API - USB Audio

7.1 配置定义

使用USB音频框架的应用程序需要为配置设置了定义。这些定义的默认值可以在devicedefines.h中的module_usb_audio中找到。

这些定义应该在 customdefines.h文件或相关构建配置的Makefile中被覆盖。

本节完整记录了所有分段定义及其默认值（在合适的地方）。

7.1.1 代码位置(tile)

Macro	AUDIO_IO_TILE
Description	Location (tile) of audio I/O. Default: 0

Macro	XUD_TILE
Description	Location (tile) of audio I/O. Default: 0

Macro	AUDIO_IO_TILE
Description	Location (tile) of IAP. Default: AUDIO_IO_TILE

Macro	AUDIO_IO_TILE
Description	Location (tile) of MIDI I/O. Default: AUDIO_IO_TILE

7.1.2 通道计数

Macro	NUM_USB_CHAN_OUT
Description	Number of output channels (host to device). Default: NONE (Must be defined by app)

Macro	NUM_USB_CHAN_IN
Description	Number of input channels (device to host). Default: NONE (Must be defined by app)

Macro	DSD_CHANS_DAC
Description	Number of DSD output channels. Default: 0 (disabled)

Macro	I2S_CHANS_DAC
Description	Number of IS2 channels to DAC/CODEC. Must be a multiple of 2. Default: NONE (Must be defined by app)

Macro	I2S_CHANS_ADC
Description	Number of I2S channels from ADC/CODEC. Must be a multiple of 2. Default: NONE (Must be defined by app)

7.1.3 频率和时钟

Macro	MAX_FREQ
Description	Max supported sample frequency for device (Hz). Default: 192000

Macro	MIN_FREQ
Description	Min supported sample frequency for device (Hz). Default 44100

Macro	DEFAULT_FREQ
Description	Default device sample frequency. A safe default should be used. Default: MIN_FREQ

Macro	MCLK_441
Description	Master clock defines for 44100 rates (in Hz). Default: NONE (Must be defined by app)

Macro	MCLK_48
Description	Master clock defines for 48000 rates (in Hz). Default: NONE (Must be defined by app)

7.1.4 Audio Class

Macro	AUDIO_CLASS
Description	USB Audio Class Version. Default: 2 (Audio Class version 2.0)

Macro	AUDIO_CLASS_FALLBACK
Description	Whether or not to fall back to Audio Class 1.0 in USB Full-speed. Default: 0 (Disabled)

Macro	FULL_SPEED_AUDIO_2
Description	Whether or not to run UAC2 in full-speed. When disabled device can either operate in UAC1 mode in full-speed (if AUDIO_CLASS_FALLBACK enabled) or return “null” descriptors. Default: 1 (Enabled) when AUDIO_CLASS_FALLBACK disabled.

7.1.5 系统功能配置

7.1.5.1 MIDI

Macro	MIDI
Description	Enable MIDI functionality including buffffering, descriptors etc. Default: DISABLED

Macro	MIDI_RX_PORT_WIDTH
Description	MIDI Rx port width (1 or 4bit). Default: 1

7.1.5.2 S/PDIF

Macro	SPDIF_TX
Description	Enables SPDIF Tx. Default: 0 (Disabled)

Macro	SPDIF_TX_INDEX
Description	Defines which output channels (stereo) should be output on S/PDIF. Note, Output channels indexed from 0. Default: 0 (i.e. channels 0 & 1)

Macro	SPDIF_RX
Description	Enables SPDIF Rx. Default: 0 (Disabled)

Macro	SPDIF_RX_INDEX
Description	S/PDIF Rx first channel index, defines which channels S/PDIF will be input on. Note, indexed from 0. Default: NONE (Must be defined by app when SPDIF_RX enabled)

7.1.5.3 ADAT

Macro	ADAT_RX
Description	Enables ADAT Rx. Default: 0 (Disabled)

Macro	ADAT_RX_INDEX
Description	ADAT Rx first channel index. defines which channels ADAT will be input on. Note, indexed from 0. Default: NONE (Must be defined by app when ADAT_RX enabled)

7.1.5.4 PDM Microphones

Macro	NUM_PDM_MICS
Description	Number of PDM microphones in the design. Default: None

7.1.5.5 DFU

Macro	DFU
Description	Enable DFU functionality. A driver required for Windows operation. Default: 1 (Enabled)

