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android_kernel_xiaomi_sm8350/arch/ppc/8xx_io/cs4218_tdm.c
Linus Torvalds 1da177e4c3 Linux-2.6.12-rc2 
Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.

Let it rip!
2005-04-16 15:20:36 -07:00

2837 lines
70 KiB
C
Raw Blame History

/* This is a modified version of linux/drivers/sound/dmasound.c to
* support the CS4218 codec on the 8xx TDM port. Thanks to everyone
* that contributed to the dmasound software (which includes me :-).
*
* The CS4218 is configured in Mode 4, sub-mode 0. This provides
* left/right data only on the TDM port, as a 32-bit word, per frame
* pulse. The control of the CS4218 is provided by some other means,
* like the SPI port.
* Dan Malek (dmalek@jlc.net)
*/
#include <linux/module.h>
#include <linux/sched.h>
#include <linux/timer.h>
#include <linux/major.h>
#include <linux/config.h>
#include <linux/fcntl.h>
#include <linux/errno.h>
#include <linux/mm.h>
#include <linux/slab.h>
#include <linux/sound.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <asm/system.h>
#include <asm/irq.h>
#include <asm/pgtable.h>
#include <asm/uaccess.h>
#include <asm/io.h>
/* Should probably do something different with this path name.....
* Actually, I should just stop using it...
*/
#include "cs4218.h"
#include <linux/soundcard.h>
#include <asm/mpc8xx.h>
#include <asm/8xx_immap.h>
#include <asm/commproc.h>
#define DMASND_CS4218 5
#define MAX_CATCH_RADIUS 10
#define MIN_BUFFERS 4
#define MIN_BUFSIZE 4
#define MAX_BUFSIZE 128
#define HAS_8BIT_TABLES
static int sq_unit = -1;
static int mixer_unit = -1;
static int state_unit = -1;
static int irq_installed = 0;
static char **sound_buffers = NULL;
static char **sound_read_buffers = NULL;
static DEFINE_SPINLOCK(cs4218_lock);
/* Local copies of things we put in the control register. Output
* volume, like most codecs is really attenuation.
*/
static int cs4218_rate_index;
/*
* Stuff for outputting a beep. The values range from -327 to +327
* so we can multiply by an amplitude in the range 0..100 to get a
* signed short value to put in the output buffer.
*/
static short beep_wform[256] = {
0, 40, 79, 117, 153, 187, 218, 245,
269, 288, 304, 316, 323, 327, 327, 324,
318, 310, 299, 288, 275, 262, 249, 236,
224, 213, 204, 196, 190, 186, 183, 182,
182, 183, 186, 189, 192, 196, 200, 203,
206, 208, 209, 209, 209, 207, 204, 201,
197, 193, 188, 183, 179, 174, 170, 166,
163, 161, 160, 159, 159, 160, 161, 162,
164, 166, 168, 169, 171, 171, 171, 170,
169, 167, 163, 159, 155, 150, 144, 139,
133, 128, 122, 117, 113, 110, 107, 105,
103, 103, 103, 103, 104, 104, 105, 105,
105, 103, 101, 97, 92, 86, 78, 68,
58, 45, 32, 18, 3, -11, -26, -41,
-55, -68, -79, -88, -95, -100, -102, -102,
-99, -93, -85, -75, -62, -48, -33, -16,
0, 16, 33, 48, 62, 75, 85, 93,
99, 102, 102, 100, 95, 88, 79, 68,
55, 41, 26, 11, -3, -18, -32, -45,
-58, -68, -78, -86, -92, -97, -101, -103,
-105, -105, -105, -104, -104, -103, -103, -103,
-103, -105, -107, -110, -113, -117, -122, -128,
-133, -139, -144, -150, -155, -159, -163, -167,
-169, -170, -171, -171, -171, -169, -168, -166,
-164, -162, -161, -160, -159, -159, -160, -161,
-163, -166, -170, -174, -179, -183, -188, -193,
-197, -201, -204, -207, -209, -209, -209, -208,
-206, -203, -200, -196, -192, -189, -186, -183,
-182, -182, -183, -186, -190, -196, -204, -213,
-224, -236, -249, -262, -275, -288, -299, -310,
-318, -324, -327, -327, -323, -316, -304, -288,
-269, -245, -218, -187, -153, -117, -79, -40,
};
#define BEEP_SPEED 5 /* 22050 Hz sample rate */
#define BEEP_BUFLEN 512
#define BEEP_VOLUME 15 /* 0 - 100 */
static int beep_volume = BEEP_VOLUME;
static int beep_playing = 0;
static int beep_state = 0;
static short *beep_buf;
static void (*orig_mksound)(unsigned int, unsigned int);
/* This is found someplace else......I guess in the keyboard driver
* we don't include.
*/
static void (*kd_mksound)(unsigned int, unsigned int);
static int catchRadius = 0;
static int numBufs = 4, bufSize = 32;
static int numReadBufs = 4, readbufSize = 32;
/* TDM/Serial transmit and receive buffer descriptors.
*/
static volatile cbd_t *rx_base, *rx_cur, *tx_base, *tx_cur;
MODULE_PARM(catchRadius, "i");
MODULE_PARM(numBufs, "i");
MODULE_PARM(bufSize, "i");
MODULE_PARM(numreadBufs, "i");
MODULE_PARM(readbufSize, "i");
#define arraysize(x) (sizeof(x)/sizeof(*(x)))
#define le2be16(x) (((x)<<8 & 0xff00) | ((x)>>8 & 0x00ff))
#define le2be16dbl(x) (((x)<<8 & 0xff00ff00) | ((x)>>8 & 0x00ff00ff))
#define IOCTL_IN(arg, ret) \
do { int error = get_user(ret, (int *)(arg)); \
if (error) return error; \
} while (0)
#define IOCTL_OUT(arg, ret) ioctl_return((int *)(arg), ret)
/* CS4218 serial port control in mode 4.
*/
#define CS_INTMASK ((uint)0x40000000)
#define CS_DO1 ((uint)0x20000000)
#define CS_LATTEN ((uint)0x1f000000)
#define CS_RATTEN ((uint)0x00f80000)
#define CS_MUTE ((uint)0x00040000)
#define CS_ISL ((uint)0x00020000)
#define CS_ISR ((uint)0x00010000)
#define CS_LGAIN ((uint)0x0000f000)
#define CS_RGAIN ((uint)0x00000f00)
#define CS_LATTEN_SET(X) (((X) & 0x1f) << 24)
#define CS_RATTEN_SET(X) (((X) & 0x1f) << 19)
#define CS_LGAIN_SET(X) (((X) & 0x0f) << 12)
#define CS_RGAIN_SET(X) (((X) & 0x0f) << 8)
#define CS_LATTEN_GET(X) (((X) >> 24) & 0x1f)
#define CS_RATTEN_GET(X) (((X) >> 19) & 0x1f)
#define CS_LGAIN_GET(X) (((X) >> 12) & 0x0f)
#define CS_RGAIN_GET(X) (((X) >> 8) & 0x0f)
/* The control register is effectively write only. We have to keep a copy
* of what we write.
