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Manual merge from experimental/sync_app_pll (not working yet)

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Ed
2024-01-15 14:08:22 +00:00
parent 103aa8840b
commit c0a844c303
16 changed files with 546 additions and 324 deletions
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7
lib_xua/api/xua_audiohub.h
View File

@@ -12,6 +12,8 @@
#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
@@ -34,6 +36,8 @@
*
* \param p_i2s_adc Nullable array of ports for I2S data input lines
*
* \param i_SoftPll Interface to software PLL task
*
* \param c_spdif_tx Channel connected to S/PDIF transmiter core from lib_spdif
*
* \param c_dig Channel connected to the clockGen() thread for
@@ -49,6 +53,9 @@ void XUA_AudioHub(chanend ?c_aud,
buffered _XUA_CLK_DIR port:32 ?p_bclk,
buffered out port:32 (&?p_i2s_dac)[I2S_WIRES_DAC],
buffered in port:32 (&?p_i2s_adc)[I2S_WIRES_ADC]
#if (XUA_USE_APP_PLL) || defined(__DOXYGEN__)
, client interface SoftPll_if i_SoftPll
#endif
#if (XUA_SPDIF_TX_EN) || defined(__DOXYGEN__)
, chanend c_spdif_tx
#endif

28
lib_xua/api/xua_buffer.h
View File

@@ -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__
@@ -26,6 +26,7 @@
* \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_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 +39,7 @@ 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 +52,13 @@ 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__)
#if (!XUA_USE_APP_PLL) || defined(__DOXYGEN__)
, client interface pll_ref_if i_pll_ref
#endif
#if (XUA_USE_APP_PLL) || defined(__DOXYGEN__)
, chanend c_swpll_update
#endif
#endif
);
@@ -66,7 +72,7 @@ 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 +87,16 @@ 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__)
#if (!XUA_USE_APP_PLL) || defined(__DOXYGEN__)
, client interface pll_ref_if i_pll_ref
#endif
#if (XUA_USE_APP_PLL) || defined(__DOXYGEN__)
, chanend c_swpll_update
#endif
#endif
);
);
/** Manage the data transfer between the USB audio buffer and the
* Audio I/O driver.

13
lib_xua/api/xua_clocking.h
View File

@@ -38,5 +38,18 @@ void clockGen( streaming chanend ?c_spdif_rx,
chanend c_clk_int,
port ?p_for_mclk_count_aud,
chanend c_mclk_change);
interface SoftPll_if
{
void init(int mclk_hz);
};
#if (XUA_SYNCMODE == XUA_SYNCMODE_ASYNC)
[[distributable]]
#endif
void XUA_SoftPll(tileref tile, server interface SoftPll_if i_softPll, chanend c_update);
#endif
#endif