7.1.5.6 HID

Macro	HID_CONTROLS
Description	Enable HID playback controls functionality. 1 for enabled, 0 for disabled. Default 0 (Disabled)

7.1.5.7 CODEC Interface

Macro	CODEC_MASTER
Description

7.1.6 USB Device Configuration

Macro	VENDOR_STR
Description	Vendor String used by the device. This is also pre-pended to various strings used by the design. Default: “XMOS”

Macro	VENDOR_ID
Description	USB Vendor ID (or VID) as assigned by the USB-IF. Default: 0x20B1 (XMOS)

Macro	PRODUCT_STR
Description	USB Product String for the device. If defined will be used for both PRODUCT_STR_A2 and PRODUCT_STR_A1. Default: Undefined

Macro	PRODUCT_STR_A2
Description	Product string for Audio Class 2.0 mode. Default: “xCore USB Audio 2.0”

Macro	PRODUCT_STR_A1
Description	Product string for Audio Class 1.0 mode. Default: “xCore USB Audio 1.0”

Macro	PID_AUDIO_1
Description	USB Product ID (PID) for Audio Class 1.0 mode. Only required if AUDIO_CLASS == 1 or AUDIO_CLASS_FALLBACK is enabled. Default: 0x0003

Macro	PID_AUDIO_2
Description	USB Product ID (PID) for Audio Class 2.0 mode. Default: 0x0002

Macro	BCD_DEVICE
Description	Device firmware version number in Binary Coded Decimal format: 0xJJMN where JJ: major, M: minor, N: sub-minor version number. NOTE: User code should not modify this but should modify BCD_DEVICE_J, BCD_DEVICE_M, BCD_DEVICE_N instead Default: XMOS USB Audio Release version (e.g. 0x0651 for 6.5.1).

7.1.7 流格式

7.1.7.1 Output/Playback

Macro	OUTPUT_FORMAT_COUNT
Description	Number of supported output stream formats. Values 1,2,3 supported. Default: 2

Macro	STREAM_FORMAT_OUTPUT_1_RESOLUTION_BITS
Description	Sample resolution (bits) of output stream Alternate 1. Default: 24 if Alternate 1 is PCM, else 32 if DSD/RAW. Note, 24 on the lowests alt in case of OUTPUT_FORMAT_COUNT = 1 leaving 24bit as the designs default resolution.

Macro	STREAM_FORMAT_OUTPUT_2_RESOLUTION_BITS
Description	Sample resolution (bits) of output stream Alternate 2. Default: 16 if Alternate 2 is PCM, else 32 if DSD/RAW.

Macro	STREAM_FORMAT_OUTPUT_3_RESOLUTION_BITS
Description	Sample resolution (bits) of output stream Alternate 3. Default: 32 if Alternate 2 is PCM, else 32 if DSD/RAW

Macro	HS_STREAM_FORMAT_OUTPUT_1_SUBSLOT_BYTES
Description	Sample sub-slot size (bytes) of output stream Alternate 1 when running in high-speed. Default: 4 if resolution for Alternate 1 is 24bits, else resolution / 8. Note, the default catchs the 24bit special case where 4-byte subslot is nicer for our 32-bit machine. Typically do not care about this extra bus overhead at High-speed.

Macro	HS_STREAM_FORMAT_OUTPUT_2_SUBSLOT_BYTES
Description	Sample sub-slot size (bytes) of output stream Alternate 2 when running in high-speed. Default: 4 if resolution for Alternate 2 is 24bits, else resolution / 8. Note, the default catchs the 24bit special case where 4-byte subslot is nicer for our 32-bit machine. Typically do not care about this extra bus overhead at High-speed.

Macro	HS_STREAM_FORMAT_OUTPUT_3_SUBSLOT_BYTES
Description	Sample sub-slot size (bytes) of output stream Alternate 3 when running in high-speed. Default: 4 if resolution for Alternate 3 is 24bits, else resolution / 8. Note, the default catchs the 24bit special case where 4-byte subslot is nicer for our 32-bit machine. Typically do not care about this extra bus overhead at High-speed.

Macro	FS_STREAM_FORMAT_OUTPUT_1_SUBSLOT_BYTES
Description	Sample sub-slot size (bytes) of output stream Alternate 1 when running in full-speed. Note, in full-speed mode bus bandwidth is at a premium, therefore pack samples into smallest possible sub-slot. Default: STREAM_FORMAT_OUTPUT_1_RESOLUTION_BITS / 8.

