log
 ==================
 
-HEAD
-----
+UNRELEASED
+----------
+
+  * FIXED:     Device fails to enumerate when ADAT and S/PDIF transmit are enabled
+
+4.0.0
+-----
 
   * ADDED:     Support for XCommon CMake build system
-  * RESOLVED:  Output volume control not enabled by default when MIXER disabled
-  * RESOLVED:  Full 32bit result of volume processing not calculated when required
-  * RESOLVED:  Input stream sending an erroneous zero-length packet when exiting underflow state
-  * RESOLVED   Build failures when XUA_USB_EN = 0
-  * RESOLVED:  Clock configuration issues when ADAT and S/PDIF receive are enabled (#352)
+  * FIXED:     Output volume control not enabled by default when MIXER disabled
+  * FIXED:     Full 32bit result of volume processing not calculated when
+    required
+  * FIXED:     Input stream sending an erroneous zero-length packet when exiting
+    underflow state
+  * FIXED      Build failures when XUA_USB_EN = 0
+  * FIXED:     Clock configuration issues when ADAT and S/PDIF receive are
+    enabled (#352)
+  * FIXED:     Repeated old S/PDIF and ADAT samples when entering underflow
+    state
   * CHANGED:   QUAD_SPI_FLASH replaced by XUA_QUAD_SPI_FLASH (default: 1)
+  * CHANGED:   UserBufferManagementInit() now takes a sample rate parameter
+  * CHANGED:   xcore.ai targets use sigma-delta software PLL for clock recovery
+    of digital Rx streams and synchronous USB audio by default
+  * CHANGED:   Windows host mixer control application now requires driver GUID
+    option
+
+  * Changes to dependencies:
+
+    - lib_dsp: 6.2.1 -> 6.3.0
+
+    - lib_mic_array: 4.5.0 -> 4.6.0
+
+    - lib_spdif: 5.0.1 -> 6.1.0
+
+    - lib_sw_pll: Added dependency 2.1.0
+
+    - lib_xud: 2.2.3 -> 2.3.1
 
 3.5.1
 -----

Jenkinsfile

@@ -49,7 +49,8 @@ pipeline {
                       withVenv {
                         runWaf('.', "configure clean build --target=xcore200")
                         viewEnv() {
-                          runPython("TARGET=XCORE200 pytest -s")
+                          runPython("TARGET=XCORE200 pytest -s --junitxml=pytest_unity.xml")
+                          junit "pytest_unity.xml"
                         }
                       }
                     }

README.rst

@@ -1,10 +1,9 @@
 lib_xua
 #######
 
-:Version: 3.5.1
+:Version: 4.0.0
 :Vendor: XMOS
 
-
 :Scope: General Use
 
 Summary
@@ -83,14 +82,15 @@ The following application notes use this library:
 Required Software (dependencies)
 ================================
 
+  * lib_adat (www.github.com/xmos/lib_adat)
   * lib_locks (www.github.com/xmos/lib_locks)
   * lib_logging (www.github.com/xmos/lib_logging)
   * lib_mic_array (www.github.com/xmos/lib_mic_array)
   * lib_xassert (www.github.com/xmos/lib_xassert)
   * lib_dsp (www.github.com/xmos/lib_dsp)
   * lib_spdif (www.github.com/xmos/lib_spdif)
+  * lib_sw_pll (www.github.com/xmos/lib_sw_pll)
   * lib_xud (www.github.com/xmos/lib_xud)
-  * lib_adat (www.github.com/xmos/lib_adat)
 
 Documentation
 =============
@@ -101,4 +101,3 @@ Support
 =======
 
 This package is supported by XMOS Ltd. Issues can be raised against the software at: http://www.xmos.com/support
-

examples/AN00246_xua_example/src/hwsupport.xc

@@ -1,9 +1,11 @@
-// Copyright 2017-2022 XMOS LIMITED.
+// Copyright 2017-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 #include <xs1.h>
 #include <platform.h>
 #include "xua.h"
-#include "../../shared/apppll.h"
+extern "C"{
+    #include "sw_pll.h"
+}
 
 on tile[0]: out port p_ctrl = XS1_PORT_8D;
 
@@ -38,19 +40,26 @@ void AudioHwInit()
     delay_milliseconds(100);
 
     /* Use xCORE Secondary PLL to generate *fixed* master clock */
-    AppPllEnable_SampleRate(DEFAULT_FREQ);
+    if(DEFAULT_FREQ % 22050 == 0)
+    {
+        sw_pll_fixed_clock(MCLK_441);
+    }
+    else
+    {
+        sw_pll_fixed_clock(MCLK_48);
+    }
 
     delay_milliseconds(100);
 
 	/* DAC setup: For basic I2S input we don't need any register setup. DACs will clock auto detect etc.
      * It holds DAC in reset until it gets clocks anyway.
-	 * Note, this example doesn't use the ADC's
+	 * Note, this example doesn't use the ADCs
 	 */
 }
 
 /* Configures the external audio hardware for the required sample frequency */
 void AudioHwConfig(unsigned samFreq, unsigned mClk, unsigned dsdMode, unsigned sampRes_DAC, unsigned sampRes_ADC)
 {
-    AppPllEnable_SampleRate(samFreq);
+    sw_pll_fixed_clock(mClk);
 }
 

examples/AN00247_xua_example_spdif_tx/src/hwsupport.xc

@@ -1,9 +1,12 @@
-// Copyright 2017-2022 XMOS LIMITED.
+// Copyright 2017-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 #include <xs1.h>
 #include <platform.h>
 #include "xua.h"
-#include "../../shared/apppll.h"
+#include "xassert.h"
+extern "C"{
+    #include "sw_pll.h"
+}
 
 on tile[0]: out port p_ctrl = XS1_PORT_8D;
 
@@ -38,19 +41,26 @@ void AudioHwInit()
     delay_milliseconds(100);
 
     /* Use xCORE Secondary PLL to generate *fixed* master clock */
-    AppPllEnable_SampleRate(DEFAULT_FREQ);
+    if(DEFAULT_FREQ % 22050 == 0)
+    {
+        sw_pll_fixed_clock(MCLK_441);
+    }
+    else
+    {
+        sw_pll_fixed_clock(MCLK_48);
+    }
 
     delay_milliseconds(100);
 
 	/* DAC setup: For basic I2S input we don't need any register setup. DACs will clock auto detect etc.
      * It holds DAC in reset until it gets clocks anyway.
-	 * Note, this example doesn't use the ADC's
+	 * Note, this example doesn't use the ADCs
 	 */
 }
 
 /* Configures the external audio hardware for the required sample frequency */
 void AudioHwConfig(unsigned samFreq, unsigned mClk, unsigned dsdMode, unsigned sampRes_DAC, unsigned sampRes_ADC)
 {
-    AppPllEnable_SampleRate(samFreq);
+    sw_pll_fixed_clock(mClk);
 }
 

examples/shared/apppll.h

@@ -1,109 +0,0 @@
-// Copyright 2022 XMOS LIMITED.
-// This Software is subject to the terms of the XMOS Public Licence: Version 1.
-#include <stdint.h>
-#include "xassert.h"
-
-// App PLL setup
-#define APP_PLL_CTL_BYPASS       (0)   // 0 = no bypass, 1 = bypass.
-#define APP_PLL_CTL_INPUT_SEL    (0)   // 0 = XTAL, 1 = sysPLL
-#define APP_PLL_CTL_ENABLE       (1)   // 0 = disabled, 1 = enabled.
-
-// 24MHz in, 24.576MHz out, integer mode
-// Found exact solution:   IN  24000000.0, OUT  24576000.0, VCO 2457600000.0, RD  5, FD  512, OD 10, FOD  10
-#define APP_PLL_CTL_OD_48        (4)   // Output divider = (OD+1)
-#define APP_PLL_CTL_F_48         (511) // FB divider = (F+1)/2
-#define APP_PLL_CTL_R_48         (4)   // Ref divider = (R+1)
-
-#define APP_PLL_CTL_48           ((APP_PLL_CTL_BYPASS << 29) | (APP_PLL_CTL_INPUT_SEL << 28) | (APP_PLL_CTL_ENABLE << 27) |\
-                                    (APP_PLL_CTL_OD_48 << 23) | (APP_PLL_CTL_F_48 << 8) | APP_PLL_CTL_R_48)
-
-// Fractional divide is M/N
-#define APP_PLL_FRAC_EN_48             (0)   // 0 = disabled
-#define APP_PLL_FRAC_NPLUS1_CYCLES_48  (0)   // M value is this reg value + 1.
-#define APP_PLL_FRAC_TOTAL_CYCLES_48   (0)   // N value is this reg value + 1.
-#define APP_PLL_FRAC_48          ((APP_PLL_FRAC_EN_48 << 31) | (APP_PLL_FRAC_NPLUS1_CYCLES_48 << 8) | APP_PLL_FRAC_TOTAL_CYCLES_48)
-
-// 24MHz in, 22.5792MHz out (44.1kHz * 512), frac mode
-// Found exact solution:   IN  24000000.0, OUT  22579200.0, VCO 2257920000.0, RD  5, FD  470.400 (m =   2, n =   5), OD  5, FOD   10
-#define APP_PLL_CTL_OD_441       (4)   // Output divider = (OD+1)
-#define APP_PLL_CTL_F_441        (469) // FB divider = (F+1)/2
-#define APP_PLL_CTL_R_441        (4)   // Ref divider = (R+1)
-
-#define APP_PLL_CTL_441          ((APP_PLL_CTL_BYPASS << 29) | (APP_PLL_CTL_INPUT_SEL << 28) | (APP_PLL_CTL_ENABLE << 27) |\
-                                    (APP_PLL_CTL_OD_441 << 23) | (APP_PLL_CTL_F_441 << 8) | APP_PLL_CTL_R_441)
-
-#define APP_PLL_FRAC_EN_44             (1)   // 1 = enabled
-#define APP_PLL_FRAC_NPLUS1_CYCLES_44  (1)   // M value is this reg value + 1.
-#define APP_PLL_FRAC_TOTAL_CYCLES_44   (4)   // N value is this reg value + 1.define APP_PLL_CTL_R_441        (4)   // Ref divider = (R+1)
-#define APP_PLL_FRAC_44   ((APP_PLL_FRAC_EN_44 << 31) | (APP_PLL_FRAC_NPLUS1_CYCLES_44 << 8) | APP_PLL_FRAC_TOTAL_CYCLES_44)
-
-#define APP_PLL_DIV_INPUT_SEL    (1)   // 0 = sysPLL, 1 = app_PLL
-#define APP_PLL_DIV_DISABLE      (0)   // 1 = disabled (pin connected to X1D11), 0 = enabled divider output to pin.
-#define APP_PLL_DIV_VALUE        (4)   // Divide by N+1 - remember there's a /2 also afterwards for 50/50 duty cycle.
-#define APP_PLL_DIV              ((APP_PLL_DIV_INPUT_SEL << 31) | (APP_PLL_DIV_DISABLE << 16) | APP_PLL_DIV_VALUE)
-
-/* TODO support more than two freqs..*/
-void AppPllEnable(int32_t clkFreq_hz)
-{
-    switch(clkFreq_hz)
-    {
-        case 44100*512:
-
-            // Disable the PLL
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, (APP_PLL_CTL_441 & 0xF7FFFFFF));
-            // Enable the PLL to invoke a reset on the appPLL.
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, APP_PLL_CTL_441);
-            // Must write the CTL register twice so that the F and R divider values are captured using a running clock.
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, APP_PLL_CTL_441);
-            // Now disable and re-enable the PLL so we get the full 5us reset time with the correct F and R values.
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, (APP_PLL_CTL_441 & 0xF7FFFFFF));
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, APP_PLL_CTL_441);
-
-            // Set the fractional divider if used
-            write_node_config_reg(tile[0], XS1_SSWITCH_SS_APP_PLL_FRAC_N_DIVIDER_NUM, APP_PLL_FRAC_44);
-
-            break;
-
-        case 48000*512:
-
-            // Disable the PLL
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, (APP_PLL_CTL_48 & 0xF7FFFFFF));
-            // Enable the PLL to invoke a reset on the appPLL.
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, APP_PLL_CTL_48);
-            // Must write the CTL register twice so that the F and R divider values are captured using a running clock.
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, APP_PLL_CTL_48);
-            // Now disable and re-enable the PLL so we get the full 5us reset time with the correct F and R values.
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, (APP_PLL_CTL_48 & 0xF7FFFFFF));
-            write_node_config_reg(tile[1], XS1_SSWITCH_SS_APP_PLL_CTL_NUM, APP_PLL_CTL_48);
-
-            // Set the fractional divider if used
-            write_node_config_reg(tile[0], XS1_SSWITCH_SS_APP_PLL_FRAC_N_DIVIDER_NUM, APP_PLL_FRAC_48);
-
-            break;
-
-        default:
-            assert(0);
-            break;
-    }
-
-    // Wait for PLL output frequency to stabilise due to fractional divider enable
-    delay_microseconds(100);
-
-    // Turn on the clock output
-    write_node_config_reg(tile[0], XS1_SSWITCH_SS_APP_CLK_DIVIDER_NUM, APP_PLL_DIV);
-}
-
-void AppPllEnable_SampleRate(int32_t sampleRate_hz)
-{
-    assert(sampleRate_hz >= 22050);
-
-    if(sampleRate_hz % 22050 == 0)
-    {
-        AppPllEnable(44100*512);
-    }
-    else
-    {
-        AppPllEnable(48000*512);
-    }
-}
-

host_usb_mixer_control/mixer_app.cpp

@@ -1,4 +1,4 @@
-// Copyright 2022-2023 XMOS LIMITED.
+// Copyright 2022-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
 #include <stdlib.h>
@@ -14,10 +14,20 @@
 // TODO
 // res, min, max
 
+#ifdef _WIN32
+int mixer_init(TCHAR guid[GUID_STR_LEN])
+#else
 int mixer_init(void)
+#endif
 {
+#ifdef _WIN32
+    int ret = usb_mixer_connect(guid);
+#else
+    int ret = usb_mixer_connect();
+#endif
+
     /* Open the connection to the USB mixer */
-    if (usb_mixer_connect() == USB_MIXER_FAILURE) 
+    if (ret == USB_MIXER_FAILURE)
     {
         return USB_MIXER_FAILURE;
     }
@@ -381,8 +391,15 @@ void print_levels(const char* levelTitle, unsigned char* levels, int levelBytes)
 
 
 void mixer_display_usage(void) {
-    fprintf(stderr, "Usage :\n");
-    fprintf(stderr, 
+    fprintf(stderr, "Usage: xmos_mixer "
+#ifdef _WIN32
+                    "-g<GUID> "
+#endif
+                    "<options>\n");
+    fprintf(stderr,
+#ifdef _WIN32
+            "     -g<GUID>                            Driver GUID string, eg. -g{E5A2658B-817D-4A02-A1DE-B628A93DDF5D}\n"
+#endif
             "     --display-info\n"
             "     --display-mixer-nodes               mixer_id\n"
             "     --display-min                       mixer_id\n"
@@ -421,97 +438,124 @@ int main (int argc, char **argv) {
   unsigned int mixer_index = 0;
   unsigned int result = 0;
 
+  int min_argc;
+  // arg_idx is the position in the arguments to start parsing to skip the "-g" GUID option on Windows
+  int arg_idx;
+#ifdef _WIN32
+  // Driver GUID string is required on Windows
+  min_argc = 3;
+  arg_idx = 2;
+#else
+  min_argc = 2;
+  arg_idx = 1;
+#endif
 
-
-  if (argc < 2) {
+  if (argc < min_argc) {
     fprintf(stderr, "ERROR :: No options passed to mixer application\n");
     mixer_display_usage();
     return -1;
   }
 
+#ifdef _WIN32
+  TCHAR driver_guid[GUID_STR_LEN];
+
+  if (strncmp(argv[1], "-g", 2) == 0) {
+    swprintf(driver_guid, GUID_STR_LEN, L"%hs", argv[1]+2);
+  } else {
+    fprintf(stderr, "ERROR :: First option must be driver GUID\n");
+    return -1;
+  }
+#endif
+
   if (strcmp(argv[1], "--help") == 0) {
     mixer_display_usage();
     return 0;
   } 
 
-  if (mixer_init() != USB_MIXER_SUCCESS) {
+#ifdef _WIN32
+  int ret = mixer_init(driver_guid);
+#else
+  int ret = mixer_init();
+#endif
+
+  if (ret != USB_MIXER_SUCCESS) {
     fprintf(stderr, "ERROR :: Cannot connect\n");
     return -1;
   }
 
-  if (strcmp(argv[1], "--display-info") == 0) 
+  if (strcmp(argv[arg_idx], "--display-info") == 0)
   {
     mixer_display_info();
   } 
-  else if (strcmp(argv[1], "--display-mixer-nodes") == 0) 
+  else if (strcmp(argv[arg_idx], "--display-mixer-nodes") == 0)
   {
-    if (argv[2]) 
+    if (argv[arg_idx+1])
     {
-      mixer_index = atoi(argv[2]);
+      mixer_index = atoi(argv[arg_idx+1]);
     } else {
       fprintf(stderr, "ERROR :: No mixer index supplied\n");
       return -1;
     }
     mixer_display(mixer_index, MIXER_UNIT_DISPLAY_VALUE);
-  } else if (strcmp(argv[1], "--display-mixer-nodes") == 0) {
+  } else if (strcmp(argv[arg_idx], "--display-mixer-nodes") == 0) {
     if (argv[2]) {
-      mixer_index = atoi(argv[2]);
+      mixer_index = atoi(argv[arg_idx+1]);
     } else {
       fprintf(stderr, "ERROR :: No mixer index supplied\n");
       return -1;
     }
     mixer_display(mixer_index, MIXER_UNIT_DISPLAY_VALUE);
-  } else if (strcmp(argv[1], "--display-min") == 0) {
-    if (argv[2]) {
-      mixer_index = atoi(argv[2]);
+  } else if (strcmp(argv[arg_idx], "--display-min") == 0) {
+    if (argv[arg_idx+1]) {
+      mixer_index = atoi(argv[arg_idx+1]);
     } else {
       fprintf(stderr, "ERROR :: No mixer index supplied\n");
       return -1;
     }
     mixer_display(mixer_index, MIXER_UNIT_DISPLAY_MIN);
-  } else if (strcmp(argv[1], "--display-max") == 0) {
-    if (argv[2]) {
-      mixer_index = atoi(argv[2]);
+  } else if (strcmp(argv[arg_idx], "--display-max") == 0) {
+    if (argv[arg_idx+1]) {
+      mixer_index = atoi(argv[arg_idx+1]);
     } else {
       fprintf(stderr, "ERROR :: No mixer index supplied\n");
       return -1;
     }
     mixer_display(mixer_index, MIXER_UNIT_DISPLAY_MAX);
-  } else if (strcmp(argv[1], "--display-res") == 0) {
-    if (argv[2]) {
-      mixer_index = atoi(argv[2]);
+  } else if (strcmp(argv[arg_idx], "--display-res") == 0) {
+    if (argv[arg_idx+1]) {
+      mixer_index = atoi(argv[arg_idx+1]);
     } else {
       fprintf(stderr, "ERROR :: No mixer index supplied\n");
       return -1;
     }
     mixer_display(mixer_index, MIXER_UNIT_DISPLAY_RES);
   } 
-  else if (strcmp(argv[1], "--set-value") == 0) {
+  else if (strcmp(argv[arg_idx], "--set-value") == 0) {
     unsigned int mixer_unit = 0;
     double value = 0;
-    if (argc < 5) {
+    if (argc - arg_idx < 4) {
       fprintf(stderr, "ERROR :: incorrect number of arguments passed\n");
       return -1;
     }
 
