📄 cs46xx.c
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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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