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📄 cs46xx.c

📁 iis s3c2410-uda1341语音系统的 开发
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	return r;}#if CSDEBUG/* DEBUG ROUTINES */#define SOUND_MIXER_CS_GETDBGLEVEL 	_SIOWR('M',120, int)#define SOUND_MIXER_CS_SETDBGLEVEL 	_SIOWR('M',121, int)#define SOUND_MIXER_CS_GETDBGMASK 	_SIOWR('M',122, int)#define SOUND_MIXER_CS_SETDBGMASK 	_SIOWR('M',123, int)#define SOUND_MIXER_CS_APM	 	_SIOWR('M',124, int)void printioctl(unsigned int x){    unsigned int i;    unsigned char vidx;	/* these values are incorrect for the ac97 driver, fix.         * Index of mixtable1[] member is Device ID          * and must be <= SOUND_MIXER_NRDEVICES.         * Value of array member is index into s->mix.vol[]         */        static const unsigned char mixtable1[SOUND_MIXER_NRDEVICES] = {                [SOUND_MIXER_PCM]     = 1,   /* voice */                [SOUND_MIXER_LINE1]   = 2,   /* AUX */                [SOUND_MIXER_CD]      = 3,   /* CD */                [SOUND_MIXER_LINE]    = 4,   /* Line */                [SOUND_MIXER_SYNTH]   = 5,   /* FM */                [SOUND_MIXER_MIC]     = 6,   /* Mic */                [SOUND_MIXER_SPEAKER] = 7,   /* Speaker */                [SOUND_MIXER_RECLEV]  = 8,   /* Recording level */                [SOUND_MIXER_VOLUME]  = 9    /* Master Volume */        };            switch(x)     {	case SOUND_MIXER_CS_GETDBGMASK:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_CS_GETDBGMASK: ") );		break;	case SOUND_MIXER_CS_GETDBGLEVEL:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_CS_GETDBGLEVEL: ") );		break;	case SOUND_MIXER_CS_SETDBGMASK:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_CS_SETDBGMASK: ") );		break;	case SOUND_MIXER_CS_SETDBGLEVEL:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_CS_SETDBGLEVEL: ") );		break;        case OSS_GETVERSION:		CS_DBGOUT(CS_IOCTL, 4, printk("OSS_GETVERSION: ") );		break;        case SNDCTL_DSP_SYNC:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SYNC: ") );		break;        case SNDCTL_DSP_SETDUPLEX:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETDUPLEX: ") );		break;        case SNDCTL_DSP_GETCAPS:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETCAPS: ") );		break;        case SNDCTL_DSP_RESET:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_RESET: ") );		break;        case SNDCTL_DSP_SPEED:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SPEED: ") );		break;        case SNDCTL_DSP_STEREO:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_STEREO: ") );		break;        case SNDCTL_DSP_CHANNELS:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_CHANNELS: ") );		break;        case SNDCTL_DSP_GETFMTS: 		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETFMTS: ") );		break;        case SNDCTL_DSP_SETFMT: 		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETFMT: ") );		break;        case SNDCTL_DSP_POST:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_POST: ") );		break;        case SNDCTL_DSP_GETTRIGGER:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETTRIGGER: ") );		break;        case SNDCTL_DSP_SETTRIGGER:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETTRIGGER: ") );		break;        case SNDCTL_DSP_GETOSPACE:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETOSPACE: ") );		break;        case SNDCTL_DSP_GETISPACE:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETISPACE: ") );		break;        case SNDCTL_DSP_NONBLOCK:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_NONBLOCK: ") );		break;        case SNDCTL_DSP_GETODELAY:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETODELAY: ") );		break;        case SNDCTL_DSP_GETIPTR:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETIPTR: ") );		break;        case SNDCTL_DSP_GETOPTR:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETOPTR: ") );		break;        case SNDCTL_DSP_GETBLKSIZE:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETBLKSIZE: ") );		break;        case SNDCTL_DSP_SETFRAGMENT:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETFRAGMENT: ") );		break;        case SNDCTL_DSP_SUBDIVIDE:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SUBDIVIDE: ") );		break;        case SOUND_PCM_READ_RATE:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_RATE: ") );		break;        case SOUND_PCM_READ_CHANNELS:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_CHANNELS: ") );		break;        case SOUND_PCM_READ_BITS:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_BITS: ") );		break;        case SOUND_PCM_WRITE_FILTER:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_WRITE_FILTER: ") );		break;        case SNDCTL_DSP_SETSYNCRO:		CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETSYNCRO: ") );		break;        case SOUND_PCM_READ_FILTER:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_FILTER: ") );		break;        case SOUND_MIXER_PRIVATE1:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE1: ") );		break;        case SOUND_MIXER_PRIVATE2:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE2: ") );		break;        case SOUND_MIXER_PRIVATE3:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE3: ") );		break;        case SOUND_MIXER_PRIVATE4:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE4: ") );		break;        case SOUND_MIXER_PRIVATE5:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE5: ") );		break;        case SOUND_MIXER_INFO:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_INFO: ") );		break;        case SOUND_OLD_MIXER_INFO:		CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_OLD_MIXER_INFO: ") );		break;	default:		switch (_IOC_NR(x)) 		{			case SOUND_MIXER_VOLUME:				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_VOLUME: ") );				break;			case SOUND_MIXER_SPEAKER:				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_SPEAKER: ") );				break;			case SOUND_MIXER_RECLEV:				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_RECLEV: ") );				break;			case SOUND_MIXER_MIC:				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_MIC: ") );				break;			case SOUND_MIXER_SYNTH:				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_SYNTH: ") );				break;			case SOUND_MIXER_RECSRC: 				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_RECSRC: ") );				break;			case SOUND_MIXER_DEVMASK:				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_DEVMASK: ") );				break;			case SOUND_MIXER_RECMASK:				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_RECMASK: ") );				break;			case SOUND_MIXER_STEREODEVS: 				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_STEREODEVS: ") );				break;			case SOUND_MIXER_CAPS:				CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_CAPS:") );				break;			default:				i = _IOC_NR(x);				if (i >= SOUND_MIXER_NRDEVICES || !(vidx = mixtable1[i]))				{					CS_DBGOUT(CS_IOCTL, 4, printk("UNKNOWN IOCTL: 0x%.8x NR=%d ",x,i) );				}				else				{					CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_IOCTL AC9x: 0x%.8x NR=%d ",							x,i) );				}				break;		}    }    CS_DBGOUT(CS_IOCTL, 4, printk("command = 0x%x IOC_NR=%d\n",x, _IOC_NR(x)) );}#endif/* *  common I/O routines */static void cs461x_poke(struct cs_card *codec, unsigned long reg, unsigned int val){	writel(val, codec->ba1.idx[(reg >> 16) & 3]+(reg&0xffff));}static unsigned int cs461x_peek(struct cs_card *codec, unsigned long reg){	return readl(codec->ba1.idx[(reg >> 16) & 3]+(reg&0xffff));}static void cs461x_pokeBA0(struct cs_card *codec, unsigned long reg, unsigned int val){	writel(val, codec->ba0+reg);}static unsigned int cs461x_peekBA0(struct cs_card *codec, unsigned long reg){	return readl(codec->ba0+reg);}static u16 cs_ac97_get(struct ac97_codec *dev, u8 reg);static void cs_ac97_set(struct ac97_codec *dev, u8 reg, u16 data);static struct cs_channel *cs_alloc_pcm_channel(struct cs_card *card){	if(card->channel[1].used==1)		return NULL;	card->channel[1].used=1;	card->channel[1].num=1;	return &card->channel[1];}static struct cs_channel *cs_alloc_rec_pcm_channel(struct cs_card *card){	if(card->channel[0].used==1)		return NULL;	card->channel[0].used=1;	card->channel[0].num=0;	return &card->channel[0];}static void cs_free_pcm_channel(struct cs_card *card, int channel){	card->channel[channel].state = NULL;	card->channel[channel].used=0;}/* * setup a divisor value to help with conversion from * 16bit Stereo, down to 8bit stereo/mono or 16bit mono. * assign a divisor of 1 if using 16bit Stereo as that is * the only format that the static image will capture. */static void cs_set_divisor(struct dmabuf *dmabuf){	if(dmabuf->type == CS_TYPE_DAC)		dmabuf->divisor = 1;	else if( !(dmabuf->fmt & CS_FMT_STEREO) && 	    (dmabuf->fmt & CS_FMT_16BIT))		dmabuf->divisor = 2;	else if( (dmabuf->fmt & CS_FMT_STEREO) && 	    !(dmabuf->fmt & CS_FMT_16BIT))		dmabuf->divisor = 2;	else if( !(dmabuf->fmt & CS_FMT_STEREO) && 	    !