codec.h File Reference

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Detailed Description

Codec module C header file.

REVISION

• 11/03/2003 M.GAO
  Initial creation.
• 02/13/2004 M.GAO
  LinkIt style MyFi hidden feature support.

Definition in file codec.h.

#include "../feature.h"
#include "../PIPE/pipe.h"
#include "../GENERIC/generic.h"
#include "../SYSTEM/sys_cpld.h"
#include "aic23.h"

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Data Structures
struct	st_PCM
	PCM_Globals structure definition. More...
Enumerations
enum	COP_MODE {   COP_MUSIC = 0x0001, COP_TUNER = 0x0002, COP_FMXMIT = 0x0004, COP_ILINK = 0x0008, COP_MIC_RECORD = 0x0010, COP_LINE_RECORD = 0x0020,   COP_FM_RECORD = 0x0040, COP_AUTOSCAN = 0x0080, COP_TUNER_MIC = 0x0100, COP_PCMHALTED = 0x0200, COP_POWERDOWN = 0x0000 }
	Codec operation mode definition. More...
Functions
void	CODEC__spiWrite (unsigned short)
	Function writes commands to Codec chip.
void	CODEC__defaultPcmGet (int *, int)
	Function serves as the default PCM data get handle.
void	CODEC__defaultPcmPut (int *, int)
	Functions serves as the default PCM data put handle.
void	CODEC_linkitPcmGet (int *, int)
	Function serves as the linkIt mode PCM data get handle.
void	CODEC_linkitPcmPut (int *, int)
	Function serves as the linkIt mode PCM data put handle.
void	CODEC_autoscanPcmGet (int *, int)
	Function serves as the autoscan PCM data get handle.
void	CODEC_autoscanPcmPut (int *, int)
	Function serves as the autoscan PCM data put handle.
void	CODEC_linePcmGet (int *, int)
	Function serves as the line-in record PCM data get handle.
void	CODEC_linePcmPut (int *, int)
	Function serves as the line-in PCM data put handle.
void	CODEC_micPcmGet (int *, int)
	Function serves as the mic record PCM data get handle.
void	CODEC_tunerPcmPut (int *, int)
	Function serves as the FM radio PCM data put handle.
void	CODEC_tunerPcmGet (int *, int)
	Function serves as the FM radio PCM data get handle.
void	CODEC_playPcmPut (int *, int)
	Function serves as the normal playback PCM data put handle.
void	CODEC_myfiPcmPut (int *, int)
	Function serves as the MyFi playback PCM data put handle.
void	CODEC_config (COP_MODE)
	Function initializes the codec given the inputs.
void	CODEC_data (void)
	Function serves as the data task handle to fetch data from codec ADC.
void	CODEC_init (void)
	Function initializes Codec hardware interface.
void	CODEC_hpVolume (int)
	Function controls headphone volume.
void	CODEC_gain ()
	Function controls channel gain.
void	CODEC_pcm (void)
	Function serves as the codec pcm task handle to output data to codec DAC.
void	CODEC_power (COP_MODE)
	Function controls the power ON/OFF of the codec DAC and ADC.
void	CODEC_setHandle (PFPCMGET, PFPCMPUT, PFPCMDEC, PFPCMENC)
	Function sets up the PCM get/put handle.
void	CODEC_mute (void)
	Function mute the codec interface.
void	CODEC_unmute (void)
	Function unmute the codec interface.
void	CODEC_recordHdlReset (void)
	Function resets the encoder codec handle controls.

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Enumeration Type Documentation

enum COP_MODE

Codec operation mode definition. Enumeration values: COP_MUSIC  Simple audio playback. COP_TUNER  Simple FM tuner. COP_FMXMIT  FM out while audio playback. COP_ILINK  FM out for line in. COP_MIC_RECORD  Record from microphone. COP_LINE_RECORD  Record from line in. COP_FM_RECORD  Record from FM radio. COP_AUTOSCAN  Autoscan mode, similar to COP_TUNER. COP_TUNER_MIC  tuner mode but with micrphone enabled...MFG test COP_PCMHALTED  PCM halted. COP_POWERDOWN  Shut down codec. Definition at line 57 of file codec.h. Referenced by AUDIO_start(). { COP_MUSIC = 0x0001, COP_TUNER = 0x0002, COP_FMXMIT = 0x0004, COP_ILINK = 0x0008, COP_MIC_RECORD = 0x0010, COP_LINE_RECORD = 0x0020, COP_FM_RECORD = 0x0040, COP_AUTOSCAN = 0x0080, COP_TUNER_MIC = 0x0100, COP_PCMHALTED = 0x0200, COP_POWERDOWN = 0x0000 } COP_MODE;

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Function Documentation

void CODEC__defaultPcmGet (  int *  pcmBuf, int  len )

Function serves as the default PCM data get handle. Parameters: pcmBuf  PCM data buffer. len  Available PCM data length in words. Definition at line 136 of file CODEC_pcm.c. { }

void CODEC__defaultPcmPut (  int *  pcmBuf, int  len )

Functions serves as the default PCM data put handle. Parameters: pcmBuf  PCM data buffer. len  Pending PCM data length in words. Definition at line 122 of file CODEC_pcm.c. References GEN_zeroOut(). { GEN_zeroOut((UINT16*)pcmBuf, len); }

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void CODEC__spiWrite (  unsigned short  cmd  )