*/
static uint cs4218_control;
/* A place to store expanding information.
*/
static int expand_bal;
static int expand_data;
/* Since I can't make the microcode patch work for the SPI, I just
* clock the bits using software.
*/
static void sw_spi_init(void);
static void sw_spi_io(u_char *obuf, u_char *ibuf, uint bcnt);
static uint cs4218_ctl_write(uint ctlreg);
/*** Some low level helpers **************************************************/
/* 16 bit mu-law */
static short ulaw2dma16[] = {
-32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956,
-23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764,
-15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412,
-11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316,
-7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140,
-5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092,
-3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004,
-2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980,
-1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436,
-1372, -1308, -1244, -1180, -1116, -1052, -988, -924,
-876, -844, -812, -780, -748, -716, -684, -652,
-620, -588, -556, -524, -492, -460, -428, -396,
-372, -356, -340, -324, -308, -292, -276, -260,
-244, -228, -212, -196, -180, -164, -148, -132,
-120, -112, -104, -96, -88, -80, -72, -64,
-56, -48, -40, -32, -24, -16, -8, 0,
32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956,
23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764,
15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412,
11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316,
7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140,
5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092,
3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004,
2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980,
1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436,
1372, 1308, 1244, 1180, 1116, 1052, 988, 924,
876, 844, 812, 780, 748, 716, 684, 652,
620, 588, 556, 524, 492, 460, 428, 396,
372, 356, 340, 324, 308, 292, 276, 260,
244, 228, 212, 196, 180, 164, 148, 132,
120, 112, 104, 96, 88, 80, 72, 64,
56, 48, 40, 32, 24, 16, 8, 0,
};
/* 16 bit A-law */
static short alaw2dma16[] = {
-5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736,
-7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784,
-2752, -2624, -3008, -2880, -2240, -2112, -2496, -2368,
-3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392,
-22016, -20992, -24064, -23040, -17920, -16896, -19968, -18944,
-30208, -29184, -32256, -31232, -26112, -25088, -28160, -27136,
-11008, -10496, -12032, -11520, -8960, -8448, -9984, -9472,
-15104, -14592, -16128, -15616, -13056, -12544, -14080, -13568,
-344, -328, -376, -360, -280, -264, -312, -296,
-472, -456, -504, -488, -408, -392, -440, -424,
-88, -72, -120, -104, -24, -8, -56, -40,
-216, -200, -248, -232, -152, -136, -184, -168,
-1376, -1312, -1504, -1440, -1120, -1056, -1248, -1184,
-1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696,
-688, -656, -752, -720, -560, -528, -624, -592,
-944, -912, -1008, -976, -816, -784, -880, -848,
5504, 5248, 6016, 5760, 4480, 4224, 4992, 4736,
7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784,
2752, 2624, 3008, 2880, 2240, 2112, 2496, 2368,
3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392,
22016, 20992, 24064, 23040, 17920, 16896, 19968, 18944,
30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136,
11008, 10496, 12032, 11520, 8960, 8448, 9984, 9472,
15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568,
344, 328, 376, 360, 280, 264, 312, 296,
472, 456, 504, 488, 408, 392, 440, 424,
88, 72, 120, 104, 24, 8, 56, 40,
216, 200, 248, 232, 152, 136, 184, 168,
1376, 1312, 1504, 1440, 1120, 1056, 1248, 1184,
1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696,
688, 656, 752, 720, 560, 528, 624, 592,
944, 912, 1008, 976, 816, 784, 880, 848,
};
/*** Translations ************************************************************/
static ssize_t cs4218_ct_law(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ct_s8(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ct_u8(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ct_s16(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ct_u16(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ctx_law(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ctx_s8(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ctx_u8(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ctx_s16(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ctx_u16(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ct_s16_read(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t cs4218_ct_u16_read(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
/*** Low level stuff *********************************************************/
struct cs_sound_settings {
MACHINE mach; /* machine dependent things */
SETTINGS hard; /* hardware settings */
SETTINGS soft; /* software settings */
SETTINGS dsp; /* /dev/dsp default settings */
TRANS *trans_write; /* supported translations for playback */
TRANS *trans_read; /* supported translations for record */
int volume_left; /* volume (range is machine dependent) */
int volume_right;
int bass; /* tone (range is machine dependent) */
int treble;
int gain;
int minDev; /* minor device number currently open */
};
static struct cs_sound_settings sound;
static void *CS_Alloc(unsigned int size, int flags);
static void CS_Free(void *ptr, unsigned int size);
static int CS_IrqInit(void);
#ifdef MODULE
static void CS_IrqCleanup(void);
#endif /* MODULE */
static void CS_Silence(void);
static void CS_Init(void);
static void CS_Play(void);
static void CS_Record(void);
static int CS_SetFormat(int format);
static int CS_SetVolume(int volume);
static void cs4218_tdm_tx_intr(void *devid);
static void cs4218_tdm_rx_intr(void *devid);
static void cs4218_intr(void *devid, struct pt_regs *regs);
static int cs_get_volume(uint reg);
static int cs_volume_setter(int volume, int mute);
static int cs_get_gain(uint reg);
static int cs_set_gain(int gain);
static void cs_mksound(unsigned int hz, unsigned int ticks);
static void cs_nosound(unsigned long xx);
/*** Mid level stuff *********************************************************/
static void sound_silence(void);
static void sound_init(void);
static int sound_set_format(int format);
static int sound_set_speed(int speed);
static int sound_set_stereo(int stereo);
static int sound_set_volume(int volume);
static ssize_t sound_copy_translate(const u_char *userPtr,
size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
static ssize_t sound_copy_translate_read(const u_char *userPtr,
size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft);
/*
* /dev/mixer abstraction
*/
struct sound_mixer {
int busy;
int modify_counter;
};
static struct sound_mixer mixer;
static struct sound_queue sq;
static struct sound_queue read_sq;
#define sq_block_address(i) (sq.buffers[i])
#define SIGNAL_RECEIVED (signal_pending(current))
#define NON_BLOCKING(open_mode) (open_mode & O_NONBLOCK)
#define ONE_SECOND HZ /* in jiffies (100ths of a second) */
#define NO_TIME_LIMIT 0xffffffff
/*
* /dev/sndstat
*/
struct sound_state {
int busy;
char buf[512];
int len, ptr;
};
static struct sound_state state;
/*** Common stuff ********************************************************/
static long long sound_lseek(struct file *file, long long offset, int orig);
/*** Config & Setup **********************************************************/
void dmasound_setup(char *str, int *ints);
/*** Translations ************************************************************/
/* ++TeSche: radically changed for new expanding purposes...
*
* These two routines now deal with copying/expanding/translating the samples
* from user space into our buffer at the right frequency. They take care about
* how much data there's actually to read, how much buffer space there is and
* to convert samples into the right frequency/encoding. They will only work on
* complete samples so it may happen they leave some bytes in the input stream
* if the user didn't write a multiple of the current sample size. They both
* return the number of bytes they've used from both streams so you may detect
* such a situation. Luckily all programs should be able to cope with that.