Macro	FS_STREAM_FORMAT_OUTPUT_2_SUBSLOT_BYTES
Description	Sample sub-slot size (bytes) of output stream Alternate 2 when running in full-speed. Note, in full-speed mode bus bandwidth is at a premium, therefore pack samples into smallest possible sub-slot. Default: STREAM_FORMAT_OUTPUT_2_RESOLUTION_BITS / 8

Macro	FS_STREAM_FORMAT_OUTPUT_3_SUBSLOT_BYTES
Description	Sample sub-slot size (bytes) of output stream Alternate 3 when running in full-speed. Note, in full-speed mode bus bandwidth is at a premium, therefore pack samples into smallest possible sub-slot. Default: STREAM_FORMAT_OUTPUT_3_RESOLUTION_BITS / 8

Macro	STREAM_FORMAT_OUTPUT_1_DATAFORMAT
Description	Sample audio data-format if output stream Alternate 1. Default: UAC_FORMAT_TYPEI_RAW_DATA when Alternate 1 is RAW/DSD else UAC_FORMAT_TYPEI_PCM

Macro	STREAM_FORMAT_OUTPUT_2_DATAFORMAT
Description	Sample audio data-format if output stream Alternate 2. Default: UAC_FORMAT_TYPEI_RAW_DATA when Alternate 2 is RAW/DSD else UAC_FORMAT_TYPEI_PCM

Macro	STREAM_FORMAT_OUTPUT_3_DATAFORMAT
Description	Sample audio data-format if output stream Alternate 3. Default: UAC_FORMAT_TYPEI_RAW_DATA when Alternate 3 is RAW/DSD else UAC_FORMAT_TYPEI_PCM

7.1.7.2 Input/Recording

Macro	INPUT_FORMAT_COUNT
Description	Number of supported input stream formats. Default: 1.

Macro	STREAM_FORMAT_INPUT_1_RESOLUTION_BITS
Description	Sample resolution (bits) of input stream Alternate 1. Default: 24.

Macro	HS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES
Description	Sample sub-slot size (bytes) of input stream Alternate 1 when running in high-speed. Default: 4 if resolution for Alternate 1 is 24bits, else resolution / 8. Note, the default catchs the 24bit special case where 4-byte subslot is nicer for our 32-bit machine. Typically do not care about this extra bus overhead at High-speed.

Macro	FS_STREAM_FORMAT_INPUT_1_SUBSLOT_BYTES
Description	Sample sub-slot size (bytes) of input stream Alternate 1 when running in full-speed. Note, in full-speed mode bus bandwidth is at a premium, therefore pack samples into smallest possible sub-slot. Default: STREAM_FORMAT_INPUT_1_RESOLUTION_BITS / 8.

Macro	STREAM_FORMAT_INPUT_1_DATAFORMAT
Description	Sample audio data-format for input stream Alternate 1. Default: UAC_FORMAT_TYPEI_PCM.

7.1.8 Volume Control

Macro	OUTPUT_VOLUME_CONTROL
Description	Enable/disable output volume control including all processing and descriptor support. Default: 1 (Enabled).

Macro	INPUT_VOLUME_CONTROL
Description	Enable/disable input volume control including all processing and descriptor support. Default: 1 (Enabled).

Macro	MIN_VOLUME
Description	The minimum volume setting above -inf. This is a signed 8.8 fixed point number that must be strictly greater than -128 (0x8000). Default: 0x8100 (-127db).

Macro	MAX_VOLUME
Description	The maximum volume setting. This is a signed 8.8 fixed point number. Default: 0x0000 (0db).

Macro	VOLUME_RES
Description	The resolution of the volume control in db as a 8.8 fixed point number. Default: 0x100 (1db).

7.1.9 Mixing Parameters

Macro	MIXER
Description	Enable “mixer” core. Default: 0 (Disabled)

Macro	MAX_MIX_COUNT
Description	Number of seperate mixes to perform. Default: 8 if MIXER enabled, else 0.

Macro	MIX_INPUTS
Description	Number of channels input into the mixer. Note, total number of mixer nodes is MIX_INPUTS * MAX_MIX_COUNT. Default: 18.

Macro	MIN_MIXER_VOLUME
Description	The minimum volume setting for the mixer unit above -inf. This is a signed 8.8 fixed point number that must be strictly greater than -128 (0x8000). Default: 0x8100 (-127db).