448
lib_xua/api/xua_conf_default.h
View File

@@ -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 +255,34 @@
#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 master-clocks.
* Only available on xcore.ai devices.
*
* Default: Enabled (for xcore.ai devices)
*/
#ifndef XUA_USE_APP_PLL
#if defined(__XS3A__)
#if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
/* Currently must use an external CS2100 device for syncing to external digital streams */
#define XUA_USE_APP_PLL (0)
#else
#define XUA_USE_APP_PLL (1)
#endif
#else
#define XUA_USE_APP_PLL (0)
#endif
#endif
/**
* @brief Default device sample frequency. A safe default should be used.
*
@@ -249,10 +292,25 @@
#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 +319,9 @@
#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 +336,7 @@
#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 +353,17 @@
#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 +377,14 @@
#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 +405,7 @@
* @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 +414,14 @@
* 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 +430,21 @@
* 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 +499,11 @@
* 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 +512,7 @@
* Default 0 (Disabled)
*/
#ifndef HID_CONTROLS
#define HID_CONTROLS (0)
#define HID_CONTROLS (0)
#endif
/**
@@ -488,12 +528,11 @@
* 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 +545,7 @@
* 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 +554,7 @@
* Default: ""
*/
#ifndef SERIAL_STR
#define SERIAL_STR ""
#define SERIAL_STR ""
#endif
/**
@@ -535,7 +563,7 @@
* Default: "XMOS"
*/
#ifndef VENDOR_STR
#define VENDOR_STR "XMOS"
#define VENDOR_STR "XMOS"
#endif
/**
@@ -544,7 +572,7 @@
* Default: 0x20B1 (XMOS)
*/
#ifndef VENDOR_ID
#define VENDOR_ID (0x20B1)
#define VENDOR_ID (0x20B1)
#endif
/**
@@ -586,7 +614,7 @@
*/
#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 +624,28 @@
* 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 +1049,7 @@
* Default: 1 (Enabled)
*/
#ifndef OUTPUT_VOLUME_CONTROL
#define OUTPUT_VOLUME_CONTROL (1)
#define OUTPUT_VOLUME_CONTROL (1)
#endif
/**
@@ -1030,7 +1058,7 @@
* Default: 1 (Enabled)
*/
#ifndef INPUT_VOLUME_CONTROL
#define INPUT_VOLUME_CONTROL (1)
#define INPUT_VOLUME_CONTROL (1)
#endif
/* Power */
@@ -1072,7 +1100,7 @@
* Default: 0 (Disabled)
*/
#ifndef MIXER
#define MIXER (0)
#define MIXER (0)
#endif
/**
@@ -1082,11 +1110,11 @@
*/
#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 +1126,7 @@
* Default: 18
*/
#ifndef MIX_INPUTS
#define MIX_INPUTS (18)
#define MIX_INPUTS (18)
#endif
/* Volume processing defines */
@@ -1110,7 +1138,7 @@
* Default: 0x8100 (-127db)
*/
#ifndef MIN_VOLUME
#define MIN_VOLUME (0x8100)
#define MIN_VOLUME (0x8100)
#endif
@@ -1120,7 +1148,7 @@
* Default: 0x0000 (0db)
*/
#ifndef MAX_VOLUME
#define MAX_VOLUME (0x0000)
#define MAX_VOLUME (0x0000)
#endif
/**
@@ -1129,7 +1157,7 @@
* Default: 0x100 (1db)
*/
#ifndef VOLUME_RES
#define VOLUME_RES (0x100)
#define VOLUME_RES (0x100)
#endif
/**
@@ -1139,7 +1167,7 @@
* Default: 0x8100 (-127db)
*/
#ifndef MIN_MIXER_VOLUME
#define MIN_MIXER_VOLUME (0x8100)
#define MIN_MIXER_VOLUME (0x8100)
#endif
/**
@@ -1148,7 +1176,7 @@
* Default: 0x0000 (0db)
*/
#ifndef MAX_MIXER_VOLUME