-    mixer_index = atoi(argv[2]);
-    mixer_unit = atoi(argv[3]);
-    if (strcmp(argv[4],"-inf")==0) 
+    mixer_index = atoi(argv[arg_idx+1]);
+    mixer_unit = atoi(argv[arg_idx+2]);
+    if (strcmp(argv[arg_idx+3],"-inf")==0)
       value = -128;
     else
-      value = atof(argv[4]);
+      value = atof(argv[arg_idx+3]);
 
     usb_mixer_set_value(mixer_index, mixer_unit, value);
-  } else if (strcmp(argv[1], "--get-value") == 0) {
+  } else if (strcmp(argv[arg_idx], "--get-value") == 0) {
     unsigned int mixer_unit = 0;
     double result = 0;
-    if (argc < 4) {
+    if (argc - arg_idx < 3) {
       fprintf(stderr, "ERROR :: incorrect number of arguments passed\n");
       return -1;
     }
 
-    mixer_index = atoi(argv[2]);
-    mixer_unit = atoi(argv[3]);
+    mixer_index = atoi(argv[arg_idx+1]);
+    mixer_unit = atoi(argv[arg_idx+2]);
 
     result = usb_mixer_get_value(mixer_index, mixer_unit);
     if (result <= -127.996)
@@ -519,99 +563,99 @@ int main (int argc, char **argv) {
     else
       printf("%g\n",result);
   }
-  else if (strcmp(argv[1], "--display-current-mixer-sources") == 0) 
+  else if (strcmp(argv[arg_idx], "--display-current-mixer-sources") == 0)
   {
-    if(argc < 3)
+    if(argc - arg_idx < 2)
     {
         usage_error();
         return -1;
     }
-    display_mixer_sources(atoi(argv[2]));
+    display_mixer_sources(atoi(argv[arg_idx+1]));
   }
-  else if (strcmp(argv[1], "--display-available-mixer-sources") == 0) 
+  else if (strcmp(argv[arg_idx], "--display-available-mixer-sources") == 0)
   {
-    if(argc < 3)
+    if(argc - arg_idx < 2)
     {
         usage_error();
         return -1;
     }
-    display_available_mixer_sources(atoi(argv[2]));
+    display_available_mixer_sources(atoi(argv[arg_idx+1]));
   }
-  else if(strcmp(argv[1], "--set-mixer-source") == 0)
+  else if(strcmp(argv[arg_idx], "--set-mixer-source") == 0)
   {
-    if(argc < 5)
+    if(argc - arg_idx < 4)
     {
         usage_error();
         return -1;
     }
-    set_mixer_source(atoi(argv[2]), atoi(argv[3]), atoi(argv[4]));
+    set_mixer_source(atoi(argv[arg_idx+1]), atoi(argv[arg_idx+2]), atoi(argv[arg_idx+3]));
   }
-    else if (strcmp(argv[1], "--display-aud-channel-map") == 0) 
+    else if (strcmp(argv[arg_idx], "--display-aud-channel-map") == 0)
     {
         /* Display the channel mapping to the devices audio outputs */
         display_aud_channel_map();
     }
-    else if (strcmp(argv[1], "--display-aud-channel-map-sources") == 0) 
+    else if (strcmp(argv[arg_idx], "--display-aud-channel-map-sources") == 0)
     {
         display_aud_channel_map_sources();
     }
-    else if (strcmp(argv[1], "--display-daw-channel-map") == 0) 
+    else if (strcmp(argv[arg_idx], "--display-daw-channel-map") == 0)
     {
         /* Display the channel mapping to the devices DAW output to host */
         display_daw_channel_map();
     }
-    else if (strcmp(argv[1], "--display-daw-channel-map-sources") == 0) 
+    else if (strcmp(argv[arg_idx], "--display-daw-channel-map-sources") == 0)
     {
         display_daw_channel_map_sources();
     }
-    else if (strcmp(argv[1], "--set-aud-channel-map") == 0) 
+    else if (strcmp(argv[arg_idx], "--set-aud-channel-map") == 0)
     { 
         unsigned int dst = 0;
         unsigned int src = 0;
-        if (argc != 4) 
+        if (argc - arg_idx != 3)
         {
             usage_error();
             return -1;
         }
-        dst = atoi(argv[2]);
-        src = atoi(argv[3]);
+        dst = atoi(argv[arg_idx+1]);
+        src = atoi(argv[arg_idx+2]);
 
         usb_set_aud_channel_map(dst, src);
     } 
-  else if (strcmp(argv[1], "--set-daw-channel-map") == 0) 
+  else if (strcmp(argv[arg_idx], "--set-daw-channel-map") == 0)
   { 
     unsigned int dst = 0;
     unsigned int src = 0;
-    if (argc != 4) 
+    if (argc - arg_idx != 3)
     {
         usage_error();
         return -1;
     }
-    dst = atoi(argv[2]);
-    src = atoi(argv[3]);
+    dst = atoi(argv[arg_idx+1]);
+    src = atoi(argv[arg_idx+2]);
 
     usb_set_usb_channel_map(dst, src);
   }
-  else if(strcmp(argv[1], "--get-mixer-levels-input") == 0 || 
-    strcmp(argv[1],"--get-mixer-levels-output") == 0) 
+  else if(strcmp(argv[arg_idx], "--get-mixer-levels-input") == 0 ||
+    strcmp(argv[arg_idx],"--get-mixer-levels-output") == 0)
   {
     unsigned int dst = 0;
     unsigned char levels[64];
     int datalength = 0;
     int offset = 0;
 
-    if (argc < 3) {
+    if (argc - arg_idx < 2) {
       fprintf(stderr, "ERROR :: incorrect number of arguments passed\n");
       return -1;
     }
 
-    if(strcmp(argv[1],"--get-mixer-levels-output") == 0) 
+    if(strcmp(argv[arg_idx],"--get-mixer-levels-output") == 0)
        offset = 1;
 
     for(int i = 0; i < 64; i++)
       levels[i] = 0;
 
-    dst = atoi(argv[2]);
+    dst = atoi(argv[arg_idx+1]);
     
     /* Mem request to mixer with offset of 0 gives input levels */
     datalength = usb_mixer_mem_get(dst, offset, levels);
@@ -628,7 +672,7 @@ int main (int argc, char **argv) {
       print_levels("Mixer Input", levels, datalength);
 
   }
-  else if(strcmp(argv[1], "--vendor-audio-request-get") == 0)
+  else if(strcmp(argv[arg_idx], "--vendor-audio-request-get") == 0)
   {
     unsigned int bRequest = 0;
     unsigned int cs = 0;
@@ -637,7 +681,7 @@ int main (int argc, char **argv) {
     int datalength = 0;
     unsigned char data[64];
 
-    if(argc < 6)
+    if(argc - arg_idx < 5)
     {
       fprintf(stderr, "ERROR :: incorrect number of arguments passed\n");
       return -1;
@@ -646,10 +690,10 @@ int main (int argc, char **argv) {
     for(int i = 0; i < 64; i++)
       data[i] = 0;
 
-    bRequest = atoi(argv[2]);
-    cs = atoi(argv[3]);
-    cn = atoi(argv[4]);
-    unitId = atoi(argv[5]);
+    bRequest = atoi(argv[arg_idx+1]);
+    cs = atoi(argv[arg_idx+2]);
+    cn = atoi(argv[arg_idx+3]);
+    unitId = atoi(argv[arg_idx+4]);
    
     /* Do request */ 
     datalength = usb_audio_request_get(bRequest, cs, cn, unitId, data);    
@@ -666,7 +710,7 @@ int main (int argc, char **argv) {
             printf("0x%02x\n" ,data[i]);
     }
   }
-  else if(strcmp(argv[1], "--vendor-audio-request-set") == 0)
+  else if(strcmp(argv[arg_idx], "--vendor-audio-request-set") == 0)
   {
   
     unsigned int bRequest = 0;
@@ -680,23 +724,23 @@ int main (int argc, char **argv) {
       data[i] = 0;
     }
 
-    if(argc < 7)
+    if(argc - arg_idx < 6)
     {
       fprintf(stderr, "ERROR :: incorrect number of arguments passed - no data passed\n");
       return -1;
     }
-    bRequest = atoi(argv[2]);
-    cs = atoi(argv[3]);
-    cn = atoi(argv[4]);
-    unitId = atoi(argv[5]);
+    bRequest = atoi(argv[arg_idx+1]);
+    cs = atoi(argv[arg_idx+2]);
+    cn = atoi(argv[arg_idx+3]);
+    unitId = atoi(argv[arg_idx+4]);
    
     /* Get data */
-    for(int i=0; i < argc-6; i++)
+    for(int i=0; i < argc-arg_idx-5; i++)
     {
-       data[i] = atoi(argv[i+6]);
+       data[i] = atoi(argv[i+arg_idx+5]);
     }
     
-    result = usb_audio_request_set(bRequest, cs, cn, unitId, data, argc-6);    
+    result = usb_audio_request_set(bRequest, cs, cn, unitId, data, argc-arg_idx-5);
    
     if(result < 0)
     {

host_usb_mixer_control/usb_mixer.cpp

@@ -1,4 +1,4 @@
-// Copyright 2022-2023 XMOS LIMITED.
+// Copyright 2022-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
 #include <stdlib.h>
@@ -800,14 +800,18 @@ static int find_xmos_device(unsigned int id)
 
 // End of libusb interface functions
 
-int usb_mixer_connect() 
+#ifdef _WIN32
+int usb_mixer_connect(TCHAR guid[GUID_STR_LEN])
+#else
+int usb_mixer_connect()
+#endif
 {
     // Allocate internal storage
     usb_mixers = (usb_mixer_handle *)malloc(sizeof(usb_mixer_handle));
     memset(usb_mixers, 0, sizeof(usb_mixer_handle));
 
 #if defined(_WIN32)
-    gDrvApi.LoadByGUID(_T("{E5A2658B-817D-4A02-A1DE-B628A93DDF5D}"));
+    gDrvApi.LoadByGUID(guid);
     TUsbAudioStatus st = gDrvApi.TUSBAUDIO_EnumerateDevices();
 #endif
 

host_usb_mixer_control/usb_mixer.h

@@ -1,4 +1,4 @@
-// Copyright 2022-2023 XMOS LIMITED.
+// Copyright 2022-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
 #define USB_MIXER_SUCCESS 0
@@ -22,7 +22,14 @@ enum usb_chan_type {
 #define RANGE (2)
 #define MEM   (3)
 
+#ifdef _WIN32
+#include <tchar.h>
+// GUID strings are 36 characters, plus a pair of braces and NUL-termination
+#define GUID_STR_LEN (36+2+1)
+int usb_mixer_connect(TCHAR guid[GUID_STR_LEN]);
+#else
 int usb_mixer_connect();
+#endif
 int usb_mixer_disconnect();
 
 /* MIXER UNIT(s) INTERFACE */

lib_xua/README.md

@@ -0,0 +1,9 @@
+# lib_xua
+
+- 版本：3.5.1
+- 私有版本维护：Vergil Wong
+
+## 主要改动
+
+- 添加了`I2S_TDM_LRCLK_EDGES`，以适配AK4438VN的TDM时序，参考[修改时序以配置TDM](https://docs.pawpaw.cn/docs/applications/USB-Audio-Interface/Advanced-Guide/app-note/dac-timing-config/)
+- 添加了`AUDIO_UNSAFE_RESRC`，以将`unsafe`资源的赋值，合并到Audio线程中，以避免编译器检查（额外占用核心）

lib_xua/api/xua_audiohub.h

@@ -1,4 +1,4 @@
-// Copyright 2011-2023 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 #ifndef _XUA_AUDIOHUB_H_
 #define _XUA_AUDIOHUB_H_
@@ -12,32 +12,42 @@
 #include "dfu_interface.h"
 #endif
 
+#include "xua_clocking.h"
+
 /** The audio driver thread.
  *
  *  This function drives I2S ports and handles samples to/from other digital
  *  I/O threads.
  *
- *  \param c_aud            Audio sample channel connected to the mixer() thread or the
- *                          decouple() thread
+ *  \param c_aud                Audio sample channel connected to the mixer() thread or the
+ *                              decouple() thread
  *
- *  \param clk_audio_mclk   Nullable clockblock to be clocked from master clock
+ *  \param clk_audio_mclk       Nullable clockblock to be clocked from master clock
  *
- *  \param clk_audio_bclk   Nullable clockblock to be clocked from i2s bit clock
+ *  \param clk_audio_bclk       Nullable clockblock to be clocked from i2s bit clock
  *
- *  \param p_mclk_in        Master clock inport port (must be 1-bit)
+ *  \param p_mclk_in            Master clock inport port (must be 1-bit)
  *
- *  \param p_lrclk          Nullable port for I2S sample clock
+ *  \param p_lrclk              Nullable port for I2S sample clock
  *
- *  \param p_bclk           Nullable port for I2S bit
+ *  \param p_bclk               Nullable port for I2S bit clock
  *
- *  \param p_i2s_dac        Nullable array of ports for I2S data output lines
+ *  \param p_i2s_dac            Nullable array of ports for I2S data output lines
  *
- *  \param p_i2s_adc        Nullable array of ports for I2S data input lines
+ *  \param p_i2s_adc            Nullable array of ports for I2S data input lines
  *
- *  \param c_spdif_tx       Channel connected to S/PDIF transmiter core from lib_spdif
+ *  \param i_SoftPll            Interface to software PLL task
  *
- *  \param c_dig            Channel connected to the clockGen() thread for
- *                          receiving/transmitting samples
+ *  \param c_spdif_tx           Channel connected to S/PDIF transmitter core from lib_spdif
+ *
+ *  \param c_dig                Channel connected to the clockGen() thread for
+ *                              receiving/transmitting samples
+ *
+ *  \param c_audio_rate_change  Channel notifying ep_buffer of an mclk frequency change and sync for stable clock
+ *
+ *  \param dfuInterface         Interface supporting DFU methods
+ *
+ *  \param c_pdm_in             Channel for receiving decimated PDM samples
  */
 void XUA_AudioHub(chanend ?c_aud,
     clock ?clk_audio_mclk,
@@ -53,10 +63,13 @@ void XUA_AudioHub(chanend ?c_aud,
 #if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN || defined(__DOXYGEN__))
     , chanend c_dig
 #endif
-#if (XUD_TILE != 0) && (AUDIO_IO_TILE == 0) && (XUA_DFU_EN == 1)
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN || defined(__DOXYGEN__))
+    , chanend c_audio_rate_change
+#endif
+#if (((XUD_TILE != 0) && (AUDIO_IO_TILE == 0) && (XUA_DFU_EN == 1)) || defined(__DOXYGEN__))
    , server interface i_dfu ?dfuInterface
 #endif
-#if (XUA_NUM_PDM_MICS > 0)
+#if (XUA_NUM_PDM_MICS > 0 || defined(__DOXYGEN__))
     , chanend c_pdm_in
 #endif
 );
@@ -76,8 +89,29 @@ void AudioHwConfig(unsigned samFreq, unsigned mClk, unsigned dsdMode,
 
 #endif // __XC__
 
-void UserBufferManagementInit();
-
+/**
+ * @brief   User buffer management code
+ *
+ * This function is called at the sample rate of the USB Audio stack (e.g,. 48 kHz) and between the two parameter arrays
+ * contain a full multi-channel audio-frame. The first array carries all the data that has been received from the USB host
+ * and is to be presented to the audio interfaces. The second array carries all the data received from the interfaces and
+ * is to be presented to the USB host. The user can chose to intercept and overwrite the samples stored in these arrays.
+ *
+ * \param sampsFromUsbToAudio    Samples received from USB host and to be presented to audio interfaces
+ *
+ * \param sampsFromAudioToUsb    Samples received from the audio interfaces and to be presented to the USB host
+*/
 void UserBufferManagement(unsigned sampsFromUsbToAudio[], unsigned sampsFromAudioToUsb[]);
 
+/**
+ * @brief   User buffer managment init code
+ *
+ * This function is called once, before the first call to UserBufferManagement(), and can be used to initialise any
+ * related user state
+ *
+ * \param sampFreq               The initial sample frequency
+ *
+ */
+void UserBufferManagementInit(unsigned sampFreq);
+
 #endif // _XUA_AUDIOHUB_H_

lib_xua/api/xua_buffer.h

@@ -1,7 +1,7 @@
-// Copyright 2011-2022 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
-#ifndef __XUA_BUFFER_H__
-#define __XUA_BUFFER_H__
+#ifndef _XUA_BUFFER_H_
+#define _XUA_BUFFER_H_
 
 #if __XC__
 
@@ -13,19 +13,21 @@
  *  Most of the chanend parameters to the function should be connected to
  *  XUD_Manager().  The uses two cores.
  *
- *  \param c_aud_out          Audio OUT endpoint channel connected to the XUD
- *  \param c_aud_in           Audio IN endpoint channel connected to the XUD
- *  \param c_aud_fb           Audio feedback endpoint channel connected to the XUD
- *  \param c_midi_from_host   MIDI OUT endpoint channel connected to the XUD
- *  \param c_midi_to_host     MIDI IN endpoint channel connected to the XUD
- *  \param c_midi             Channel connected to MIDI core
- *  \param c_int              Audio clocking interrupt endpoint channel connected to the XUD
- *  \param c_clk_int          Optional chanend connected to the clockGen() thread if present
- *  \param c_sof              Start of frame channel connected to the XUD
- *  \param c_aud_ctl          Audio control channel connected to  Endpoint0()
- *  \param p_off_mclk         A port that is clocked of the MCLK input (not the MCLK input itself)
- *  \param c_aud              Channel connected to XUA_AudioHub() core
- *  \param i_pll_ref          Interface to task that toggles reference pin to CS2100
+ *  \param c_aud_out            Audio OUT endpoint channel connected to the XUD
+ *  \param c_aud_in             Audio IN endpoint channel connected to the XUD
+ *  \param c_aud_fb             Audio feedback endpoint channel connected to the XUD
+ *  \param c_midi_from_host     MIDI OUT endpoint channel connected to the XUD
+ *  \param c_midi_to_host       MIDI IN endpoint channel connected to the XUD
+ *  \param c_midi               Channel connected to MIDI core
+ *  \param c_int                Audio clocking interrupt endpoint channel connected to the XUD
+ *  \param c_clk_int            Optional chanend connected to the clockGen() thread if present
+ *  \param c_sof                Start of frame channel connected to the XUD
+ *  \param c_aud_ctl            Audio control channel connected to  Endpoint0()
+ *  \param p_off_mclk           A port that is clocked of the MCLK input (not the MCLK input itself)
+ *  \param c_aud                Channel connected to XUA_AudioHub() core
+ *  \param c_audio_rate_change  Channel to notify and synchronise on audio rate change
+ *  \param i_pll_ref            Interface to task that toggles reference pin to CS2100
+ *  \param c_swpll_update       Channel connected to software PLL task. Expects master clock counts based on USB frames.
  */
 void XUA_Buffer(
             chanend c_aud_out,
@@ -38,7 +40,7 @@ void XUA_Buffer(
 #if defined(MIDI) || defined(__DOXYGEN__)
             chanend c_midi_from_host,
             chanend c_midi_to_host,
-			chanend c_midi,
+            chanend c_midi,
 #endif
 #if XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN || defined(__DOXYGEN__)
             chanend ?c_int,
@@ -51,8 +53,14 @@ void XUA_Buffer(
             , chanend c_hid
 #endif
             , chanend c_aud
-#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC) || defined(__DOXYGEN__)
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC) || defined(__DOYXGEN__)
+            , chanend c_audio_rate_change
+    #if (!XUA_USE_SW_PLL) || defined(__DOXYGEN__)
             , client interface pll_ref_if i_pll_ref
+    #endif
+    #if (XUA_USE_SW_PLL) || defined(__DOXYGEN__)
+            , chanend c_swpll_update
+    #endif
 #endif
         );
 