Macro	MAX_MIXER_VOLUME
Description	The maximum volume setting for the mixer. This is a signed 8.8 fixed point number. Default: 0x0000 (0db).

Macro	VOLUME_RES_MIXER
Description	The resolution of the volume control in db as a 8.8 fixed point number. Default: 0x100 (1db).

7.1.10 Power

Macro	SELF_POWERED
Description	Report as self to the host when enabled, else reports as bus-powered. This affffects descriptors and XUD usage. Default: 0 (Disabled).

Macro	BMAX_POWER
Description	Power drawn from the host (in mA x 2). Default: 0 when SELF_POWERED enabled else 250 (500mA).

7.2 所需的用户函数定义

以下功能需要由一个使用XMOS USB音频框架的应用程序来定义。

7.2.1 外部音频硬件配置功能

Function	AudioHwInit
Description	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.
Type	void AudioHwInit(chanend ?c_codec)
Parameters	c_codec An optional chanend that was original passed into audio() that can be used to communicate with other cores.

Function	AudioHwConfig
Description	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.
Type	void AudioHwConfig(unsigned samFreq, unsigned mclk, chanend ?c_codec, unsigned dsdMode, unsigned sampRes_DAC, unsigned sampRes_ADC)
Parameters	samFreq The sample frequency in Hz that the hardware should be configured to (in Hz). mclkThe master clock frequency that is required in Hz. c_codecAn optional chanend that was original passed into audio() that can be used to communicate with other cores. dsdModeSignifies if the audio hardware should be configured for DSD operation. sampRes_DACThe sample resolution of the DAC stream. sampRes_ADC The sample resolution of the ADC stream.

7.2.2 音频流功能

设计人员可选择使用以下功能。它们可以用于静音线等。

Function	AudioStreamStart
Description	This function is called when the audio stream from device to host starts.
Type	void AudioStreamStart(void)
Parameters

Function	AudioStreamStop
Description	This function is called when the audio stream from device to host stops.
Type	void AudioStreamStop(void)

7.2.3 主机激活

以下功能可用于指示设备已连接到有效主机的信号。这是对状态的改变的。

Function	AudioStreamStart
Description
Type	void AudioStreamStart(int active)
Parameters	active Indicates if the host is active or not. 1 for active else 0.

7.2.4 HID Controls

当设备希望读取物理用户输入（按钮等）时，将调用以下函数。

Function	UserReadHIDButtons
Description
Type	void UserReadHIDButtons(unsigned char hidData[])
Parameters	hidData The function should write relevant HID bits into this array. The bit ordering and functionality is defined by the HID report descriptor used.

7.3 组件API

可以从应用程序的顶级主节点调用以下函数，并实现3.1中描述的各种组件。

Function	XUD_Manager
Description	This performs the low-level USB I/O operations. Note that this needs to run in a thread with at least 80 MIPS worst case execution speed.
Type	int XUD_Manager(chanend c_epOut[],int noEpOut,chanend c_epIn[],int noEpIn,chanend ?c_sof,XUD_EpType epTypeTableOut[],XUD_EpType epTypeTableIn[],out port ?p_usb_rst,clock ?clk,unsigned rstMask,XUD_BusSpeed_t desiredSpeed,XUD_PwrConfig pwrConfig)
Parameters	c_epOutAn array of channel ends, one channel end per output endpoint (USB OUT transaction); this includes a channel to obtain requests on Endpoint 0. noEpOutThe number of output endpoints, should be at least 1 (for Endpoint 0). c_epInAn array of channel ends, one channel end per input endpoint (USB IN transaction); this includes a channel to respond to requests on Endpoint 0. noEpInThe number of input endpoints, should be at least 1 (for Endpoint 0). c_sofA channel to receive SOF tokens on. This channel must be connected to a process that can receive a token once every 125 ms. If tokens are not read, the USB layer will lock up. If no SOF tokens are required null should be used for this parameter. epTypeTableOutSeeepTypeTableIn. epTypeTableIn.This andepTypeTableOut are two arrays indicating the type of the endpoint. Legal types include: XUD_EPTYPE_CTL (Endpoint 0), XUD_EPTYPE_BUL (Bulk endpoint), XUD_EPTYPE_ISO (Isochronous endpoint), XUD_EPTYPE_INT (Interrupt endpoint), XUD_EPTYPE_DIS (Endpoint not used). The first array contains the endpoint types for each of the OUT endpoints, the second array contains the endpoint types for each of the IN endpoints. p_usb_rstThe port to used to connect to an external phy reset line. Should be null for U-Series. clkThe clock block to use for the p_usb_rst port - this should not be clock block 0. Should be null for U-Series. rstMaskThe mask to use when taking an external phy into/out of reset. The mask is ORed into the port to disable reset, and unset when deasserting reset. Use ‘-1’ as a default mask if this port is not shared. desiredSpeedThis parameter specifies what speed the device will attempt to run at i.e. full-speed (ie 12Mbps) or high-speed (480Mbps) if supported by the host. Pass XUD_SPEED_HS if high-speed is desired or XUD_SPEED_FS if not. Low speed USB is not supported by XUD. pwrConfigSpecifies whether the device is bus or self-powered. When self-powered the XUD will monitor the VBUS line for host disconnections. This is required for compliance reasons. Valid values are XUD_PWR_SELF and XUD_PWR_BUS.