*
* I think I've optimized anything as far as one can do in plain C, all
* variables should fit in registers and the loops are really short. There's
* one loop for every possible situation. Writing a more generalized and thus
* parameterized loop would only produce slower code. Feel free to optimize
* this in assembler if you like. :)
*
* I think these routines belong here because they're not yet really hardware
* independent, especially the fact that the Falcon can play 16bit samples
* only in stereo is hardcoded in both of them!
*
* ++geert: split in even more functions (one per format)
*/
static ssize_t cs4218_ct_law(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
short *table = sound.soft.format == AFMT_MU_LAW ? ulaw2dma16: alaw2dma16;
ssize_t count, used;
short *p = (short *) &frame[*frameUsed];
int val, stereo = sound.soft.stereo;
frameLeft >>= 2;
if (stereo)
userCount >>= 1;
used = count = min(userCount, frameLeft);
while (count > 0) {
u_char data;
if (get_user(data, userPtr++))
return -EFAULT;
val = table[data];
*p++ = val;
if (stereo) {
if (get_user(data, userPtr++))
return -EFAULT;
val = table[data];
}
*p++ = val;
count--;
}
*frameUsed += used * 4;
return stereo? used * 2: used;
}
static ssize_t cs4218_ct_s8(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t count, used;
short *p = (short *) &frame[*frameUsed];
int val, stereo = sound.soft.stereo;
frameLeft >>= 2;
if (stereo)
userCount >>= 1;
used = count = min(userCount, frameLeft);
while (count > 0) {
u_char data;
if (get_user(data, userPtr++))
return -EFAULT;
val = data << 8;
*p++ = val;
if (stereo) {
if (get_user(data, userPtr++))
return -EFAULT;
val = data << 8;
}
*p++ = val;
count--;
}
*frameUsed += used * 4;
return stereo? used * 2: used;
}
static ssize_t cs4218_ct_u8(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t count, used;
short *p = (short *) &frame[*frameUsed];
int val, stereo = sound.soft.stereo;
frameLeft >>= 2;
if (stereo)
userCount >>= 1;
used = count = min(userCount, frameLeft);
while (count > 0) {
u_char data;
if (get_user(data, userPtr++))
return -EFAULT;
val = (data ^ 0x80) << 8;
*p++ = val;
if (stereo) {
if (get_user(data, userPtr++))
return -EFAULT;
val = (data ^ 0x80) << 8;
}
*p++ = val;
count--;
}
*frameUsed += used * 4;
return stereo? used * 2: used;
}
/* This is the default format of the codec. Signed, 16-bit stereo
* generated by an application shouldn't have to be copied at all.
* We should just get the phsical address of the buffers and update
* the TDM BDs directly.
*/
static ssize_t cs4218_ct_s16(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t count, used;
int stereo = sound.soft.stereo;
short *fp = (short *) &frame[*frameUsed];
frameLeft >>= 2;
userCount >>= (stereo? 2: 1);
used = count = min(userCount, frameLeft);
if (!stereo) {
short *up = (short *) userPtr;
while (count > 0) {
short data;
if (get_user(data, up++))
return -EFAULT;
*fp++ = data;
*fp++ = data;
count--;
}
} else {
if (copy_from_user(fp, userPtr, count * 4))
return -EFAULT;
}
*frameUsed += used * 4;
return stereo? used * 4: used * 2;
}
static ssize_t cs4218_ct_u16(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t count, used;
int mask = (sound.soft.format == AFMT_U16_LE? 0x0080: 0x8000);
int stereo = sound.soft.stereo;
short *fp = (short *) &frame[*frameUsed];
short *up = (short *) userPtr;
frameLeft >>= 2;
userCount >>= (stereo? 2: 1);
used = count = min(userCount, frameLeft);
while (count > 0) {
int data;
if (get_user(data, up++))
return -EFAULT;
data ^= mask;
*fp++ = data;
if (stereo) {
if (get_user(data, up++))
return -EFAULT;
data ^= mask;
}
*fp++ = data;
count--;
}
*frameUsed += used * 4;
return stereo? used * 4: used * 2;
}
static ssize_t cs4218_ctx_law(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
unsigned short *table = (unsigned short *)
(sound.soft.format == AFMT_MU_LAW ? ulaw2dma16: alaw2dma16);
unsigned int data = expand_data;
unsigned int *p = (unsigned int *) &frame[*frameUsed];
int bal = expand_bal;
int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
int utotal, ftotal;
int stereo = sound.soft.stereo;
frameLeft >>= 2;
if (stereo)
userCount >>= 1;
ftotal = frameLeft;
utotal = userCount;
while (frameLeft) {
u_char c;
if (bal < 0) {
if (userCount == 0)
break;
if (get_user(c, userPtr++))
return -EFAULT;
data = table[c];
if (stereo) {
if (get_user(c, userPtr++))
return -EFAULT;
data = (data << 16) + table[c];
} else
data = (data << 16) + data;
userCount--;
bal += hSpeed;
}
*p++ = data;
frameLeft--;
bal -= sSpeed;
}
expand_bal = bal;
expand_data = data;
*frameUsed += (ftotal - frameLeft) * 4;
utotal -= userCount;
return stereo? utotal * 2: utotal;
}
static ssize_t cs4218_ctx_s8(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
unsigned int *p = (unsigned int *) &frame[*frameUsed];
unsigned int data = expand_data;
int bal = expand_bal;
int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
int stereo = sound.soft.stereo;
int utotal, ftotal;
frameLeft >>= 2;
if (stereo)
userCount >>= 1;
ftotal = frameLeft;
utotal = userCount;
while (frameLeft) {
u_char c;
if (bal < 0) {
if (userCount == 0)
break;
if (get_user(c, userPtr++))
return -EFAULT;
data = c << 8;
if (stereo) {
if (get_user(c, userPtr++))
return -EFAULT;
data = (data << 16) + (c << 8);
} else
data = (data << 16) + data;
userCount--;
bal += hSpeed;
}
*p++ = data;
frameLeft--;
bal -= sSpeed;
}
expand_bal = bal;
expand_data = data;
*frameUsed += (ftotal - frameLeft) * 4;
utotal -= userCount;
return stereo? utotal * 2: utotal;
}
static ssize_t cs4218_ctx_u8(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
unsigned int *p = (unsigned int *) &frame[*frameUsed];
unsigned int data = expand_data;
int bal = expand_bal;
int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
int stereo = sound.soft.stereo;
int utotal, ftotal;
frameLeft >>= 2;
if (stereo)
userCount >>= 1;
ftotal = frameLeft;
utotal = userCount;
while (frameLeft) {
u_char c;
if (bal < 0) {
if (userCount == 0)
break;
if (get_user(c, userPtr++))
return -EFAULT;
data = (c ^ 0x80) << 8;