Function writes commands to Codec chip. Parameters: cmd  command sent to Codec chip. Definition at line 35 of file CODEC__spi.c. References SYS_delayMicrosecond(). Referenced by CODEC_gain(), CODEC_hpVolume(), CODEC_init(), CODEC_mute(), and CODEC_power(). { unsigned short i; unsigned short iTmp; SPI_lock(SYS_FOREVER); // clear the clock. All data is latched in on the rising edge. iTmp = SPI_CLK; // clear tuner data line to set to input. //iTmp = TUNER_DAT; // clear the CODEC enable to enable. iTmp = CODEC_EN; SYS_delayMicrosecond(1); for (i = 0; i < 16; i++) { iTmp = SPI_CLK; if (cmd & 0x8000) SPI_DATA = 0x0; else iTmp = SPI_DATA; SYS_delayMicrosecond(1); SPI_CLK = 0x0; SYS_delayMicrosecond(1); cmd = cmd<<1; } // disable the EEPROM; // asm (" ssbx XF"); // clear SPI clock iTmp = SPI_CLK; // clear the DATA line, set to input. iTmp = SPI_DATA; // clear the DATA line on the tuner to set to input //iTmp = TUNER_DAT; // set codec enable to disable CODEC_EN = 0x0; // put tuner in read mode. //iTmp = TUNER_WR; SPI_unlock(); }

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void CODEC_autoscanPcmGet (  int *  pcmBuf, int  len )

Function serves as the autoscan PCM data get handle. Parameters: pcmBuf  PCM data buffer. len  Available PCM data length in words. Definition at line 87 of file CODEC_autoscanHdl.c. { leftChannel = pcmBuf; rightChannel = pcmBuf + len/2; }

void CODEC_autoscanPcmPut (  int *  pcmBuf, int  len )

Function serves as the autoscan PCM data put handle. Parameters: pcmBuf  PCM data buffer. len  Pending PCM data length in words. Definition at line 48 of file CODEC_autoscanHdl.c. References st_AUTOSCAN::pLeftInput, and st_AUTOSCAN::pRightInput. { unsigned int i; int * pSrcLeft; int * pSrcRight; int * pDstLeft; int * pDstRight; int dataLen = len/2; pSrcLeft = leftChannel; pSrcRight = rightChannel; pDstLeft = pcmBuf; pDstRight = pcmBuf + dataLen; AutoscanGlobals.pLeftInput = pDstLeft; AutoscanGlobals.pRightInput = pDstRight; for (i = 0; i < dataLen; i++) { FIRDeemphasisLeftTaps[0] = *pSrcLeft++; FIRDeemphasisRightTaps[0] = *pSrcRight++; // apply demphasis filter *pDstLeft++ = TUNER_deemphasisFIR(&FIRDeemphasisLeft); *pDstRight++ = TUNER_deemphasisFIR(&FIRDeemphasisRight); } AUTOSCAN_genMetrics(&AutoscanGlobals); }

void CODEC_config (  COP_MODE  eMode  )