当使用USB音频框架时，c_ep_in数组总是按以下顺序组成：

Endpoint 0 (in)

• Audio Feedback endpoint (if output enabled)

• Audio IN endpoint (if input enabled)

• MIDI IN endpoint (if MIDI enabled)

• Clock Interrupt endpoint

数组c_ep_out总是按以下顺序组成：

• Endpoint 0 (out)

• Audio OUT endpoint (if output enabled)

• MIDI OUT endpoint (if MIDI enabled)

Function	Endpoint0
Description	Function implementing Endpoint 0 for enumeration, control and configuration of USB audio devices. It uses the descriptors defined in descriptors_2.h.
Type	void Endpoint0(chanend c_ep0_out,chanend c_ep0_in,chanend c_audioCtrl,chanend ?c_mix_ctl,chanend ?c_clk_ctl,chanend ?c_EANativeTransport_ctr,client interface i_dfu dfuInterface)
Parameters	c_ep0_outChanend connected to the XUD_Manager() out endpoint array c_ep0_inChanend connected to the XUD_Manager() in endpoint array c_audioCtrlChanend connected to the decouple thread for control audio (sample rate changes etc.) c_mix_ctlOptional chanend to be connected to the mixer thread if present c_clk_ctlOptional chanend to be connected to the clockgen thread if present. c_usb_testOptional chanend to be connected to XUD if test modes required. c_EANativeTransport_ctrlOptional chanend to be connected to EA Native endpoint manager if present

Function	buffffer
Description	USB Audio Buffffering Thread. This function buffffers USB audio data between the XUD layer and the decouple thread. Most of the chanend parameters to the function should be connected to XUD_Manager()
Type	void buffer(chanend c_aud_out,chanend c_aud_in,chanend c_aud_fb,chanend c_sof,chanend c_aud_ctl,in port p_off_mclk)
Parameters	c_aud_outAudio OUT endpoint channel connected to the XUD c_aud_inAudio IN endpoint channel connected to the XUD c_aud_fbAudio feedback endpoint channel connected to the XUD c_midi_from_hostMIDI OUT endpoint channel connected to the XUD c_midi_to_hostMIDI IN endpoint channel connected to the XUD c_intAudio clocking interrupt endpoint channel connected to the XUD c_clk_intOptional chanend connected to the clockGen() thread if present c_sofStart of frame channel connected to the XUD c_aud_ctlAudio control channel connected to Endpoint0() p_off_mclkA port that is clocked of the MCLK input (not the MCLK input itself)

Function	decouple
Description	Manage the data transfer between the USB audio buffffer and the Audio I/O audio driver.
Type	void decouple(chanend c_audio_out)
Parameters	c_audio_outChannel connected to the audio() or mixer() threads

Function	mixer
Description	Digital sample mixer. This thread mixes audio streams between the decouple() and the audio() thread.
Type	void mixer(chanend c_to_host, chanend c_to_audio, chanend c_mix_ctl)
Parameters	c_to_hosta chanend connected to the decouple() thread for receiving/transmitting samples c_to_audioa chanend connected to the audio() thread for receiving/transmitting samples c_mix_ctla chanend connected to the Endpoint0() thread for receiving control commands

Function	audio
Description	The audio driver thread. This function drives I2S ports and handles samples to/from other digital I/O threads.
Type	void audio(chanend c_in, chanend ?c_config, chanend ?c_adc)
Parameters	c_inAudio sample channel connected to the mixer() thread or the decouple() thread c_digchannel connected to the clockGen() thread for receiving/transmitting samples c_configAn optional channel that will be passed on to the CODEC configuration functions.