@@ -66,7 +74,7 @@ void XUA_Buffer_Ep(chanend c_aud_out,
 #ifdef MIDI
             chanend c_midi_from_host,
             chanend c_midi_to_host,
-			chanend c_midi,
+            chanend c_midi,
 #endif
 #if (XUA_SPDIF_RX_EN) || (XUA_ADAT_RX_EN)
             chanend ?c_int,
@@ -81,10 +89,17 @@ void XUA_Buffer_Ep(chanend c_aud_out,
 #ifdef CHAN_BUFF_CTRL
             , chanend c_buff_ctrl
 #endif
-#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC) || defined(__DOXYGEN__)
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC) || defined(__DOYXGEN__)
+            , chanend c_audio_rate_change
+    #if (!XUA_USE_SW_PLL) || defined(__DOXYGEN__)
             , client interface pll_ref_if i_pll_ref
+    #endif
+    #if (XUA_USE_SW_PLL) || defined(__DOXYGEN__)
+            , chanend c_swpll_update
+    #endif
 #endif
-        );
+    );
+
 
 /** Manage the data transfer between the USB audio buffer and the
  *  Audio I/O driver.

lib_xua/api/xua_clocking.h

@@ -1,4 +1,4 @@
-// Copyright 2011-2022 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
 #ifndef _CLOCKING_H_
@@ -6,6 +6,8 @@
 
 #include <xs1.h>
 
+#include "sw_pll_wrapper.h"
+
 interface pll_ref_if
 {
     void toggle();
@@ -18,15 +20,31 @@ void PllRefPinTask(server interface pll_ref_if i_pll_ref, out port p_sync);
 
 /** Clock generation and digital audio I/O handling.
  *
- *  \param c_spdif_rx channel connected to S/PDIF receive thread
- *  \param c_adat_rx channel connect to ADAT receive thread
- *  \param i_pll_ref interface to taslk that outputs clock signal to drive external frequency synthesizer
- *  \param c_audio channel connected to the audio() thread
- *  \param c_clk_ctl channel connected to Endpoint0() for configuration of the
- *                   clock
- *  \param c_clk_int channel connected to the decouple() thread for clock
-                     interrupts
+ *  \param c_spdif_rx           channel connected to S/PDIF receive thread
+ *  \param c_adat_rx            channel connect to ADAT receive thread
+ *  \param i_pll_ref            interface to taslk that outputs clock signal to drive external frequency synthesizer
+ *  \param c_audio              channel connected to the audio() thread
+ *  \param c_clk_ctl            channel connected to Endpoint0() for configuration of the
+ *                              clock
+ *  \param c_clk_int            channel connected to the decouple() thread for clock
+ *                              interrupts
+ *  \param c_audio_rate_change  channel to notify of master clock change
+ *  \param p_for_mclk_count_aud port used for counting mclk and providing a timestamp
+ *  \param c_sw_pll             channel used to communicate with software PLL task
+ *
  */
-void clockGen(streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interface pll_ref_if i_pll_ref, chanend c_audio, chanend c_clk_ctl, chanend c_clk_int);
+void clockGen(  streaming chanend ?c_spdif_rx,
+                chanend ?c_adat_rx,
+                client interface pll_ref_if i_pll_ref,
+                chanend c_audio,
+                chanend c_clk_ctl,
+                chanend c_clk_int,
+                chanend c_audio_rate_change
+#if XUA_USE_SW_PLL
+                , port p_for_mclk_count_aud
+                , chanend c_sw_pll
+#endif
+                );
+
 #endif
 

lib_xua/api/xua_conf_default.h

@@ -1,4 +1,4 @@
-// Copyright 2011-2023 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 /*
  * @brief       Defines relating to device configuration and customisation of lib_xua
@@ -11,56 +11,55 @@
     #include "xua_conf.h"
 #endif
 
-/* Default tile arrangement */
+/*
+ * Tile arrangement defines
+ */
 
 /**
  * @brief Location (tile) of audio I/O. Default: 0
  */
 #ifndef AUDIO_IO_TILE
-#define AUDIO_IO_TILE           (0)
+#define AUDIO_IO_TILE   (0)
 #endif
 
 /**
  * @brief Location (tile) of audio I/O. Default: 0
  */
 #ifndef XUD_TILE
-#define XUD_TILE                (0)
+#define XUD_TILE        (0)
 #endif
 
 /**
  * @brief Location (tile) of MIDI I/O. Default: AUDIO_IO_TILE
  */
 #ifndef MIDI_TILE
-#define MIDI_TILE               AUDIO_IO_TILE
+#define MIDI_TILE       AUDIO_IO_TILE
 #endif
 
 /**
  * @brief Location (tile) of SPDIF Tx. Default: AUDIO_IO_TILE
  */
 #ifndef SPDIF_TX_TILE
-#define SPDIF_TX_TILE           AUDIO_IO_TILE
+#define SPDIF_TX_TILE   AUDIO_IO_TILE
 #endif
 
 /**
  * @brief Location (tile) of PDM Rx. Default: AUDIO_IO_TILE
  */
 #ifndef PDM_TILE
-#define PDM_TILE                AUDIO_IO_TILE
+#define PDM_TILE        AUDIO_IO_TILE
 #endif
 
 /**
  * @brief Location (tile) of reference signal to CS2100. Default: AUDIO_IO_TILE
  */
 #ifndef PLL_REF_TILE
-#define PLL_REF_TILE            AUDIO_IO_TILE
+#define PLL_REF_TILE    AUDIO_IO_TILE
 #endif
 
-/**
- * @brief Disable USB functionalty just leaving AudioHub
+/*
+ * Channel based defines
  */
-#ifndef XUA_USB_EN
-#define XUA_USB_EN              (1)
-#endif
 
 /**
  * @brief Number of input channels (device to host). Default: NONE (Must be defined by app)
@@ -79,18 +78,57 @@
 #endif
 
 /**
- * @brief Number of DSD output channels. Default: 0 (disabled)
+ * @brief Number of PDM microphones in the design.
+ *
+ * Default: 0
+ */
+#ifndef XUA_NUM_PDM_MICS
+#define XUA_NUM_PDM_MICS         (0)
+#endif
+
+/**
+ * @brief Number of DSD output channels.
+ *
+ * Default: 0 (disabled)
  */
 #if defined(DSD_CHANS_DAC) && (DSD_CHANS_DAC != 0)
     #if defined(NATIVE_DSD) && (NATIVE_DSD == 0)
         #undef NATIVE_DSD
     #else
-        #define NATIVE_DSD      (1)  /* Always enable Native DSD when DSD mode is enabled */
+        #define NATIVE_DSD       1  /* Always enable Native DSD when DSD mode is enabled */
     #endif
 #else
-    #define DSD_CHANS_DAC       (0)
+    #define DSD_CHANS_DAC        0
 #endif
 
+/**
+ * @brief Number of I2S channesl to DAC/CODEC. Must be a multiple of 2.
+ *
+ * Default: NONE (Must be defined by app)
+ */
+#ifndef I2S_CHANS_DAC
+    #error I2S_CHANS_DAC not defined
+    #define I2S_CHANS_DAC 2          /* Define anyway for doxygen */
+#else
+#define I2S_WIRES_DAC            (I2S_CHANS_DAC / I2S_CHANS_PER_FRAME)
+#endif
+
+/**
+ * @brief Number of I2S channels from ADC/CODEC. Must be a multiple of 2.
+ *
+ * Default: NONE (Must be defined by app)
+ */
+#ifndef I2S_CHANS_ADC
+    #error I2S_CHANS_ADC not defined
+    #define I2S_CHANS_ADC 2      /* Define anyway for doxygen */
+#else
+#define I2S_WIRES_ADC            (I2S_CHANS_ADC / I2S_CHANS_PER_FRAME)
+#endif
+
+/*
+ * Defines relating to the interface to external audio hardware i.e. DAC/ADC
+ */
+
 #define XUA_PCM_FORMAT_I2S      (0)
 #define XUA_PCM_FORMAT_TDM      (1)
 /**
@@ -103,7 +141,7 @@
         #error Bad value for XUA_PCM_FORMAT
     #endif
 #else
-    #define XUA_PCM_FORMAT      XUA_PCM_FORMAT_I2S
+    #define XUA_PCM_FORMAT        XUA_PCM_FORMAT_I2S
 #endif
 
 /**
@@ -114,89 +152,12 @@
  **/
 #ifndef I2S_CHANS_PER_FRAME
     #if (XUA_PCM_FORMAT == XUA_PCM_FORMAT_TDM)
-        #define I2S_CHANS_PER_FRAME (8)
+        #define I2S_CHANS_PER_FRAME 8
     #else
-        #define I2S_CHANS_PER_FRAME (2)
+        #define I2S_CHANS_PER_FRAME 2
     #endif
 #endif
 
-
-/**
- * @brief Number of IS2 channesl to DAC/CODEC. Must be a multiple of 2.
- *
- * Default: NONE (Must be defined by app)
- */
-#ifndef I2S_CHANS_DAC
-    #error I2S_CHANS_DAC not defined
-    #define I2S_CHANS_DAC 2          /* Define anyway for doxygen */
-#else
-#define I2S_WIRES_DAC            (I2S_CHANS_DAC / I2S_CHANS_PER_FRAME)
-#endif
-
-
-/**
- * @brief Number of I2S channels from ADC/CODEC. Must be a multiple of 2.
- *
- * Default: NONE (Must be defined by app)
- */
-#ifndef I2S_CHANS_ADC
-    #error I2S_CHANS_ADC not defined
-    #define I2S_CHANS_ADC 2      /* Define anyway for doxygen */
-#else
-#define I2S_WIRES_ADC           (I2S_CHANS_ADC / I2S_CHANS_PER_FRAME)
-#endif
-
-/**
- * @brief Ratio of the I2S sample rate to the USB Audio sample rate. Up and
- *        down-sampling will be enabled as necessary when the rates differ.
- *
- * Default: 1 i.e. I2S and USB Audio are running at the same sample rate.
- */
-#ifndef AUD_TO_USB_RATIO
-#define AUD_TO_USB_RATIO        (1)
-#else
-    #if (AUD_TO_USB_RATIO != 3) && (AUD_TO_USB_RATIO != 1)
-        #error Unsupported I2S to USB Audio sample rate ratio
-    #endif
-#endif
-
-/**
- * @brief Ratio of the I2S sample rate to the PDM microphone decimator sample
- *        rate.
- *
- * Default: 1 i.e. I2S and PDM microphone decimators are running at the same sample rate.
- */
-#ifndef AUD_TO_MICS_RATIO
-#define AUD_TO_MICS_RATIO       (1)
-#else
-    #if (AUD_TO_MICS_RATIO != 3) && (AUD_TO_MICS_RATIO != 1)
-        #error Unsupported I2S to PDM microphone decimator sample rate ratio
-    #endif
-#endif
-
-/**
- * @brief Only downsample one channel per input I2S frame.
- *
- * Default: 0 i.e. mono mode is disabled, all input channels will be downsampled.
- */
-#ifndef I2S_DOWNSAMPLE_MONO_IN
-#define I2S_DOWNSAMPLE_MONO_IN  (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_IN == 1)
-    #define I2S_DOWNSAMPLE_CHANS_IN (I2S_CHANS_ADC / 2)
-    #if ((I2S_DOWNSAMPLE_FACTOR_IN > 1) && (XUA_PCM_FORMAT == XUA_PCM_FORMAT_TDM))
-        #error Mono I2S input downsampling is not avaliable in TDM mode
-    #endif
-#else
-    #define I2S_DOWNSAMPLE_CHANS_IN I2S_CHANS_ADC
-#endif
-
 /**
  * @brief Number of bits per channel for I2S/TDM. Supported values: 16/32-bit.
  *
@@ -211,21 +172,82 @@
 #endif
 
 /**
- * @brief Max supported sample frequency for device (Hz). Default: 192000
+ * @brief Ratio of the I2S sample rate to the USB Audio sample rate. Up and
+ *        down-sampling will be enabled as necessary when the rates differ.
+ *
+ * Default: 1 i.e. I2S and USB Audio are running at the same sample rate.
+ */
+#ifndef AUD_TO_USB_RATIO
+#define AUD_TO_USB_RATIO (1)
+#else
+    #if (AUD_TO_USB_RATIO != 3) && (AUD_TO_USB_RATIO != 1)
+        #error Unsupported I2S to USB Audio sample rate ratio
+    #endif
+#endif
+
+/**
+ * @brief Ratio of the I2S sample rate to the PDM microphone decimator sample
+ *        rate.
+ *
+ * Default: 1 i.e. I2S and PDM microphone decimators are running at the same sample rate.
+ */
+#ifndef AUD_TO_MICS_RATIO
+#define AUD_TO_MICS_RATIO (1)
+#else
+    #if (AUD_TO_MICS_RATIO != 3) && (AUD_TO_MICS_RATIO != 1)
+        #error Unsupported I2S to PDM microphone decimator sample rate ratio
+    #endif
+#endif
+
+/**
+ * @brief Only downsample one channel per input I2S frame.
+ *
+ * Default: 0 i.e. mono mode is disabled, all input channels will be downsampled.
+ */
+#ifndef I2S_DOWNSAMPLE_MONO_IN
+#define I2S_DOWNSAMPLE_MONO_IN (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_IN == 1)
+    #define I2S_DOWNSAMPLE_CHANS_IN (I2S_CHANS_ADC / 2)
+    #if ((I2S_DOWNSAMPLE_FACTOR_IN > 1) && (XUA_PCM_FORMAT == XUA_PCM_FORMAT_TDM))
+        #error Mono I2S input downsampling is not avaliable in TDM mode
+    #endif
+#else
+    #define I2S_DOWNSAMPLE_CHANS_IN I2S_CHANS_ADC
+#endif
+
+/*
+ * Clocking related defines
+ */
+
+/**
+ * @brief Max supported sample frequency for device (Hz).
+ *
+ * Default: 192000Hz
  */
 #ifndef MAX_FREQ
 #define MAX_FREQ                 (192000)
 #endif
 
 /**
- * @brief Min supported sample frequency for device (Hz). Default 44100
+ * @brief Min supported sample frequency for device (Hz).
+ *
+ * Default: 44100Hz
  */
 #ifndef MIN_FREQ
 #define MIN_FREQ                 (44100)
 #endif
 
 /**
- * @brief Master clock defines for 44100 rates (in Hz). Default: NONE (Must be defined by app)
+ * @brief Master clock defines for 44100 rates (in Hz).
+ *
+ * Default: NONE (Must be defined by app)
  */
 #ifndef MCLK_441
     #error MCLK_441 not defined
@@ -233,13 +255,29 @@
 #endif
 
 /**
- * @brief Master clock defines for 48000 rates (in Hz). Default: NONE (Must be defined by app)
+ * @brief Master clock defines for 48000 rates (in Hz).
+ *
+ * Default: NONE (Must be defined by app)
  */
 #ifndef MCLK_48
     #error MCLK_48 not defined
     #define MCLK_48              (256 * 48000) /* Define anyway for doygen */
 #endif
 
+/**
+ * @brief Enable/disable the use of the secondary/application PLL for generating and recovering master-clocks.
+ *        Only available on xcore.ai devices.
+ *
+ * Default: Enabled (for xcore.ai devices)
+ */
+#ifndef XUA_USE_SW_PLL
+    #if defined(__XS3A__)
+        #define XUA_USE_SW_PLL        (1)
+    #else
+        #define XUA_USE_SW_PLL        (0)
+    #endif
+#endif
+
 /**
  * @brief Default device sample frequency. A safe default should be used.
  *
@@ -249,10 +287,25 @@
 #define DEFAULT_FREQ             (MIN_FREQ)
 #endif
 
-/* Audio Class Defines */
+#define DEFAULT_MCLK       (((DEFAULT_FREQ % 7350) == 0) ? MCLK_441 : MCLK_48)
 
 /**
- * @brief USB Audio Class Version.
+ * @brief Defines whether XMOS device runs as master (i.e. drives LR and Bit clocks)
+ *
+ * 0: XMOS is I2S master. 1: CODEC is I2s master.
+ *
+ * Default: 0 (XMOS is master)
+ */
+#ifndef CODEC_MASTER
+#define CODEC_MASTER       (0)
+#endif
+
+/*
+ * Audio Class defines
+ */
+
+/**
+ * @brief USB Audio Class Version
  *
  * Default: 2 (Audio Class version 2.0)
  */
@@ -261,9 +314,9 @@
 #endif
 
 /**
- * @brief Whether or not to fall back to Audio Class 1.0 in USB Full-speed.
+ * @brief Enable/disable fall back to Audio Class 1.0 in USB Full-speed.
  *
- * Default: 0 (Disabled i.e. do not fall back to UAC 1.0
+ * Default: Disabled
  */
 #ifndef AUDIO_CLASS_FALLBACK
 #define AUDIO_CLASS_FALLBACK     (0)
@@ -278,7 +331,7 @@
 #if (AUDIO_CLASS == 2)
     /* Whether to run in Audio Class 2.0 mode in USB Full-speed */
     #if !defined(FULL_SPEED_AUDIO_2) && (AUDIO_CLASS_FALLBACK == 0)
-        #define FULL_SPEED_AUDIO_2 (1)     /* Default to falling back to UAC2 */
+        #define FULL_SPEED_AUDIO_2    1     /* Default to falling back to UAC2 */
     #endif
 #endif
 
@@ -295,16 +348,17 @@
 #error AUDIO_CLASS set to 1 and FULL_SPEED_AUDIO_2 enabled!
 #endif
 
-
-/* Feature defines */
+/*
+ * Feature defines
+ */
 
 /**
- * @brief Number of PDM microphones in the design.
+ * @brief Disable USB functionalty just leaving AudioHub
  *
- * Default: None
+ * Default: Enabled
  */
-#ifndef XUA_NUM_PDM_MICS
-#define XUA_NUM_PDM_MICS        (0)
+#ifndef XUA_USB_EN
+#define XUA_USB_EN               (1)
 #endif
 
 /**
@@ -318,18 +372,14 @@
 #endif
 
 /**
- * @brief Size of a frame of microphone data samples.
- *
- * Default: 1
+ * @brief Size of a frame of microphone data samples. Default: 1
  */
 #ifndef XUA_MIC_FRAME_SIZE
 #define XUA_MIC_FRAME_SIZE      (1)
 #endif
 
 /**
- * @brief Enable MIDI functionality including buffering, descriptors etc.
- *
- * Default: 0 (Disabled)
+ * @brief Enable MIDI functionality including buffering, descriptors etc. Default: DISABLED
  */
 #ifndef MIDI
 #define MIDI                    (0)
@@ -350,7 +400,7 @@
  * @brief Enables SPDIF Tx. Default: 0 (Disabled)
  */
 #ifndef XUA_SPDIF_TX_EN
-#define XUA_SPDIF_TX_EN         (0)
+#define XUA_SPDIF_TX_EN          (0)
 #endif
 
 /**
@@ -359,16 +409,14 @@
  * Default: 0 (i.e. channels 0 & 1)
  * */
 #ifndef SPDIF_TX_INDEX
-#define SPDIF_TX_INDEX          (0)
+#define SPDIF_TX_INDEX        (0)
 #endif
 
