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分类: LINUX

2008-09-22 00:00:14


最近做了博创2410开发板上uda1341的驱动,从网上下了一些源代码,直接拿过来在2.6.24上行不通。然后就自己去看内核中和dma相关的代码,最终把驱动弄好了,能顺利播放音乐了。把移植过程介绍一下,供大家参考。
 
遇到的第一个问题是dma通道的映射问题。
通道映射不正确的话,uda1341就没有输出。这个问题我觉得是移植过程中解决的主要问题。关于2.6.24中dma的基本建立过程在我的另一篇文章中有介绍,大家可以看一下:
 
然后直接介绍移植过程中需要修改的地方。首先还是dma通道的问题。在内核的arch/arm/plat-s3c24xx/dma.c的s3c2410_dma_map_channel函数中修改如下:
static struct s3c2410_dma_chan *s3c2410_dma_map_channel(int channel)
{
 struct s3c24xx_dma_order_ch *ord = NULL;
 struct s3c24xx_dma_map *ch_map;
 struct s3c2410_dma_chan *dmach;
 int ch;
 if (dma_sel.map == NULL || channel > dma_sel.map_size)
  return NULL;
 ch_map = dma_sel.map + channel;
 /* first, try the board mapping */
#if 0                                            //这里
 if (dma_order) {
  ord = &dma_order->channels[channel];
  for (ch = 0; ch < dma_channels; ch++) {
   if (!is_channel_valid(ord->list[ch]))
    continue;
   if (s3c2410_chans[ord->list[ch]].in_use == 0) {
    ch = ord->list[ch] & ~DMA_CH_VALID;
    goto found;
   }
  }
  if (ord->flags & DMA_CH_NEVER)
   return NULL;
 }
#endif                                           //这里
 /* second, search the channel map for first free */
就是注释掉一段代码就行了。
然后,还是同一个文件中,修改这个函数:s3c2410_dma_enqueue
这个函数的结尾:
 } else if (chan->state == S3C2410_DMA_IDLE) {
  if (chan->flags & S3C2410_DMAF_AUTOSTART) {
   s3c2410_dma_ctrl(channel, S3C2410_DMAOP_START); //修改的只有这一行
  }
 }
 local_irq_restore(flags);
 return 0;
}
 