#define MAX_MIXER_VOLUME (0x0000)
#define MAX_MIXER_VOLUME (0x0000)
#endif
/**
@@ -1157,36 +1185,32 @@
* 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)
#else
#define OUT_VOLUME_IN_MIXER (0)
#endif
#define OUT_VOLUME_IN_MIXER (1)
#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 */
/* Apply in volume controls after the mix. Only relebant 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 +1218,9 @@
#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 +1300,8 @@ 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)
@@ -1378,7 +1404,7 @@ enum USBEndpointNumber_Out
/* Some defines that allow us to remove unused code */
/* Useful for dropping lower part of macs in volume processing... */
#if (FS_STREAM_FORMAT_OUTPUT_1_RESOLUTION_BITS > 24) || (HS_STREAM_FORMAT_OUTPUT_1_RESOLUTION_BITS > 24) || \
#if (FS_STREAM_FORMAT_OUTPUT_1_RESOLUTION_BITS > 24) || (HS_STREAM_FORMAT_OUTPUT_2_RESOLUTION_BITS > 24) || \
(((FS_STREAM_FORMAT_OUTPUT_2_RESOLUTION_BITS > 24) || (HS_STREAM_FORMAT_OUTPUT_2_RESOLUTION_BITS > 24)) && (OUTPUT_FORMAT_COUNT > 1)) || \
(((FS_STREAM_FORMAT_OUTPUT_3_RESOLUTION_BITS > 24) || (HS_STREAM_FORMAT_OUTPUT_3_RESOLUTION_BITS > 24)) && (OUTPUT_FORMAT_COUNT > 2))
#define STREAM_FORMAT_OUTPUT_RESOLUTION_32BIT_USED 1
@@ -1412,29 +1438,29 @@ enum USBEndpointNumber_Out
#endif
/* Useful for dropping lower part of macs in volume processing... */
#if (FS_STREAM_FORMAT_INPUT_1_RESOLUTION_BITS > 24) || (HS_STREAM_FORMAT_INPUT_1_RESOLUTION_BITS > 24)
#define STREAM_FORMAT_INPUT_RESOLUTION_32BIT_USED 1
#else
#define STREAM_FORMAT_INPUT_RESOLUTION_32BIT_USED 0
#endif
#if (FS_STREAM_FORMAT_INPUT_1_RESOLUTION_BITS > 24) || (FS_STREAM_FORMAT_INPUT_2_RESOLUTION_BITS > 24)
#define STREAM_FORMAT_INPUT_RESOLUTION_32BIT_USED 1
#else
#define STREAM_FORMAT_INPUT_RESOLUTION_32BIT_USED 0
#endif
#if((FS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 4) || (HS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 4))
#define STREAM_FORMAT_INPUT_SUBSLOT_4_USED 1
#else
#define STREAM_FORMAT_INPUT_SUBSLOT_4_USED 0
#endif
#if((FS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 4) || (HS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 4))
#define STREAM_FORMAT_INPUT_SUBSLOT_4_USED 1
#else
#define STREAM_FORMAT_INPUT_SUBSLOT_4_USED 0
#endif
#if((FS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 3) || (HS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 3))
#define STREAM_FORMAT_INPUT_SUBSLOT_3_USED 1
#else
#define STREAM_FORMAT_INPUT_SUBSLOT_3_USED 0
#endif
#if((FS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 3) || (HS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 3))
#define STREAM_FORMAT_INPUT_SUBSLOT_3_USED 1
#else
#define STREAM_FORMAT_INPUT_SUBSLOT_3_USED 0
#endif
#if((FS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 2) || (HS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 2))
#define STREAM_FORMAT_INPUT_SUBSLOT_2_USED 1
#else
#define STREAM_FORMAT_INPUT_SUBSLOT_2_USED 0
#endif
#if((FS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 2) || (HS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES == 2))
#define STREAM_FORMAT_INPUT_SUBSLOT_2_USED 1
#else
#define STREAM_FORMAT_INPUT_SUBSLOT_2_USED 0
#endif
#if MAX_FREQ < MIN_FREQ
#error MAX_FREQ should be >= MIN_FREQ!!