 /**
- * @brief Enables ADAT Tx.
- *
- * Default: 0 (Disabled)
+ * @brief Enables ADAT Tx. Default: 0 (Disabled)
  */
 #ifndef XUA_ADAT_TX_EN
-#define XUA_ADAT_TX_EN          (0)
+#define XUA_ADAT_TX_EN           (0)
 #endif
 
 /**
@@ -377,25 +425,21 @@
  * Default: 0 (i.e. channels [0:7])
  * */
 #ifndef ADAT_TX_INDEX
-#define ADAT_TX_INDEX           (0)
+#define ADAT_TX_INDEX         (0)
 #endif
 
 /**
- * @brief Enables SPDIF Rx.
- *
- * Default: 0 (Disabled)
+ * @brief Enables SPDIF Rx. Default: 0 (Disabled)
  */
 #ifndef XUA_SPDIF_RX_EN
-#define XUA_SPDIF_RX_EN         (0)
+#define XUA_SPDIF_RX_EN       (0)
 #endif
 
 /**
- * @brief Enables ADAT Rx.
- *
- * Default: 0 (Disabled)
+ * @brief Enables ADAT Rx. Default: 0 (Disabled)
  */
 #ifndef XUA_ADAT_RX_EN
-#define XUA_ADAT_RX_EN          (0)
+#define XUA_ADAT_RX_EN        (0)
 #endif
 
 /**
@@ -450,20 +494,11 @@
  * Default: 1 (Enabled)
  */
 #if !defined(XUA_DFU_EN)
-#define XUA_DFU_EN              (1)
+#define XUA_DFU_EN                   (1)
 #elif defined(XUA_DFU_EN) && (XUA_DFU_EN == 0)
 #undef XUA_DFU_EN
 #endif
 
-/**
- * @brief Use a QSPI (rather than SPI) flash for DFU (and boot)
- *
- * Default: 1 (True i.e use QSPI flash)
-*/
-#if !defined(XUA_QUAD_SPI_FLASH)
-#define XUA_QUAD_SPI_FLASH      (1)
-#endif
-
 /**
  * @brief Enable HID playback controls functionality.
  *
@@ -472,7 +507,7 @@
  * Default 0 (Disabled)
  */
 #ifndef HID_CONTROLS
-#define HID_CONTROLS            (0)
+#define HID_CONTROLS       (0)
 #endif
 
 /**
@@ -488,12 +523,11 @@
  * You must also supply your own function to deal with the HID endpoint(s)
  * in this case.
  */
-#if( 0 < HID_CONTROLS )
+#if (HID_CONTROLS) || defined (__DOXYGEN__)
 #define   XUA_HID_ENABLED            (1)
 #define   XUA_OR_STATIC_HID_ENABLED  (1)
 #endif
 
-
 #if defined(__static_hid_report_h_exists__)
 #define   XUA_OR_STATIC_HID_ENABLED  (1)
 #endif
@@ -506,18 +540,7 @@
  * Default 0 (Disabled)
  */
 #ifndef HID_OUT_REQUIRED
-#define HID_OUT_REQUIRED        (0)
-#endif
-
-/**
- * @brief Defines whether XMOS device runs as master (i.e. drives LR and Bit clocks)
- *
- * 0: XMOS is I2S master. 1: CODEC is I2s master.
- *
- * Default: 0 (XMOS is master)
- */
-#ifndef CODEC_MASTER
-#define CODEC_MASTER            (0)
+#define HID_OUT_REQUIRED       (0)
 #endif
 
 /**
@@ -526,7 +549,7 @@
  * Default: ""
  */
 #ifndef SERIAL_STR
-#define SERIAL_STR              ""
+#define SERIAL_STR               ""
 #endif
 
 /**
@@ -535,7 +558,7 @@
  * Default: "XMOS"
  */
 #ifndef VENDOR_STR
-#define VENDOR_STR              "XMOS"
+#define VENDOR_STR               "XMOS"
 #endif
 
 /**
@@ -544,7 +567,7 @@
  * Default: 0x20B1 (XMOS)
  */
 #ifndef VENDOR_ID
-#define VENDOR_ID               (0x20B1)
+#define VENDOR_ID                (0x20B1)
 #endif
 
 /**
@@ -586,7 +609,7 @@
  */
 #if (AUDIO_CLASS == 1) || (AUDIO_CLASS_FALLBACK) || defined(__DOXYGEN__)
 #ifndef PID_AUDIO_1
-#define PID_AUDIO_1             (0x0003)
+#define PID_AUDIO_1              (0x0003)
 #endif
 #endif
 
@@ -596,28 +619,28 @@
  * Default: 0x0002
  */
 #ifndef PID_AUDIO_2
-#define PID_AUDIO_2             (0x0002)
+#define PID_AUDIO_2              (0x0002)
 #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_J
-#define BCD_DEVICE_J            (1)
+#define BCD_DEVICE_J             (1)
 #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            (2)
+#define BCD_DEVICE_M             (2)
 #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_N
-#define BCD_DEVICE_N            (0)
+#define BCD_DEVICE_N             (0)
 #endif
 
 /**
@@ -1021,7 +1044,7 @@
  * Default: 1 (Enabled)
  */
 #ifndef OUTPUT_VOLUME_CONTROL
-#define OUTPUT_VOLUME_CONTROL 	    (1)
+#define OUTPUT_VOLUME_CONTROL       (1)
 #endif
 
 /**
@@ -1030,7 +1053,7 @@
  * Default: 1 (Enabled)
  */
 #ifndef INPUT_VOLUME_CONTROL
-#define INPUT_VOLUME_CONTROL 	    (1)
+#define INPUT_VOLUME_CONTROL        (1)
 #endif
 
 /* Power */
@@ -1072,7 +1095,7 @@
  * Default: 0 (Disabled)
  */
 #ifndef MIXER
-#define MIXER                   (0)
+#define MIXER              (0)
 #endif
 
 /**
@@ -1082,11 +1105,11 @@
  */
 #if (MIXER)
     #ifndef MAX_MIX_COUNT
-    	#define MAX_MIX_COUNT   (8)
+        #define MAX_MIX_COUNT          (8)
     #endif
 #else
     #ifndef MAX_MIX_COUNT
-        #define MAX_MIX_COUNT   (0)
+        #define MAX_MIX_COUNT          (0)
     #endif
 #endif
 
@@ -1098,7 +1121,7 @@
  * Default: 18
  */
 #ifndef MIX_INPUTS
-    #define MIX_INPUTS          (18)
+    #define MIX_INPUTS                 (18)
 #endif
 
 /* Volume processing defines */
@@ -1110,7 +1133,7 @@
  * Default: 0x8100 (-127db)
  */
 #ifndef MIN_VOLUME
-#define MIN_VOLUME              (0x8100)
+#define MIN_VOLUME                     (0x8100)
 #endif
 
 
@@ -1120,7 +1143,7 @@
  * Default: 0x0000 (0db)
  */
 #ifndef MAX_VOLUME
-#define MAX_VOLUME              (0x0000)
+#define MAX_VOLUME                     (0x0000)
 #endif
 
 /**
@@ -1129,7 +1152,7 @@
  * Default: 0x100 (1db)
  */
 #ifndef VOLUME_RES
-#define VOLUME_RES              (0x100)
+#define VOLUME_RES                     (0x100)
 #endif
 
 /**
@@ -1139,7 +1162,7 @@
  * Default: 0x8100 (-127db)
  */
 #ifndef MIN_MIXER_VOLUME
-#define MIN_MIXER_VOLUME        (0x8100)
+#define MIN_MIXER_VOLUME             (0x8100)
 #endif
 
 /**
@@ -1148,7 +1171,7 @@
  * Default: 0x0000 (0db)
  */
 #ifndef MAX_MIXER_VOLUME
-#define MAX_MIXER_VOLUME        (0x0000)
+#define MAX_MIXER_VOLUME            (0x0000)
 #endif
 
 /**
@@ -1157,36 +1180,36 @@
 * Default: 0x100 (1db)
 */
 #ifndef VOLUME_RES_MIXER
-#define VOLUME_RES_MIXER        (0x100)
+#define VOLUME_RES_MIXER            (0x100)
 #endif
 
-/* Handle out volume control in the mixer - enabled by default if mixer enabled */
+/* Handle out volume control in the mixer - enabled by default */
 #ifndef OUT_VOLUME_IN_MIXER
 #if MIXER
-    #define OUT_VOLUME_IN_MIXER (1)
+    #define OUT_VOLUME_IN_MIXER     (1)
 #else
-    #define OUT_VOLUME_IN_MIXER (0)
+    #define OUT_VOLUME_IN_MIXER     (0)
 #endif
 #endif
 
 /* Apply out volume controls after the mix. Only relevant when OUT_VOLUME_IN_MIXER enabled. Enabled by default */
 #ifndef OUT_VOLUME_AFTER_MIX
-#define OUT_VOLUME_AFTER_MIX    (1)
+#define OUT_VOLUME_AFTER_MIX        (1)
 #endif
 
 /* Handle in volume control in the mixer - disabled by default */
 #ifndef IN_VOLUME_IN_MIXER
-#define IN_VOLUME_IN_MIXER      (0)
+#define IN_VOLUME_IN_MIXER          (0)
 #endif
 
 /* Apply in volume controls after the mix. Only relevant when IN_VOLUMNE_IN MIXER enabled. Enabled by default */
 #ifndef IN_VOLUME_AFTER_MIX
-#define IN_VOLUME_AFTER_MIX     (1)
+#define IN_VOLUME_AFTER_MIX         (1)
 #endif
 
 /* Always enable explicit feedback EP, even when input stream is present */
 #ifndef UAC_FORCE_FEEDBACK_EP
-#define UAC_FORCE_FEEDBACK_EP   (1)
+#define UAC_FORCE_FEEDBACK_EP       (1)
 #endif
 
 #if (defined(UAC_FORCE_FEEDBACK_EP) && UAC_FORCE_FEEDBACK_EP == 0)
@@ -1194,9 +1217,9 @@
 #endif
 
 /* Synchronisation defines */
-#define XUA_SYNCMODE_ASYNC      (1) // USB_ENDPOINT_SYNCTYPE_ASYNC
-#define XUA_SYNCMODE_ADAPT      (2) // USB_ENDPOINT_SYNCTYPE_ADAPT
-#define XUA_SYNCMODE_SYNC       (3) // USB_ENDPOINT_SYNCTYPE_SYNC
+#define XUA_SYNCMODE_ASYNC (1) // USB_ENDPOINT_SYNCTYPE_ASYNC
+#define XUA_SYNCMODE_ADAPT (2) // USB_ENDPOINT_SYNCTYPE_ADAPT
+#define XUA_SYNCMODE_SYNC  (3) // USB_ENDPOINT_SYNCTYPE_SYNC
 
 #ifndef XUA_SYNCMODE
 #define XUA_SYNCMODE XUA_SYNCMODE_ASYNC
@@ -1276,6 +1299,8 @@ enum USBEndpointNumber_Out
 #endif /* __ASSEMBLER__ */
 
 #define AUDIO_STOP_FOR_DFU                (0x12345678)
+#define AUDIO_START_FROM_DFU              (0x87654321)
+#define AUDIO_REBOOT_FROM_DFU             (0xa5a5a5a5)
 
 /* Result of db_to_mult(MAX_VOLUME, 8, 29) */
 #define MAX_VOLUME_MULT                   (0x20000000)

lib_xua/doc/rst/api_defines.rst

@@ -8,9 +8,9 @@ An application using the USB audio framework needs to have defines set for confi
 Defaults for these defines are found in ``xua_conf_default.h``.
 
 These defines should be over-ridden in an optional header file  ``xua_conf.h`` file or in the ``Makefile``
-for a relevant build configuration. 
+for a relevant build configuration.
 
-This section fully documents all of the settable defines and their default values (where appropriate).  
+This section fully documents all of the settable defines and their default values (where appropriate).
 
 Code Location (tile)
 --------------------
@@ -25,12 +25,12 @@ Code Location (tile)
 Channel Counts
 --------------
 
-.. doxygendefine:: NUM_USB_CHAN_OUT 
-.. doxygendefine:: NUM_USB_CHAN_IN 
-.. doxygendefine:: I2S_CHANS_DAC 
-.. doxygendefine:: I2S_CHANS_ADC 
+.. doxygendefine:: NUM_USB_CHAN_OUT
+.. doxygendefine:: NUM_USB_CHAN_IN
+.. doxygendefine:: I2S_CHANS_DAC
+.. doxygendefine:: I2S_CHANS_ADC
 
-Frequencies and Clocks 
+Frequencies and Clocks
 ----------------------
 
 .. doxygendefine:: MAX_FREQ
@@ -38,6 +38,7 @@ Frequencies and Clocks
 .. doxygendefine:: DEFAULT_FREQ
 .. doxygendefine:: MCLK_441
 .. doxygendefine:: MCLK_48
+.. doxygendefine:: XUA_USE_SW_PLL
 
 Audio Class
 -----------

lib_xua/doc/rst/api_user_functions.rst

@@ -1,74 +1,49 @@
-Required User Function Definitions
-==================================
+|newpage|
 
-The following functions need to be defined by an application using the XMOS USB Audio framework.
+User Function Definitions
+=========================
+
+The following functions can be defined by an application using `lib_xua`.
+
+.. note:: Default, empty, implementations of these functions are provided in `lib_xua`. These are marked
+   as weak symbols so the application can simply define its own version of them.
 
 External Audio Hardware Configuration Functions
 -----------------------------------------------
 
-.. c:function:: void AudioHwInit(chanend ?c_codec)
+The following functions can be optionally used by the design to configure external audio hardware.
+As a minimum, in most applications, it is expected that a implementation of `AudioHwConfig()` will need
+to be provided.
 
-   This function is called when the audio core starts after the
-   device boots up and should initialize the external audio harware e.g. clocking, DAC, ADC etc
+.. doxygenfunction:: AudioHwInit
+.. doxygenfunction:: AudioHwConfig
+.. doxygenfunction:: AudioHwConfig_Mute
+.. doxygenfunction:: AudioHwConfig_UnMute
 
-   :param c_codec: An optional chanend that was original passed into
-                   :c:func:`audio` that can be used to communicate 
-                   with other cores.
-  
 
-.. c:function:: void AudioHwConfig(unsigned samFreq, unsigned mclk, chanend ?c_codec, unsigned dsdMode, unsigned sampRes_DAC, unsigned sampRes_ADC)
-
-   This function is called when the audio core starts or changes
-   sample rate. It should configure the extenal audio hardware to run at the specified
-   sample rate given the supplied master clock frequency.
-
-   :param samFreq: The sample frequency in Hz that the hardware should be configured to (in Hz).
-                   
-   :param mclk: The master clock frequency that is required in Hz.
-   
-   :param c_codec: An optional chanend that was original passed into
-                   :c:func:`audio` that can be used to communicate 
-                   with other cores.
-
-   :param dsdMode: Signifies if the audio hardware should be configured for DSD operation
-
-   :param sampRes_DAC: The sample resolution of the DAC stream
-   
-   :param sampRes_ADC: The sample resolution of the ADC stream
-  
-
-Audio Streaming Functions
--------------------------
+Audio Stream Start/Stop Functions
+---------------------------------
 
 The following functions can be optionally used by the design. They can be useful for mute lines etc.
 
-.. c:function:: void AudioStreamStart(void)
+.. doxygenfunction:: UserAudioStreamStart
+.. doxygenfunction:: UserAudioStreamStop
+.. doxygenfunction:: UserAudioInputStreamStart
+.. doxygenfunction:: UserAudioInputStreamStop
+.. doxygenfunction:: UserAudioOutputStreamStart
+.. doxygenfunction:: UserAudioOutputStreamStop
 
-  This function is called when the audio stream from device to host
-  starts. 
-
-.. c:function:: void AudioStreamStop(void)
-
-  This function is called when the audio stream from device to host stops.
-
-Host Active
------------
+Host Active Functions
+---------------------
 
 The following function can be used to signal that the device is connected to a valid host.
 
-This is called on a change in state.
-
-.. c:function:: void AudioStreamStart(int active)
-
-   :param active: Indicates if the host is active or not. 1 for active else 0.
-
+.. doxygenfunction:: UserHostActive
 
 HID Controls
 ------------
 
 The following function is called when the device wishes to read physical user input (buttons etc).
+The function should write relevant HID bits into this array. The bit ordering and functionality is defined by the HID report descriptor used.
 
-.. c:function:: void UserReadHIDButtons(unsigned char hidData[])
-
-    :param hidData: The function should write relevant HID bits into this array. The bit ordering and functionality is defined by the HID report descriptor used.
-
+.. doxygenfunction:: UserHIDGetData

lib_xua/doc/rst/arch.rst

@@ -54,7 +54,9 @@ In addition :ref:`usb_audio_optional_components` shows optional components that
  * - Clockgen
    - Drives an external frequency generator (PLL) and manages
      changes between internal clocks and external clocks arising
-     from digital input.
+     from digital input. On xcore.ai Clockgen may also work in
+     conjunction with lib_sw_pll to produce a local clock from 
+     the XCORE which is locked to the incoming digital stream.
  * - MIDI
    - Outputs and inputs MIDI over a serial UART interface.
 

lib_xua/doc/rst/feat_spdif_rx.rst

@@ -29,10 +29,11 @@ The S/PDIF receiver should be called on the appropriate tile::
 With the steps above an S/PDIF stream can be captured by the xCORE. To be functionally useful the audio
 master clock must be able to synchronise to this external digital stream. Additionally, the host can be 
 notified regarding changes in the validity of this stream, it's frequency etc. To synchronise to external 
-streams the codebase assumes the use of an external Cirrus Logic CS2100 device.
+streams the codebase assumes the use of an external Cirrus Logic CS2100 device or lib_sw_pll on xcore.ai designs.
 
-The ``ClockGen()`` task from ``lib_xua`` provides the reference signal to the CS2100 device and also handles
-recording of clock validity etc. See :ref:`usb_audio_sec_clock_recovery` for full details regarding ``ClockGen()``.
+The ``ClockGen()`` task from ``lib_xua`` provides the reference signal to the CS2100 device or timing information
+to lib_sw_pll and also handles recording of clock validity etc.
+See :ref:`usb_audio_sec_clock_recovery` for full details regarding ``ClockGen()``.
 
 It also provides a small FIFO for S/PDIF samples before they are forwarded to the ``AudioHub`` core.
 As such it requires to be inserted in the communication path between the S/PDIF receiver and the 

lib_xua/doc/rst/hw.rst

@@ -52,11 +52,11 @@ Three methods of generating an audio master clock are provided on the board:
 
     * A Skyworks Si5351B PLL device. The Si5351 is an I2C configurable clock generator that is ideally suited for replacing crystals, crystal oscillators, VCXOs, phase-locked loops (PLLs), and fanout buffers.
 
-    * xCORE.ai devices are equipped with a secondary (or 'application') PLL which can be used to generate audio clocks
+    * xCORE.ai devices are equipped with a secondary (or 'application') PLL which can be used to generate fixed audio clocks or recover external clocks using lib_sw_pll.
 
 Selection between these methods is done via writing to bits 6 and 7 of PORT 8D on tile[0]. 
 
-Either the locally generated clock (from the PL611) or the recovered low jitter clock (from the CS2100) may be selected to clock the audio stages; the xCORE-200, the ADC/DAC and Digital output stages. Selection is controlled via an additional I/O, bit 5 of PORT 8C, see :ref:`hw_316_ctrlport`.
+Either the locally generated clock (from the PL611) or the recovered low jitter clock (from the CS2100) may be selected to clock the audio stages; the xcore.ai, the ADC/DAC and Digital output stages. Selection is controlled via an additional I/O, bit 5 of PORT 8C, see :ref:`hw_316_ctrlport`.
 
