Merge commit 'cb690527c9907dda021101f2bcfa28fa5a3f844a' into fetaure/multi_interface

# Conflicts:
#	CHANGELOG.rst

CHANGELOG.rst

@@ -13,6 +13,9 @@ UNRELEASED
   * FIXED:     ADAT Tx called too frequently
   * CHANGED:   ADAT Tx presents different channel count interfaces based on sample
     rate 
+  * CHANGED:   aud_to_host buffer size and the condition to come out of underflow
+    in decoupler to fix buffer underflow seen in ADAT tests
+  * FIXED:     Initialise SMUX based on DEFAULT_FREQ in clockgen
 
 4.0.0
 -----

lib_xua/api/xua_conf_default.h

@@ -1318,7 +1318,7 @@
 #endif
 
 #if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
-    #if (XUA_SPDIF_RX_EN|| ADAT_RX)
+    #if (XUA_SPDIF_RX_EN|| XUA_ADAT_RX_EN)
         #error "Digital input streams not supported in Sync mode"
     #endif
 #endif

lib_xua/src/core/buffer/decouple/decouple.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"
 
@@ -42,8 +42,13 @@
 #define MAX_DEVICE_AUD_PACKET_SIZE_OUT (MAX(MAX_DEVICE_AUD_PACKET_SIZE_OUT_FS, MAX_DEVICE_AUD_PACKET_SIZE_OUT_HS))
 
 /*** BUFFER SIZES ***/
-
+/* How many packets too allow for in buffer - minimum is 5.
-#define BUFFER_PACKET_COUNT (4)    /* How many packets too allow for in buffer - minimum is 4 */
+2 for having in the aud_to_host buffer when it comes out of underflow, space available for 2 more for to accomodate cases when
+2 pkts from audio hub get written into the aud_to_host buffer within 1 SOF period, and space for 1 extra packet to ensure that
+when the 4th packet gets written to the buffer, there's space to accomodate the next packet, otherwise handle_audio_request() will
+drop packets after writing the 4th packet in the buffer
+*/
+#define BUFFER_PACKET_COUNT (5)
 
 #define BUFF_SIZE_OUT_HS    MAX_DEVICE_AUD_PACKET_SIZE_OUT_HS * BUFFER_PACKET_COUNT
 #define BUFF_SIZE_OUT_FS    MAX_DEVICE_AUD_PACKET_SIZE_OUT_FS * BUFFER_PACKET_COUNT
@@ -1054,7 +1059,21 @@ void XUA_Buffer_Decouple(chanend c_mix_out
                     assert(fillLevel <= BUFF_SIZE_IN);
 
                     /* Check if we have come out of underflow */
-                    if (fillLevel >= IN_BUFFER_PREFILL)
+                    unsigned sampFreq;
+                    GET_SHARED_GLOBAL(sampFreq, g_freqChange_sampFreq);
+                    int min, mid, max;
+                    GetADCCounts(sampFreq, min, mid, max);
+                    const int min_pkt_size = ((min * g_curSubSlot_In * g_numUsbChan_In + 3) & ~0x3) + 4;
+
+                    /*
+                        Come out of underflow if there are exactly 2 packets in the buffer.
+                        This ensures that handle_audio_request() does not drop packets when writing packets into the aud_to_host buffer
+                        when aud_to_host buffer is not in underflow.
+                        For example, coming out of underflow with 3 packets in the buffer would mean handle_audio_request()
+                        drops packets if 2 pkts are received from audio hub in 1 SOF period. Coming out of underflow with 4
+                        packets would mean handle_audio_request would drop packets after writing 1 packet to the aud_to_host buffer.
+                    */
+                    if ((fillLevel >= (min_pkt_size*2)) && (fillLevel < (min_pkt_size*3)))
                     {
                         int aud_to_host_rdptr;
                         GET_SHARED_GLOBAL(aud_to_host_rdptr, g_aud_to_host_rdptr);

lib_xua/src/core/clocking/clockgen.xc

@@ -235,7 +235,23 @@ void clockGen ( streaming chanend ?c_spdif_rx,
     unsigned tmp;
 
     /* Start in no-SMUX (8-channel) mode */
-    int smux = 0;
+    int smux;
+    // Initialise smux based based on the DEFAULT_FREQ
+    if(DEFAULT_FREQ < 88200)
+    {
+        /* No SMUX */
+        smux = 0;
+    }
+    else if(DEFAULT_FREQ < 176400)
+    {
+        /* SMUX */
+        smux = 1;
+    }
+    else
+    {
+        /* SMUX II */
+        smux = 2;
+    }
 
 #ifdef LEVEL_METER_LEDS
     timer t_level;

tests/test_i2s_loopback/xk_216_mc/audiohw.xc

@@ -1,4 +1,4 @@
-// Copyright 2016-2022 XMOS LIMITED.
+// Copyright 2016-2024 XMOS LIMITED.
 // This Software is subject to the terms of the XMOS Public Licence: Version 1.
 #ifdef HARDWARE
 
@@ -19,7 +19,7 @@
 /* CS2100 lists typical lock time as 100 * input period */
 #define     AUDIO_PLL_LOCK_DELAY     (40000000)
 
-#if defined(SPDIF_RX) || defined(ADAT_RX)
+#if defined(XUA_SPDIF_RX_EN) || defined(XUA_ADAT_RX_EN)
 #define USE_FRACTIONAL_N 1
 #endif
 
@@ -32,7 +32,7 @@ port p_i2c = on tile[0]:PORT_I2C;
 
 #ifdef USE_FRACTIONAL_N
 
-#if !(defined(SPDIF_RX) || defined(ADAT_RX))
+#if !(defined(XUA_SPDIF_RX_EN) || defined(XUA_ADAT_RX_EN))
 /* Choose a frequency the xcore can easily generate internally */
 #define PLL_SYNC_FREQ 1000000
 #else
@@ -95,7 +95,7 @@ void PllMult(unsigned output, unsigned ref, client interface i2c_master_if i2c)
 }
 #endif
 
-#if !(defined(SPDIF_RX) || defined(ADAT_RX)) && defined(USE_FRACTIONAL_N)
+#if !(defined(XUA_SPDIF_RX_EN) || defined(XUA_ADAT_RX_EN)) && defined(USE_FRACTIONAL_N)
 on tile[AUDIO_IO_TILE] : out port p_pll_clk = PORT_PLL_REF;
 on tile[AUDIO_IO_TILE] : clock clk_pll_sync = XS1_CLKBLK_5;
 #endif
@@ -111,7 +111,7 @@ void wait_us(int microseconds)
 
 void AudioHwInit(chanend ?c_codec)
 {
-#if !(defined(SPDIF_RX) || defined(ADAT_RX)) && defined(USE_FRACTIONAL_N)
+#if !(defined(XUA_SPDIF_RX_EN) || defined(XUA_ADAT_RX_EN)) && defined(USE_FRACTIONAL_N)
     /* Output a fixed sync clock to the pll */
     configure_clock_rate(clk_pll_sync, 100, 100/(PLL_SYNC_FREQ/1000000));
     configure_port_clock_output(p_pll_clk, clk_pll_sync);