iekc64_audio.h

一个小的测试程序

#ifndef __AUDIO_H__
#define __AUDIO_H__

#include <csl_stdinc.h>
#include <csl_edma.h>
#include "iekc64_error.h"
#include "iekc64.h"

#ifdef __cplusplus
extern "C" {
#endif


/*=======================================================================*//*!
\defgroup AUDIO AUDIO - Audio recording and playing

	\brief This module concerns for capturing and generating audio.

   <h3> Audio module </h3>

	<b> How does it work </b>
	
	The AUDIO module captures audio into frames and play frames.
	
	For record, frames are allocated contiguously in a circular buffer 
	allocated by the user before openning the module. This buffer must be large
	enough to allow at least 4 frames to be stored. Using a 4 frames buffer is the
	best way to insure that you won't have any overlapping between the
	background capture and the processing.
	If your process needs to work on more than 1 frame, you don't need to copy
	frames to another  buffer, you just have to create a buffer large enough
	and tell the AUDIO module to always leave the last N frames unmodified.
	
	The AUDIO playing generates audio from frames provided by the user.
	Frames are allocated by the user. Their memory location is non important,
	as far as each frame start is aligned on a 32 bit word boudary. Frames
	for audio generation don't need to be contiguously allocated in the
	same buffer like the ones used for audio record. They even doesn't need
	to be allocated before openning the module. The user can allocate any
	number of frames from 1 to memory limits.
	When using 1 single frame, it is possible to work into the frame while it
	is played. Depending on the refresh rate that can cause some artifacts.
	
	<b> Openning the module </b>

	You open the module by calling the AUDIO_open() function. This function
	initializes hardware and software with parameters given in the
	IEKC64_AUDIO structure. You must use the default configuration, 
	IEKC64_AUDIO_CONFIG_DEFAULT, to initialize your own configuration but the 
	fields FramesLength, nFramesInBuf, nFramesToKeep and pBuffer	must be set 
	before calling the AUDIO_open() function.
	The default configuration structure field description is:

<small>
  dwSize = sizeof(IEKC64_AUDIO) <br>
  Source = SOURCE_LINE <br>
	{ <br>
		isDACSoftMuteEnabled 				= FALSE 					// No software mute <br>
		DeEmphasisCtrl 						 	= AUDIO_DEEMP48K	// de-emphasis filter <br>
		isADCHighPassFilterEnabled 	= TRUE						// analog to digital filter enabled <br>
	}<br>
  SampleRate				= AUDIO_SAMPLERATE_44K1 <br>
  GainInLeft				= 6 <br>
  GainInRigth				= 6 <br>
  GainOutLeft				= 6 <br>
  GainOutRigth			= 6 <br>
  LoopBack					= FALSE <br>
  FramesLength			= 0 <br>
  nFramesInBuf			= 0 <br>
  nFramesToKeep			= 0 <br>
  pBuffer						= NULL <br>
  dwAudioInEdmaPri  = EDMA_OPT_PRI_URGENT <br>
  dwAudioOutEdmaPri = EDMA_OPT_PRI_URGENT <br>
</small>
	
	<b> Starting to record </b>
	
	A call to AUDIO_startRecord() will activate the record process. The audio signal
	is digitized. The DMA mechanism is hidden to the user by IEKLIB.
	Each new audio frame is stored into next frame in the buffer.

	<b> Getting next frame </b>
	
	AUDIO_getFrame() will give the next frame pointer from the input buffer.
	If no new frame is avalable, depending on the timeout value, you will
	get an error or wait until the timeout ends or a new frame is captured.
	
	<b> Starting to play </b>
	
	A call to AUDIO_startPlay() will activate the playing process.
	It will pre-activate the playing process. Their will be no audio generation
	until the first call to AUDIO_putFrame().
	
	<b> Outputing a frame </b>

	With AUDIO_putFrame() the user provides the pointer to the next frame
	to be generated.
	The frame data will be send to the hardware using DMA. DMA mechanism is
	hidden to the user by IEKLIB.

	<b> Ending audio record and play </b>
	
	The AUDIO_close() free all DMA and interrupt used by the IEKLIB.


