⭐ 欢迎来到虫虫下载站! | 📦 资源下载 📁 资源专辑 ℹ️ 关于我们
⭐ 虫虫下载站

📄 s3c2410-uda1341.c

📁 《嵌入式系统设计与实例开发实验教材II:基于ARM9微处理器与Linux操作系统》——音频驱动及应用实验
💻 C
📖 第 1 页 / 共 3 页
字号:
		y |= y << 16;		*dst++ = x;		*dst++ = y;	}	if (from < end) {		u_int v;		__get_user(v, (const u_short *)from);		*dst = v | (v << 16);	}	return 0;}static ssize_t smdk2410_audio_write(struct file *file, const char *buffer, 				    size_t count, loff_t * ppos){	const char *buffer0 = buffer;	audio_stream_t *s = &output_stream;	int chunksize, ret = 0;	DPRINTK("audio_write : start count=%d\n", count);	switch (file->f_flags & O_ACCMODE) {	  	case O_WRONLY:	  	case O_RDWR:			break;	  	default:		  	return -EPERM;	}	if (!s->buffers && audio_setup_buf(s))		return -ENOMEM;	count &= ~0x03;	while (count > 0) {		audio_buf_t *b = s->buf;		if (file->f_flags & O_NONBLOCK) {			ret = -EAGAIN;			if (down_trylock(&b->sem))				break;		} else {			ret = -ERESTARTSYS;			if (down_interruptible(&b->sem))				break;		}		if (audio_channels == 2) {			chunksize = s->fragsize - b->size;			if (chunksize > count)				chunksize = count;			DPRINTK("write %d to %d\n", chunksize, s->buf_idx);			if (copy_from_user(b->start + b->size, buffer, chunksize)) {				up(&b->sem);				return -EFAULT;			}			b->size += chunksize;		} else {			chunksize = (s->fragsize - b->size) >> 1;			if (chunksize > count)				chunksize = count;			DPRINTK("write %d to %d\n", chunksize*2, s->buf_idx);			if (copy_from_user_mono_stereo(b->start + b->size, 				    		       buffer, chunksize)) {				up(&b->sem);				return -EFAULT;			}			b->size += chunksize*2;		}		buffer += chunksize;		count -= chunksize;		if (b->size < s->fragsize) {			up(&b->sem);			break;		}		s3c2410_dma_queue_buffer(s->dma_ch, (void *) b,					   b->dma_addr, b->size, DMA_BUF_WR);		b->size = 0;		NEXT_BUF(s, buf);	}	if ((buffer - buffer0))		ret = buffer - buffer0;	DPRINTK("audio_write : end count=%d\n\n", ret);	return ret;}static ssize_t smdk2410_audio_read(struct file *file, char *buffer,                                         size_t count, loff_t * ppos){	const char *buffer0 = buffer;        audio_stream_t *s = &input_stream;         int chunksize, ret = 0;        DPRINTK("audio_read: count=%d\n", count);        if (ppos != &file->f_pos)                return -ESPIPE;			if (!s->buffers) {		int i;                  if (audio_setup_buf(s))                         return -ENOMEM;                  for (i = 0; i < s->nbfrags; i++) {                         audio_buf_t *b = s->buf;                          down(&b->sem);                          s3c2410_dma_queue_buffer(s->dma_ch, (void *) b,                                         b->dma_addr, s->fragsize, DMA_BUF_RD);                         NEXT_BUF(s, buf);                 }         }	//mdelay(1);	if((IISFIFOC & IISFCON_RX_EN)==0){//start RX FIFO	  //IISFIFOC&=~(IISFCON_RX_DMA|IISFCON_RX_EN);	  //udelay(1);	  //IISFIFOC|=(IISFCON_RX_DMA|IISFCON_RX_EN);	  IISFIFOC |=IISFCON_RX_EN;	  DPRINTK("audio_read: start RX\n");	  	}			DPRINTK("audio_read: DISRCC1=%x, DIDSTC1=%x, DCON1=%x,DSTAT1=%x,DMSKTRIG1=%x, DISRC1=%x, DIDST1=%x, DCDST1=%x, DCSRC1=%x\n", DISRCC1,DIDSTC1, DCON1, DSTAT1, DMTRIG1, DISRC1, DIDST1, DCDST1, DCSRC1);        while (count > 0) {                audio_buf_t *b = s->buf;                /* Wait for a buffer to become full */                if (file->f_flags & O_NONBLOCK) {                        ret = -EAGAIN;                        if (down_trylock(&b->sem))                                break;                } else {                        ret = -ERESTARTSYS;                        