Function	clockGen
Description	Clock generation and digital audio I/O handling.
Type	void clockGen(streaming chanend ?c_spdif_rx,chanend ?c_adat_rx,out port p,chanend c_audio,chanend c_clk_ctl,chanend c_clk_int)
Parameters	c_spdif_rxchannel connected to S/PDIF receive thread. c_adat_rxchannel connect to ADAT receive thread. pport to output clock signal to drive external frequency synthesizer. c_audiochannel connected to the audio() thread. c_clk_ctlchannel connected to Endpoint0() for configuration of the clock. c_clk_intchannel connected to the decouple() thread for clock interrupts

Function	SpdifReceive
Description	S/PDIF receive function. This function needs 1 thread and no memory other than ~2800 bytes of program code. It can do 11025, 12000, 22050, 24000, 44100, 48000, 88200, 96000, and 192000 Hz. When the decoder encounters a long series of zeros it will lower the divider; when it encounters a short series of 0-1 transitions it will increase the divider. Output: the received 24-bit sample values are output as a word on the streaming channel end. Each value is shifted up by 4-bits with the bottom four bits being one of FRAME_X, FRAME_Y, or FRAME_Z. The bottom four bits should be removed whereupon the sample value should be sign extended. The function does not return unless compiled with TEST defined in which case it returns any time that it loses synchronisation.
Type	void SpdifReceive(in buffered port:4 p,streaming chanend c,int initial_divider,clock clk)
Parameters	pS/PDIF input port. This port must be 4-bit buffffered, declared as in buffered port:4 cchannel to output samples to initial_dividerinitial divide for initial estimate of sample rate For a 100Mhz reference clock, use an initial divider of 1 for 192000, 2 for 96000/88200, and 4 for 48000/44100. clkclock block sourced from the 100 MHz reference clock.

Function	adatReceiver48000
Description	ADAT Receive Thread (48kHz sample rate). When a data rame is received, samples will be output onto the streaming channel At first a word 0x000000Z1 will be output, where Z are the user data; after that eight words 0x0ZZZZZZ0 will be output where ZZZZZZ is a 24-bit sample value. The eight words may refer to sample values on eight channels, or on fewer channels if muxing is used. The function will return if it cannot lock onto a 48,000 Hz signal. Normally the 48000 function is called in a while(1) loop. If both 44,100 and 48,000 need to be supported, they should be called in sequence in a while(1) loop. Note that the functions are large, and that 44,100 should not be called if 44.1 KHz does not need to be supported.
Type	void adatReceiver48000(buffered in port:32 p, chanend oChan)
Parameters	pADAT port - should be 1-bit and clocked at 100MHz oChanchannel on which decoded samples are output

Function	adatReceiver44100
Description	ADAT Receive Thread (44.1kHz sample rate). When a data rame is received, samples will be output onto the streaming channel At first a word 0x000000Z1 will be output, where Z are the user data; after that eight words 0x0ZZZZZZ0 will be output where ZZZZZZ is a 24-bit sample value. The eight words may refer to sample values on eight channels, or on fewer channels if muxing is used. The function will return if it cannot lock onto a 44,100 Hz signal. Normally the 44,100 function is called in a while(1) loop. If both 44,100 and 48,000 need to be supported, they should be called in sequence in a while(1) loop. Note that the functions are large, and that 48,000 should not be called if 48 Khz does not need to be supported.
Type	void adatReceiver44100(buffered in port:32 p, chanend oChan)
Parameters	pADAT port - should be 1-bit and clocked at 100MHz oChanchannel on which decoded samples are output

Function	usb_midi
Description	USB MIDI I/O thread. This function passes MIDI data from USB to UART I/O.
Type	void usb_midi(buffered in port:1 ?p_midi_in, port ?p_midi_out, clock ?clk_midi, chanend ?c_midi, unsigned cable_number, chanend ?c_iap, chanend ?c_i2c, port ?p_scl, port ?p_sda)
Parameters	p_midi_in1-bit input port for MIDI p_midi_out1-bit output port for MIDI clk_midiclock block used for clockin the UART; should have a rate of 100MHz c_midichanend connected to the decouple() thread cable_numberthe cable number of the MIDI implementation. This should be set to 0.