 .. _hw_316_ctrlport:
 

lib_xua/doc/rst/opt_spdif_rx.rst

@@ -33,8 +33,8 @@ This must be a 1-bit port, for example::
 
     <Port Location="XS1_PORT_1A"  Name="PORT_SPDIF_IN"/>
 
-When S/PDIF receive is enabled the codebase expects to drive a synchronisation signal to an external 
-Cirrus Logic CS2100 device for master-clock generation.
+When S/PDIF receive is enabled the codebase expects to either drive a synchronisation signal to an external 
+Cirrus Logic CS2100 device or use lib_swp_pll (xcore.ai only) for master-clock generation.
 
 The programmer should ensure the define in :ref:`opt_spdif_rx_ref_defines` is set appropriately.
 

lib_xua/doc/rst/opt_sync.rst

@@ -39,8 +39,11 @@ Setting the synchronisation mode of the device is done using the define in :ref:
      - USB synchronisation mode
      - ``XUA_SYNCMODE_ASYNC``
 
-When operating in synchronous mode an external Cirrus Logic CS2100 device is required for master clock 
-generation. The codebase expects to drive a synchronisation signal to this external device
+When operating in synchronous mode a local master clock must be generated that is synchronised to the incoming 
+SoF rate from USB. Either an external Cirrus Logic CS2100 device is required for this purpose 
+or, on xcore.ai devices, the on-chip application PLL may be used via lib_sw_pll.
+In the case of using the CS2100, the codebase expects to drive a synchronisation signal to this external device
+as a reference.
 
 The programmer should ensure the define in :ref:`opt_sync_ref_defines` is set appropriately.
 
@@ -56,8 +59,11 @@ The programmer should ensure the define in :ref:`opt_sync_ref_defines` is set ap
    * - ``PLL_REF_TILE``
      - Tile location of reference to CS2100 device
      - ``AUDIO_IO_TILE``
+   * - ``XUA_USE_SW_PLL``
+     - Whether or not to use sw_pll to recover the clock (xcore.ai only)
+     - 1 for xcore.ai targets. May be overridden to 0 in ``xua_conf.h``
 
-The codebase expects this reference signal port to be defined in the application XN file as ``PORT_PLL_REF``. 
+The codebase expects the CS2100 reference signal port to be defined in the application XN file as ``PORT_PLL_REF``. 
 This may be a port of any bit-width, however, connection to bit[0] is assumed::
 
     <Port Location="XS1_PORT_1A"  Name="PORT_PLL_REF"/>

lib_xua/doc/rst/sw_clocking.rst

@@ -15,32 +15,35 @@ the xCORE.
 
 Using an external PLL/Clock Multiplier allows an Asynchronous mode design to lock to an external
 clock source from a digital stream (e.g. S/PDIF or ADAT input).  The codebase supports the Cirrus
-Logic CS2100 device for this purpose. Other devices may be supported via code modification.
+Logic CS2100 device or use of lib_sw_pll (xcore.ai only) for this purpose. Other devices may be 
+supported via code modification.
 
-.. note::
-   
-    It is expected that in a future release  the secondary PLL in xCORE.ai devices, coupled with
-    associated software changes, will be capable of replacing the CS2100 part for most designs.
+The Clock Recovery core (Clock Gen) is responsible for either generating the reference frequency 
+to the CS2100 device or driving lib_sw_pll from time measurements based on the local master clock
+and the time of received samples. Clock Gen (via CS2100 or lib_sw_pll) generates the master clock 
+used over the whole design. This core also serves as a smaller buffer between ADAT and S/PDIF 
+receiving cores and the Audio Hub core.
 
-The Clock Recovery core (Clock Gen) is responsible for generating the reference frequency 
-to the CS2100 device. This, in turn, generates the master clock used over the whole design.
-This core also serves as a smaller buffer between ADAT and S/PDIF receiving cores and the Audio Hub
-core.
+When using lib_sw_pll (xcore.ai only) an further core is instantiated which performs the sigma-delta
+modulation of the xCORE PLL to ensure the lowest jitter over the audio band. See lib_sw_pll
+documentation for further details.
 
 When running in *Internal Clock* mode this core simply generates this clock using a local
 timer, based on the XMOS reference clock.
 
 When running in an external clock mode (i.e. S/PDIF Clock" or "ADAT Clock" mode) samples are 
-received from the S/PDIF and/or ADAT receive core.  The external frequency is calculated through 
-counting samples in a given period. The reference clock to the CS2100 is then generated based on
-the reception of these samples.
+received from the S/PDIF and/or ADAT receive core. The external frequency is calculated through 
+counting samples in a given period. Either the reference clock to the CS2100 is then generated based on
+the reception of these samples or the timing information is provided to lib_sw_pll to generate
+the phase-locked clock on-chip (xcore.ai only).
 
 If an external stream becomes invalid, the *Internal Clock* timer event will fire to ensure that 
 valid master clock generation continues regardless of cable unplugs etc. Efforts are made to 
 ensure the transition between these clocks are relatively seamless. Additionally efforts are also
-made to try and keep the jitter on the reference clock as low as possibly, regardless of activity
+made to try and keep the jitter on the reference clock as low as possible, regardless of activity
 level of the Clock Gen core. The is achieved though the use of port times to schedule pin toggling
-rather than directly outputting to the port.
+rather than directly outputting to the port in the case of using the CS2100. For lib_sw_pll cases the
+last setting is kept for the sigma-delta modulator ensuring clock continuity.
 
 The Clock Gen core gets clock selection Get/Set commands from Endpoint 0 via the ``c_clk_ctl`` 
 channel.  This core also records the validity of external clocks, which is also queried 

lib_xua/doc/rst/xdoc.conf

@@ -1,5 +1,5 @@
 XMOSNEWSTYLE = 2
-DOXYGEN_DIRS=../../api
+DOXYGEN_DIRS=../../api ../../src/core/user/audiostream ../../src/core/user/hostactive ../../src/core/user/hid ../../src/core/user/audiohw
 SOURCE_INCLUDE_DIRS=../../../lib_xua
 SPHINX_MASTER_DOC=lib_xua
 

lib_xua/host/xmosdfu/Makefile.Pi

@@ -6,4 +6,4 @@
 
 xmosdfu: xmosdfu.cpp
 	mkdir -p bin
-	g++ -D_GNU_SOURCE -Wall -g -o bin/xmosdfu -Ilibusb/Rasp -lusb-1.0 -x c xmosdfu.cpp -std=c99
+	g++ -D_GNU_SOURCE -Wall -g -o bin/xmosdfu -Ilibusb/Rasp -x c xmosdfu.cpp -std=c99 -lusb-1.0

lib_xua/lib_build_info.cmake

@@ -1,5 +1,5 @@
 set(LIB_NAME lib_xua)
-set(LIB_VERSION 3.5.1)
+set(LIB_VERSION 4.0.0)
 set(LIB_INCLUDES api
                  src/core
                  src/core/audiohub
@@ -14,18 +14,20 @@ set(LIB_INCLUDES api
                  src/core/support
                  src/core/user
                  src/core/user/audiostream
+                 src/core/user/audiohw
                  src/core/user/hid
                  src/core/user/hostactive
                  src/hid
                  src/midi)
 set(LIB_OPTIONAL_HEADERS xua_conf.h static_hid_report.h)
-set(LIB_DEPENDENT_MODULES "lib_locks"
-                          "lib_logging"
-                          "lib_mic_array(feature/xcommon_cmake)"
-                          "lib_spdif"
-                          "lib_xassert"
-                          "lib_xud"
-                          "lib_adat")
+set(LIB_DEPENDENT_MODULES "lib_adat(1.1.0)"
+                          "lib_locks(2.2.0)"
+                          "lib_logging(3.2.0)"
+                          "lib_mic_array(4.6.0)"
+                          "lib_spdif(6.1.0)"
+                          "lib_sw_pll(2.1.0)"
+                          "lib_xassert(4.2.0)"
+                          "lib_xud(2.3.1)")
 
 set(LIB_COMPILER_FLAGS -O3 -DREF_CLK_FREQ=100 -fasm-linenum -fcomment-asm)
 

lib_xua/module_build_info

@@ -1,4 +1,4 @@
-VERSION = 3.5.1
+VERSION = 4.0.0
 
 DEBUG ?= 0
 
@@ -8,13 +8,14 @@ else
 DEBUG_FLAGS = -DXASSERT_ENABLE_ASSERTIONS=0 -DXASSERT_ENABLE_DEBUG=0 -DXASSERT_ENABLE_LINE_NUMBERS=0
 endif
 
-DEPENDENT_MODULES = lib_locks(>=2.1.0) \
-                    lib_logging(>=3.1.1) \
-                    lib_mic_array(>=4.5.0) \
-                    lib_spdif(>=5.0.0) \
-                    lib_xassert(>=4.1.0) \
-                    lib_xud(>=2.2.3) \
-                    lib_adat(>=1.0.0)
+DEPENDENT_MODULES = lib_adat(>=1.1.0) \
+                    lib_locks(>=2.2.0) \
+                    lib_logging(>=3.2.0) \
+                    lib_mic_array(>=4.6.0) \
+                    lib_spdif(>=6.1.0) \
+                    lib_sw_pll(>=2.1.0) \
+                    lib_xassert(>=4.2.0) \
+                    lib_xud(>=2.3.1)
 
 MODULE_XCC_FLAGS = $(XCC_FLAGS) \
                    -O3 \
@@ -53,6 +54,7 @@ INCLUDE_DIRS = $(EXPORT_INCLUDE_DIRS) \
                src/core/support \
                src/core/user \
                src/core/user/audiostream \
+               src/core/user/audiohw \
                src/core/user/hid \
                src/core/user/hostactive \
                src/hid \
@@ -69,6 +71,7 @@ SOURCE_DIRS = src/core \
               src/core/ports \
               src/core/support \
               src/core/user/audiostream \
+              src/core/user/audiohw \
               src/core/user/hostactive \
               src/core/xuduser \
               src/dfu \

lib_xua/src/core/audiohub/audiohub_adat.h

@@ -1,13 +1,11 @@
-// Copyright 2018-2022 XMOS LIMITED.
+// Copyright 2018-2023 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
-
 unsigned adatCounter = 0;
 unsigned adatSamples[8];
 
 #pragma unsafe arrays
 static inline void TransferAdatTxSamples(chanend c_adat_out, const unsigned samplesFromHost[], int smux, int handshake)
 {
-
     /* Do some re-arranging for SMUX.. */
     unsafe
     {
@@ -29,7 +27,6 @@ static inline void TransferAdatTxSamples(chanend c_adat_out, const unsigned samp
 
     if(adatCounter == smux)
     {
-
 #ifdef ADAT_TX_USE_SHARED_BUFF
         unsafe
         {

lib_xua/src/core/audiohub/xua_audiohub.xc

@@ -1,4 +1,4 @@
-// Copyright 2011-2023 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 /**
  * @file xua_audiohub.xc
@@ -20,6 +20,7 @@
 
 #include "xua.h"
 
+#include "audiohw.h"
 #include "audioports.h"
 #include "mic_array_conf.h"
 #if (XUA_SPDIF_TX_EN)
@@ -241,12 +242,12 @@ unsigned static AudioHub_MainLoop(chanend ?c_out, chanend ?c_spd_out
     }
 #endif // ((DEBUG_MIC_ARRAY == 1) && (XUA_NUM_PDM_MICS > 0))
 
-    UserBufferManagementInit();
+    UserBufferManagementInit(curSamFreq);
 
     unsigned command = DoSampleTransfer(c_out, readBuffNo, underflowWord);
 
     // Reinitialise user state before entering the main loop
-    UserBufferManagementInit();
+    UserBufferManagementInit(curSamFreq);
 
 #if (XUA_ADAT_TX_EN)
     unsafe{
@@ -641,6 +642,9 @@ void XUA_AudioHub(chanend ?c_aud, clock ?clk_audio_mclk, clock ?clk_audio_bclk,
 #if (XUA_ADAT_RX_EN || XUA_SPDIF_RX_EN)
     , chanend c_dig_rx
 #endif
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
+    , chanend c_audio_rate_change
+#endif
 #if (XUD_TILE != 0) && (AUDIO_IO_TILE == 0) && (XUA_DFU_EN == 1)
     , server interface i_dfu ?dfuInterface
 #endif
@@ -666,7 +670,6 @@ void XUA_AudioHub(chanend ?c_aud, clock ?clk_audio_mclk, clock ?clk_audio_bclk,
     /* Note, marked unsafe since other cores may be using this mclk port */
     configure_clock_src(clk_audio_mclk, p_mclk_in);
 
-    start_clock(clk_audio_mclk);
 
 #if (DSD_CHANS_DAC > 0)
     /* Make sure the DSD ports are on and buffered - just in case they are not shared with I2S */
@@ -678,15 +681,12 @@ void XUA_AudioHub(chanend ?c_aud, clock ?clk_audio_mclk, clock ?clk_audio_bclk,
 #endif
 
 #if (XUA_ADAT_TX_EN)
-    /* Share SPDIF clk blk */
-    configure_clock_src(clk_mst_spd, p_mclk_in);
-    configure_out_port_no_ready(p_adat_tx, clk_mst_spd, 0);
-    set_clock_fall_delay(clk_mst_spd, 7);
-#if (XUA_SPDIF_TX_EN == 0)
-    start_clock(clk_mst_spd);
-#endif
+    configure_out_port_no_ready(p_adat_tx, clk_audio_mclk, 0);
+    set_clock_fall_delay(clk_audio_mclk, 7);
 #endif
 
+    start_clock(clk_audio_mclk);
+
     /* Perform required CODEC/ADC/DAC initialisation */
     AudioHwInit();
 
@@ -799,7 +799,23 @@ void XUA_AudioHub(chanend ?c_aud, clock ?clk_audio_mclk, clock ?clk_audio_bclk,
             }
 #endif
             /* Configure Clocking/CODEC/DAC/ADC for SampleFreq/MClk */
+
+            /* User should mute audio hardware */
+            AudioHwConfig_Mute();
+
+            /* User code should configure audio harware for SampleFreq/MClk etc */
             AudioHwConfig(curFreq, mClk, dsdMode, curSamRes_DAC, curSamRes_ADC);
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
+            /* Notify clockgen of new mCLk */
+            c_audio_rate_change <: mClk;
+            c_audio_rate_change <: curFreq;
+
+            /* Wait for ACK back from clockgen or ep_buffer to signal clocks all good */
+            c_audio_rate_change :> int _;
+#endif
+
+            /* User should unmute audio hardware */
+            AudioHwConfig_UnMute();
         }
 
         if(!firstRun)

lib_xua/src/core/audiohub/xua_buffman_default.c

@@ -1,11 +1,11 @@
-// Copyright 2016-2021 XMOS LIMITED.
+// Copyright 2016-2023 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
 #include "xccompat.h"
 #include "xua_audiohub.h"
 
 /* Default implementation for UserBufferManagementInit() */
-void __attribute__ ((weak)) UserBufferManagementInit()
+void __attribute__ ((weak)) UserBufferManagementInit(unsigned sampFreq)
 {
     /* Do nothing */
 }

lib_xua/src/core/buffer/ep/ep_buffer.xc

@@ -1,4 +1,4 @@
-// Copyright 2011-2023 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 #include "xua.h"
 #if XUA_USB_EN
@@ -105,7 +105,12 @@ void XUA_Buffer(
 #endif
     , chanend c_aud
 #if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
+    , chanend c_audio_rate_change
+    #if(XUA_USE_SW_PLL)
+    , chanend c_sw_pll
+    #else
     , client interface pll_ref_if i_pll_ref
+    #endif
 #endif
 )
 {
@@ -141,7 +146,12 @@ void XUA_Buffer(
                 , c_buff_ctrl
 #endif
 #if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
-                , i_pll_ref
+                , c_audio_rate_change
+    #if(XUA_USE_SW_PLL)
+               , c_sw_pll
+    #else
+               , i_pll_ref
+    #endif
 #endif
             );
 
@@ -190,8 +200,13 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
 #ifdef CHAN_BUFF_CTRL
     , chanend c_buff_ctrl
 #endif
-#if XUA_SYNCMODE == XUA_SYNCMODE_SYNC
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
+    , chanend c_audio_rate_change
+    #if (XUA_USE_SW_PLL)
+    , chanend c_sw_pll
+    #else
     , client interface pll_ref_if i_pll_ref
+    #endif
 #endif
     )
 {
@@ -247,7 +262,7 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
 #if (NUM_USB_CHAN_IN > 0)
     unsigned bufferIn = 1;
 #endif
-    unsigned sofCount = 0;
+    int sofCount = 0;
 
     unsigned mod_from_last_time = 0;
 #ifdef FB_TOLERANCE_TEST
@@ -294,7 +309,6 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
     unsigned iap_ea_native_interface_alt_setting = 0;
     unsigned iap_ea_native_control_to_send = 0;
     unsigned iap_ea_native_incoming = 0;
-
 #endif
 #endif
 
@@ -357,6 +371,24 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
 #ifndef LOCAL_CLOCK_MARGIN
 #define LOCAL_CLOCK_MARGIN          (1000)
 #endif
+
+#if (XUA_USE_SW_PLL)
+    /* Setup the phase frequency detector */
+    const unsigned controller_rate_hz = 100;
+    const unsigned pfd_ppm_max = 2000;                      /* PPM range before we assume unlocked */
+
+    sw_pll_pfd_state_t sw_pll_pfd;
+    sw_pll_pfd_init(&sw_pll_pfd,
+                    1,                                      /* How often the PFD is invoked per call */
+                    masterClockFreq / controller_rate_hz,   /* pll ratio integer */
+                    0,                                      /* Assume precise timing of sampling */
+                    pfd_ppm_max);
+    outuint(c_sw_pll, masterClockFreq);
+    outct(c_sw_pll, XS1_CT_END);
+    inuint(c_sw_pll); /* receive ACK */
+    inct(c_sw_pll);
+
+#else /* XUA_USE_SW_PLL */
     timer t_sofCheck;
     unsigned timeLastEdge;
     unsigned timeNextEdge;
@@ -365,6 +397,8 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
     i_pll_ref.toggle();
 #endif
 
+#endif /* (XUA_SYNCMODE == XUA_SYNCMODE_SYNC) */
+
     while(1)
     {
         XUD_Result_t result;
@@ -427,7 +461,7 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
                             /* Reset FB */
                             /* Note, Endpoint 0 will hold off host for a sufficient period to allow our feedback
                              * to stabilise (i.e. sofCount == 128 to fire) */
-                            sofCount = 1;
+                            sofCount = 0;
                             clocks = 0;
                             clockcounter = 0;
                             mod_from_last_time = 0;
@@ -450,7 +484,7 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
                                 masterClockFreq = MCLK_441;
                             }
                         }
-#endif
+#endif /* (MAX_FREQ != MIN_FREQ) */
                         /* Ideally we want to wait for handshake (and pass back up) here.  But we cannot keep this
                         * core locked, it must stay responsive to packets (MIDI etc) and SOFs.  So, set a flag and check for
                         * handshake elsewhere */
@@ -502,13 +536,13 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
                     }
 #endif
                     /* Pass on sample freq change to decouple() via global flag (saves a chanend) */
-                    /* Note: freqChange flags now used to communicate other commands also */
+                    /* Note: freqChange_flag now used to communicate other commands also */
                     SET_SHARED_GLOBAL0(g_freqChange, cmd);                /* Set command */
                     SET_SHARED_GLOBAL(g_freqChange_flag, cmd);  /* Set Flag */
                 }
                 break;
             }
-#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC) && (!XUA_USE_SW_PLL)
             case t_sofCheck when timerafter(timeNextEdge) :> void:
                 i_pll_ref.toggle();
                 timeLastEdge = timeNextEdge;
@@ -523,28 +557,61 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
             /* SOF notification from XUD_Manager() */
             case inuint_byref(c_sof, u_tmp):
 #if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
-                /* This really could (should) be done in decouple. However, for a quick demo this is okay
-                 * Decouple expects a 16:16 number in fixed point stored in the global g_speed */
                 unsigned usbSpeed;
-                int framesPerSec;
                 GET_SHARED_GLOBAL(usbSpeed, g_curUsbSpeed);
                 static int sofCount = 0;
+#if (XUA_USE_SW_PLL)
+                /* Run PFD and sw_pll controller at 100Hz */
+                const int sofFreqDivider = (usbSpeed == XUD_SPEED_HS) ? (8000 / controller_rate_hz) : (1000 / controller_rate_hz);
+#else /* (XUA_USE_SW_PLL) */
+                /* 1000 toggles per second for CS2100 reference -> 500 Hz */
+                const int toggleRateHz = 1000;
+                const int sofFreqDivider = (usbSpeed == XUD_SPEED_HS) ? (8000 / toggleRateHz) : (1000 / toggleRateHz);
+#endif /* (XUA_USE_SW_PLL) */
 