上面注释的那行,把第一个参数chan->number,换成了现在的channel。
 
ok,这个文件就改这些。
下面贴出uda1341.c这个文件:
#include  
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#include
#define DMA_BUF_WR 1 
#define DMA_BUF_RD 0
#define DMA_CH1 DMACH_I2S_IN
#define DMA_CH2 DMACH_I2S_OUT
static struct clk *iis_clock; 
static void __iomem *iis_base;
static struct s3c2410_dma_client s3c2410iis_dma_out= { 
 .name = "I2SSDO", 
};
static struct s3c2410_dma_client s3c2410iis_dma_in = { 
 .name = "I2SSDI", 
};
static void init_s3c2410_iis_bus(void);
#define DEF_VOLUME 65
/* UDA1341 Register bits */ 
#define UDA1341_ADDR 0x14
#define UDA1341_REG_DATA0 (UDA1341_ADDR + 0) 
#define UDA1341_REG_STATUS (UDA1341_ADDR + 2)
/* status control */ 
#define STAT0 (0x00) 
#define STAT0_RST (1 << 6) 
#define STAT0_SC_MASK (3 << 4) 
#define STAT0_SC_512FS (0 << 4) 
#define STAT0_SC_384FS (1 << 4) 
#define STAT0_SC_256FS (2 << 4) 
#define STAT0_IF_MASK (7 << 1) 
#define STAT0_IF_I2S (0 << 1) 
#define STAT0_IF_LSB16 (1 << 1) 
#define STAT0_IF_LSB18 (2 << 1) 
#define STAT0_IF_LSB20 (3 << 1) 
#define STAT0_IF_MSB (4 << 1) 
#define STAT0_IF_LSB16MSB (5 << 1) 
#define STAT0_IF_LSB18MSB (6 << 1) 
#define STAT0_IF_LSB20MSB (7 << 1) 
#define STAT0_DC_FILTER (1 << 0) 
#define STAT0_DC_NO_FILTER (0 << 0)
#define STAT1 (0x80) 
#define STAT1_DAC_GAIN (1 << 6) /* gain of DAC */ 
#define STAT1_ADC_GAIN (1 << 5) /* gain of ADC */ 
#define STAT1_ADC_POL (1 << 4) /* polarity of ADC */ 
#define STAT1_DAC_POL (1 << 3) /* polarity of DAC */ 
#define STAT1_DBL_SPD (1 << 2) /* double speed playback */ 
#define STAT1_ADC_ON (1 << 1) /* ADC powered */ 
#define STAT1_DAC_ON (1 << 0) /* DAC powered */
/* data0 direct control */ 
#define DATA0 (0x00) 
#define DATA0_VOLUME_MASK (0x3f) 
#define DATA0_VOLUME(x) (x)
#define DATA1 (0x40) 
#define DATA1_BASS(x) ((x) << 2) 
#define DATA1_BASS_MASK (15 << 2) 
#define DATA1_TREBLE(x) ((x)) 
#define DATA1_TREBLE_MASK (3)
#define DATA2 (0x80) 
#define DATA2_PEAKAFTER (0x1 << 5) 
#define DATA2_DEEMP_NONE (0x0 << 3) 
#define DATA2_DEEMP_32KHz (0x1 << 3) 
#define DATA2_DEEMP_44KHz (0x2 << 3) 
#define DATA2_DEEMP_48KHz (0x3 << 3) 
#define DATA2_MUTE (0x1 << 2) 
#define DATA2_FILTER_FLAT (0x0 << 0) 
#define DATA2_FILTER_MIN (0x1 << 0) 
#define DATA2_FILTER_MAX (0x3 << 0) 
/* data0 extend control */ 
#define EXTADDR(n) (0xc0 | (n)) 
#define EXTDATA(d) (0xe0 | (d))
#define EXT0 0 
#define EXT0_CH1_GAIN(x) (x) 
#define EXT1 1 
#define EXT1_CH2_GAIN(x) (x) 
#define EXT2 2 
#define EXT2_MIC_GAIN_MASK (7 << 2) 
#define EXT2_MIC_GAIN(x) ((x) << 2) 
#define EXT2_MIXMODE_DOUBLEDIFF (0) 
#define EXT2_MIXMODE_CH1 (1) 
#define EXT2_MIXMODE_CH2 (2) 
#define EXT2_MIXMODE_MIX (3) 
#define EXT4 4 
#define EXT4_AGC_ENABLE (1 << 4) 
#define EXT4_INPUT_GAIN_MASK (3) 
#define EXT4_INPUT_GAIN(x) ((x) & 3) 
#define EXT5 5 
#define EXT5_INPUT_GAIN(x) ((x) >> 2) 
#define EXT6 6 
#define EXT6_AGC_CONSTANT_MASK (7 << 2) 
#define EXT6_AGC_CONSTANT(x) ((x) << 2) 
#define EXT6_AGC_LEVEL_MASK (3) 
#define EXT6_AGC_LEVEL(x) (x)
#define AUDIO_NAME "UDA1341" 
#define AUDIO_NAME_VERBOSE "UDA1341 audio driver"
#define AUDIO_FMT_MASK (AFMT_S16_LE) 
#define AUDIO_FMT_DEFAULT (AFMT_S16_LE)
#define AUDIO_CHANNELS_DEFAULT 2 
#define AUDIO_RATE_DEFAULT 8000
#define AUDIO_NBFRAGS_DEFAULT 8 
#define AUDIO_FRAGSIZE_DEFAULT 8192
#define S_CLOCK_FREQ 384

typedef struct { 
 int size; /* buffer size */ 
 char *start; /* point to actual buffer */ 
 dma_addr_t dma_addr; /* physical buffer address */ 
 struct semaphore sem; /* down before touching the buffer */ 
 int master; /* owner for buffer allocation, contain size when true */ 
} audio_buf_t;
typedef struct { 
 audio_buf_t *buffers; /* pointer to audio buffer structures */ 
 audio_buf_t *buf; /* current buffer used by read/write */ 
 u_int buf_idx; /* index for the pointer above */ 
 u_int fragsize; /* fragment i.e. buffer size */ 
 u_int nbfrags; /* nbr of fragments */ 
        enum dma_ch dmach; 
} audio_stream_t;
static audio_stream_t output_stream; 
static audio_stream_t input_stream;
#define NEXT_BUF(_s_,_b_) { \
        (_s_)->_b_##_idx++; \
        (_s_)->_b_##_idx %= (_s_)->nbfrags; \
        (_s_)->_b_ = (_s_)->buffers + (_s_)->_b_##_idx; }
 
static u_int audio_rate; 
static int audio_channels; 
static int audio_fmt; 
static u_int audio_fragsize; 
static u_int audio_nbfrags;