15
lib_xua/doc/rst/api_defines.rst
View File

@@ -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:: USE_SW_PLL
Audio Class
-----------

81
lib_xua/doc/rst/api_user_functions.rst
View File

@@ -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

2
lib_xua/doc/rst/xdoc.conf
View File

@@ -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

22
lib_xua/src/core/audiohub/xua_audiohub.xc
View File

@@ -20,6 +20,7 @@
#include "xua.h"
#include "audiohw.h"
#include "audioports.h"
#include "mic_array_conf.h"
#if (XUA_SPDIF_TX_EN)
@@ -635,6 +636,9 @@ void XUA_AudioHub(chanend ?c_aud, clock ?clk_audio_mclk, clock ?clk_audio_bclk,
buffered _XUA_CLK_DIR port:32 ?p_bclk,
buffered out port:32 (&?p_i2s_dac)[I2S_WIRES_DAC],
buffered in port:32 (&?p_i2s_adc)[I2S_WIRES_ADC]
#if (XUA_USE_APP_PLL)
, client interface SoftPll_if i_softPll
#endif
#if (XUA_SPDIF_TX_EN) //&& (SPDIF_TX_TILE != AUDIO_IO_TILE)
, chanend c_spdif_out
#endif
@@ -663,6 +667,12 @@ void XUA_AudioHub(chanend ?c_aud, clock ?clk_audio_mclk, clock ?clk_audio_bclk,
unsigned divide;
unsigned firstRun = 1;
#if (XUA_USE_APP_PLL)
/* Use xCORE.ai Secondary PLL to generate master clock
* This could be "fixed" for async mode or adjusted if in sync mode */
i_softPll.init(DEFAULT_MCLK);
#endif
/* Clock master clock-block from master-clock port */
/* Note, marked unsafe since other cores may be using this mclk port */
configure_clock_src(clk_audio_mclk, p_mclk_in);
@@ -800,6 +810,15 @@ 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();
#if (XUA_USE_APP_PLL)
i_softPll.init(mClk);
#endif
/* User code should configure audio harware for SampleFreq/MClk etc */
AudioHwConfig(curFreq, mClk, dsdMode, curSamRes_DAC, curSamRes_ADC);
#if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
/* Notify clockgen of new mCLk */
@@ -809,6 +828,9 @@ void XUA_AudioHub(chanend ?c_aud, clock ?clk_audio_mclk, clock ?clk_audio_bclk,
/* Wait for ACK back from clockgen to signal clocks all good */
c_mclk_change :> int _;
#endif
/* User should unmute audio hardware */
AudioHwConfig_UnMute();
}
if(!firstRun)

56
lib_xua/src/core/buffer/ep/ep_buffer.xc
View File

@@ -105,7 +105,11 @@ void XUA_Buffer(
#endif
, chanend c_aud
#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
#if(XUA_USE_APP_PLL)
, chanend c_swpll_update
#else
, client interface pll_ref_if i_pll_ref
#endif
#endif
)
{
@@ -141,7 +145,11 @@ void XUA_Buffer(
, c_buff_ctrl
#endif
#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
, i_pll_ref
#if(XUA_USE_APP_PLL)
, c_swpll_update
#else
, i_pll_ref
#endif
#endif
);
@@ -190,8 +198,12 @@ 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)
#if (XUA_USE_APP_PLL)
, chanend c_swpll_update
#else
, client interface pll_ref_if i_pll_ref
#endif
#endif
)
{
@@ -247,7 +259,8 @@ 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;
int pllUpdate = 0;
unsigned mod_from_last_time = 0;
#ifdef FB_TOLERANCE_TEST
@@ -294,7 +307,6 @@ 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
@@ -357,12 +369,16 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
#ifndef LOCAL_CLOCK_MARGIN
#define LOCAL_CLOCK_MARGIN (1000)
#endif
#if (!XUA_USE_APP_PLL)
timer t_sofCheck;
unsigned timeLastEdge;
unsigned timeNextEdge;
t_sofCheck :> timeLastEdge;
timeNextEdge + LOCAL_CLOCK_INCREMENT;
i_pll_ref.toggle();
#endif
#endif
while(1)
@@ -427,7 +443,8 @@ 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;
pllUpdate = 0;
clocks = 0;
clockcounter = 0;
mod_from_last_time = 0;
@@ -502,13 +519,13 @@ 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_APP_PLL)
case t_sofCheck when timerafter(timeNextEdge) :> void:
i_pll_ref.toggle();
timeLastEdge = timeNextEdge;
@@ -528,7 +545,6 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
unsigned usbSpeed;
int framesPerSec;
GET_SHARED_GLOBAL(usbSpeed, g_curUsbSpeed);
static int sofCount = 0;
framesPerSec = (usbSpeed == XUD_SPEED_HS) ? 8000 : 1000;
@@ -539,12 +555,27 @@ void XUA_Buffer_Ep(register chanend c_aud_out,
sofCount += 1000;
if (sofCount == framesPerSec)
{
sofCount = 0;
pllUpdate++;
#if (!XUA_USE_APP_PLL)
/* 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
}
#if (XUA_USE_APP_PLL)
// Update PLL @ 100Hz
if(pllUpdate == 10)
{
pllUpdate = 0;
unsigned short mclk_pt;
asm volatile("getts %0, res[%1]" : "=r" (mclk_pt) : "r" (p_off_mclk));
outuint(c_swpll_update, mclk_pt);
outct(c_swpll_update, XS1_CT_END);
}
#endif
#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 +677,6 @@ 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.
@@ -897,8 +927,8 @@ 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 +941,7 @@ 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)
{