if (stereo) {
if (get_user(c, userPtr++))
return -EFAULT;
data = (data << 16) + ((c ^ 0x80) << 8);
} else
data = (data << 16) + data;
userCount--;
bal += hSpeed;
}
*p++ = data;
frameLeft--;
bal -= sSpeed;
}
expand_bal = bal;
expand_data = data;
*frameUsed += (ftotal - frameLeft) * 4;
utotal -= userCount;
return stereo? utotal * 2: utotal;
}
static ssize_t cs4218_ctx_s16(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
unsigned int *p = (unsigned int *) &frame[*frameUsed];
unsigned int data = expand_data;
unsigned short *up = (unsigned short *) userPtr;
int bal = expand_bal;
int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
int stereo = sound.soft.stereo;
int utotal, ftotal;
frameLeft >>= 2;
userCount >>= (stereo? 2: 1);
ftotal = frameLeft;
utotal = userCount;
while (frameLeft) {
unsigned short c;
if (bal < 0) {
if (userCount == 0)
break;
if (get_user(data, up++))
return -EFAULT;
if (stereo) {
if (get_user(c, up++))
return -EFAULT;
data = (data << 16) + c;
} else
data = (data << 16) + data;
userCount--;
bal += hSpeed;
}
*p++ = data;
frameLeft--;
bal -= sSpeed;
}
expand_bal = bal;
expand_data = data;
*frameUsed += (ftotal - frameLeft) * 4;
utotal -= userCount;
return stereo? utotal * 4: utotal * 2;
}
static ssize_t cs4218_ctx_u16(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
int mask = (sound.soft.format == AFMT_U16_LE? 0x0080: 0x8000);
unsigned int *p = (unsigned int *) &frame[*frameUsed];
unsigned int data = expand_data;
unsigned short *up = (unsigned short *) userPtr;
int bal = expand_bal;
int hSpeed = sound.hard.speed, sSpeed = sound.soft.speed;
int stereo = sound.soft.stereo;
int utotal, ftotal;
frameLeft >>= 2;
userCount >>= (stereo? 2: 1);
ftotal = frameLeft;
utotal = userCount;
while (frameLeft) {
unsigned short c;
if (bal < 0) {
if (userCount == 0)
break;
if (get_user(data, up++))
return -EFAULT;
data ^= mask;
if (stereo) {
if (get_user(c, up++))
return -EFAULT;
data = (data << 16) + (c ^ mask);
} else
data = (data << 16) + data;
userCount--;
bal += hSpeed;
}
*p++ = data;
frameLeft--;
bal -= sSpeed;
}
expand_bal = bal;
expand_data = data;
*frameUsed += (ftotal - frameLeft) * 4;
utotal -= userCount;
return stereo? utotal * 4: utotal * 2;
}
static ssize_t cs4218_ct_s8_read(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t count, used;
short *p = (short *) &frame[*frameUsed];
int val, stereo = sound.soft.stereo;
frameLeft >>= 2;
if (stereo)
userCount >>= 1;
used = count = min(userCount, frameLeft);
while (count > 0) {
u_char data;
val = *p++;
data = val >> 8;
if (put_user(data, (u_char *)userPtr++))
return -EFAULT;
if (stereo) {
val = *p;
data = val >> 8;
if (put_user(data, (u_char *)userPtr++))
return -EFAULT;
}
p++;
count--;
}
*frameUsed += used * 4;
return stereo? used * 2: used;
}
static ssize_t cs4218_ct_u8_read(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t count, used;
short *p = (short *) &frame[*frameUsed];
int val, stereo = sound.soft.stereo;
frameLeft >>= 2;
if (stereo)
userCount >>= 1;
used = count = min(userCount, frameLeft);
while (count > 0) {
u_char data;
val = *p++;
data = (val >> 8) ^ 0x80;
if (put_user(data, (u_char *)userPtr++))
return -EFAULT;
if (stereo) {
val = *p;
data = (val >> 8) ^ 0x80;
if (put_user(data, (u_char *)userPtr++))
return -EFAULT;
}
p++;
count--;
}
*frameUsed += used * 4;
return stereo? used * 2: used;
}
static ssize_t cs4218_ct_s16_read(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t count, used;
int stereo = sound.soft.stereo;
short *fp = (short *) &frame[*frameUsed];
frameLeft >>= 2;
userCount >>= (stereo? 2: 1);
used = count = min(userCount, frameLeft);
if (!stereo) {
short *up = (short *) userPtr;
while (count > 0) {
short data;
data = *fp;
if (put_user(data, up++))
return -EFAULT;
fp+=2;
count--;
}
} else {
if (copy_to_user((u_char *)userPtr, fp, count * 4))
return -EFAULT;
}
*frameUsed += used * 4;
return stereo? used * 4: used * 2;
}
static ssize_t cs4218_ct_u16_read(const u_char *userPtr, size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t count, used;
int mask = (sound.soft.format == AFMT_U16_LE? 0x0080: 0x8000);
int stereo = sound.soft.stereo;
short *fp = (short *) &frame[*frameUsed];
short *up = (short *) userPtr;
frameLeft >>= 2;
userCount >>= (stereo? 2: 1);
used = count = min(userCount, frameLeft);
while (count > 0) {
int data;
data = *fp++;
data ^= mask;
if (put_user(data, up++))
return -EFAULT;
if (stereo) {
data = *fp;
data ^= mask;
if (put_user(data, up++))
return -EFAULT;
}
fp++;
count--;
}
*frameUsed += used * 4;
return stereo? used * 4: used * 2;
}
static TRANS transCSNormal = {
cs4218_ct_law, cs4218_ct_law, cs4218_ct_s8, cs4218_ct_u8,
cs4218_ct_s16, cs4218_ct_u16, cs4218_ct_s16, cs4218_ct_u16
};
static TRANS transCSExpand = {
cs4218_ctx_law, cs4218_ctx_law, cs4218_ctx_s8, cs4218_ctx_u8,
cs4218_ctx_s16, cs4218_ctx_u16, cs4218_ctx_s16, cs4218_ctx_u16
};
static TRANS transCSNormalRead = {
NULL, NULL, cs4218_ct_s8_read, cs4218_ct_u8_read,
cs4218_ct_s16_read, cs4218_ct_u16_read,
cs4218_ct_s16_read, cs4218_ct_u16_read
};
/*** Low level stuff *********************************************************/
static void *CS_Alloc(unsigned int size, int flags)
{
int order;
size >>= 13;
for (order=0; order < 5; order++) {
if (size == 0)
break;
size >>= 1;
}
return (void *)__get_free_pages(flags, order);
}
static void CS_Free(void *ptr, unsigned int size)
{
int order;
size >>= 13;
for (order=0; order < 5; order++) {
if (size == 0)
break;
size >>= 1;
}
free_pages((ulong)ptr, order);
}
static int __init CS_IrqInit(void)
{
cpm_install_handler(CPMVEC_SMC2, cs4218_intr, NULL);
return 1;
}
#ifdef MODULE
static void CS_IrqCleanup(void)
{
volatile smc_t *sp;
volatile cpm8xx_t *cp;
/* First disable transmitter and receiver.
*/
sp = &cpmp->cp_smc[1];
sp->smc_smcmr &= ~(SMCMR_REN | SMCMR_TEN);
/* And now shut down the SMC.
*/
cp = cpmp; /* Get pointer to Communication Processor */
cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2,
CPM_CR_STOP_TX) | CPM_CR_FLG;
while (cp->cp_cpcr & CPM_CR_FLG);
/* Release the interrupt handler.