static int audio_rd_refcount; 
static int audio_wr_refcount; 
#define audio_active (audio_rd_refcount | audio_wr_refcount)
static int audio_dev_dsp; 
static int audio_dev_mixer; 
static int audio_mix_modcnt;
static int uda1341_volume; 
static u8 uda_sampling; 
static int uda1341_boost; 
static int mixer_igain=0x4; /* -6db*/
static void uda1341_l3_address(u8 data) 

 int i; 
 unsigned long flags; 
 
 local_irq_save(flags); 
 
 s3c2410_gpio_setpin(S3C2410_GPG9,1);
 s3c2410_gpio_setpin(S3C2410_GPG0,0);
 udelay(10);
 s3c2410_gpio_setpin(S3C2410_GPG8,0);
 udelay(5);
 
 for (i = 0; i < 8; i++) { 
  if (data & 0x1) { 
   s3c2410_gpio_setpin(S3C2410_GPG9,0);
   s3c2410_gpio_setpin(S3C2410_GPG0,1); 
   udelay(1); 
   s3c2410_gpio_setpin(S3C2410_GPG9,1);
   udelay(1);
  } else { 
   s3c2410_gpio_setpin(S3C2410_GPG9,0);
   s3c2410_gpio_setpin(S3C2410_GPG0,0); 
   udelay(1); 
   s3c2410_gpio_setpin(S3C2410_GPG9,1);
   udelay(1);
  } 
  data >>= 1; 
 } 
 udelay(5);
 s3c2410_gpio_setpin(S3C2410_GPG8,1); 
 udelay(1);
 local_irq_restore(flags); 
}
static void uda1341_l3_data(u8 data) 

 int i; 
 unsigned long flags;
 local_irq_save(flags);
 s3c2410_gpio_setpin(S3C2410_GPG8,1);
 udelay(5);
 
 for (i = 0; i < 8; i++) { 
  if (data & 0x1) { 
   s3c2410_gpio_setpin(S3C2410_GPG9,0);
   s3c2410_gpio_setpin(S3C2410_GPG0,1); 
   udelay(1); 
   s3c2410_gpio_setpin(S3C2410_GPG9,1);
   udelay(1);
  } else { 
   s3c2410_gpio_setpin(S3C2410_GPG9,0);
   s3c2410_gpio_setpin(S3C2410_GPG0,0); 
   udelay(1); 
   s3c2410_gpio_setpin(S3C2410_GPG9,1);
   udelay(1);
  }   
  data >>= 1; 
 } 
 udelay(1);
 s3c2410_gpio_setpin(S3C2410_GPG8, 0);
 udelay(2);
 s3c2410_gpio_setpin(S3C2410_GPG8, 1);       
 local_irq_restore(flags); 
}
static void audio_clear_buf(audio_stream_t * s) 

   s3c2410_dma_ctrl(s->dmach, S3C2410_DMAOP_FLUSH); 
 
 if (s->buffers) { 
  int frag; 
  
  for (frag = 0; frag < s->nbfrags; frag++) { 
   if (!s->buffers[frag].master) 
    continue; 
   dma_free_coherent(NULL, 
    s->buffers[frag].master, 
    s->buffers[frag].start, 
    s->buffers[frag].dma_addr); 
  } 
  kfree(s->buffers); 
  s->buffers = NULL; 
 } 
 
 s->buf_idx = 0; 
 s->buf = NULL; 
}
static int audio_setup_buf(audio_stream_t * s) 

 int frag; 
 int dmasize = 0; 
 char *dmabuf = 0; 
 dma_addr_t dmaphys = 0; 
 
 if (s->buffers)
   return -EBUSY; 
 
 s->nbfrags = audio_nbfrags; 
 s->fragsize = audio_fragsize; 
 
 s->buffers = (audio_buf_t *) 
  kmalloc(sizeof(audio_buf_t) * s->nbfrags, GFP_KERNEL); 
 if (!s->buffers) 
  goto err; 
 memset(s->buffers, 0, sizeof(audio_buf_t) * s->nbfrags); 
 
 for (frag = 0; frag < s->nbfrags; frag++) { 
  audio_buf_t *b = &s->buffers[frag]; 
  
  if (!dmasize) { 
   dmasize = (s->nbfrags - frag) * s->fragsize; 
   do { 
    dmabuf = dma_alloc_coherent(NULL, dmasize, &dmaphys, GFP_KERNEL|GFP_DMA); 
    if (!dmabuf) 
     dmasize -= s->fragsize; 
   } while (!dmabuf && dmasize); 
   if (!dmabuf) 
    goto err; 
   b->master = dmasize; 
  } 
  