33
lib_xua/src/core/main.xc
View File

@@ -316,6 +316,9 @@ void usb_audio_io(chanend ?c_aud_in,
, client interface pll_ref_if i_pll_ref
, port ?p_for_mclk_count_aud
#endif
#if (XUA_USE_APP_PLL)
, client interface SoftPll_if i_softPll
#endif
)
{
#if (MIXER)
@@ -373,6 +376,9 @@ void usb_audio_io(chanend ?c_aud_in,
#define AUDIO_CHANNEL c_aud_in
#endif
XUA_AudioHub(AUDIO_CHANNEL, clk_audio_mclk, clk_audio_bclk, p_mclk_in, p_lrclk, p_bclk, p_i2s_dac, p_i2s_adc
if (XUA_USE_APP_PLL)
, i_softPll
#endif
#if (XUA_SPDIF_TX_EN) //&& (SPDIF_TX_TILE != AUDIO_IO_TILE)
, c_spdif_tx
#endif
@@ -483,9 +489,14 @@ 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_APP_PLL) || XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
interface pll_ref_if i_pll_ref;
#endif
#if (XUA_USE_APP_PLL)
interface SoftPll_if i_softPll;
chan c_swpll_update;
#endif
chan c_sof;
chan c_xud_out[ENDPOINT_COUNT_OUT]; /* Endpoint channels for XUD */
chan c_xud_in[ENDPOINT_COUNT_IN];
@@ -507,7 +518,7 @@ 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_APP_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]:
@@ -520,6 +531,10 @@ int main()
DFUHandler(dfuInterface, null);
#endif
#if (XUA_USE_APP_PLL)
//XUA_SoftPll(tile[0], i_softPll, c_swpll_update);
#endif
/* Core USB task, buffering, USB etc */
{
#ifdef XUD_PRIORITY_HIGH
@@ -578,7 +593,13 @@ int main()
#endif
, c_mix_out
#if (XUA_SYNCMODE == XUA_SYNCMODE_SYNC)
#if (!XUA_USE_APP_PLL)
, i_pll_ref
#else
, c_swpll_update
#endif
#endif
#endif
);
//:
@@ -593,6 +614,10 @@ int main()
#endif /* XUA_USB_EN */
}
#if(XUA_USE_APP_PLL)
on tile[AUDIO_IO_TILE]: XUA_SoftPll(tile[0], i_softPll, c_swpll_update);
#endif
on tile[AUDIO_IO_TILE]:
{
@@ -616,6 +641,10 @@ int main()
#endif
#if (XUA_SPDIF_RX_EN || XUA_ADAT_RX_EN)
, i_pll_ref
#endif
if (XUA_USE_APP_PLL)
, i_softPll
#endif
, p_for_mclk_count_audio
#endif
);

29
lib_xua/src/core/user/audiohw/audiohw.c Normal file
View File

@@ -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;
}

54
lib_xua/src/core/user/audiohw/audiohw.h Normal file
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@@ -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

45
lib_xua/src/core/user/audiostream/audiostream.h
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@@ -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

25
lib_xua/src/core/user/hid/user_hid.h
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@@ -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_ */

10
lib_xua/src/core/user/hostactive/hostactive.h
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@@ -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);

2
tests/test_sync_clk_basic/Makefile
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@@ -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_APP_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_APP_PLL=0 \
$(TEST_FLAGS)
TARGET = test_xs3_600.xn