Function initializes the codec given the inputs. This function will set up the corresponding DMA engines and initialize McBSP. DAC and ADC of the Codec will also be set to given state determined by eMode. Parameters: eMode  New Codec operation mode. Definition at line 56 of file CODEC_config.c. References CODEC_power(), st_PCM::codecMode, st_PCM::flag, GEN_zeroOut(), st_PCM::getFunc, McBSP_init(), st_PCM::newGetFunc, st_PCM::newPutFunc, st_PCM::pipe, PIPE_reset(), PIPE_writeNext(), and st_PCM::putFunc. Referenced by AUDIO_start(). { if(Pcm.codecMode != eMode) { /* Halt DMA3 and CODEC_McBSP */ CODEC_haltPcm(); Pcm.codecMode = eMode; Pcm.getFunc = Pcm.newGetFunc; Pcm.putFunc = Pcm.newPutFunc; CODEC_power(eMode); if( (eMode != COP_MIC_RECORD) ) { /* Reset the pipe object, initialize to zero frames. */ PIPE_reset(&Pcm.pipe); GEN_zeroOut((UINT16 *)Pcm.pipe.WrFrame.Frame, PCM_PIPE_FRAMELEN); PIPE_writeNext(&Pcm.pipe); GEN_zeroOut((UINT16 *)Pcm.pipe.WrFrame.Frame, PCM_PIPE_FRAMELEN); PIPE_writeNext(&Pcm.pipe); Pcm.flag |= HAVE_NEXT_BLOCK; } /* Initialize the pcm globals */ Pcm.flag &= ~(UPPER_HALF_BUF|STARTED_NEXT_BLOCK); Pcm.flag |= IS_MUTED; /* DMA3/DMA4 has higher priority, Interrupt source selected. */ DMPREC |= 0x1840; /* Set the FREE bit so that transfers not stop with emulator */ //DMPREC |= 0x8000; /* Setup DMA3 to connect to McBSP transmitter to send out audio data. */ /* point to DMSRC3 register */ DMSBAR = DMSRC3_SUBADDR; /* DMSRC3 = Transmit buffer pong address */ DMSBAI = (int)Pcm.pipe.RdFrame.Frame; /* DMDST3 = CODEC_McBSP data transmit register */ DMSBAI = MCBSP_DXR1_ADDR(CODEC_McBSP); /* DMCTR3 = Transmit buffer size (2 words in sample--left/right) ** 2 words per frame, 64 frames per block. */ DMSBAI = 1; /* Modification of CODEC_McBSP definition will cause changes here. */ /* DMSFC3 = McBSP0 transmit sync event frames = # of samples */ DMSBAI = 0x2000+(PCM_CHANNEL_BUFLEN-1); /* DMMCR3 = Auto-init, ** interrupt at block complete, ** DMA interrupt enabled, ** decrement mode (auto-init) */ DMSBAI = 0xc541; DMSBAR = DMIDX0_SUBADDR; /* DMIDX0 = step to next sample in frame */ DMSBAN = PCM_CHANNEL_BUFLEN; DMSBAR = DMFRI0_SUBADDR; /* DMFRI0 = step to next frame */ DMSBAN = (unsigned short)(1 - PCM_CHANNEL_BUFLEN); DMSBAR = DMGSA_SUBADDR; /* DMGSA = Transmit buffer pong address */ DMSBAI = (int)Pcm.pipe.RdFrame.NextFrame; /* DMGDA = McBSP0 data transmit register */ DMSBAI = MCBSP_DXR1_ADDR(CODEC_McBSP); /* DMGCR = Transmit buffer size (2 words in sample--left/right) */ DMSBAI = 1; /* DMGFR = McBSP0 transmit sync event frames = # of samples */ DMSBAI = 0x2000+(PCM_CHANNEL_BUFLEN-1); /* Setup DMA4 to connect to McBSP receiver to receive audio data. */ DMSBAR = DMSRC4_SUBADDR; /* DMA source: CODEC_McBSP receiver address */ DMSBAI = MCBSP_DRR1_ADDR(CODEC_McBSP); /* DMA destination: BUF_input, which MUST be a circular buffer.*/ DMSBAI = (int)BUF_output; /* DMCTR4 = Transmit buffer size (2 words in sample--left/right) */ DMSBAI = 4*PCM_CHANNEL_BUFLEN; /* Modification of CODEC_McBSP definition will cause changes here. */ /* DMSFC4 = McBSP0 receiver sync event */ DMSBAI = 0x1000; /* DMMCR4 = No Auto-init ** DMA in ABU mode ** DMA interrupt occurs at half-buffer filled ** DMA index address autoincrement step: one word ** DMA in single word mode ** DMA postincrement wiht index offset, similiar with DMA3. */ DMSBAI = 0x7045; /* clear the DMA3/4 interrupt here to prevent malfunction. */ IFR |= 0x1800; /* Unmute Codec & start DMA3/4. */ CODEC_setActiveState(eMode); /* Initialize CODEC_McBSP. */ McBSP_init(CODEC_McBSP, /* port number */ /* Digital Loop back, just for testing. */ //FIELD_VAL(DLB, 1, 1 )| /* spcr1 */ FIELD_VAL(RINTM, RINTM_SZ, INTM_RDY )| /* spcr1 */ FIELD_VAL(RJUST, RJUST_SZ, RXJUST_RJSE )| /* spcr1 */ FIELD_VAL(CLKSTP, CLKSTP_SZ, 0 )| /* spcr1 */ FIELD_VAL(RRST, 1, 1 ), /* spcr1 - receiver is enabled */ FIELD_VAL(FREE, 1, 1 )| /* spcr2 - free running mode is enabled */ FIELD_VAL(SOFT, 1, 1 )| /* spcr2 - don't care. */ FIELD_VAL(XINTM, XINTM_SZ, INTM_RDY )| /* spcr2 */ FIELD_VAL(XRST, 1, 1 )| /* spcr2 - transmittor is enabled */ FIELD_VAL(GRST, 1, 1 )| /* spcr2 */ FIELD_VAL(FRST, 1, 0 ), /* spcr2 - Frame sync is provided by Codec */ FIELD_VAL(RWDLEN1, RWDLEN1_SZ, WORD_LENGTH_16 )| /* rcr1 - 16 bits per word */ FIELD_VAL(RFRLEN1, RFRLEN1_SZ, 1 ), /* rcr1 - 2 words per frame */ FIELD_VAL(RDATDLY, RDATDLY_SZ, DATA_DELAY0 )| /* rcr2 - N/A */ FIELD_VAL(RFIG, 1, FRAME_IGNORE )| /* rcr2 - N/A */ FIELD_VAL(RCOMPAND, RCOMPAND_SZ,NO_COMPAND_MSB_1ST )| /* rcr2 - N/A */ FIELD_VAL(RWDLEN2, RWDLEN2_SZ, WORD_LENGTH_16 )| /* rcr2 - N/A */ FIELD_VAL(RFRLEN2, RFRLEN2_SZ, 0 )| /* rcr2 - N/A */ FIELD_VAL(RPHASE, 1, SINGLE_PHASE ), /* rcr2 - N/A */ FIELD_VAL(XWDLEN1, XWDLEN1_SZ, WORD_LENGTH_16 )| /* xcr1 - 16-bit words in the first phase */ FIELD_VAL(XFRLEN1, XFRLEN1_SZ, 1 ), /* xcr1 - 2 words in the first phase */ FIELD_VAL(XDATDLY, XDATDLY_SZ, DATA_DELAY0 )| /* xcr2 - delay data clocks from frame sync. */ FIELD_VAL(XFIG, 1, FRAME_IGNORE )| /* xcr2 - ignore extraneous frame syncs */ FIELD_VAL(XCOMPAND, XCOMPAND_SZ,NO_COMPAND_MSB_1ST )| /* xcr2 - MSB align data (no companding) */ FIELD_VAL(XWDLEN2, XWDLEN2_SZ, WORD_LENGTH_16 )| /* xcr2 - N/A */ FIELD_VAL(XFRLEN2, XFRLEN2_SZ, 0 )| /* xcr2 - N/A */ FIELD_VAL(XPHASE, 1, SINGLE_PHASE ), /* xcr2 - single phase frame */ 0, /* srgr1 - N/A */ 0, /* srgr2 - N/A */ FIELD_VAL(RMCM, 1, 0 ), /* mcr1 - all 128 RX channels are enabled */ FIELD_VAL(XMCM, XMCM_SZ, 0 ), /* mcr2 - all 128 TX channels are enabled */ 0, /* rcera - N/A */ 0, /* rcerb - N/A */ 0, /* xcera - N/A */ 0, /* xcerb - N/A */ FIELD_VAL(CLKRP, 1, CLKR_POL_FALLING )| /* pcr - N/A */ FIELD_VAL(CLKXP, 1, CLKX_POL_FALLING )| /* pcr - transmit data sampled on falling edge of CLKX */ FIELD_VAL(FSRP, 1, FSYNC_POL_HIGH )| /* pcr - N/A */ FIELD_VAL(FSXP, 1, FSYNC_POL_HIGH )| /* pcr - frame sync pulse (FSX) is active high */ FIELD_VAL(CLKRM, 1, CLK_MODE_EXT )| /* pcr - CLKR is an input pin */ FIELD_VAL(CLKXM, 1, CLK_MODE_EXT )| /* pcr - CLKX is an intput pin */ FIELD_VAL(FSRM, 1, FSYNC_MODE_EXT )| /* pcr - FSR is an input pin */ FIELD_VAL(FSXM, 1, FSYNC_MODE_EXT ) /* pcr - FSX is an input */ ); // Wait till codec interface stablized before bringing up volume. We'll // lose 50ms music here. //if( eMode & (COP_MUSIC|COP_TUNER|COP_ILINK|COP_LINE_RECORD|COP_FM_RECORD|COP_AUTOSCAN|COP_TUNER_MIC) ) //{ // SYS_delayMicrosecond(50000); // CODEC_hpVolume(sysParam.volume); //} } else { SWI_disable(); Pcm.getFunc = Pcm.newGetFunc; Pcm.putFunc = Pcm.newPutFunc; SWI_enable(); } }