  b->start = dmabuf; 
  b->dma_addr = dmaphys; 
  sema_init(&b->sem, 1); 
  
  dmabuf += s->fragsize; 
  dmaphys += s->fragsize; 
  dmasize -= s->fragsize; 
 } 
 
 s->buf_idx = 0; 
 s->buf = &s->buffers[0]; 
 return 0; 
 
err: 
 printk(AUDIO_NAME ": unable to allocate audio memory\n "); 
 audio_clear_buf(s); 
 return -ENOMEM; 
}
static void audio_dmaout_done_callback(struct s3c2410_dma_chan *ch, void *buf, int size,enum s3c2410_dma_buffresult result) 

 audio_buf_t *b = (audio_buf_t *) buf; 
 up(&b->sem); 
 wake_up(&b->sem.wait); 
}
static void audio_dmain_done_callback(struct s3c2410_dma_chan *ch, void *buf, int size,enum s3c2410_dma_buffresult result) 

 audio_buf_t *b = (audio_buf_t *) buf; 
 b->size = size; 
 up(&b->sem); 
 wake_up(&b->sem.wait); 

/* using when write */ 
static int audio_sync(struct file *file) 

 audio_stream_t *s = &output_stream; 
 audio_buf_t *b = s->buf; 
 
 printk("audio_sync\n");
 
 if (!s->buffers) 
  return 0; 
 
 if (b->size != 0) { 
  down(&b->sem); 
  s3c2410_dma_enqueue(s->dmach, (void *) b, b->dma_addr, b->size); 
  b->size = 0; 
  NEXT_BUF(s, buf); 
 } 
 
 b = s->buffers + ((s->nbfrags + s->buf_idx - 1) % s->nbfrags); 
 if (down_interruptible(&b->sem)) 
  return -EINTR; 
 up(&b->sem); 
 
 return 0; 
}
static inline int copy_from_user_mono_stereo(char *to, const char *from, int count) 

 u_int *dst = (u_int *)to; 
 const char *end = from + count; 
 
 if (!access_ok(VERIFY_READ, from, count)) 
  return -EFAULT; 
 
 if ((int)from & 0x2) { 
  u_int v; 
  __get_user(v, (const u_short *)from); from += 2; 
  *dst++ = v | (v << 16); 
 } 
 
 while (from < end-2) { 
  u_int v, x, y; 
  __get_user(v, (const u_int *)from); from += 4; 
  x = v << 16; 
  x |= x >> 16; 
  y = v >> 16; 
  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;
 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;
   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;
   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;
  }
                ret = s3c2410_dma_enqueue(s->dmach, (void *)b, b->dma_addr, b->size);
  if(ret) {
   printk("dma enqueue failed.\n");
   return ret;
  }
  b->size = 0;
  NEXT_BUF(s, buf);
 }
 if ((buffer - buffer0))
 ret = buffer - buffer0;
 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; 
 
 printk("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_enqueue(s->dmach, (void *) b, b->dma_addr, s->fragsize); 
   NEXT_BUF(s, buf); 
  } 
 } 
 
 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; 
  
  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_enqueue(s->dmach, (void *) b, b->dma_addr, s->fragsize); 
  
  NEXT_BUF(s, buf); 
 } 
 
 if ((buffer - buffer0)) 
  ret = buffer - buffer0; 
 
 printk("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; 
 
 printk("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; 
  } 
 } 
 
 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: 
   
  return -ENOSYS; 
 } 
 
 audio_mix_modcnt++; 
 return 0; 
}
 
static int iispsr_value(int s_bit_clock, int sample_rate)
{
 unsigned int i,j,prescaler = 0;
 i = s_bit_clock*sample_rate;
 prescaler = clk_get_rate(iis_clock)/i;
 j = clk_get_rate(iis_clock)%i;
 if(j<(i/2))
   return (prescaler - 1);
 return prescaler;
}
static int audio_set_dsp_speed(long val) 

  unsigned long iispsr = 0;
  unsigned long value = 0;
  value = iispsr_value(S_CLOCK_FREQ, val);
  iispsr = (value<<5)|value;
    