</P>*//*==============================================================*//*@{*/

// AUDIO register, I2C register on Byte for writing
#define AUDIO_REG_INPUTVOLL     0x00
#define AUDIO_REG_INPUTVOLR     0x01
#define AUDIO_REG_HPVOLL        0x02
#define AUDIO_REG_HPVOLR        0x03
#define AUDIO_REG_ANALOGCTRL    0x04
#define AUDIO_REG_DIGITALCTRL   0x05
#define AUDIO_REG_POWERCTRL     0x06
#define AUDIO_REG_DIGITALFORMAT 0x07
#define AUDIO_REG_SAMPLERATE    0x08
#define AUDIO_REG_DIGITALACTIVE 0x09
#define AUDIO_REG_RESET         0x0F

// Register bits for  AUDIO_REG_INPUTVOLL and AUDIO_REG_INPUTVOLR
#define AUDIO_LIV           0x003F
#define AUDIO_LIM           0x0080
#define AUDIO_LRS           0x0100

#define AUDIO_POS_LIV           0

#define AUDIO_ZERO_DB_IN	0x17

// Register bits for  AUDIO_REG_HPVOLL and AUDIO_REG_HPVOLR
#define AUDIO_LHV           0x007F
#define AUDIO_LZC           0x0080
#define AUDIO_HLRS          0x0100

#define AUDIO_POS_LHV           0

#define AUDIO_ZERO_DB_HP	0x79

// Register bits for  AUDIO_REG_ANALOGCTRL 
#define AUDIO_MICB          0x0001
#define AUDIO_MICM          0x0002
#define AUDIO_INSEL         0x0004
#define AUDIO_BYP           0x0008
#define AUDIO_DACSEL        0x0010
#define AUDIO_STE           0x0020
#define AUDIO_STA           0x00C0

#define AUDIO_STA6DB        0x0000
#define AUDIO_STA9DB        0x0040
#define AUDIO_STA12DB       0x0080
#define AUDIO_STA15DB       0x00C0

// Register bits for  AUDIO_REG_DIGITALCTRL 
#define AUDIO_ADCHP         0x0001
#define AUDIO_DEEMP         0x0006
#define AUDIO_DACM          0x0008

#define AUDIO_DEEMPNONE	        0x0000
#define AUDIO_DEEMP32K	        0x0001
#define AUDIO_DEEMP44K1 	    0x0002
#define AUDIO_DEEMP48K     		0x0003

// Register bits for AUDIO_REG_POWERCTRL
#define AUDIO_LINE          0x0001
#define AUDIO_MIC           0x0002
#define AUDIO_ADC           0x0004
#define AUDIO_DAC           0x0008
#define AUDIO_OUT           0x0010
#define AUDIO_OSC           0x0020
#define AUDIO_CLK           0x0040
#define AUDIO_OFF           0x0080

// Register bits for  AUDIO_REG_DIGITALFORMAT 
#define AUDIO_FOR           0x0003
#define AUDIO_IWL           0x000C
#define AUDIO_LRP           0x0010
#define AUDIO_LRSWAP        0x0020
#define AUDIO_MS            0x0040

#define AUDIO_FORRIGHT      0x0000
#define AUDIO_FORLEFT       0x0001
#define AUDIO_FORI2S        0x0002
#define AUDIO_FORDSP        0x0003

#define AUDIO_IWL16         0x0000
#define AUDIO_IWL20         0x0004
#define AUDIO_IWL24         0x0008
#define AUDIO_IWL32         0x000C

// Register bits for  AUDIO_REG_SAMPLERATE 
#define AUDIO_USB           0x0001
#define AUDIO_BOSR          0x0002
#define AUDIO_SR0           0x0004
#define AUDIO_SR1           0x0008
#define AUDIO_SR2           0x0010
#define AUDIO_SR3           0x0020
#define AUDIO_CLKIN         0x0040
#define AUDIO_CLKOUT        0x0080

#define AUDIO_96KHZ			AUDIO_SR0 | AUDIO_SR1 | AUDIO_SR2 
#define AUDIO_48KHZ			0x0000
#define AUDIO_32KHZ			AUDIO_SR1 | AUDIO_SR2 
#define AUDIO_8KHZ			AUDIO_SR0 | AUDIO_SR1 