if (down_interruptible(&b->sem))                                break;                }                chunksize = b->size;		if (chunksize > count)                        chunksize = count;                DPRINTK("read %d from %d\n", chunksize, s->buf_idx);                if (copy_to_user(buffer, b->start + s->fragsize - b->size,				       	chunksize)) {                        up(&b->sem);                        return -EFAULT;                }		                b->size -= chunksize;                buffer += chunksize;                count -= chunksize;                if (b->size > 0) {                        up(&b->sem);                        break;                }                /* Make current buffer available for DMA again */                s3c2410_dma_queue_buffer(s->dma_ch, (void *) b,					 b->dma_addr, s->fragsize, DMA_BUF_RD);                NEXT_BUF(s, buf);        }	        if ((buffer - buffer0))                ret = buffer - buffer0;     //   DPRINTK("audio_read: return=%d\n", ret);        return ret;}static unsigned int smdk2410_audio_poll(struct file *file, 					struct poll_table_struct *wait){	unsigned int mask = 0;	int i;	DPRINTK("audio_poll(): mode=%s\n",		(file->f_mode & FMODE_WRITE) ? "w" : "");	if (file->f_mode & FMODE_READ) {                if (!input_stream.buffers && audio_setup_buf(&input_stream))                        return -ENOMEM;                poll_wait(file, &input_stream.buf->sem.wait, wait);                for (i = 0; i < input_stream.nbfrags; i++) {                        if (atomic_read(&input_stream.buffers[i].sem.count) > 0)                                mask |= POLLIN | POLLWRNORM;				break;                }        }	if (file->f_mode & FMODE_WRITE) {		if (!output_stream.buffers && audio_setup_buf(&output_stream))			return -ENOMEM;		poll_wait(file, &output_stream.buf->sem.wait, wait);		for (i = 0; i < output_stream.nbfrags; i++) {			if (atomic_read(&output_stream.buffers[i].sem.count) > 0)				mask |= POLLOUT | POLLWRNORM;				break;		}	}	DPRINTK("audio_poll() returned mask of %s\n",		(mask & POLLOUT) ? "w" : "");	return mask;}static loff_t smdk2410_audio_llseek(struct file *file, loff_t offset, 				    int origin){            return -ESPIPE;}static int smdk2410_mixer_ioctl(struct inode *inode, struct file *file,                                 unsigned int cmd, unsigned long arg){        int ret;        long val = 0;	switch (cmd) {		case SOUND_MIXER_INFO:		{			mixer_info info;			strncpy(info.id, "UDA1341", sizeof(info.id));			strncpy(info.name,"Philips UDA1341", sizeof(info.name));			info.modify_counter = audio_mix_modcnt;			return copy_to_user((void *)arg, &info, sizeof(info));		}			case SOUND_OLD_MIXER_INFO:		{			_old_mixer_info info;			strncpy(info.id, "UDA1341", sizeof(info.id));			strncpy(info.name,"Philips UDA1341", sizeof(info.name));			return copy_to_user((void *)arg, &info, sizeof(info));		}		case SOUND_MIXER_READ_STEREODEVS:			return put_user(0, (long *) arg);		case SOUND_MIXER_READ_CAPS:			val = SOUND_CAP_EXCL_INPUT;			return put_user(val, (long *) arg);		case SOUND_MIXER_WRITE_VOLUME:			ret = get_user(val, (long *) arg);			if (ret)				return ret;			uda1341_volume = 63 - (((val & 0xff) + 1) * 63) / 100;			uda1341_l3_address(UDA1341_REG_DATA0);			uda1341_l3_data(uda1341_volume);			break;				case SOUND_MIXER_READ_VOLUME:			val = ((63 - uda1341_volume) * 100) / 63;			val |= val << 8;			return put_user(val, (long *) arg);			case SOUND_MIXER_READ_IGAIN:			val = ((31- mixer_igain) * 100) / 31;			return put_user(val, (int *) arg);		case