  writel(iispsr, iis_base + S3C2410_IISPSR);
  iispsr = readl(iis_base + S3C2410_IISPSR);
  return (audio_rate = val);
}
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)
     audio_nbfrags = 2;
    if (audio_nbfrags * audio_fragsize > 128 * 1024)
     audio_nbfrags = 128 * 1024 / audio_fragsize;
    if (audio_setup_buf(&output_stream))
     return -ENOMEM;
   }
   if (file->f_mode & FMODE_READ) {
    if (input_stream.buffers)
     return -EBUSY;
    get_user(val, (int *) 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)
     audio_nbfrags = 2;
    if (audio_nbfrags * audio_fragsize > 128 * 1024)
     audio_nbfrags = 128 * 1024 / audio_fragsize;
    if (audio_setup_buf(&input_stream))
     return -ENOMEM;
   }
   break;
  case SNDCTL_DSP_SYNC:
   return audio_sync(file);
  case SNDCTL_DSP_GETOSPACE:
  {
   audio_stream_t *s = &output_stream;
   audio_buf_info *inf = (audio_buf_info *) arg;
   int err = access_ok(VERIFY_WRITE, inf, sizeof(*inf));
   int i;
   int frags = 0, bytes = 0;
   if (err)
    return err;
   for (i = 0; i < s->nbfrags; i++) {
    if (atomic_read(&s->buffers[i].sem.count) > 0) {
     if (s->buffers[i].size == 0) frags++;
     bytes += s->fragsize - s->buffers[i].size;
    }
   }
   put_user(frags, &inf->fragments);
   put_user(s->nbfrags, &inf->fragstotal);
   put_user(s->fragsize, &inf->fragsize);
   put_user(bytes, &inf->bytes);
   break;
  }
  case SNDCTL_DSP_GETISPACE:
  {
   audio_stream_t *s = &input_stream;
   audio_buf_info *inf = (audio_buf_info *) arg;
   int err = access_ok(VERIFY_WRITE, inf, sizeof(*inf));
   int i;
   int frags = 0, bytes = 0;
   if (!(file->f_mode & FMODE_READ))
    return -EINVAL;
   if (err)
    return err;
   for(i = 0; i < s->nbfrags; i++){
    if (atomic_read(&s->buffers[i].sem.count) > 0)
    {
     if (s->buffers[i].size == s->fragsize)
      frags++;
     bytes += s->buffers[i].size;
    }
   }
   put_user(frags, &inf->fragments);
   put_user(s->nbfrags, &inf->fragstotal);
   put_user(s->fragsize, &inf->fragsize);
   put_user(bytes, &inf->bytes);
   break;
  }
  case SNDCTL_DSP_RESET:
   if (file->f_mode & FMODE_READ) {
    audio_clear_buf(&input_stream);
   }
   if (file->f_mode & FMODE_WRITE) {
    audio_clear_buf(&output_stream);
   }
   return 0;
  case SNDCTL_DSP_NONBLOCK:
   file->f_flags |= O_NONBLOCK;
   return 0;
  case SNDCTL_DSP_POST:
  case SNDCTL_DSP_SUBDIVIDE:
  case SNDCTL_DSP_GETCAPS:
  case SNDCTL_DSP_GETTRIGGER:
  case SNDCTL_DSP_SETTRIGGER:
  case SNDCTL_DSP_GETIPTR:
  case SNDCTL_DSP_GETOPTR:
  case SNDCTL_DSP_MAPINBUF:
  case SNDCTL_DSP_MAPOUTBUF:
  case SNDCTL_DSP_SETSYNCRO:
  case SNDCTL_DSP_SETDUPLEX:
   return -ENOSYS;
  default:
   return smdk2410_mixer_ioctl(inode, file, cmd, arg);
 }
 return 0;
}

static int smdk2410_audio_open(struct inode *inode, struct file *file) 

 int cold = !audio_active; 
 printk("audio open\n");
 if ((file->f_flags & O_ACCMODE) == O_RDONLY) { 
  if (audio_rd_refcount || audio_wr_refcount) 
   return -EBUSY; 
  audio_rd_refcount++; 
 } else if ((file->f_flags & O_ACCMODE) == O_WRONLY) { 
  if (audio_wr_refcount) 
   return -EBUSY; 
  audio_wr_refcount++; 
 } else if ((file->f_flags & O_ACCMODE) == O_RDWR) { 
  if (audio_rd_refcount || audio_wr_refcount) 
   return -EBUSY; 
  audio_rd_refcount++; 
  audio_wr_refcount++; 
 } else 
  return -EINVAL; 
 
 if (cold) { 
  audio_rate = AUDIO_RATE_DEFAULT; 
  audio_channels = AUDIO_CHANNELS_DEFAULT; 
  audio_fragsize = AUDIO_FRAGSIZE_DEFAULT; 
  audio_nbfrags = AUDIO_NBFRAGS_DEFAULT;
 
  init_s3c2410_iis_bus(); 
  if ((file->f_mode & FMODE_WRITE)){
    audio_clear_buf(&output_stream);
  }
 
  if ((file->f_mode & FMODE_READ)) 
    audio_clear_buf(&input_stream); 
 