(dmabuf->fmt & CS_FMT_16BIT))		dmabuf->divisor = 4;	else		dmabuf->divisor = 1;	CS_DBGOUT(CS_PARMS | CS_FUNCTION, 8, printk(		"cs46xx: cs_set_divisor()- %s %d\n",			(dmabuf->type == CS_TYPE_ADC) ? "ADC" : "DAC", 			dmabuf->divisor) );}/** mute some of the more prevalent registers to avoid popping.*/static void cs_mute(struct cs_card *card, int state) {	struct ac97_codec *dev=card->ac97_codec[0];	CS_DBGOUT(CS_FUNCTION, 2, printk(KERN_INFO "cs46xx: cs_mute()+ %s\n",		(state == CS_TRUE) ? "Muting" : "UnMuting") );	if(state == CS_TRUE)	{	/*	* fix pops when powering up on thinkpads	*/		card->pm.u32AC97_master_volume = (u32)cs_ac97_get( dev, 				(u8)BA0_AC97_MASTER_VOLUME); 		card->pm.u32AC97_headphone_volume = (u32)cs_ac97_get(dev, 				(u8)BA0_AC97_HEADPHONE_VOLUME); 		card->pm.u32AC97_master_volume_mono = (u32)cs_ac97_get(dev, 				(u8)BA0_AC97_MASTER_VOLUME_MONO); 		card->pm.u32AC97_pcm_out_volume = (u32)cs_ac97_get(dev, 				(u8)BA0_AC97_PCM_OUT_VOLUME);					cs_ac97_set(dev, (u8)BA0_AC97_MASTER_VOLUME, 0x8000);		cs_ac97_set(dev, (u8)BA0_AC97_HEADPHONE_VOLUME, 0x8000);		cs_ac97_set(dev, (u8)BA0_AC97_MASTER_VOLUME_MONO, 0x8000);		cs_ac97_set(dev, (u8)BA0_AC97_PCM_OUT_VOLUME, 0x8000);	}	else	{		cs_ac97_set(dev, (u8)BA0_AC97_MASTER_VOLUME, card->pm.u32AC97_master_volume);		cs_ac97_set(dev, (u8)BA0_AC97_HEADPHONE_VOLUME, card->pm.u32AC97_headphone_volume);		cs_ac97_set(dev, (u8)BA0_AC97_MASTER_VOLUME_MONO, card->pm.u32AC97_master_volume_mono);		cs_ac97_set(dev, (u8)BA0_AC97_PCM_OUT_VOLUME, card->pm.u32AC97_pcm_out_volume);	}	CS_DBGOUT(CS_FUNCTION, 2, printk(KERN_INFO "cs46xx: cs_mute()-\n"));}/* set playback sample rate */static unsigned int cs_set_dac_rate(struct cs_state * state, unsigned int rate){		struct dmabuf *dmabuf = &state->dmabuf;	unsigned int tmp1, tmp2;	unsigned int phiIncr;	unsigned int correctionPerGOF, correctionPerSec;	unsigned long flags;	CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: cs_set_dac_rate()+ %d\n",rate) );	/*	 *  Compute the values used to drive the actual sample rate conversion.	 *  The following formulas are being computed, using inline assembly	 *  since we need to use 64 bit arithmetic to compute the values:	 *	 *  phiIncr = floor((Fs,in * 2^26) / Fs,out)	 *  correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /         *                                   GOF_PER_SEC)         *  ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -M         *                       GOF_PER_SEC * correctionPerGOF	 *	 *  i.e.	 *	 *  phiIncr:other = dividend:remainder((Fs,in * 2^26) / Fs,out)	 *  correctionPerGOF:correctionPerSec =	 *      dividend:remainder(ulOther / GOF_PER_SEC)	 */	tmp1 = rate << 16;	phiIncr = tmp1 / 48000;	tmp1 -= phiIncr * 48000;	tmp1 <<= 10;	phiIncr <<= 10;	tmp2 = tmp1 / 48000;	phiIncr += tmp2;	tmp1 -= tmp2 * 48000;	correctionPerGOF = tmp1 / GOF_PER_SEC;	tmp1 -= correctionPerGOF * GOF_PER_SEC;	correctionPerSec = tmp1;	/*	 *  Fill in the SampleRateConverter control block.	 */	 	spin_lock_irqsave(&state->card->lock, flags);	cs461x_poke(state->card, BA1_PSRC,	  ((correctionPerSec << 16) & 0xFFFF0000) | (correctionPerGOF & 0xFFFF));	cs461x_poke(state->card, BA1_PPI, phiIncr);	spin_unlock_irqrestore(&state->card->lock, flags);	dmabuf->rate = rate;		CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: cs_set_dac_rate()- %d\n",rate) );	return rate;}/* set recording sample rate */static unsigned int cs_set_adc_rate(struct cs_state * state, unsigned int rate){	struct dmabuf *dmabuf = &state->dmabuf;	struct cs_card *card = state->card;	unsigned int phiIncr, coeffIncr, tmp1, tmp2;	unsigned int correctionPerGOF, correctionPerSec, initialDelay;	unsigned int frameGroupLength, cnt;	unsigned long flags;	CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: cs_set_adc_rate()+ %d\n",rate) );	/*	 *  We can only decimate by up to a factor of 1/9th the hardware rate.	 *  Correct the value if an attempt is made to stray outside that limit.	 */	if ((rate * 9) < 48000)		rate = 48000 / 9;	/*	 *  We can not capture at at rate greater than the Input Rate (48000).	 *  Return an error if an attempt is made to stray outside that limit.	 */	if (rate > 48000)		rate = 48000;	/*	 *  Compute the values used to drive the actual sample rate conversion.	 *  The following formulas are being computed, using inline assembly	 *  since we need to use 64 bit arithmetic to compute the values:	 *	 *     coeffIncr = -floor((Fs,out * 2^23) / Fs,in)	 *     phiIncr = floor((Fs,in * 2^26) / Fs,out)	 *     correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /	 *                                GOF_PER_SEC)

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