*/
cpm_free_handler(CPMVEC_SMC2);
if (beep_buf)
kfree(beep_buf);
kd_mksound = orig_mksound;
}
#endif /* MODULE */
static void CS_Silence(void)
{
volatile smc_t *sp;
/* Disable transmitter.
*/
sp = &cpmp->cp_smc[1];
sp->smc_smcmr &= ~SMCMR_TEN;
}
/* Frequencies depend upon external oscillator. There are two
* choices, 12.288 and 11.2896 MHz. The RPCG audio supports both through
* and external control register selection bit.
*/
static int cs4218_freqs[] = {
/* 12.288 11.2896 */
48000, 44100,
32000, 29400,
24000, 22050,
19200, 17640,
16000, 14700,
12000, 11025,
9600, 8820,
8000, 7350
};
static void CS_Init(void)
{
int i, tolerance;
switch (sound.soft.format) {
case AFMT_S16_LE:
case AFMT_U16_LE:
sound.hard.format = AFMT_S16_LE;
break;
default:
sound.hard.format = AFMT_S16_BE;
break;
}
sound.hard.stereo = 1;
sound.hard.size = 16;
/*
* If we have a sample rate which is within catchRadius percent
* of the requested value, we don't have to expand the samples.
* Otherwise choose the next higher rate.
*/
i = (sizeof(cs4218_freqs) / sizeof(int));
do {
tolerance = catchRadius * cs4218_freqs[--i] / 100;
} while (sound.soft.speed > cs4218_freqs[i] + tolerance && i > 0);
if (sound.soft.speed >= cs4218_freqs[i] - tolerance)
sound.trans_write = &transCSNormal;
else
sound.trans_write = &transCSExpand;
sound.trans_read = &transCSNormalRead;
sound.hard.speed = cs4218_freqs[i];
cs4218_rate_index = i;
/* The CS4218 has seven selectable clock dividers for the sample
* clock. The HIOX then provides one of two external rates.
* An even numbered frequency table index uses the high external
* clock rate.
*/
*(uint *)HIOX_CSR4_ADDR &= ~(HIOX_CSR4_AUDCLKHI | HIOX_CSR4_AUDCLKSEL);
if ((i & 1) == 0)
*(uint *)HIOX_CSR4_ADDR |= HIOX_CSR4_AUDCLKHI;
i >>= 1;
*(uint *)HIOX_CSR4_ADDR |= (i & HIOX_CSR4_AUDCLKSEL);
expand_bal = -sound.soft.speed;
}
static int CS_SetFormat(int format)
{
int size;
switch (format) {
case AFMT_QUERY:
return sound.soft.format;
case AFMT_MU_LAW:
case AFMT_A_LAW:
case AFMT_U8:
case AFMT_S8:
size = 8;
break;
case AFMT_S16_BE:
case AFMT_U16_BE:
case AFMT_S16_LE:
case AFMT_U16_LE:
size = 16;
break;
default: /* :-) */
printk(KERN_ERR "dmasound: unknown format 0x%x, using AFMT_U8\n",
format);
size = 8;
format = AFMT_U8;
}
sound.soft.format = format;
sound.soft.size = size;
if (sound.minDev == SND_DEV_DSP) {
sound.dsp.format = format;
sound.dsp.size = size;
}
CS_Init();
return format;
}
/* Volume is the amount of attenuation we tell the codec to impose
* on the outputs. There are 32 levels, with 0 the "loudest".
*/
#define CS_VOLUME_TO_MASK(x) (31 - ((((x) - 1) * 31) / 99))
#define CS_MASK_TO_VOLUME(y) (100 - ((y) * 99 / 31))
static int cs_get_volume(uint reg)
{
int volume;
volume = CS_MASK_TO_VOLUME(CS_LATTEN_GET(reg));
volume |= CS_MASK_TO_VOLUME(CS_RATTEN_GET(reg)) << 8;
return volume;
}
static int cs_volume_setter(int volume, int mute)
{
uint tempctl;
if (mute && volume == 0) {
tempctl = cs4218_control | CS_MUTE;
} else {
tempctl = cs4218_control & ~CS_MUTE;
tempctl = tempctl & ~(CS_LATTEN | CS_RATTEN);
tempctl |= CS_LATTEN_SET(CS_VOLUME_TO_MASK(volume & 0xff));
tempctl |= CS_RATTEN_SET(CS_VOLUME_TO_MASK((volume >> 8) & 0xff));
volume = cs_get_volume(tempctl);
}
if (tempctl != cs4218_control) {
cs4218_ctl_write(tempctl);
}
return volume;
}
/* Gain has 16 steps from 0 to 15. These are in 1.5dB increments from
* 0 (no gain) to 22.5 dB.
*/
#define CS_RECLEVEL_TO_GAIN(v) \
((v) < 0 ? 0 : (v) > 100 ? 15 : (v) * 3 / 20)
#define CS_GAIN_TO_RECLEVEL(v) (((v) * 20 + 2) / 3)
static int cs_get_gain(uint reg)
{
int gain;
gain = CS_GAIN_TO_RECLEVEL(CS_LGAIN_GET(reg));
gain |= CS_GAIN_TO_RECLEVEL(CS_RGAIN_GET(reg)) << 8;
return gain;
}
static int cs_set_gain(int gain)
{
uint tempctl;
tempctl = cs4218_control & ~(CS_LGAIN | CS_RGAIN);
tempctl |= CS_LGAIN_SET(CS_RECLEVEL_TO_GAIN(gain & 0xff));
tempctl |= CS_RGAIN_SET(CS_RECLEVEL_TO_GAIN((gain >> 8) & 0xff));
gain = cs_get_gain(tempctl);
if (tempctl != cs4218_control) {
cs4218_ctl_write(tempctl);
}
return gain;
}
static int CS_SetVolume(int volume)
{
return cs_volume_setter(volume, CS_MUTE);
}
static void CS_Play(void)
{
int i, count;
unsigned long flags;
volatile cbd_t *bdp;
volatile cpm8xx_t *cp;
/* Protect buffer */
spin_lock_irqsave(&cs4218_lock, flags);
#if 0
if (awacs_beep_state) {
/* sound takes precedence over beeps */
out_le32(&awacs_txdma->control, (RUN|PAUSE|FLUSH|WAKE) << 16);
out_le32(&awacs->control,
(in_le32(&awacs->control) & ~0x1f00)
| (awacs_rate_index << 8));
out_le32(&awacs->byteswap, sound.hard.format != AFMT_S16_BE);
out_le32(&awacs_txdma->cmdptr, virt_to_bus(&(awacs_tx_cmds[(sq.front+sq.active) % sq.max_count])));
beep_playing = 0;
awacs_beep_state = 0;
}
#endif
i = sq.front + sq.active;
if (i >= sq.max_count)
i -= sq.max_count;
while (sq.active < 2 && sq.active < sq.count) {
count = (sq.count == sq.active + 1)?sq.rear_size:sq.block_size;
if (count < sq.block_size && !sq.syncing)
/* last block not yet filled, and we're not syncing. */
break;
bdp = &tx_base[i];
bdp->cbd_datlen = count;
flush_dcache_range((ulong)sound_buffers[i],
(ulong)(sound_buffers[i] + count));
if (++i >= sq.max_count)
i = 0;
if (sq.active == 0) {
/* The SMC does not load its fifo until the first
* TDM frame pulse, so the transmit data gets shifted
* by one word. To compensate for this, we incorrectly
* transmit the first buffer and shorten it by one
* word. Subsequent buffers are then aligned properly.