 } 
 return 0; 
}

static int smdk2410_mixer_open(struct inode *inode, struct file *file) 

 return 0; 
}

static int smdk2410_audio_release(struct inode *inode, struct file *file) 
{
 if (file->f_mode & FMODE_READ) { 
  if (audio_rd_refcount == 1) 
   audio_clear_buf(&input_stream); 
  audio_rd_refcount = 0; 
 } 
 
 if(file->f_mode & FMODE_WRITE) { 
  if (audio_wr_refcount == 1) { 
   audio_sync(file); 
   audio_clear_buf(&output_stream); 
   audio_wr_refcount = 0; 
  } 
 } 
 
 return 0; 
}

static int smdk2410_mixer_release(struct inode *inode, struct file *file) 

 return 0; 
}

static struct file_operations smdk2410_audio_fops = { 
 .llseek = smdk2410_audio_llseek, 
 .write = smdk2410_audio_write, 
 .read = smdk2410_audio_read, 
 .poll = smdk2410_audio_poll, 
 .ioctl = smdk2410_audio_ioctl, 
 .open = smdk2410_audio_open, 
 .release = smdk2410_audio_release 
};
static struct file_operations smdk2410_mixer_fops = { 
 .ioctl = smdk2410_mixer_ioctl, 
 .open =  smdk2410_mixer_open, 
 .release = smdk2410_mixer_release 
};
static void init_uda1341(void) 

        unsigned long flags;
 uda1341_volume = 62 - ((DEF_VOLUME * 61) / 100); 
 uda1341_boost = 0; 
 uda_sampling = DATA2_DEEMP_NONE; 
 uda_sampling &= ~(DATA2_MUTE);
 local_irq_save(flags);
 s3c2410_gpio_setpin(S3C2410_GPG9, 1);
 s3c2410_gpio_setpin(S3C2410_GPG8, 1);
 local_irq_restore(flags);
 uda1341_l3_address(UDA1341_REG_STATUS);
 uda1341_l3_data(STAT0_RST); 
    
 uda1341_l3_address(UDA1341_REG_STATUS);
 uda1341_l3_data(STAT0_SC_384FS | STAT0_IF_MSB | STAT0_DC_FILTER);    
 uda1341_l3_data(STAT1 | STAT1_DAC_GAIN | STAT1_ADC_GAIN | STAT1_ADC_ON | STAT1_DAC_ON);
 uda1341_l3_address(UDA1341_REG_DATA0);
 uda1341_l3_data(DATA0 |DATA0_VOLUME(uda1341_volume));  
 uda1341_l3_data(DATA1 |DATA1_BASS(uda1341_boost)| DATA1_TREBLE(0));
 uda1341_l3_data(DATA2 |uda_sampling);
 uda1341_l3_data(EXTADDR(EXT2));
 uda1341_l3_data(EXTDATA(EXT2_MIC_GAIN(0x6)| EXT2_MIXMODE_CH2));
 uda1341_l3_data(EXTADDR(EXT5));
 uda1341_l3_data(EXTDATA(0x7));
 local_irq_save(flags);
 s3c2410_gpio_setpin(S3C2410_GPG9, 1);
 s3c2410_gpio_setpin(S3C2410_GPG8, 0);
 local_irq_restore(flags);
}
static void init_s3c2410_iis_bus(void){ 
 unsigned long iiscon, iismod, iisfcon; 
 
 clk_enable(iis_clock); 
 
 iiscon = iismod = iisfcon = 0; 
 
 iiscon |= S3C2410_IISCON_PSCEN; // Enable prescaler 
 iiscon |= S3C2410_IISCON_IISEN; // Enable interface
 iiscon |= S3C2410_IISCON_RXDMAEN; //Enable RX DMA service request 
 iiscon |= S3C2410_IISCON_TXDMAEN; //Enable TX DMA service request 
 
 iismod |= S3C2410_IISMOD_LR_LLOW; // Low for left channel 
 iismod |= S3C2410_IISMOD_MSB;
 iismod |= S3C2410_IISMOD_16BIT; // Serial data bit/channel is 16 bit 
 iismod |= S3C2410_IISMOD_384FS; // Master clock freq = 384 fs 
 iismod |= S3C2410_IISMOD_TXRXMODE; //Set RX Mode 
 iismod |= S3C2410_IISMOD_32FS; // 32 fs 
 