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void CODEC_data (  void   )

Function serves as the data task handle to fetch data from codec ADC. Also function checkes to feed data into pcm pipe. Definition at line 40 of file CODEC_data.c. References st_PCM::codecMode, COP_AUTOSCAN, COP_FM_RECORD, COP_FMXMIT, COP_ILINK, COP_LINE_RECORD, COP_MIC_RECORD, COP_MUSIC, COP_TUNER, st_PCM::flag, st_PCM::getFunc, st_PCM::pipe, PIPE_writeNext(), and st_PCM::putFunc. { PIPE_Obj * pPipe = &Pcm.pipe; int * pData; // Use input to sync. if ( PIPE_writeFrameNum(pPipe) ) { // Check output first. #ifdef MIXER_SUPPORT if( Pcm.codecMode & (COP_TUNER|COP_LINE_RECORD|COP_MIC_RECORD|COP_FM_RECORD|COP_AUTOSCAN|COP_ILINK|COP_MUSIC|COP_FMXMIT) ) #else if( Pcm.codecMode & (COP_TUNER|COP_LINE_RECORD|COP_MIC_RECORD|COP_FM_RECORD|COP_AUTOSCAN|COP_ILINK) ) #endif { /* which part of the buffer is valid??? */ if( Pcm.flag & UPPER_HALF_BUF ) { /* upper half. */ pData = &BUF_output[0]; } else { /* lower half. */ pData = &BUF_output[BUF_OUTPUTSIZE/2]; } _sortChannel(pData); Pcm.getFunc(ChannelBuf, BUF_OUTPUTSIZE/2); } if( Pcm.codecMode &(COP_MUSIC|COP_FMXMIT|COP_TUNER|COP_LINE_RECORD|COP_FM_RECORD|COP_AUTOSCAN|COP_ILINK) ) { Pcm.putFunc(pPipe->WrFrame.Frame, pPipe->wFrameLen); } /* validate frame. */ PIPE_writeNext(pPipe); } }

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void CODEC_hpVolume (  int  volume  )

Function controls headphone volume. Parameters: volume  New headphone volume. Definition at line 267 of file CODEC_config.c. References CODEC__spiWrite(). Referenced by CODEC_mute(), CODEC_unmute(), UI_CSF_volume(), and UI_PSF_radio(). { /* Simultaneously update left/right channel???. */ if(volume) volume += VOLUME_OFFSET; CODEC__spiWrite(AIC23REG_LHDPHONE | 0x100 | volume); //CODEC__spiWrite(AIC23REG_RHDPHONE | 0x100 | volume); }

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void CODEC_linePcmGet (  int *  pcmBuf, int  len )

Function serves as the line-in record PCM data get handle. Parameters: pcmBuf  PCM data buffer. len  Available PCM data length in words. Definition at line 99 of file CODEC_recordHdl.c. References ENCODER_getInputBuf(), and ENCODER_putInputBuf(). Referenced by AUDIO_start(). { int * pEncDstLeft; int * pEncDstRight; int dataLen = len/2; leftChannel = pcmBuf; rightChannel = pcmBuf + dataLen; if(FALSE == ENCODER_getInputBuf(&pEncDstLeft, &pEncDstRight)) { // Encoder can not keep up, out of sync. asm(" nop"); } else { memcpy(pEncDstLeft, leftChannel, dataLen); memcpy(pEncDstRight, rightChannel, dataLen); ENCODER_putInputBuf(); } }