*/
bdp->cbd_datlen -= 2;
/* Start up the SMC Transmitter.
*/
cp = cpmp;
cp->cp_smc[1].smc_smcmr |= SMCMR_TEN;
cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2,
CPM_CR_RESTART_TX) | CPM_CR_FLG;
while (cp->cp_cpcr & CPM_CR_FLG);
}
/* Buffer is ready now.
*/
bdp->cbd_sc |= BD_SC_READY;
++sq.active;
}
spin_unlock_irqrestore(&cs4218_lock, flags);
}
static void CS_Record(void)
{
unsigned long flags;
volatile smc_t *sp;
if (read_sq.active)
return;
/* Protect buffer */
spin_lock_irqsave(&cs4218_lock, flags);
/* This is all we have to do......Just start it up.
*/
sp = &cpmp->cp_smc[1];
sp->smc_smcmr |= SMCMR_REN;
read_sq.active = 1;
spin_unlock_irqrestore(&cs4218_lock, flags);
}
static void
cs4218_tdm_tx_intr(void *devid)
{
int i = sq.front;
volatile cbd_t *bdp;
while (sq.active > 0) {
bdp = &tx_base[i];
if (bdp->cbd_sc & BD_SC_READY)
break; /* this frame is still going */
--sq.count;
--sq.active;
if (++i >= sq.max_count)
i = 0;
}
if (i != sq.front)
WAKE_UP(sq.action_queue);
sq.front = i;
CS_Play();
if (!sq.active)
WAKE_UP(sq.sync_queue);
}
static void
cs4218_tdm_rx_intr(void *devid)
{
/* We want to blow 'em off when shutting down.
*/
if (read_sq.active == 0)
return;
/* Check multiple buffers in case we were held off from
* interrupt processing for a long time. Geeze, I really hope
* this doesn't happen.
*/
while ((rx_base[read_sq.rear].cbd_sc & BD_SC_EMPTY) == 0) {
/* Invalidate the data cache range for this buffer.
*/
invalidate_dcache_range(
(uint)(sound_read_buffers[read_sq.rear]),
(uint)(sound_read_buffers[read_sq.rear] + read_sq.block_size));
/* Make buffer available again and move on.
*/
rx_base[read_sq.rear].cbd_sc |= BD_SC_EMPTY;
read_sq.rear++;
/* Wrap the buffer ring.
*/
if (read_sq.rear >= read_sq.max_active)
read_sq.rear = 0;
/* If we have caught up to the front buffer, bump it.
* This will cause weird (but not fatal) results if the
* read loop is currently using this buffer. The user is
* behind in this case anyway, so weird things are going
* to happen.
*/
if (read_sq.rear == read_sq.front) {
read_sq.front++;
if (read_sq.front >= read_sq.max_active)
read_sq.front = 0;
}
}
WAKE_UP(read_sq.action_queue);
}
static void cs_nosound(unsigned long xx)
{
unsigned long flags;
/* not sure if this is needed, since hardware command is #if 0'd */
spin_lock_irqsave(&cs4218_lock, flags);
if (beep_playing) {
#if 0
st_le16(&beep_dbdma_cmd->command, DBDMA_STOP);
#endif
beep_playing = 0;
}
spin_unlock_irqrestore(&cs4218_lock, flags);
}
static struct timer_list beep_timer = TIMER_INITIALIZER(cs_nosound, 0, 0);
};
static void cs_mksound(unsigned int hz, unsigned int ticks)
{
unsigned long flags;
int beep_speed = BEEP_SPEED;
int srate = cs4218_freqs[beep_speed];
int period, ncycles, nsamples;
int i, j, f;
short *p;
static int beep_hz_cache;
static int beep_nsamples_cache;
static int beep_volume_cache;
if (hz <= srate / BEEP_BUFLEN || hz > srate / 2) {
#if 1
/* this is a hack for broken X server code */
hz = 750;
ticks = 12;
#else
/* cancel beep currently playing */
awacs_nosound(0);
return;
#endif
}
/* lock while modifying beep_timer */
spin_lock_irqsave(&cs4218_lock, flags);
del_timer(&beep_timer);
if (ticks) {
beep_timer.expires = jiffies + ticks;
add_timer(&beep_timer);
}
if (beep_playing || sq.active || beep_buf == NULL) {
spin_unlock_irqrestore(&cs4218_lock, flags);
return; /* too hard, sorry :-( */
}
beep_playing = 1;
#if 0
st_le16(&beep_dbdma_cmd->command, OUTPUT_MORE + BR_ALWAYS);
#endif
spin_unlock_irqrestore(&cs4218_lock, flags);
if (hz == beep_hz_cache && beep_volume == beep_volume_cache) {
nsamples = beep_nsamples_cache;
} else {
period = srate * 256 / hz; /* fixed point */
ncycles = BEEP_BUFLEN * 256 / period;
nsamples = (period * ncycles) >> 8;
f = ncycles * 65536 / nsamples;
j = 0;
p = beep_buf;
for (i = 0; i < nsamples; ++i, p += 2) {
p[0] = p[1] = beep_wform[j >> 8] * beep_volume;
j = (j + f) & 0xffff;
}
beep_hz_cache = hz;
beep_volume_cache = beep_volume;
beep_nsamples_cache = nsamples;
}
#if 0
st_le16(&beep_dbdma_cmd->req_count, nsamples*4);
st_le16(&beep_dbdma_cmd->xfer_status, 0);
st_le32(&beep_dbdma_cmd->cmd_dep, virt_to_bus(beep_dbdma_cmd));
st_le32(&beep_dbdma_cmd->phy_addr, virt_to_bus(beep_buf));
awacs_beep_state = 1;
spin_lock_irqsave(&cs4218_lock, flags);
if (beep_playing) { /* i.e. haven't been terminated already */
out_le32(&awacs_txdma->control, (RUN|WAKE|FLUSH|PAUSE) << 16);
out_le32(&awacs->control,
(in_le32(&awacs->control) & ~0x1f00)
| (beep_speed << 8));
out_le32(&awacs->byteswap, 0);
out_le32(&awacs_txdma->cmdptr, virt_to_bus(beep_dbdma_cmd));
out_le32(&awacs_txdma->control, RUN | (RUN << 16));
}
spin_unlock_irqrestore(&cs4218_lock, flags);
#endif
}
static MACHINE mach_cs4218 = {
.owner = THIS_MODULE,
.name = "HIOX CS4218",
.name2 = "Built-in Sound",
.dma_alloc = CS_Alloc,
.dma_free = CS_Free,
.irqinit = CS_IrqInit,
#ifdef MODULE
.irqcleanup = CS_IrqCleanup,
#endif /* MODULE */
.init = CS_Init,
.silence = CS_Silence,
.setFormat = CS_SetFormat,
.setVolume = CS_SetVolume,
.play = CS_Play
};
/*** Mid level stuff *********************************************************/
static void sound_silence(void)
{