 iisfcon|= S3C2410_IISFCON_RXDMA; //Set RX FIFO acces mode to DMA 
 iisfcon|= S3C2410_IISFCON_RXENABLE; //Enable RX Fifo 
 iisfcon|= S3C2410_IISFCON_TXDMA; //Set RX FIFO acces mode to DMA 
 iisfcon|= S3C2410_IISFCON_TXENABLE;
 audio_set_dsp_speed(44100); 
 
 writel(iismod, iis_base + S3C2410_IISMOD); 
 writel(iisfcon, iis_base + S3C2410_IISFCON); 
 writel(iiscon, iis_base + S3C2410_IISCON);
}
static int __init audio_init_dma(audio_stream_t * s, char *desc) 

 int ret ; 
 enum s3c2410_dmasrc source; 
 int hwcfg; 
 unsigned long devaddr; 
 int dcon; 
 unsigned int flags = 0; 
 
 if(s->dmach == DMA_CH2){ 
  source = S3C2410_DMASRC_MEM; 
  hwcfg = 3; 
  devaddr = 0x55000010; 
  dcon = 0xa0800000; 
  flags = S3C2410_DMAF_AUTOSTART; 
  
  ret = s3c2410_dma_request(s->dmach, &s3c2410iis_dma_out, NULL);
  s3c2410_dma_devconfig(s->dmach, source, hwcfg, devaddr); 
  s3c2410_dma_config(s->dmach, 2, dcon); 
  s3c2410_dma_set_buffdone_fn(s->dmach, audio_dmaout_done_callback); 
  s3c2410_dma_setflags(s->dmach, flags);
  return ret; 
 } 
 else if(s->dmach == DMA_CH1){ 
  source =S3C2410_DMASRC_HW;
  hwcfg =3;
  devaddr = 0x55000010;
  dcon = 0xa2900000;
  flags = S3C2410_DMAF_AUTOSTART;
  
  ret = s3c2410_dma_request(s->dmach, &s3c2410iis_dma_in, NULL);
  s3c2410_dma_devconfig(s->dmach, source, hwcfg, devaddr);
  s3c2410_dma_config(s->dmach, 2, dcon);
  s3c2410_dma_set_buffdone_fn(s->dmach, audio_dmain_done_callback);
  s3c2410_dma_setflags(s->dmach, flags);
  return ret ;
 } 
 else 
  return 1; 
}
static int audio_clear_dma(audio_stream_t * s,struct s3c2410_dma_client *client) 

 s3c2410_dma_set_buffdone_fn(s->dmach, NULL); 
 s3c2410_dma_free(s->dmach, client); 
 return 0; 
}
static int __init s3c2410iis_probe(struct platform_device *pdev) { 
 struct resource *res;
 int size;
        unsigned long flags;
 res = platform_get_resource(pdev, IORESOURCE_MEM, 0); 
 if (res == NULL) { 
  printk(KERN_INFO "failed to get memory region resouce\n"); 
  return -ENOENT; 
 }
 
 if (!request_mem_region(res->start, 0x20, pdev->name)) {
   printk(KERN_INFO "failed to get memory region\n"); 
          return -EBUSY;       
  }
 
 size = (res->end - res->start) + 1;
 iis_base = ioremap(res->start, size); 
 if (iis_base == NULL) { 
  printk(KERN_INFO "failed to ioremap() region\n"); 
  return -EINVAL; 
 } 
 
 iis_clock = clk_get(&pdev->dev, "iis"); 
 if (iis_clock == NULL) { 
  printk(KERN_INFO "failed to find clock source\n"); 
  return -ENOENT; 
 } 
 
 clk_enable(iis_clock); 
 
 local_irq_save(flags); 
 
 /* GPB 4: L3CLOCK, OUTPUT */ 
 s3c2410_gpio_cfgpin(S3C2410_GPG9, S3C2410_GPG9_OUTP);
 s3c2410_gpio_pullup(S3C2410_GPG9, 1);
 /* GPB 3: L3DATA, OUTPUT */ 
 s3c2410_gpio_cfgpin(S3C2410_GPG0,S3C2410_GPG0_OUTP);
 s3c2410_gpio_pullup(S3C2410_GPG0, 1);
 /* GPB 2: L3MODE, OUTPUT */ 
 s3c2410_gpio_cfgpin(S3C2410_GPG8,S3C2410_GPG8_OUTP);
 s3c2410_gpio_pullup(S3C2410_GPG8, 1);