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void CODEC_linePcmPut (  int *  pcmBuf, int  len )

Function serves as the line-in PCM data put handle. Parameters: pcmBuf  PCM data buffer. len  Pending PCM data length in words. Definition at line 130 of file CODEC_recordHdl.c. References st_AUDIOENERGY::pFrame. Referenced by AUDIO_start(). { unsigned int i; int dataLen = len/2; int * pTmp; int * left; int * right; pTmp = pcmBuf + dataLen; left = leftChannel; right = rightChannel; for (i = 0; i < dataLen; i++) { *pcmBuf++ = *left; *pTmp++ = *right; LeftInputEnergy.pFrame = left++; RightInputEnergy.pFrame = right++; AUDIO_energy(&LeftInputEnergy); AUDIO_energy(&RightInputEnergy); } }

void CODEC_linkitPcmGet (  int *  pcmBuf, int  len )

Function serves as the linkIt mode PCM data get handle. Parameters: pcmBuf  PCM data buffer. len  Available PCM data length in words. Definition at line 367 of file CODEC_playHdl.c. Referenced by AUDIO_start(). { leftChannel = pcmBuf; rightChannel = pcmBuf + len/2; }

void CODEC_linkitPcmPut (  int *  pcmBuf, int  len )

Function serves as the linkIt mode PCM data put handle. Parameters: pcmBuf  PCM data buffer. len  Pending PCM data length in words. Definition at line 305 of file CODEC_playHdl.c. Referenced by AUDIO_start(). { int i; /*if(SYS_HF_myfiTest()) { st_AUDIO_OSCILLATOR LeftTestTone; st_AUDIO_OSCILLATOR RightTestTone; // set up oscillators for the test tones // 500 Hz, 95% full-scale sine wave (to prevent clipping due to // rounding) LeftTestTone.Coeff = RightTestTone.Coeff = 32753; LeftTestTone.Tap1 = RightTestTone.Tap1 = 887; LeftTestTone.Tap2 = RightTestTone.Tap2 = 0; LeftTestTone.pOutput = leftChannel; RightTestTone.pOutput = rightChannel; AUDIO_toneGen(&LeftTestTone); AUDIO_toneGen(&RightTestTone); }*/ StereoMod.pSum = leftChannel; StereoMod.pDiff = rightChannel; StereoMod.OutputLeft = (int *)pcmBuf; StereoMod.OutputRight = (int *)pcmBuf + 64; //if(SYS_HF_myfiTest()) { FIRPremphasisLeft.pOutput = leftChannel; FIRPremphasisRight.pOutput = rightChannel; } for (i = 0; i < PCM_CHANNEL_BUFLEN; i++) { //XMIT_Energy(&EnergyInputLeft); //XMIT_Energy(&EnergyInputRight); CarrierGen(&Carrier); //if(SYS_HF_myfiTest()) { XMIT_preemphasisFIR(&FIRPremphasisLeft); XMIT_preemphasisFIR(&FIRPremphasisRight); } XMIT_Modulator(&StereoMod); } }

void CODEC_micPcmGet (  int *  pcmBuf, int  len )

Function serves as the mic record PCM data get handle. Parameters: pcmBuf  PCM data buffer. len  Available PCM data length in words. Definition at line 58 of file CODEC_recordHdl.c. References ENCODER_getInputBuf(), ENCODER_putInputBuf(), and st_AUDIOENERGY::pFrame. Referenced by AUDIO_start(). { unsigned short i; int * pDstLeft; int * pDstRight; if(FALSE == ENCODER_getInputBuf(&pDstLeft, &pDstRight)) { // Encoder can not keep up, out of sync. asm(" nop"); } else { len = len/2; for (i = 0; i < len; i++) { // put the input data into the beginning of the taps array. //FIRMicTaps[0] = *pSrcLeft++; FIRMicTaps[2] = *pcmBuf++; // apply LPF filter *pDstLeft = *pDstRight++ = CODEC_micLPF(&FIRMicRecord); LeftInputEnergy.pFrame = RightInputEnergy.pFrame = pDstLeft++; AUDIO_energy(&LeftInputEnergy); AUDIO_energy(&RightInputEnergy); } ENCODER_putInputBuf(); } }

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void CODEC_myfiPcmPut (  int *  pcmBuf, int  len )