#define AUDIO_POS_SR            2

// Register bits for  AUDIO_REG_DIGITALACTIVE
#define AUDIO_ACT           0x0001



/*--------------------------------------------------------------------------*/
/*! Special timeout values 
*/
enum IEKC64_AUDIO_WAIT
{
	//! Tell the API not to wait until the function is complete
	IEKC64_AUDIO_NO_WAIT = 0,
	//! Tell the API to wait until the function complete
	IEKC64_AUDIO_WAIT_INFINITE = -1  
};


/*--------------------------------------------------------------------------*/
/*! Defines the audio input used
*/
typedef enum
{
	//! \Select line source input mode
  SOURCE_LINE,
	//! \Select mic source mode
  SOURCE_MIC,
	//! \Select mic
  SOURCE_MIC20DB
} 
IEKC64_AUDIO_SOURCE;


/*--------------------------------------------------------------------------*/
/*! Defines the audio sample rate used
*/
typedef enum
{
  //! \Select sample rate to 8Khz (voice)
  AUDIO_SAMPLERATE_8K,
  //! \Select sample rate to 11.025Khz
  AUDIO_SAMPLERATE_11K025,
  //! \Select sample rate to 12Khz
  AUDIO_SAMPLERATE_12K,
  //! \Select sample rate to 16Khz
  AUDIO_SAMPLERATE_16K,
  //! \Select sample rate to 22,050Khz
  AUDIO_SAMPLERATE_22K05,
  //! \Select sample rate to 24Khz
  AUDIO_SAMPLERATE_24K,
  //! \Select sample rate to 32Khz
  AUDIO_SAMPLERATE_32K,
  //! \Select sample rate to 44.1Khz
  AUDIO_SAMPLERATE_44K1,
  //! \Select sample rate to 48Khz
  AUDIO_SAMPLERATE_48K,
  //! \Select sample rate to 96Khz
  AUDIO_SAMPLERATE_96K
} 
IEKC64_AUDIO_SAMPLERATE;


/*--------------------------------------------------------------------------*/
/*! Codec specific parameters structure
*/
typedef struct
{
	/*! This field define if software mute should be enabled
	*/
	Bool		isDACSoftMuteEnabled;
	/*! This field define de-emphasis filter
	*/
	Uint8		DeEmphasisCtrl;				
	/*! This field define if the analog to digital filter should be enabled
	*/
	Bool		isADCHighPassFilterEnabled;
} 
IEKC64_AUDIO_CODEC_CONF;


/*--------------------------------------------------------------------------*/
/*! Initialization parameters for AUDIO_open().
	 All fields have to be set before the call.
*/
typedef struct 
{
	/*! This field must contains the IEKC64_AUDIO structure size 
	*/
	Uint32 dwSize;
	/*! This field defines the input source from which you want to acquire audio
	*/
  IEKC64_AUDIO_SOURCE	Source;
	/*! This field defines all parameters for digital audio
	*/
  IEKC64_AUDIO_CODEC_CONF	DigitalAudio;
	/*! This field defines the sample rate
	*/
  IEKC64_AUDIO_SAMPLERATE	SampleRate;
	/*! Left line input channel volume control +12 dB down -34.5 dB in 1.5-dB steps
	*/
  Int32				GainInLeft;
	/*! Right line input channel volume control +12 dB down -34.5 dB in 1.5-dB steps
	*/
  Int32				GainInRigth;
	/*! Left channel headphone volume control +6 dB down to -73 dB in 1-dB steps
	*/
  Int32				GainOutLeft;
	/*! Right channel headphone volume control +6 dB down to -73 dB in 1-dB steps
	*/
  Int32				GainOutRigth;
	/*! This field define if the audio codec is in loopback mode
	*/
  Uint8					LoopBack;
	/*! This field tells the AUDIO module the size to capture before overrun
	*/
  Uint32				FramesLength;
	/*! This field defines the length of the capture buffer in number of frames size
	*/
  Uint32 		    	nFramesInBuffer;	
	/*! This field tells the AUDIO module how many frames have to be kept
	    in the capture buffer before overrun.  It must be greater or eaqual 
	    than nFramesInBuffer + 3.
	*/
  Uint32 				nFramesToKeep;
	/*! This field tells to the AUDIO module where begins the buffer to
	    store captured frames.
	*/
  Uint32*				pBuffer;	
  /*! This field defines to the AUDIO input EDMA priority