-                framesPerSec = (usbSpeed == XUD_SPEED_HS) ? 8000 : 1000;
-
-                clocks =  ((int64_t) sampleFreq << 16) / framesPerSec;
-
-                asm volatile("stw %0, dp[g_speed]"::"r"(clocks));
-
-                sofCount += 1000;
-                if (sofCount == framesPerSec)
+                sofCount++;
+                if (sofCount == sofFreqDivider)
                 {
+#if (XUA_USE_SW_PLL)
+                    /* Grab port timer count, run through PFD and send to sw_pll */
+                    unsigned short mclk_pt;
+                    asm volatile("getts %0, res[%1]" : "=r" (mclk_pt) : "r" (p_off_mclk));
+
+                    uint8_t first_loop = 0;
+                    unsafe{
+                        sw_pll_calc_error_from_port_timers(&sw_pll_pfd, &first_loop, mclk_pt, 0);
+                    }
+
+                    int error = 0;
+                    if(!first_loop)
+                    {
+                        error = sw_pll_pfd.mclk_diff;
+                    }
+                    sw_pll_pfd.mclk_pt_last = mclk_pt;
+
+                    /* Send error to sw_pll */
+                    outuint(c_sw_pll, error);
+                    outct(c_sw_pll, XS1_CT_END);
+
+#else /* (XUA_USE_SW_PLL) */
+                    /* Do toggle for CS2100 reference clock */
                     /* Port is accessed via interface to allow flexibilty with location */
                     i_pll_ref.toggle();
                     t_sofCheck :> timeLastEdge;
-                    sofCount = 0;
                     timeNextEdge = timeLastEdge + LOCAL_CLOCK_INCREMENT + LOCAL_CLOCK_MARGIN;
+#endif /* (XUA_USE_SW_PLL) */
+                    sofCount = 0;
                 }
+
+                /* This really could (should) be done in decouple. However, for a quick demo this is okay
+                 * Decouple expects a 16:16 number in fixed point stored in the global g_speed */
+
+                const int framesPerSec = (usbSpeed == XUD_SPEED_HS) ? 8000 : 1000;
+
+                clocks = ((int64_t) sampleFreq << 16) / framesPerSec;
+                asm volatile("stw %0, dp[g_speed]"::"r"(clocks));
+
+
 #elif (XUA_SYNCMODE == XUA_SYNCMODE_ASYNC)
 
                 /* NOTE our feedback will be wrong for a couple of SOF's after a SF change due to
@@ -646,7 +713,6 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
                         clockcounter = 0;
                     }
 #else
-
                     /* Assuming 48kHz from a 24.576 master clock (0.0407uS period)
                      * MCLK ticks per SOF = 125uS / 0.0407 = 3072 MCLK ticks per SOF.
                      * expected Feedback is 48000/8000 = 6 samples. so 0x60000 in 16:16 format.
@@ -691,7 +757,6 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
                             clocks < (expected_fb + FB_TOLERANCE))
 #endif
                         {
-                            int usb_speed;
                             asm volatile("stw %0, dp[g_speed]"::"r"(clocks));   // g_speed = clocks
 
                             GET_SHARED_GLOBAL(usb_speed, g_curUsbSpeed);
@@ -897,8 +962,8 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
 #endif
 #endif
 
-#if XUA_HID_ENABLED
-                /* HID Report Data */
+#if (XUA_HID_ENABLED)
+            /* HID Report Data */
             case XUD_SetData_Select(c_hid, ep_hid, result):
                 hid_ready_flag = 0U;
                 unsigned reportTime;
@@ -911,7 +976,7 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
 #endif
 
 #ifdef MIDI
-                /* Received word from MIDI thread - Check for ACK or Data */
+            /* Received word from MIDI thread - Check for ACK or Data */
             case midi_get_ack_or_data(c_midi, is_ack, datum):
                 if (is_ack)
                 {
@@ -963,6 +1028,33 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
                 break;
 #endif  /* ifdef MIDI */
 
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
+            case c_audio_rate_change :> u_tmp:
+                unsigned selected_mclk_rate = u_tmp;
+                c_audio_rate_change :> u_tmp;                       /* Sample rate is discarded as only care about mclk */
+#if (XUA_USE_SW_PLL)
+                sw_pll_pfd_init(&sw_pll_pfd,
+                                1,                                          /* How often the PFD is invoked per call */
+                                selected_mclk_rate / controller_rate_hz,    /* pll muliplication ratio integer */
+                                0,                                          /* Assume precise timing of sampling */
+                                pfd_ppm_max);
+                restart_sigma_delta(c_sw_pll, selected_mclk_rate);
+                                                                    /* Delay ACK until sw_pll says it is ready */
+#else
+                c_audio_rate_change <: 0;                           /* ACK back to audio to release I2S immediately */
+#endif /* XUA_USE_SW_PLL */
+                break;
+
+#if (XUA_USE_SW_PLL)
+            /* This is fired when sw_pll has completed initialising a new mclk_rate */
+            case inuint_byref(c_sw_pll, u_tmp):
+                inct(c_sw_pll);
+                c_audio_rate_change <: 0;     /* ACK back to audio to release */
+
+                break;
+#endif /* (XUA_USE_SW_PLL) */
+#endif /* (XUA_SYNCMODE == XUA_SYNCMODE_SYNC) */
+
 #ifdef IAP
             /* Received word from iap thread - Check for ACK or Data */
             case iap_get_ack_or_reset_or_data(c_iap, is_ack_iap, is_reset, datum_iap):

lib_xua/src/core/clocking/clockgen.xc

@@ -1,4 +1,4 @@
-// Copyright 2011-2023 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 #include <xs1.h>
 #include <assert.h>
@@ -25,11 +25,11 @@ unsigned g_digData[10];
 
 typedef struct
  {
-    int receivedSamples;
-    int samples;
-    int savedSamples;
-    int lastDiff;
-    unsigned identicaldiffs;
+    int receivedSamples;        /* Uses by clockgen to count number of dig rx samples to ascertain clock specs */
+    int samples;                /* Raw sample count - rolling int and never reset */
+    int savedSamples;           /* Used by validSamples() to store state of last raw sample count */
+    int lastDiff;               /* Used by validSamples() to store state of last sample count diff */
+    unsigned identicaldiffs;    /* Used by validSamples() to store state of number of identical diffs */
     int samplesPerTick;
 } Counter;
 
@@ -38,6 +38,7 @@ static int clockValid[NUM_CLOCKS];                          /* Store current val
 static int clockInt[NUM_CLOCKS];                            /* Interupt flag for clocks */
 static int clockId[NUM_CLOCKS];
 
+
 [[distributable]]
 void PllRefPinTask(server interface pll_ref_if i_pll_ref, out port p_pll_ref)
 {
@@ -87,27 +88,10 @@ static int abs(int x)
     return x;
 }
 
-static int channelContainsControlToken(chanend x)
-{
-    unsigned char tmpc;
-
-    select
-    {
-        case inct_byref(x, tmpc):
-            return 1;
-        default:
-            return 0;
-    }
-}
-
 static void outInterrupt(chanend c_interruptControl, int value)
 {
-    /* Non-blocking check for control token */
-    //if (channelContainsControlToken(c_interruptControl))
-    {
-        outuint(c_interruptControl, value);
-        outct(c_interruptControl, XS1_CT_END);
-    }
+    outuint(c_interruptControl, value);
+    outct(c_interruptControl, XS1_CT_END);
 }
 #endif
 
@@ -206,7 +190,7 @@ static inline int validSamples(Counter &counter, int clockIndex)
             }
         }
     }
-    else
+    else /* No valid frequency found - reset state */
     {
         counter.identicaldiffs = 0;
         counter.lastDiff = diff;
@@ -215,6 +199,13 @@ static inline int validSamples(Counter &counter, int clockIndex)
 }
 #endif
 
+#if XUA_USE_SW_PLL
+unsafe
+{
+    unsigned * unsafe selected_mclk_rate_ptr = NULL;
+}
+#endif
+
 #ifdef LEVEL_METER_LEDS
 void VendorLedRefresh(unsigned levelData[]);
 unsigned g_inputLevelData[NUM_USB_CHAN_IN];
@@ -225,14 +216,25 @@ extern int samples_to_host_inputs_buff[NUM_USB_CHAN_IN];
 int VendorAudCoreReqs(unsigned cmd, chanend c);
 
 #pragma unsafe arrays
-void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interface pll_ref_if i_pll_ref, chanend c_dig_rx, chanend c_clk_ctl, chanend c_clk_int)
+void clockGen ( streaming chanend ?c_spdif_rx,
+                chanend ?c_adat_rx,
+                client interface pll_ref_if i_pll_ref,
+                chanend c_dig_rx,
+                chanend c_clk_ctl,
+                chanend c_clk_int,
+                chanend c_audio_rate_change
+#if XUA_USE_SW_PLL
+                , port p_for_mclk_count_aud
+                , chanend c_sw_pll
+#endif
+)
 {
     timer t_local;
     unsigned timeNextEdge, timeLastEdge, timeNextClockDetection;
     unsigned clkMode = CLOCK_INTERNAL;              /* Current clocking mode in operation */
     unsigned tmp;
 
-    /* start in no-SMUX (8-channel) mode */
+    /* Start in no-SMUX (8-channel) mode */
     int smux = 0;
 
 #ifdef LEVEL_METER_LEDS
@@ -242,6 +244,16 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
 
 #if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
     timer t_external;
+    unsigned selected_mclk_rate = MCLK_48; // Assume 24.576MHz initial clock
+    unsigned selected_sample_rate = 0;
+#if XUA_USE_SW_PLL
+
+    unsigned mclks_per_sample = 0;
+    unsigned short mclk_time_stamp = 0;
+
+    /* Get MCLK count */
+    asm volatile(" getts %0, res[%1]" : "=r" (mclk_time_stamp) : "r" (p_for_mclk_count_aud));
+#endif
 #endif
 
 #if (XUA_SPDIF_RX_EN)
@@ -281,21 +293,21 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
     }
 
     /* Init clock unit state */
+    clockFreq[CLOCK_INTERNAL] = 0;
+    clockId[CLOCK_INTERNAL] = ID_CLKSRC_INT;
+    clockValid[CLOCK_INTERNAL] = 0;
+    clockInt[CLOCK_INTERNAL] = 0;
 #if (XUA_SPDIF_RX_EN)
-    clockFreq[CLOCK_SPDIF_INDEX] = 0;
-    clockValid[CLOCK_SPDIF_INDEX] = 0;
-    clockInt[CLOCK_SPDIF_INDEX] = 0;
-    clockId[CLOCK_SPDIF_INDEX] = ID_CLKSRC_SPDIF;
+    clockFreq[CLOCK_SPDIF] = 0;
+    clockValid[CLOCK_SPDIF] = 0;
+    clockInt[CLOCK_SPDIF] = 0;
+    clockId[CLOCK_SPDIF] = ID_CLKSRC_SPDIF;
 #endif
-    clockFreq[CLOCK_INTERNAL_INDEX] = 0;
-    clockId[CLOCK_INTERNAL_INDEX] = ID_CLKSRC_INT;
-    clockValid[CLOCK_INTERNAL_INDEX] = 0;
-    clockInt[CLOCK_INTERNAL_INDEX] = 0;
 #if (XUA_ADAT_RX_EN)
-    clockFreq[CLOCK_ADAT_INDEX] = 0;
-    clockInt[CLOCK_ADAT_INDEX] = 0;
-    clockValid[CLOCK_ADAT_INDEX] = 0;
-    clockId[CLOCK_ADAT_INDEX] = ID_CLKSRC_ADAT;
+    clockFreq[CLOCK_ADAT] = 0;
+    clockInt[CLOCK_ADAT] = 0;
+    clockValid[CLOCK_ADAT] = 0;
+    clockId[CLOCK_ADAT] = ID_CLKSRC_ADAT;
 #endif
 #if (XUA_SPDIF_RX_EN)
     spdifCounters.receivedSamples = 0;
@@ -333,6 +345,12 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
     /* Initial ref clock output and get timestamp */
     i_pll_ref.init();
 
+#if ((XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN) && XUA_USE_SW_PLL)
+    int reset_sw_pll_pfd = 1;
+    int require_ack_to_audio = 0;
+    restart_sigma_delta(c_sw_pll, MCLK_48); /* default to 48kHz - this will be reset shortly when host selects rate */
+#endif
+
     while(1)
     {
         select
@@ -389,13 +407,9 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
 
                     case SET_SEL:
                         /* Update clock mode */
-                        tmp = inuint(c_clk_ctl);
+                        clkMode = inuint(c_clk_ctl);
                         chkct(c_clk_ctl, XS1_CT_END);
 
-                        if(tmp!=0)
-                        {
-                            clkMode = tmp;
-                        }
 #ifdef CLOCK_VALIDITY_CALL
                         switch(clkMode)
                         {
@@ -404,12 +418,12 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                                 break;
 #if (XUA_ADAT_RX_EN)
                             case CLOCK_ADAT:
-                                VendorClockValidity(clockValid[CLOCK_ADAT_INDEX]);
+                                VendorClockValidity(clockValid[CLOCK_ADAT]);
                                 break;
 #endif
 #if (XUA_SPDIF_RX_EN)
                             case CLOCK_SPDIF:
-                                VendorClockValidity(clockValid[CLOCK_SPDIF_INDEX]);
+                                VendorClockValidity(clockValid[CLOCK_SPDIF]);
                                 break;
 #endif
                         }
@@ -454,9 +468,12 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
             /* Generate local clock from timer */
             case t_local when timerafter(timeNextEdge) :> void:
 
+#if XUA_USE_SW_PLL
+                /* Do nothing - hold the most recent sw_pll setting */
+#else
                 /* Setup next local clock edge */
                 i_pll_ref.toggle_timed(0);
-
+#endif
                 /* Record time of edge */
                 timeLastEdge = timeNextEdge;
 
@@ -483,32 +500,63 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
 #endif
                 break;
 
-
 #if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
             case t_external when timerafter(timeNextClockDetection) :> void:
-
-                timeNextClockDetection += (LOCAL_CLOCK_INCREMENT);
+                {
+                    int valid;
+                    timeNextClockDetection += (LOCAL_CLOCK_INCREMENT);
 #if (XUA_SPDIF_RX_EN)
-                tmp = spdifCounters.samplesPerTick;
-
-                /* Returns 1 if valid clock found */
-                tmp = validSamples(spdifCounters, CLOCK_SPDIF_INDEX);
-                setClockValidity(c_clk_int, CLOCK_SPDIF_INDEX, tmp, clkMode);
+                    /* Returns 1 if valid clock found */
+                    valid = validSamples(spdifCounters, CLOCK_SPDIF);
+                    setClockValidity(c_clk_int, CLOCK_SPDIF, valid, clkMode);
 #endif
 #if (XUA_ADAT_RX_EN)
-                tmp = validSamples(adatCounters, CLOCK_ADAT_INDEX);
-                setClockValidity(c_clk_int, CLOCK_ADAT_INDEX, tmp, clkMode);
+                    /* Returns 1 if valid clock found */
+                    valid = validSamples(adatCounters, CLOCK_ADAT);
+                    setClockValidity(c_clk_int, CLOCK_ADAT, valid, clkMode);
+#endif
+                }
+                break;
 #endif
 
+#if ((XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN) && XUA_USE_SW_PLL)
+            case inuint_byref(c_sw_pll, tmp):
+                inct(c_sw_pll);
+                /* Send ACK back to audiohub to allow I2S to start
+                   This happens only on SDM restart and only once */
+                if(require_ack_to_audio)
+                {
+                    c_audio_rate_change <: tmp;
+                    require_ack_to_audio = 0;
+                }
                 break;
+#endif
 
+#if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
+                /* Receive notification of audio streaming settings change and store */
+            case c_audio_rate_change :> selected_mclk_rate:
+                c_audio_rate_change :> selected_sample_rate;
+#if XUA_USE_SW_PLL
+                mclks_per_sample = selected_mclk_rate / selected_sample_rate;
+                restart_sigma_delta(c_sw_pll, selected_mclk_rate);
+                reset_sw_pll_pfd = 1;
+                /* We will shedule an ACK when sigma delta is up and running */
+                require_ack_to_audio = 1;
+#else
+                /* Send ACK immediately as we are good to go if not using SW_PLL */
+                c_audio_rate_change <: 0;
+#endif
+                break;
 #endif
 
 #if (XUA_SPDIF_RX_EN)
             /* Receive sample from S/PDIF RX thread (streaming chan) */
             case c_spdif_rx :> spdifRxData:
 
+#if XUA_USE_SW_PLL
                 /* Record time of sample */
+                asm volatile(" getts %0, res[%1]" : "=r" (mclk_time_stamp) : "r" (p_for_mclk_count_aud));
+#endif
                 t_local :> spdifRxTime;
 
                 /* Check parity and ignore if bad */
@@ -530,7 +578,7 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                     case SPDIF_FRAME_Y:
 
                         /* Only store sample if not in overflow and stream is reasonably valid */
-                        if(!spdifOverflow && clockValid[CLOCK_SPDIF_INDEX])
+                        if(!spdifOverflow && clockValid[CLOCK_SPDIF])
                         {
                             /* Store left and right sample pair to buffer */
                             spdifSamples[spdifWr] = spdifLeft;
@@ -562,7 +610,7 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
 
                 spdifCounters.samples += 1;
 
-                if(clkMode == CLOCK_SPDIF && clockValid[CLOCK_SPDIF_INDEX])
+                if(clkMode == CLOCK_SPDIF && clockValid[CLOCK_SPDIF])
                 {
                     spdifCounters.receivedSamples+=1;
 
@@ -578,9 +626,16 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                             /* Setup for next edge */
                             timeNextEdge = spdifRxTime + LOCAL_CLOCK_INCREMENT + LOCAL_CLOCK_MARGIN;
 
+#if XUA_USE_SW_PLL
+                            do_sw_pll_phase_frequency_detector_dig_rx(  mclk_time_stamp,
+                                                                        mclks_per_sample,
+                                                                        c_sw_pll,
+                                                                        spdifCounters.receivedSamples,
+                                                                        reset_sw_pll_pfd);
+#else
                             /* Toggle edge */
                             i_pll_ref.toggle_timed(1);
-
+#endif
                             /* Reset counters */
                             spdifCounters.receivedSamples = 0;
                         }
@@ -591,7 +646,11 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
 #if (XUA_ADAT_RX_EN)
                 /* receive sample from ADAT rx thread (streaming channel with CT_END) */
                 case inuint_byref(c_adat_rx, tmp):
+
+#if XUA_USE_SW_PLL
                     /* record time of sample */
+                    asm volatile(" getts %0, res[%1]" : "=r" (mclk_time_stamp) : "r" (p_for_mclk_count_aud));
+#endif
                     t_local :> adatReceivedTime;
 