 /* GPE 3: I2SSDI */ 
 s3c2410_gpio_cfgpin(S3C2410_GPE3,S3C2410_GPE3_I2SSDI);
 s3c2410_gpio_pullup(S3C2410_GPE3, 0);
 /* GPE 0: I2SLRCK */ 
 s3c2410_gpio_cfgpin(S3C2410_GPE0,S3C2410_GPE0_I2SLRCK); 
 s3c2410_gpio_pullup(S3C2410_GPE0, 0);
 /* GPE 1: I2SSCLK */ 
 s3c2410_gpio_cfgpin(S3C2410_GPE1,S3C2410_GPE1_I2SSCLK);
 s3c2410_gpio_pullup(S3C2410_GPE1, 0);
 /* GPE 2: CDCLK */ 
 s3c2410_gpio_cfgpin(S3C2410_GPE2,S3C2410_GPE2_CDCLK);
 s3c2410_gpio_pullup(S3C2410_GPE2, 0); 
 /* GPE 4: I2SSDO */ 
 s3c2410_gpio_cfgpin(S3C2410_GPE4,S3C2410_GPE4_I2SSDO);
 s3c2410_gpio_pullup(S3C2410_GPE4, 0);
 
 local_irq_restore(flags);
 
 init_uda1341(); 
 
 output_stream.dmach = DMA_CH2; 
 
 if (audio_init_dma(&output_stream, "UDA1341 out")) { 
  audio_clear_dma(&output_stream,&s3c2410iis_dma_out); 
  printk( KERN_WARNING AUDIO_NAME_VERBOSE 
   ": unable to get DMA channels\n" ); 
  return -EBUSY; 
 }
 
 input_stream.dmach = DMA_CH1;
 
 if (audio_init_dma(&input_stream, "UDA1341 in")) { 
  audio_clear_dma(&input_stream,&s3c2410iis_dma_in); 
  printk( KERN_WARNING AUDIO_NAME_VERBOSE 
   ": unable to get DMA channels\n" ); 
  return -EBUSY; 
 }
 
 audio_dev_dsp = register_sound_dsp(&smdk2410_audio_fops, -1); 
 audio_dev_mixer = register_sound_mixer(&smdk2410_mixer_fops, -1); 
 
 printk(AUDIO_NAME_VERBOSE " initialized\n"); 
 
 return 0; 
}

static int s3c2410iis_remove(struct platform_device *dev) { 
 unregister_sound_dsp(audio_dev_dsp); 
 unregister_sound_mixer(audio_dev_mixer); 
 audio_clear_dma(&output_stream,&s3c2410iis_dma_out); 
 audio_clear_dma(&input_stream,&s3c2410iis_dma_in); /* input */ 
 printk(AUDIO_NAME_VERBOSE " unloaded\n"); 
 return 0; 
}

static struct platform_driver s3c2410iis_driver = { 
  .probe = s3c2410iis_probe, 
  .remove = s3c2410iis_remove,
  .driver = {
    .name = "s3c2410-iis",
    .owner = THIS_MODULE,
  },
};
static int __init s3c2410_uda1341_init(void) { 
 memzero(&input_stream, sizeof(audio_stream_t)); 
 memzero(&output_stream, sizeof(audio_stream_t)); 
 return platform_driver_register(&s3c2410iis_driver); 
}
static void __exit s3c2410_uda1341_exit(void) { 
 platform_driver_unregister(&s3c2410iis_driver); 
}

module_init(s3c2410_uda1341_init); 
module_exit(s3c2410_uda1341_exit);
MODULE_LICENSE("GPL"); 
MODULE_DESCRIPTION("S3C2410 uda1341 sound driver"); 
把上面的文件保存到sound/oss中,然后修改sound/oss中的kconfig和makefile,使内核加入该文件,然后编译就可以了。
 
值得注意的地方是,在驱动程序中DMA_CH1和DMA_CH2我是定义成DMACH_I2S_IN和DMACH_I2S_OUT,与其他人的程序中有一定的区别。上面两个量是枚举类型,大家可以查看include/asm-arm/plat-s3c24xx/dma.h文件,里面有它们的定义。
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给主人留下些什么吧!~~

younger02008-09-23 15:24:22

其中sound/oss中的kconfig添加以下代码 # drivers/sound/Config.in # # 18 Apr 1998, Michael Elizabeth Chastain, # More hacking for modularisation. # # Prompt user for primary drivers. #added by yg 08-09-21 config SOUND_UDA1341 tristate "S3C2410 UDA1341 driver (S3C2410)" depends on SOUND_PRIME!=n && SOUND && ARM && ARCH_SMDK2410 #added end config SOUND_BCM_CS4297A makefile添加以下代码: obj-$(CONFIG_SOUND_CS4232) += cs4232.o ad1848.o #added by yg 08-0