Function serves as the MyFi playback PCM data put handle. Parameters: pcmBuf  PCM data buffer. len  Pending PCM data length in words. Definition at line 144 of file CODEC_playHdl.c. References st_PCM::decode, st_PCM::flag, GEN_zeroOut(), and st_PCM::sampleRate. Referenced by AUDIO_start(). { int i; int * pData; int * pDstLeft; int * pDstRight; if (XMIT_CompressBuffer( &EncoderBuffer )) // if the input buffer has less than the required number of samples // and the output buffer has less than 64 samples ready to go, we need // to decode more MP3 and perform SRC on it. // For the input buffer, there are always 11 extra samples left over // from the previous frame for the taps of the FIR filter. { pDstLeft = EncoderBuffer.pSumInput; pDstRight = EncoderBuffer.pDiffInput; /* Setup sample rate conversion output buffer. */ SRC_setOutput(&LeftSRC, pDstLeft); SRC_setOutput(&RightSRC, pDstRight); /* check for a change in sample rate and reset SRC if needed */ if (Pcm.flag&SAMPLE_RATE_CHANGED) { /* set sample rate */ SRC_setInputRate(&LeftSRC, Pcm.sampleRate); SRC_setInputRate(&RightSRC, Pcm.sampleRate); PCM_samplerateNotChanged(); } /* loop over sample rate converter until output buffers are filled */ while (SRC_isOutputBufferFull(&LeftSRC) == 0) { /* get another input block if needed */ if (SRC_isInputBufferEmpty(&LeftSRC)) { pData = Pcm.decode(); if( NULL == pData) { // Decoder data is not ready, // Zero sending out. GEN_zeroOut((UINT16*)pcmBuf, len); return; } else { SRC_setInput(&LeftSRC, pData); SRC_setInput(&RightSRC, pData + SRC_inputBufferSize()); } } /* run the left and right channels */ SRC_apply(&LeftSRC); SRC_apply(&RightSRC); } EncoderBuffer.Samples += PCM_CHANNEL_BUFLEN; } // by the time we get here, we should have at least 26 samples (+11 previous samples) // available in the input buffers. DifSRCData.Input = EncoderBuffer.pDiffOutput; SumSRCData.Input = EncoderBuffer.pSumOutput; DifSRCData.Output = SRCBuffer.pDiffInput; SumSRCData.Output = SRCBuffer.pSumInput; while (SRCBuffer.Samples < PCM_CHANNEL_BUFLEN) /* If there are not enough samples after the SRC, then * we need to run the SRC until there are enough samples */ { /* SRC Automatically updates the input and output pointers. */ XMIT_SRC(&SumSRCData); XMIT_SRC(&DifSRCData); SRCBuffer.Samples += XMIT_OUTPUT; EncoderBuffer.Samples -= XMIT_SAMPLES; } /* Update pointers */ EncoderBuffer.pDiffOutput = DifSRCData.Input; EncoderBuffer.pSumOutput = SumSRCData.Input; SRCBuffer.pDiffInput = DifSRCData.Output; SRCBuffer.pSumInput = SumSRCData.Output; /* Now, enough up-sampled data is in the SRC Output buffers. * we can not process the data and put it in the output buffer. * Then, we need to compress the SRC Output Buffers. */ EnergyInputLeft.pInput = StereoMod.pSum = SRCBuffer.pSumOutput; EnergyInputRight.pInput = StereoMod.pDiff = SRCBuffer.pDiffOutput; SRCBuffer.pSumOutput += PCM_CHANNEL_BUFLEN; SRCBuffer.pDiffOutput += PCM_CHANNEL_BUFLEN; SRCBuffer.Samples -= PCM_CHANNEL_BUFLEN; pDstLeft = pcmBuf; pDstRight = pcmBuf + len/2; //' Now, let's take this data and put it into the EncoderBuffers //' Apply all filters (15kHz and pre-emphasis here) //' Also apply all summing of the left and right channels //' and the difference of the left and right channels. StereoMod.OutputRight = pDstRight; StereoMod.OutputLeft = pDstLeft; #ifdef MIXER_SUPPORT { int ii, *pL, *pR; pL = StereoMod.pDiff; pR = StereoMod.pSum; for (ii = 0; ii < PCM_CHANNEL_BUFLEN; ii++) { *pL = (*pL+*leftChannel)/2; *pR = (*pR+*rightChannel)/2; pL++; pR++; leftChannel++; rightChannel++; } } #endif for (i = 0; i < PCM_CHANNEL_BUFLEN; i++) { // Not enought mips for this. //XMIT_Energy(&EnergyInputLeft); //XMIT_Energy(&EnergyInputRight); #if 0 // 11/13/02 FIRPremphasisLeftTaps[0] = *pSrcLeft; FIRPremphasisRightTaps[0] = *pSrcRight; *pSrcLeft++ = XMIT_PreemphasisFIR(&FIRPremphasisLeft); *pSrcRight++ = XMIT_PreemphasisFIR(&FIRPremphasisRight); #endif CarrierGen(&Carrier); //XMIT_SumDiff(&StereoMod); XMIT_Modulator(&StereoMod); } XMIT_CompressBuffer(&SRCBuffer); /*remove the EQ apply for MIPs limit*/ /* Let's apply EQ here. */ /* EQ_apply(&LeftEQ, pDstLeft, pDstLeft); EQ_apply(&RightEQ, pDstRight, pDstRight); */ }

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void CODEC_playPcmPut (  int *  pcmBuf, int  len )