/* update hardware settings one more */
(*sound.mach.init)();
(*sound.mach.silence)();
}
static void sound_init(void)
{
(*sound.mach.init)();
}
static int sound_set_format(int format)
{
return(*sound.mach.setFormat)(format);
}
static int sound_set_speed(int speed)
{
if (speed < 0)
return(sound.soft.speed);
sound.soft.speed = speed;
(*sound.mach.init)();
if (sound.minDev == SND_DEV_DSP)
sound.dsp.speed = sound.soft.speed;
return(sound.soft.speed);
}
static int sound_set_stereo(int stereo)
{
if (stereo < 0)
return(sound.soft.stereo);
stereo = !!stereo; /* should be 0 or 1 now */
sound.soft.stereo = stereo;
if (sound.minDev == SND_DEV_DSP)
sound.dsp.stereo = stereo;
(*sound.mach.init)();
return(stereo);
}
static int sound_set_volume(int volume)
{
return(*sound.mach.setVolume)(volume);
}
static ssize_t sound_copy_translate(const u_char *userPtr,
size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t (*ct_func)(const u_char *, size_t, u_char *, ssize_t *, ssize_t) = NULL;
switch (sound.soft.format) {
case AFMT_MU_LAW:
ct_func = sound.trans_write->ct_ulaw;
break;
case AFMT_A_LAW:
ct_func = sound.trans_write->ct_alaw;
break;
case AFMT_S8:
ct_func = sound.trans_write->ct_s8;
break;
case AFMT_U8:
ct_func = sound.trans_write->ct_u8;
break;
case AFMT_S16_BE:
ct_func = sound.trans_write->ct_s16be;
break;
case AFMT_U16_BE:
ct_func = sound.trans_write->ct_u16be;
break;
case AFMT_S16_LE:
ct_func = sound.trans_write->ct_s16le;
break;
case AFMT_U16_LE:
ct_func = sound.trans_write->ct_u16le;
break;
}
if (ct_func)
return ct_func(userPtr, userCount, frame, frameUsed, frameLeft);
else
return 0;
}
static ssize_t sound_copy_translate_read(const u_char *userPtr,
size_t userCount,
u_char frame[], ssize_t *frameUsed,
ssize_t frameLeft)
{
ssize_t (*ct_func)(const u_char *, size_t, u_char *, ssize_t *, ssize_t) = NULL;
switch (sound.soft.format) {
case AFMT_MU_LAW:
ct_func = sound.trans_read->ct_ulaw;
break;
case AFMT_A_LAW:
ct_func = sound.trans_read->ct_alaw;
break;
case AFMT_S8:
ct_func = sound.trans_read->ct_s8;
break;
case AFMT_U8:
ct_func = sound.trans_read->ct_u8;
break;
case AFMT_S16_BE:
ct_func = sound.trans_read->ct_s16be;
break;
case AFMT_U16_BE:
ct_func = sound.trans_read->ct_u16be;
break;
case AFMT_S16_LE:
ct_func = sound.trans_read->ct_s16le;
break;
case AFMT_U16_LE:
ct_func = sound.trans_read->ct_u16le;
break;
}
if (ct_func)
return ct_func(userPtr, userCount, frame, frameUsed, frameLeft);
else
return 0;
}
/*
* /dev/mixer abstraction
*/
static int mixer_open(struct inode *inode, struct file *file)
{
mixer.busy = 1;
return nonseekable_open(inode, file);
}
static int mixer_release(struct inode *inode, struct file *file)
{
mixer.busy = 0;
return 0;
}
static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd,
u_long arg)
{
int data;
uint tmpcs;
if (_SIOC_DIR(cmd) & _SIOC_WRITE)
mixer.modify_counter++;
if (cmd == OSS_GETVERSION)
return IOCTL_OUT(arg, SOUND_VERSION);
switch (cmd) {
case SOUND_MIXER_INFO: {
mixer_info info;
strlcpy(info.id, "CS4218_TDM", sizeof(info.id));
strlcpy(info.name, "CS4218_TDM", sizeof(info.name));
info.name[sizeof(info.name)-1] = 0;
info.modify_counter = mixer.modify_counter;
if (copy_to_user((int *)arg, &info, sizeof(info)))
return -EFAULT;
return 0;
}
case SOUND_MIXER_READ_DEVMASK:
data = SOUND_MASK_VOLUME | SOUND_MASK_LINE
| SOUND_MASK_MIC | SOUND_MASK_RECLEV
| SOUND_MASK_ALTPCM;
return IOCTL_OUT(arg, data);
case SOUND_MIXER_READ_RECMASK:
data = SOUND_MASK_LINE | SOUND_MASK_MIC;
return IOCTL_OUT(arg, data);
case SOUND_MIXER_READ_RECSRC:
if (cs4218_control & CS_DO1)
data = SOUND_MASK_LINE;
else
data = SOUND_MASK_MIC;
return IOCTL_OUT(arg, data);
case SOUND_MIXER_WRITE_RECSRC:
IOCTL_IN(arg, data);
data &= (SOUND_MASK_LINE | SOUND_MASK_MIC);
if (data & SOUND_MASK_LINE)
tmpcs = cs4218_control |
(CS_ISL | CS_ISR | CS_DO1);
if (data & SOUND_MASK_MIC)
tmpcs = cs4218_control &
~(CS_ISL | CS_ISR | CS_DO1);
if (tmpcs != cs4218_control)
cs4218_ctl_write(tmpcs);
return IOCTL_OUT(arg, data);
case SOUND_MIXER_READ_STEREODEVS:
data = SOUND_MASK_VOLUME | SOUND_MASK_RECLEV;
return IOCTL_OUT(arg, data);
case SOUND_MIXER_READ_CAPS:
return IOCTL_OUT(arg, 0);
case SOUND_MIXER_READ_VOLUME:
data = (cs4218_control & CS_MUTE)? 0:
cs_get_volume(cs4218_control);
return IOCTL_OUT(arg, data);
case SOUND_MIXER_WRITE_VOLUME:
IOCTL_IN(arg, data);
return IOCTL_OUT(arg, sound_set_volume(data));
case SOUND_MIXER_WRITE_ALTPCM: /* really bell volume */
IOCTL_IN(arg, data);
beep_volume = data & 0xff;
/* fall through */
case SOUND_MIXER_READ_ALTPCM:
return IOCTL_OUT(arg, beep_volume);
case SOUND_MIXER_WRITE_RECLEV:
IOCTL_IN(arg, data);
data = cs_set_gain(data);
return IOCTL_OUT(arg, data);
case SOUND_MIXER_READ_RECLEV:
data = cs_get_gain(cs4218_control);
return IOCTL_OUT(arg, data);
}
return -EINVAL;
}
static struct file_operations mixer_fops =
{
.owner = THIS_MODULE,
.llseek = sound_lseek,
.ioctl = mixer_ioctl,
.open = mixer_open,
.release = mixer_release,
};
static void __init mixer_init(void)
{
mixer_unit = register_sound_mixer(&mixer_fops, -1);
if (mixer_unit < 0)
return;
mixer.busy = 0;
sound.treble = 0;
sound.bass = 0;
/* Set Line input, no gain, no attenuation.