                     /* Sync is: 1 | (user_byte << 4) */
@@ -611,7 +670,7 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                         if (adatChannel == 8)
                         {
                             /* only store left samples if not in overflow and stream is reasonably valid */
-                            if (!adatOverflow && clockValid[CLOCK_ADAT_INDEX])
+                            if (!adatOverflow && clockValid[CLOCK_ADAT])
                             {
                                 /* Unpick the SMUX.. */
                                 if(smux == 2)
@@ -668,7 +727,7 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                         {
                             adatCounters.samples += 1;
 
-                                if (clkMode == CLOCK_ADAT && clockValid[CLOCK_ADAT_INDEX])
+                                if (clkMode == CLOCK_ADAT && clockValid[CLOCK_ADAT])
                                 {
                                     adatCounters.receivedSamples += 1;
 
@@ -684,8 +743,16 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                                             /* Setup for next edge */
                                             timeNextEdge = adatReceivedTime + LOCAL_CLOCK_INCREMENT + LOCAL_CLOCK_MARGIN;
 
+#if XUA_USE_SW_PLL
+                                            do_sw_pll_phase_frequency_detector_dig_rx(  mclk_time_stamp,
+                                                                                        mclks_per_sample,
+                                                                                        c_sw_pll,
+                                                                                        adatCounters.receivedSamples,
+                                                                                        reset_sw_pll_pfd);
+#else
                                             /* Toggle edge */
                                             i_pll_ref.toggle_timed(1);
+#endif
 
                                             /* Reset counters */
                                             adatCounters.receivedSamples = 0;
@@ -724,7 +791,7 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                         spdifSamps -= 2;
 
                         /* spdifSamps could go to -1 */
-                        if(spdifSamps < 0)
+                        if(spdifSamps <= 0)
                         {
                             /* We're out of S/PDIF samples, mark underflow condition */
                             spdifUnderflow = 1;
@@ -738,7 +805,6 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                             spdifOverflow = 0;
                         }
                     }
-
 #endif
 #if (XUA_ADAT_RX_EN)
                 if (adatUnderflow)
@@ -806,7 +872,7 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                     }
 
                     /* adatSamps could go to -1 */
-                    if (adatSamps < 0)
+                    if (adatSamps <= 0)
                     {
                         /* we're out of ADAT samples, mark underflow condition */
                         adatUnderflow = 1;
@@ -823,7 +889,7 @@ void clockGen (streaming chanend ?c_spdif_rx, chanend ?c_adat_rx, client interfa
                 outuint(c_dig_rx, 1);
                 break;
 #endif
-        }
-    }
-}
+        } /* select */
+    } /* while(1) */
+} /* clkgen task scope */
 

lib_xua/src/core/clocking/sw_pll_wrapper.h

@@ -0,0 +1,57 @@
+// Copyright 2024 XMOS LIMITED.
+// This Software is subject to the terms of the XMOS Public Licence: Version 1.
+
+#ifndef _SW_PLL_WRAPPPER_H_
+#define _SW_PLL_WRAPPPER_H_
+
+#include "xua.h"
+
+#if XUA_USE_SW_PLL
+extern "C"
+{
+    #include "sw_pll.h"
+}
+
+/* Special control value to disable SDM. Outside of normal range which is less than 16b.*/
+#define DISABLE_SDM     0x10000000
+
+
+/** Task that receives an error term, passes it through a PI controller and periodically
+ *  calclulates a sigma delta output value and sends it to the PLL fractional register.
+ *
+ *  \param c_sw_pll                 Channel connected to the clocking thread to pass raw error terms.
+ */
+void sw_pll_task(chanend c_sw_pll);
+
+/** Helper function that sends a special restart command. It causes the SDM task
+ *  to quit and restart using the new mclk.
+ *
+ *  \param c_sw_pll                 Channel connected to the clocking thread to pass raw error terms.
+ *  \param mclk_Rate                The mclk frequency in Hz.
+ */
+void restart_sigma_delta(chanend c_sw_pll, unsigned mclk_rate);
+
+/** Performs a frequency comparsion between the incoming digital Rx stream and the local mclk.
+ *
+ *  \param mclk_time_stamp  The captured mclk count (using port timer) at the time of sample Rx.
+ *  \param mclks_per_sample The nominal number of mclks per audio sample.
+ *  \param c_sw_pll         Channel connected to the sigma delta and controller thread.
+ *  \param receivedSamples  The number of received samples since tha last call to this function.
+ *  \param reset_sw_pll_pfd Reference to a flag which will be used to signal reset of this function's state.
+ */
+void do_sw_pll_phase_frequency_detector_dig_rx( unsigned short mclk_time_stamp,
+                                                unsigned mclks_per_sample,
+                                                chanend c_sw_pll,
+                                                int receivedSamples,
+                                                int &reset_sw_pll_pfd);
+
+/** Initilaises the software PLL both hardware and state. Sets the mclk frequency to a nominal point.
+ *
+ *  \param sw_pll   Reference to a software pll state struct to be initialised.
+ *  \param mClk     The current nominal mClk frequency.
+ *
+ *  returns         The SDM update interval in ticks and the initial DCO setting for nominal frequency */
+{unsigned, unsigned} InitSWPLL(sw_pll_state_t &sw_pll, unsigned mClk);
+
+#endif /* XUA_USE_SW_PLL */
+#endif /* _SW_PLL_WRAPPPER_H_ */

lib_xua/src/core/clocking/sw_pll_wrapper.xc

@@ -0,0 +1,202 @@
+// Copyright 2024 XMOS LIMITED.
+// This Software is subject to the terms of the XMOS Public Licence: Version 1.
+#include <xs1.h>
+#include <assert.h>
+#include <print.h>
+
+#include "sw_pll_wrapper.h"
+#include "xua.h"
+
+#if XUA_USE_SW_PLL
+
+
+{unsigned, unsigned} init_sw_pll(sw_pll_state_t &sw_pll, unsigned mClk)
+{
+    /* Autogenerated SDM App PLL setup by dco_model.py using 22.5792_1M profile */
+    /* Input freq: 24000000
+       F: 134
+       R: 0
+       f: 8
+       p: 18
+       OD: 5
+       ACD: 5
+    */
+
+    #define APP_PLL_CTL_REG_22 0x0A808600
+    #define APP_PLL_DIV_REG_22 0x80000005
+    #define APP_PLL_FRAC_REG_22 0x80000812
+    #define SW_PLL_SDM_CTRL_MID_22 498283
+    #define SW_PLL_SDM_RATE_22 1000000
+
+    /* Autogenerated SDM App PLL setup by dco_model.py using 24.576_1M profile */
+    /* Input freq: 24000000
+       F: 146
+       R: 0
+       f: 4
+       p: 10
+       OD: 5
+       ACD: 5
+    */
+
+    #define APP_PLL_CTL_REG_24 0x0A809200
+    #define APP_PLL_DIV_REG_24 0x80000005
+    #define APP_PLL_FRAC_REG_24 0x8000040A
+    #define SW_PLL_SDM_CTRL_MID_24 478151
+    #define SW_PLL_SDM_RATE_24 1000000
+
+
+    const uint32_t app_pll_ctl_reg[2] = {APP_PLL_CTL_REG_22, APP_PLL_CTL_REG_24};
+    const uint32_t app_pll_div_reg[2] = {APP_PLL_DIV_REG_22, APP_PLL_DIV_REG_24};
+    const uint32_t app_pll_frac_reg[2] = {APP_PLL_FRAC_REG_22, APP_PLL_FRAC_REG_24};
+    const uint32_t sw_pll_sdm_ctrl_mid[2] = {SW_PLL_SDM_CTRL_MID_22, SW_PLL_SDM_CTRL_MID_24};
+    const uint32_t sw_pll_sdm_rate[2] = {SW_PLL_SDM_RATE_22, SW_PLL_SDM_RATE_24};
+
+    const int clkIndex = mClk == MCLK_48 ? 1 : 0;
+
+    sw_pll_sdm_init(&sw_pll,
+                SW_PLL_15Q16(0.0),
+                SW_PLL_15Q16(32.0),
+                SW_PLL_15Q16(0.25),
+                0, /* LOOP COUNT Don't care for this API */
+                0, /* PLL_RATIO  Don't care for this API */
+                0, /* No jitter compensation needed */
+                app_pll_ctl_reg[clkIndex],
+                app_pll_div_reg[clkIndex],
+                app_pll_frac_reg[clkIndex],
+                sw_pll_sdm_ctrl_mid[clkIndex],
+                3000 /* PPM_RANGE (FOR PFD) Don't care for this API*/ );
+
+    /* Reset SDM too */
+    sw_pll_init_sigma_delta(&sw_pll.sdm_state);
+
+    return {XS1_TIMER_HZ / sw_pll_sdm_rate[clkIndex], sw_pll_sdm_ctrl_mid[clkIndex]};
+}
+
+void do_sw_pll_phase_frequency_detector_dig_rx( unsigned short mclk_time_stamp,
+                                                unsigned mclks_per_sample,
+                                                chanend c_sw_pll,
+                                                int receivedSamples,
+                                                int &reset_sw_pll_pfd)
+{
+    const unsigned control_loop_rate_divider = 6; /* 300Hz * 2 edges / 6 -> 100Hz loop rate */
+    static unsigned control_loop_counter = 0;
+    static unsigned total_received_samples = 0;
+
+    /* Keep a store of the last mclk time stamp so we can work out the increment */
+    static unsigned short last_mclk_time_stamp = 0;
+
+    control_loop_counter++;
+
+    total_received_samples += receivedSamples;
+
+    if(control_loop_counter == control_loop_rate_divider)
+    {
+        /* Calculate what the zero-error mclk count increment should be for this many samples */
+        const unsigned expected_mclk_inc = mclks_per_sample * total_received_samples / 2; /* divide by 2 because this fn is called per edge */
+
+        /* Calculate actualy time-stamped mclk count increment is */
+        const unsigned short actual_mclk_inc = mclk_time_stamp - last_mclk_time_stamp;
+
+        /* The difference is the raw error in terms of mclk counts */
+        short f_error = (int)actual_mclk_inc - (int)expected_mclk_inc;
+        if(reset_sw_pll_pfd)
+        {
+            f_error = 0;            /* Skip first measurement as it will likely be very out */
+            reset_sw_pll_pfd = 0;
+        }
+
+        /* send PFD output to the sigma delta thread */
+        outuint(c_sw_pll, (int) f_error);
+        outct(c_sw_pll, XS1_CT_END);
+
+        last_mclk_time_stamp = mclk_time_stamp;
+        control_loop_counter = 0;
+        total_received_samples = 0;
+    }
+}
+
+void sw_pll_task(chanend c_sw_pll){
+    /* Zero is an invalid number and the SDM will not write the frac reg until
+       the first control value has been received. This avoids issues with
+       channel lockup if two tasks (eg. init and SDM) try to write at the same time. */
+
+    while(1)
+    {
+        unsigned selected_mclk_rate = inuint(c_sw_pll);
+        inct(c_sw_pll);
+
+        int f_error = 0;
+        int dco_setting = 0;        /* gets set at init_sw_pll */
+        unsigned sdm_interval = 0;  /* gets set at init_sw_pll */
+        sw_pll_state_t sw_pll;
+
+        /* initialse the SDM and gather SDM initial settings */
+        {sdm_interval, dco_setting} = init_sw_pll(sw_pll, selected_mclk_rate);
+
+        tileref_t this_tile = get_local_tile_id();
+
+        timer tmr;
+        int32_t time_trigger;
+        tmr :> time_trigger;
+        time_trigger += sdm_interval; /* ensure first loop has correct delay */
+        int running = 1;
+
+        outuint(c_sw_pll, 0); /* Signal back via clockgen to audio to start I2S */
+        outct(c_sw_pll, XS1_CT_END);
+
+        unsigned rx_word = 0;
+        while(running)
+        {
+            /* Poll for new SDM control value */
+            select
+            {
+                case inuint_byref(c_sw_pll, rx_word):
+                    inct(c_sw_pll);
+                    if(rx_word == DISABLE_SDM)
+                    {
+                        f_error = 0;
+                        running = 0;
+                    }
+                    else
+                    {
+                        f_error = (int32_t)rx_word;
+                        unsafe
+                        {
+                            sw_pll_sdm_do_control_from_error(&sw_pll, -f_error);
+                            dco_setting = sw_pll.sdm_state.current_ctrl_val;
+                        }
+                    }
+                break;
+
+                /* Do nothing & fall-through. Above case polls only once per loop */
+                default:
+                break;
+            }
+
+            /* Wait until the timer value has been reached
+               This implements a timing barrier and keeps
+               the loop rate constant. */
+            select
+            {
+                case tmr when timerafter(time_trigger) :> int _:
+                    time_trigger += sdm_interval;
+                break;
+            }
+
+            unsafe {
+                sw_pll_do_sigma_delta(&sw_pll.sdm_state, this_tile, dco_setting);
+            }
+        } /* while running */
+    } /* while(1) */
+}
+
+
+void restart_sigma_delta(chanend c_sw_pll, unsigned selected_mclk_rate)
+{
+    outuint(c_sw_pll, DISABLE_SDM); /* Resets SDM */
+    outct(c_sw_pll, XS1_CT_END);
+    outuint(c_sw_pll, selected_mclk_rate);
+    outct(c_sw_pll, XS1_CT_END);
+}
+
+#endif /* XUA_USE_SW_PLL */

lib_xua/src/core/endpoint0/xua_ep0_descriptors.h

@@ -1,4 +1,4 @@
-// Copyright 2011-2023 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 /**
  * @file    xua_ep0_descriptors.h
@@ -772,6 +772,11 @@ typedef struct
     unsigned char configDesc_DFU[DFU_LENGTH];
 #endif
 
+#ifdef USB_CONTROL_DESCS
+    /* Inferface descriptor for control */
+    unsigned char itfDesc_control[9];
+#endif
+
 #ifdef IAP
     USB_Descriptor_Interface_t                  iAP_Interface;
     USB_Descriptor_Endpoint_t                   iAP_Out_Endpoint;
@@ -2104,6 +2109,21 @@ USB_Config_Descriptor_Audio2_t cfgDesc_Audio2=
 #endif
 #endif /* (XUA_DFU_EN == 1) */
 
+#ifdef USB_CONTROL_DESCS
+    {
+    /* Control interface descriptor */
+    0x09,                                                /* 0 bLength : Size of this descriptor, in bytes. (field size 1 bytes) */
+    0x04,                                                /* 1 bDescriptorType : INTERFACE descriptor. (field size 1 bytes) */
+    (INTERFACE_NUMBER_MISC_CONTROL),                     /* 2 bInterfaceNumber */
+    0x00,                                                /* 3 bAlternateSetting : Index of this setting. (field size 1 bytes) */
+    0x00,                                                /* 4 bNumEndpoints : 0 endpoints. (field size 1 bytes) */
+    USB_CLASS_VENDOR_SPECIFIC,                           /* 5 bInterfaceClass : Vendor specific. (field size 1 bytes) */
+    0xFF,                                                /* 6 bInterfaceSubclass : (field size 1 bytes) */
+    0xFF,                                                /* 7 bInterfaceProtocol : Unused. (field size 1 bytes) */
+    offsetof(StringDescTable_t, ctrlStr)/sizeof(char *), /* 8 iInterface */
+    },
+#endif
+
 #ifdef IAP
     /* Interface descriptor */
     .iAP_Interface =

lib_xua/src/core/endpoint0/xua_ep0_uacreqs.xc

@@ -456,34 +456,35 @@ int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
                                     (buffer, unsigned char[])[0] = 1;
                                     return XUD_DoGetRequest(ep0_out, ep0_in, (buffer, unsigned char[]), 1, sp.wLength);
                                     break;
-
+#if (XUA_SPDIF_RX_EN)
                                 case ID_CLKSRC_SPDIF:
 
                                     /* Interogate clockgen thread for validity */
                                     if (!isnull(c_clk_ctl))
                                     {
                                         outuint(c_clk_ctl, GET_VALID);
-                                        outuint(c_clk_ctl, CLOCK_SPDIF_INDEX);
+                                        outuint(c_clk_ctl, CLOCK_SPDIF);
                                         outct(c_clk_ctl, XS1_CT_END);
                                         (buffer, unsigned char[])[0] = inuint(c_clk_ctl);
                                         chkct(c_clk_ctl, XS1_CT_END);
                                         return XUD_DoGetRequest(ep0_out, ep0_in, (buffer, unsigned char[]), 1, sp.wLength);
                                     }
-
                                     break;
-
+#endif
+#if (XUA_ADAT_RX_EN)
                                  case ID_CLKSRC_ADAT:
 
                                     if (!isnull(c_clk_ctl))
                                     {
                                         outuint(c_clk_ctl, GET_VALID);
-                                        outuint(c_clk_ctl, CLOCK_ADAT_INDEX);
+                                        outuint(c_clk_ctl, CLOCK_ADAT);
                                         outct(c_clk_ctl, XS1_CT_END);
                                         (buffer, unsigned char[])[0] = inuint(c_clk_ctl);
                                         chkct(c_clk_ctl, XS1_CT_END);
                                         return XUD_DoGetRequest(ep0_out, ep0_in, (buffer, unsigned char[]), 1, sp.wLength);
                                     }
                                     break;
+#endif
 
                                 default:
                                     //Unknown Unit ID in Clock Valid Control Request
@@ -513,19 +514,23 @@ int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
                                 return result;
                             }
 
-                            /* Check for correct datalength for clock sel */
                             if(datalength == 1)
                             {
-                                if (!isnull(c_clk_ctl))
-                                {
-                                    outuint(c_clk_ctl, SET_SEL);
-                                    outuint(c_clk_ctl, (buffer, unsigned char[])[0]);
-                                    outct(c_clk_ctl, XS1_CT_END);
-                                }
-                                /* Send 0 Length as status stage */
-                                return XUD_DoSetRequestStatus(ep0_in);
-                            }
+                                int clockIndex = (int) (buffer, unsigned char[])[0];
+                                clockIndex -= 1; /* Index to/from host is 1-based */
 
+                                if((clockIndex >= 0) && (clockIndex < CLOCK_COUNT))
+                                {
+                                    if(!isnull(c_clk_ctl))
+                                    {
+                                        outuint(c_clk_ctl, SET_SEL);
+                                        outuint(c_clk_ctl, clockIndex);
+                                        outct(c_clk_ctl, XS1_CT_END);
+                                    }
+                                    /* Send 0 Length as status stage */
+                                    return XUD_DoSetRequestStatus(ep0_in);
+                                }
+                            }
                         }
                         else
                         {
@@ -533,13 +538,15 @@ int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
                             (buffer, unsigned char[])[0] = 1;
                             if (!isnull(c_clk_ctl))
                             {
+                                int clockIndex;
                                 outuint(c_clk_ctl, GET_SEL);
                                 outct(c_clk_ctl, XS1_CT_END);
-                                (buffer, unsigned char[])[0] = inuint(c_clk_ctl);
+                                clockIndex = inuint(c_clk_ctl);
+                                clockIndex += 1; /* Index to/from host is 1-based */
+                                (buffer, unsigned char[])[0] = (unsigned char) clockIndex;
                                 chkct(c_clk_ctl, XS1_CT_END);
                             }
-                            return XUD_DoGetRequest( ep0_out, ep0_in, (buffer, unsigned char[]), 1, sp.wLength );
-
+                            return XUD_DoGetRequest( ep0_out, ep0_in, (buffer, unsigned char[]), 1, sp.wLength);
                         }
                     }
                     break;
@@ -916,17 +923,20 @@ int AudioClassRequests_2(XUD_ep ep0_out, XUD_ep ep0_in, USB_SetupPacket_t &sp, c
                                     {
                                         storeFreq((buffer, unsigned char[]), i, currentFreq44);
                                         num_freqs++;
-                                        currentFreq44*=2;
                                     }
 