SOUND_MIXER_WRITE_IGAIN:			ret = get_user(val, (int *) arg);			if (ret)				return ret;			mixer_igain = 31 - (val * 31 / 100);					/* use mixer gain channel 1*/			uda1341_l3_address(UDA1341_REG_DATA0);			uda1341_l3_data(EXTADDR(EXT0));			uda1341_l3_data(EXTDATA(EXT0_CH1_GAIN(mixer_igain)));						break;		default:			DPRINTK("mixer ioctl %u unknown\n", cmd);			return -ENOSYS;	}				audio_mix_modcnt++;	return 0;}static int iispsr_value(int s_bit_clock, int sample_rate){        int i, prescaler = 0;        unsigned long tmpval;        unsigned long tmpval384;        unsigned long tmpval384min = 0xffff; 	tmpval384 = s3c2410_get_bus_clk(GET_PCLK) / s_bit_clock;        for (i = 0; i < 32; i++) {                tmpval = tmpval384/(i+1);                if (PCM_ABS((sample_rate - tmpval)) < tmpval384min) {                        tmpval384min = PCM_ABS((sample_rate - tmpval));                        prescaler = i;                }        }        DPRINTK("prescaler = %d\n", prescaler);        return prescaler;}static long audio_set_dsp_speed(long val){	switch (val) {		case 48000:		case 44100:			IISPSR = (IISPSR_A(iispsr_value(S_CLOCK_FREQ, 44100)) 				| IISPSR_B(iispsr_value(S_CLOCK_FREQ, 44100)));			break;		case 22050:			IISPSR = (IISPSR_A(iispsr_value(S_CLOCK_FREQ, 22050)) 				| IISPSR_B(iispsr_value(S_CLOCK_FREQ, 22050)));			break;		case 11025:			IISPSR = (IISPSR_A(iispsr_value(S_CLOCK_FREQ, 11025)) 				| IISPSR_B(iispsr_value(S_CLOCK_FREQ, 11025)));			break;		case 8000:			IISPSR = (IISPSR_A(iispsr_value(S_CLOCK_FREQ, 8000)) 				| IISPSR_B(iispsr_value(S_CLOCK_FREQ, 8000)));			break;		default:	//add by threewater		{			unsigned long psr;                        psr= (IISPSR_A(iispsr_value(S_CLOCK_FREQ, val))                                | IISPSR_B(iispsr_value(S_CLOCK_FREQ, val)));			if(psr==0)				return -1;			IISPSR = psr;		}	}	audio_rate = val;		return audio_rate;}static int smdk2410_audio_ioctl(struct inode *inode, struct file *file,                                 uint cmd, ulong arg){	long val;	switch (cmd) {	  	case SNDCTL_DSP_SETFMT:			get_user(val, (long *) arg);		  	if (val & AUDIO_FMT_MASK) {			    	audio_fmt = val;			    	break;		  	} else				return -EINVAL;	  	case SNDCTL_DSP_CHANNELS:	  	case SNDCTL_DSP_STEREO:		  	get_user(val, (long *) arg);		  	if (cmd == SNDCTL_DSP_STEREO)			  	val = val ? 2 : 1;		  	if (val != 1 && val != 2)			  	return -EINVAL;		  	audio_channels = val;		  	break;	  	case SOUND_PCM_READ_CHANNELS:		  	put_user(audio_channels, (long *) arg);		 	break;	  	case SNDCTL_DSP_SPEED:		  	get_user(val, (long *) arg);		  	val = audio_set_dsp_speed(val);                        if (val < 0) 				return -EINVAL;		  	put_user(val, (long *) arg);		  	break;	  	case SOUND_PCM_READ_RATE:		  	put_user(audio_rate, (long *) arg);		  	break;	  	case SNDCTL_DSP_GETFMTS:		  	put_user(AUDIO_FMT_MASK, (long *) arg);		  	break;	  	case SNDCTL_DSP_GETBLKSIZE:			if(file->f_mode & FMODE_WRITE)		  		return put_user(audio_fragsize, (long *) arg);			else						return put_user(audio_fragsize, (int *) arg);	  	case SNDCTL_DSP_SETFRAGMENT:		        if (file->f_mode & FMODE_WRITE) {			  		if (output_stream.buffers)			  		return -EBUSY;		  		get_user(val, (long *) arg);		  		audio_fragsize = 1 << (val & 0xFFFF);		  		if (audio_fragsize < 16)			  		audio_fragsize = 16;		  		if (audio_fragsize > 16384)			  		audio_fragsize = 16384;		  		audio_nbfrags = (val >> 16) & 0x7FFF;				if (audio_nbfrags < 2)

⌨️ 快捷键说明

复制代码 Ctrl + C
搜索代码 Ctrl + F
全屏模式 F11
切换主题 Ctrl + Shift + D
显示快捷键 ?
增大字号 Ctrl + =
减小字号 Ctrl + -