Function serves as the normal playback PCM data put handle. Parameters: pcmBuf  PCM data buffer. len  Pending PCM data length in words. Definition at line 50 of file CODEC_playHdl.c. References AUDIO_speedIsOutputBufferFull(), st_PCM::decode, st_PCM::flag, GEN_zeroOut(), and st_PCM::sampleRate. Referenced by AUDIO_start(). { int * pData; int * pRight = pcmBuf+len/2; #ifdef DJ_SUPPORT // loop over speed control. while(1) #endif { #ifdef DJ_SUPPORT if(AUDIO_speedIsOutputBufferFull()) break; #endif /* Setup sample rate conversion output buffer. */ // This could be opted out in the future, to have PCM put to a buffer at fixed // position. SRC_setOutput(&LeftSRC, pcmBuf); SRC_setOutput(&RightSRC, pRight); /* check for a change in sample rate and reset SRC if needed */ if (Pcm.flag&SAMPLE_RATE_CHANGED) { /* set sample rate */ SRC_setInputRate(&LeftSRC, Pcm.sampleRate); SRC_setInputRate(&RightSRC, Pcm.sampleRate); PCM_samplerateNotChanged(); } /* loop over sample rate converter until output buffers are filled */ while (SRC_isOutputBufferFull(&LeftSRC) == 0) { /* get another input block if needed */ if (SRC_isInputBufferEmpty(&LeftSRC)) { pData = Pcm.decode(); if( NULL == pData) { // Decoder data is not ready, // Zero sending out. GEN_zeroOut((UINT16*)pcmBuf, len); return; } else { SRC_setInput(&LeftSRC, pData); SRC_setInput(&RightSRC, pData + SRC_inputBufferSize()); } } /* run the left and right channels */ SRC_apply(&LeftSRC); SRC_apply(&RightSRC); } #ifdef DJ_SUPPORT if(AUDIO_speedFIR(pcmBuf, pRight)) break; #endif } #ifdef DJ_SUPPORT AUDIO_speedGetSample(pcmBuf, pRight); #endif EQ_apply(&LeftEQ, pcmBuf, pcmBuf); EQ_apply(&RightEQ, pRight, pRight); #ifdef MIXER_SUPPORT { unsigned short ii; for (ii = 0; ii < PCM_CHANNEL_BUFLEN; ii++) { *pcmBuf = (*pcmBuf+*leftChannel)/2; *pRight = (*pRight+*rightChannel)/2; pcmBuf++; pRight++; leftChannel++; rightChannel++; } } #endif }

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void CODEC_power (  COP_MODE  eMode  )

Function controls the power ON/OFF of the codec DAC and ADC. Parameters: eMode  New Codec operation mode. Definition at line 334 of file CODEC_config.c. References CODEC__spiWrite(), COP_AUTOSCAN, COP_FM_RECORD, COP_FMXMIT, COP_ILINK, COP_LINE_RECORD, COP_MIC_RECORD, COP_MUSIC, COP_POWERDOWN, COP_TUNER, and COP_TUNER_MIC. Referenced by CODEC_config(). { // Shut down DMA before Codec to prevent "pops". CODEC_setActiveState(COP_POWERDOWN); if(eMode & COP_POWERDOWN) { // Turn off DSP clock out, disable codec clock. CPLD_CODECClk(2); } else if( eMode & (COP_ILINK|COP_FMXMIT) ) { /* Choose to use internal clock. */ CPLD_CODECClk(1); } // Always use external clock else, thus to disable clock out pin and // conserve power. else { /* Choose to use external clock. */ CPLD_CODECClk(0); } switch(eMode) { //case COP_POWERDOWN: // Do NOT power down codec, thus to prevent power up pop. // break; case COP_MIC_RECORD: //CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x069); CODEC__spiWrite(AIC23REG_APATH | 0x004); CODEC__spiWrite(AIC23REG_DPATH | 0x000); CODEC__spiWrite(AIC23REG_SRC | 0x23); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x63); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); //SYS_delayMicrosecond(200000); break; case COP_TUNER_MIC: //CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x060); CODEC__spiWrite(AIC23REG_APATH | 0x0015); CODEC__spiWrite(AIC23REG_DPATH | 0x0001); CODEC__spiWrite(AIC23REG_SRC | 0x23); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x63); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); break; case COP_AUTOSCAN: //CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x062); CODEC__spiWrite(AIC23REG_APATH | 0x012); CODEC__spiWrite(AIC23REG_DPATH | 0x000); CODEC__spiWrite(AIC23REG_SRC | 0x23); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x63); CODEC__spiWrite(AIC23REG_LLNVOL | 0x11f); // 0dB headphone output. CODEC__spiWrite(AIC23REG_LHDPHONE | 0x100 | 0x079); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); break; case COP_TUNER: //CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x062); CODEC__spiWrite(AIC23REG_APATH | 0x012); CODEC__spiWrite(AIC23REG_DPATH | 0x000); CODEC__spiWrite(AIC23REG_SRC | 0x23); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x63); CODEC__spiWrite(AIC23REG_LLNVOL | 0x11f); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); break; case COP_LINE_RECORD: //CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x062); CODEC__spiWrite(AIC23REG_APATH | 0x012); CODEC__spiWrite(AIC23REG_DPATH | 0x000); CODEC__spiWrite(AIC23REG_SRC | 0x23); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x63); CODEC__spiWrite(AIC23REG_LLNVOL | 0x117); CODEC__spiWrite(AIC23REG_RLNVOL | 0x117); // 0dB headphone output. CODEC__spiWrite(AIC23REG_LHDPHONE | 0x100 | 0x079); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); break; case COP_FM_RECORD: //CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x062); CODEC__spiWrite(AIC23REG_APATH | 0x012); CODEC__spiWrite(AIC23REG_DPATH | 0x000); CODEC__spiWrite(AIC23REG_SRC | 0x23); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x63); CODEC__spiWrite(AIC23REG_LLNVOL | 0x11f); CODEC__spiWrite(AIC23REG_RLNVOL | 0x11f); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); break; case COP_MUSIC: //CODEC__spiWrite(AIC23REG_RESET); #ifdef MIXER_SUPPORT CODEC__spiWrite(AIC23REG_PWRDOWN | 0x061); CODEC__spiWrite(AIC23REG_APATH | 0x015); CODEC__spiWrite(AIC23REG_DPATH | 0x000); CODEC__spiWrite(AIC23REG_SRC | 0x23); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x63); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); #else CODEC__spiWrite(AIC23REG_PWRDOWN | 0x067); CODEC__spiWrite(AIC23REG_APATH | 0x012); CODEC__spiWrite(AIC23REG_DPATH | 0x000); CODEC__spiWrite(AIC23REG_SRC | 0x23); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x63); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); #endif break; case COP_ILINK: CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x062); CODEC__spiWrite(AIC23REG_APATH | 0x012); CODEC__spiWrite(AIC23REG_DPATH | 0x001); CODEC__spiWrite(AIC23REG_SRC | 0x3f); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x43); CODEC__spiWrite(AIC23REG_LLNVOL | 0x17); CODEC__spiWrite(AIC23REG_RLNVOL | 0x17); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); break; case COP_FMXMIT: #ifdef MIXER_SUPPORT CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x061); CODEC__spiWrite(AIC23REG_APATH | 0x014); CODEC__spiWrite(AIC23REG_DPATH | 0x001); CODEC__spiWrite(AIC23REG_SRC | 0x3f); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x43); CODEC__spiWrite(AIC23REG_LLNVOL | 0x17); CODEC__spiWrite(AIC23REG_RLNVOL | 0x17); CODEC__spiWrite(AIC23REG_LHDPHONE | 0x100); CODEC__spiWrite(AIC23REG_RHDPHONE | 0x100); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); #else CODEC__spiWrite(AIC23REG_RESET); CODEC__spiWrite(AIC23REG_PWRDOWN | 0x062); CODEC__spiWrite(AIC23REG_APATH | 0x012); CODEC__spiWrite(AIC23REG_DPATH | 0x001); CODEC__spiWrite(AIC23REG_SRC | 0x3f); CODEC__spiWrite(AIC23REG_DAUDIOI | 0x43); CODEC__spiWrite(AIC23REG_LLNVOL | 0x17); CODEC__spiWrite(AIC23REG_RLNVOL | 0x17); CODEC__spiWrite(AIC23REG_LHDPHONE | 0x100); CODEC__spiWrite(AIC23REG_RHDPHONE | 0x100); CODEC__spiWrite(AIC23REG_DIACTV | 0x01); #endif break; } }