-                                    if((currentFreq48 <= maxFreq))
+                                    currentFreq44*=2;
+
+                                    if((currentFreq48 <= maxFreq) && (currentFreq48 >= MIN_FREQ))
                                     {
                                         /* Note i passed byref here */
                                         storeFreq((buffer, unsigned char[]), i, currentFreq48);
                                         num_freqs++;
-                                        currentFreq48*=2;
                                     }
-                                    else
+
+                                    currentFreq48*=2;
+
+                                    if((currentFreq48 > MAX_FREQ) && (currentFreq44 > MAX_FREQ))
                                     {
                                         break;
                                     }

lib_xua/src/core/main.xc

@@ -1,4 +1,4 @@
-// Copyright 2012-2023 XMOS LIMITED.
+// Copyright 2012-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
 #include "xua.h"                          /* Device specific defines */
@@ -144,6 +144,11 @@ on tile[XUD_TILE] : in port p_spdif_rx                      = PORT_SPDIF_IN;
 #if (XUA_SPDIF_RX_EN) || (XUA_ADAT_RX_EN) || (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
 /* Reference to external clock multiplier */
 on tile[PLL_REF_TILE] : out port p_pll_ref                  = PORT_PLL_REF;
+#ifdef __XS3A__
+on tile[AUDIO_IO_TILE] : port p_for_mclk_count_audio        = PORT_MCLK_COUNT_2;
+#else /* __XS3A__ */
+#define p_for_mclk_count_audio                              null
+#endif /* __XS3A__ */
 #endif
 
 #ifdef MIDI
@@ -310,6 +315,13 @@ void usb_audio_io(chanend ?c_aud_in,
 #if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
     , client interface pll_ref_if i_pll_ref
 #endif
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
+    , chanend c_audio_rate_change
+#endif
+#if ((XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN) && XUA_USE_SW_PLL)
+    , port p_for_mclk_count_aud
+    , chanend c_sw_pll
+#endif
 )
 {
 #if (MIXER)
@@ -318,9 +330,14 @@ void usb_audio_io(chanend ?c_aud_in,
 
 #if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
     chan c_dig_rx;
-#else
-    #define c_dig_rx null
-#endif
+    chan c_audio_rate_change; /* Notification of new mclk freq to clockgen and synch */
+#if XUA_USE_SW_PLL
+    /* Connect p_for_mclk_count_aud to clk_audio_mclk so we can count mclks/timestamp in digital rx*/
+    unsigned x = 0;
+    asm("ldw %0, dp[clk_audio_mclk]":"=r"(x));
+    asm("setclk res[%0], %1"::"r"(p_for_mclk_count_aud), "r"(x));
+#endif /* XUA_USE_SW_PLL */
+#endif /* (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN) */
 
 #if (XUA_NUM_PDM_MICS > 0) && (PDM_TILE == AUDIO_IO_TILE)
     /* Configure clocks ports - sharing mclk port with I2S */
@@ -366,6 +383,9 @@ void usb_audio_io(chanend ?c_aud_in,
 #if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
                 , c_dig_rx
 #endif
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
+                , c_audio_rate_change
+#endif
 #if (XUD_TILE != 0) && (AUDIO_IO_TILE == 0) && (XUA_DFU_EN == 1)
                 , dfuInterface
 #endif
@@ -385,12 +405,22 @@ void usb_audio_io(chanend ?c_aud_in,
              * However, due to the use of an interface the pll reference signal port can be on another tile
              */
             thread_speed();
-            clockGen(c_spdif_rx, c_adat_rx, i_pll_ref, c_dig_rx, c_clk_ctl, c_clk_int);
+            clockGen(   c_spdif_rx,
+                        c_adat_rx,
+                        i_pll_ref,
+                        c_dig_rx,
+                        c_clk_ctl,
+                        c_clk_int,
+                        c_audio_rate_change
+#if XUA_USE_SW_PLL
+                        , p_for_mclk_count_aud
+                        , c_sw_pll
+#endif
+                        );
         }
 #endif
 
-        //:
-    }
+    } // par
 }
 
 #ifndef USER_MAIN_DECLARATIONS
@@ -437,7 +467,7 @@ int main()
 #define c_adat_rx null
 #endif
 
-#if (XUA_SPDIF_TX_EN) //&& (SPDIF_TX_TILE != AUDIO_IO_TILE)
+#if (XUA_SPDIF_TX_EN) && (SPDIF_TX_TILE != AUDIO_IO_TILE)
     chan c_spdif_tx;
 #endif
 
@@ -463,9 +493,16 @@ int main()
 #endif
 #endif
 
-#if ((XUA_SYNCMODE == XUA_SYNCMODE_SYNC) || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
+#if (((XUA_SYNCMODE == XUA_SYNCMODE_SYNC && !XUA_USE_SW_PLL) || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN) )
     interface pll_ref_if i_pll_ref;
 #endif
+
+#if ((XUA_SYNCMODE == XUA_SYNCMODE_SYNC || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN) && XUA_USE_SW_PLL)
+    chan c_sw_pll;
+#endif
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
+    chan c_audio_rate_change; /* Notification of new mclk freq to ep_buffer */
+#endif
     chan c_sof;
     chan c_xud_out[ENDPOINT_COUNT_OUT];              /* Endpoint channels for XUD */
     chan c_xud_in[ENDPOINT_COUNT_IN];
@@ -487,7 +524,7 @@ int main()
     {
         USER_MAIN_CORES
 
-#if ((XUA_SYNCMODE == XUA_SYNCMODE_SYNC) || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
+#if (((XUA_SYNCMODE == XUA_SYNCMODE_SYNC  && !XUA_USE_SW_PLL) || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN))
         on tile[PLL_REF_TILE]: PllRefPinTask(i_pll_ref, p_pll_ref);
 #endif
         on tile[XUD_TILE]:
@@ -558,7 +595,12 @@ int main()
 #endif
                            , c_mix_out
 #if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
+                           , c_audio_rate_change
+    #if (!XUA_USE_SW_PLL)
                            , i_pll_ref
+    #else
+                           , c_sw_pll
+    #endif
 #endif
                     );
                 //:
@@ -573,8 +615,16 @@ int main()
 #endif /* XUA_USB_EN */
         }
 
+#if ((XUA_SYNCMODE == XUA_SYNCMODE_SYNC || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN) && XUA_USE_SW_PLL)
+        on tile[AUDIO_IO_TILE]: sw_pll_task(c_sw_pll);
+#endif
+
         on tile[AUDIO_IO_TILE]:
         {
+            #ifndef AUDIO_UNSAFE_RESRC
+            #define AUDIO_UNSAFE_RESRC
+            #endif
+            AUDIO_UNSAFE_RESRC
             /* Audio I/O task, includes mixing etc */
             usb_audio_io(c_mix_out
 #if (XUA_SPDIF_TX_EN) && (SPDIF_TX_TILE != AUDIO_IO_TILE)
@@ -588,13 +638,16 @@ int main()
                 , dfuInterface
 #endif
 #if (XUA_NUM_PDM_MICS > 0)
-#if (PDM_TILE == AUDIO_IO_TILE)
-                , c_ds_output
-#endif
-                , c_pdm_pcm
 #endif
 #if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
                 , i_pll_ref
+#endif
+#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
+                , c_audio_rate_change
+#endif
+#if ((XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN) && XUA_USE_SW_PLL)
+                , p_for_mclk_count_audio
+                , c_sw_pll
 #endif
             );
         }

lib_xua/src/core/ports/audioports.xc

@@ -73,6 +73,9 @@ void ConfigAudioPorts(
     if(!isnull(p_lrclk))
     {
         /* Clock LR clock from bit clock-block */
+        #if((XUA_PCM_FORMAT == XUA_PCM_FORMAT_TDM)&&(I2S_TDM_LRCLK_EDGES==0))
+        set_port_inv(p_lrclk);//反向lrclk的输出
+        #endif
         configure_out_port_no_ready(p_lrclk, clk_audio_bclk, 0);
     }
 

lib_xua/src/core/user/audiohw/audiohw.c

@@ -0,0 +1,29 @@
+// Copyright 2023-2024 XMOS LIMITED.
+// This Software is subject to the terms of the XMOS Public Licence: Version 1.
+
+/* Default implementations of AudioHwInit(), AudioHwConfig(), AudioHwConfig_Mute() and AudioHwConfig_UnMute() */
+
+void AudioHwInit() __attribute__ ((weak));
+void AudioHwInit()
+{
+    return;
+}
+
+void AudioHwConfig(unsigned samFreq, unsigned mClk, unsigned dsdMode, unsigned sampRes_DAC, unsigned sampRes_ADC) __attribute__ ((weak));
+void AudioHwConfig(unsigned samFreq, unsigned mClk, unsigned dsdMode, unsigned sampRes_DAC, unsigned sampRes_ADC)
+{
+    return;
+}
+
+void AudioHwConfig_Mute() __attribute__ ((weak));
+void AudioHwConfig_Mute()
+{
+    return;
+}
+
+void AudioHwConfig_UnMute() __attribute__ ((weak));
+void AudioHwConfig_UnMute()
+{
+    return;
+}
+

lib_xua/src/core/user/audiohw/audiohw.h

@@ -0,0 +1,54 @@
+// Copyright 2023-2024 XMOS LIMITED.
+// This Software is subject to the terms of the XMOS Public Licence: Version 1.
+#ifndef _AUDIO_HW_H_
+#define _AUDIO_HW_H_
+
+/* The functions below should be implemented for the external audio hardware arrangement of a specific design.
+ * Note, default (empty) implementations of these are provided in audiohub_user.c
+ */
+
+/**
+ * @brief   User audio hardware initialisation code
+ *
+ * This function is called when the device starts up and should contain user code to perform any required audio hardware initialisation
+ */
+void AudioHwInit(void);
+
+/**
+ * @brief   User audio hardware configuration code
+ *
+ * This function is called when on sample rate change and should contain user code to configure audio hardware
+ *  (clocking, CODECs etc) for a specific mClk/Sample frequency
+ *
+ * \param samFreq       The new sample frequency (in Hz)
+ *
+ * \param mClk          The new master clock frequency (in Hz)
+ *
+ * \param dsdMode       DSD mode, DSD_MODE_NATIVE, DSD_MODE_DOP or DSD_MODE_OFF
+ *
+ * \param sampRes_DAC   Playback sample resolution (in bits)
+ *
+ * \param sampRes_ADC   Record sample resolution (in bits)
+ */
+void AudioHwConfig(unsigned samFreq, unsigned mClk, unsigned dsdMode, unsigned sampRes_DAC, unsigned sampRes_ADC);
+
+/**
+ * @brief   User code mute audio hardware
+ *
+ * This function is called before AudioHwConfig() and should contain user code to mute audio hardware before a
+ * sample rate change in order to reduced audible pops/clicks
+ *
+ *  Note, if using the application PLL of a xcore.ai device this function will be called before the master-clock is
+ *  changed
+ */
+void AudioHwConfig_Mute(void);
+
+/**
+ * @brief   User code to un-mute audio hardware
+ *
+ * This function is called after AudioHwConfig() and should contain user code to un-mute audio hardware after a
+ *  sample rate change
+ */
+void AudioHwConfig_UnMute(void);
+
+#endif

lib_xua/src/core/user/audiostream/audiostream.h

@@ -1,30 +1,51 @@
-// Copyright 2011-2021 XMOS LIMITED.
+// Copyright 2011-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 #ifndef _AUDIOSTREAM_H_
 #define _AUDIOSTREAM_H_
 
-/* Functions that handle functions that must occur on stream start/stop e.g. DAC mute/un-mute
- *
- * THESE NEED IMPLEMENTING FOR A SPECIFIC DESIGN
- *
- * */
+/* Functions that handle functionality that occur on stream start/stop e.g. DAC mute/un-mute.
+ * They should be implemented for the external audio hardware arrangement of a specific design.
+ */
 
-/* Any actions required for stream start e.g. DAC un-mute - run every stream start */
+/**
+ * @brief   User stream start code
+ *
+ * User code to perform any actions required at every stream start - either input or output
+ */
 void UserAudioStreamStart(void);
 
-/* Any actions required on stream stop e.g. DAC mute - run every steam stop  */
+/**
+ * @brief   User stream stop code
+ *
+ * User code to perform any actions required on every stream stop - either input or output*/
 void UserAudioStreamStop(void);
 
-/* Any actions required on input stream start */
+/**
+ * @brief   User input stream stop code
+ *
+ * User code to perform any actions required on input stream start i.e. device to host
+ */
 void UserAudioInputStreamStart(void);
 
-/* Any actions required on input stream stop */
+/**
+ * @brief   User input stream stop code
+ *
+ * User code to perform any actions required on input stream stop i.e. device to host
+ */
 void UserAudioInputStreamStop(void);
 
-/* Any actions required on output stream start */
+/**
+ * @brief   User output stream start code
+ *
+ * User code to perform any actions required on output stream start i.e. host to device
+ */
 void UserAudioOutputStreamStart(void);
 
-/* Any actions required on output stream stop */
+/**
+ * @brief   User output stream stop code
+ *
+ * User code to perfrom any actions required on output stream stop i.e. host to device
+ */
 void UserAudioOutputStreamStop(void);
 
 #endif

lib_xua/src/core/user/hid/user_hid.h

@@ -1,17 +1,17 @@
-// Copyright 2013-2023 XMOS LIMITED.
+// Copyright 2013-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
 /**
  * @brief Human Interface Device (HID) API
  *
  * This file defines the Application Programming Interface (API) used to record HID
- *   events and retrieve a HID Report for sending to a host.
+ * events and retrieve a HID Report for sending to a host.
  * The using application has the responsibility to fulfill this API.
  * Document section numbers refer to the HID Device Class Definition, version 1.11.
  */
 
-#ifndef __USER_HID_H__
-#define __USER_HID_H__
+#ifndef _USER_HID_H_
+#define _USER_HID_H_
 
 #include <stddef.h>
 
@@ -34,22 +34,16 @@ typedef struct hidEvent_t {
 #define HID_MAX_DATA_BYTES ( 4 )
 #define HID_EVENT_INVALID_ID ( 0x100 )
 
-#if XUA_HID_ENABLED
-
 /**
  *  \brief  Get the data for the next HID Report
  *
- *  \note This function returns the HID data as a list of unsigned char because the
- *        \c XUD_SetReady_In() accepts data for transmission to the USB Host using
- *        this type.
- *
  *  \param[in]  id       The HID Report ID (see 5.6, 6.2.2.7, 8.1 and 8.2)
  *                       Set to zero if the application provides only one HID Report
- *                         which does not include a Report ID
+ *                       which does not include a Report ID
  *  \param[out] hidData  The HID data
  *                       If using Report IDs, this function places the Report ID in
- *                         the first element; otherwise the first element holds the
- *                         first byte of HID event data.
+ *                       the first element; otherwise the first element holds the
+ *                       first byte of HID event data.
  *
  *  \returns  The length of the HID Report in the \a hidData argument
  *  \retval   Zero means no new HID event data has been recorded for the given \a id
@@ -57,9 +51,8 @@ typedef struct hidEvent_t {
 size_t UserHIDGetData( const unsigned id, unsigned char hidData[ HID_MAX_DATA_BYTES ]);
 
 /**
- *	\brief  Initialize HID processing
+ *  \brief  Initialize HID processing
  */
 void UserHIDInit( void );
 
-#endif /* ( 0 < HID_CONTROLS ) */
-#endif /* __USER_HID_H__ */
+#endif /* _USER_HID_H_ */

lib_xua/src/core/user/hostactive/hostactive.h

@@ -1,4 +1,12 @@
-// Copyright 2013-2021 XMOS LIMITED.
+// Copyright 2013-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
+/**
+ * @brief   User host active code
+ *
+ * This function can be used to perform user defined actions based on host present/not-present events.
+ * This function is called on a change in state.
+ *
+ * \param active  Indicates if the host is active or not. 1 for active, else 0
+ */
 void UserHostActive(int active);

lib_xua/src/core/xua_commands.h

@@ -10,20 +10,19 @@
 #define SET_SEL                 1       /* Set value of clock selector */
 #define GET_FREQ                2       /* Get current freq */
 #define GET_VALID               3       /* Get current validity */
+#define SET_SMUX                7       /* Set SMUX mode (ADAT) */
 
-#define CLOCK_INTERNAL          1
-#define CLOCK_SPDIF             2
-#if (XUA_SPDIF_RX_EN)
-#define CLOCK_ADAT              3
-#else
-#define CLOCK_ADAT              2
+enum
+{
+    CLOCK_INTERNAL = 0,
+#if XUA_SPDIF_RX_EN
+    CLOCK_SPDIF,
 #endif
-
-#define CLOCK_INTERNAL_INDEX    (CLOCK_INTERNAL - 1)
-#define CLOCK_ADAT_INDEX        (CLOCK_ADAT - 1)
-#define CLOCK_SPDIF_INDEX       (CLOCK_SPDIF - 1)
-
-#define SET_SMUX                7
+#if XUA_ADAT_RX_EN
+    CLOCK_ADAT,
+#endif
+    CLOCK_COUNT
+};
 
 /* c_audioControl */
 #define SET_SAMPLE_FREQ         4

tests/test_sync_clk_basic/Makefile

@@ -3,10 +3,12 @@ TEST_FLAGS ?=
 
 XCC_FLAGS_HS = -O3 -g -DXUD_CORE_CLOCK=600 -save-temps -DUSB_TILE=tile[0] -DLOCAL_CLOCK_INCREMENT=10000 -DLOCAL_CLOCK_MARGIN=100 \
 							-DBUS_SPEED=2 \
+							-DXUA_USE_SW_PLL=0 \
 							$(TEST_FLAGS)
 
 XCC_FLAGS_FS = -O3 -g -DXUD_CORE_CLOCK=600 -save-temps -DUSB_TILE=tile[0] -DLOCAL_CLOCK_INCREMENT=10000 -DLOCAL_CLOCK_MARGIN=100 \
 							-DBUS_SPEED=1 \
+							-DXUA_USE_SW_PLL=0 \
 							$(TEST_FLAGS)
 
 TARGET = test_xs3_600.xn

tests/test_sync_clk_basic/src/main.xc

@@ -1,4 +1,4 @@
-// Copyright 2022 XMOS LIMITED.
+// Copyright 2022-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 
 /* Simple test to ensure reference clock to CS2100 device continues when SOF clock not available
@@ -173,6 +173,7 @@ int main()
     chan c_in[EP_COUNT_IN];
     chan c_sof;
     chan c_aud_ctl;
+    chan c_audio_rate_change;
 
     interface pll_ref_if i_pll_ref;
 
@@ -185,7 +186,7 @@ int main()
 
             XUA_Buffer_Ep(c_out[1],      /* USB Audio Out*/
                 c_in[1],                 /* USB Audio In */
-                c_sof, c_aud_ctl, p_off_mclk, i_pll_ref
+                c_sof, c_aud_ctl, p_off_mclk, c_audio_rate_change, i_pll_ref
             );
         }
 
@@ -193,4 +194,6 @@ int main()
 
         checker();
     }
+
+    return 0;
 }