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void CODEC_setHandle (  PFPCMGET  getHdl, PFPCMPUT  putHdl, PFPCMDEC  decode, PFPCMENC  encode )

Function sets up the PCM get/put handle. Parameters: getHdl  PCM get handle. putHdl  PCM put handle. decode  PCM decoder handle. encode  PCM encoder handle. Definition at line 313 of file CODEC_config.c. References st_PCM::decode, st_PCM::encode, st_PCM::newGetFunc, and st_PCM::newPutFunc. Referenced by AUDIO_start(). { if(getHdl) Pcm.newGetFunc = getHdl; else Pcm.newGetFunc = CODEC__defaultPcmGet; if(putHdl) Pcm.newPutFunc = putHdl; else Pcm.newPutFunc = CODEC__defaultPcmPut; if(decode) Pcm.decode = decode; if(encode) Pcm.encode = encode; }

void CODEC_tunerPcmGet (  int *  pcmBuf, int  len )

Function serves as the FM radio PCM data get handle. Parameters: pcmBuf  PCM data buffer. len  Available PCM data length in words. Definition at line 112 of file CODEC_radioHdl.c. Referenced by AUDIO_start(). { leftChannel = pcmBuf; rightChannel = pcmBuf + len/2; }

void CODEC_tunerPcmPut (  int *  pcmBuf, int  len )

Function serves as the FM radio PCM data put handle. Parameters: pcmBuf  PCM data buffer. len  Pending PCM data length in words. Definition at line 46 of file CODEC_radioHdl.c. References st_PCM::codecMode, COP_FM_RECORD, ENCODER_getInputBuf(), ENCODER_putInputBuf(), and st_AUDIOENERGY::pFrame. Referenced by AUDIO_start(). { unsigned int i; int dataLen = len/2; int * pSrcLeft; int * pSrcRight; int * pDstLeft; int * pDstRight; int * pEncDstLeft; int * pEncDstRight; pSrcLeft = leftChannel; pSrcRight = rightChannel; pDstLeft = pcmBuf; pDstRight = pcmBuf + dataLen; if(COP_FM_RECORD&Pcm.codecMode) { if(FALSE == ENCODER_getInputBuf(&pEncDstLeft, &pEncDstRight)) { // Encoder can not catch up. asm(" nop"); } } else pEncDstLeft = NULL; for (i = 0; i < dataLen; i++) { FIRDeemphasisLeftTaps[0] = *pSrcLeft++; FIRDeemphasisRightTaps[0] = *pSrcRight++; // apply demphasis filter *pDstLeft = TUNER_deemphasisFIR(&FIRDeemphasisLeft); *pDstRight = TUNER_deemphasisFIR(&FIRDeemphasisRight); if(pEncDstLeft) { *pEncDstLeft++ = *pDstLeft; *pEncDstRight++ = *pDstRight; LeftInputEnergy.pFrame = pDstRight; RightInputEnergy.pFrame = pDstLeft; AUDIO_energy(&LeftInputEnergy); AUDIO_energy(&RightInputEnergy); } pDstLeft++; pDstRight++; } if(pEncDstLeft) ENCODER_putInputBuf(); else { EQ_apply(&LeftEQ, pcmBuf, pcmBuf); pcmBuf += dataLen; EQ_apply(&RightEQ, pcmBuf, pcmBuf); } }

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