*/
cs4218_control = CS_ISL | CS_ISR | CS_DO1;
cs4218_control |= CS_LGAIN_SET(0) | CS_RGAIN_SET(0);
cs4218_control |= CS_LATTEN_SET(0) | CS_RATTEN_SET(0);
cs4218_ctl_write(cs4218_control);
}
/*
* Sound queue stuff, the heart of the driver
*/
static int sq_allocate_buffers(void)
{
int i;
if (sound_buffers)
return 0;
sound_buffers = kmalloc (numBufs * sizeof(char *), GFP_KERNEL);
if (!sound_buffers)
return -ENOMEM;
for (i = 0; i < numBufs; i++) {
sound_buffers[i] = sound.mach.dma_alloc (bufSize << 10, GFP_KERNEL);
if (!sound_buffers[i]) {
while (i--)
sound.mach.dma_free (sound_buffers[i], bufSize << 10);
kfree (sound_buffers);
sound_buffers = 0;
return -ENOMEM;
}
}
return 0;
}
static void sq_release_buffers(void)
{
int i;
if (sound_buffers) {
for (i = 0; i < numBufs; i++)
sound.mach.dma_free (sound_buffers[i], bufSize << 10);
kfree (sound_buffers);
sound_buffers = 0;
}
}
static int sq_allocate_read_buffers(void)
{
int i;
if (sound_read_buffers)
return 0;
sound_read_buffers = kmalloc(numReadBufs * sizeof(char *), GFP_KERNEL);
if (!sound_read_buffers)
return -ENOMEM;
for (i = 0; i < numBufs; i++) {
sound_read_buffers[i] = sound.mach.dma_alloc (readbufSize<<10,
GFP_KERNEL);
if (!sound_read_buffers[i]) {
while (i--)
sound.mach.dma_free (sound_read_buffers[i],
readbufSize << 10);
kfree (sound_read_buffers);
sound_read_buffers = 0;
return -ENOMEM;
}
}
return 0;
}
static void sq_release_read_buffers(void)
{
int i;
if (sound_read_buffers) {
cpmp->cp_smc[1].smc_smcmr &= ~SMCMR_REN;
for (i = 0; i < numReadBufs; i++)
sound.mach.dma_free (sound_read_buffers[i],
bufSize << 10);
kfree (sound_read_buffers);
sound_read_buffers = 0;
}
}
static void sq_setup(int numBufs, int bufSize, char **write_buffers)
{
int i;
volatile cbd_t *bdp;
volatile cpm8xx_t *cp;
volatile smc_t *sp;
/* Make sure the SMC transmit is shut down.
*/
cp = cpmp;
sp = &cpmp->cp_smc[1];
sp->smc_smcmr &= ~SMCMR_TEN;
sq.max_count = numBufs;
sq.max_active = numBufs;
sq.block_size = bufSize;
sq.buffers = write_buffers;
sq.front = sq.count = 0;
sq.rear = -1;
sq.syncing = 0;
sq.active = 0;
bdp = tx_base;
for (i=0; i<numBufs; i++) {
bdp->cbd_bufaddr = virt_to_bus(write_buffers[i]);
bdp++;
}
/* This causes the SMC to sync up with the first buffer again.
*/
cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2, CPM_CR_INIT_TX) | CPM_CR_FLG;
while (cp->cp_cpcr & CPM_CR_FLG);
}
static void read_sq_setup(int numBufs, int bufSize, char **read_buffers)
{
int i;
volatile cbd_t *bdp;
volatile cpm8xx_t *cp;
volatile smc_t *sp;
/* Make sure the SMC receive is shut down.
*/
cp = cpmp;
sp = &cpmp->cp_smc[1];
sp->smc_smcmr &= ~SMCMR_REN;
read_sq.max_count = numBufs;
read_sq.max_active = numBufs;
read_sq.block_size = bufSize;
read_sq.buffers = read_buffers;
read_sq.front = read_sq.count = 0;
read_sq.rear = 0;
read_sq.rear_size = 0;
read_sq.syncing = 0;
read_sq.active = 0;
bdp = rx_base;
for (i=0; i<numReadBufs; i++) {
bdp->cbd_bufaddr = virt_to_bus(read_buffers[i]);
bdp->cbd_datlen = read_sq.block_size;
bdp++;
}
/* This causes the SMC to sync up with the first buffer again.
*/
cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2, CPM_CR_INIT_RX) | CPM_CR_FLG;
while (cp->cp_cpcr & CPM_CR_FLG);
}
static void sq_play(void)
{
(*sound.mach.play)();
}
/* ++TeSche: radically changed this one too */
static ssize_t sq_write(struct file *file, const char *src, size_t uLeft,
loff_t *ppos)
{
ssize_t uWritten = 0;
u_char *dest;
ssize_t uUsed, bUsed, bLeft;
/* ++TeSche: Is something like this necessary?
* Hey, that's an honest question! Or does any other part of the
* filesystem already checks this situation? I really don't know.
*/
if (uLeft == 0)
return 0;
/* The interrupt doesn't start to play the last, incomplete frame.
* Thus we can append to it without disabling the interrupts! (Note
* also that sq.rear isn't affected by the interrupt.)
*/
if (sq.count > 0 && (bLeft = sq.block_size-sq.rear_size) > 0) {
dest = sq_block_address(sq.rear);
bUsed = sq.rear_size;
uUsed = sound_copy_translate(src, uLeft, dest, &bUsed, bLeft);
if (uUsed <= 0)
return uUsed;
src += uUsed;
uWritten += uUsed;
uLeft -= uUsed;
sq.rear_size = bUsed;
}
do {
while (sq.count == sq.max_active) {
sq_play();
if (NON_BLOCKING(sq.open_mode))
return uWritten > 0 ? uWritten : -EAGAIN;
SLEEP(sq.action_queue);
if (SIGNAL_RECEIVED)
return uWritten > 0 ? uWritten : -EINTR;
}
/* Here, we can avoid disabling the interrupt by first
* copying and translating the data, and then updating
* the sq variables. Until this is done, the interrupt
* won't see the new frame and we can work on it
* undisturbed.
*/
dest = sq_block_address((sq.rear+1) % sq.max_count);
bUsed = 0;
bLeft = sq.block_size;
uUsed = sound_copy_translate(src, uLeft, dest, &bUsed, bLeft);
if (uUsed <= 0)
break;
src += uUsed;
uWritten += uUsed;
uLeft -= uUsed;
if (bUsed) {
sq.rear = (sq.rear+1) % sq.max_count;
sq.rear_size = bUsed;
sq.count++;
}
} while (bUsed); /* uUsed may have been 0 */
sq_play();
return uUsed < 0? uUsed: uWritten;
}
/***********/
/* Here is how the values are used for reading.
* The value 'active' simply indicates the DMA is running. This is
* done so the driver semantics are DMA starts when the first read is
* posted. The value 'front' indicates the buffer we should next
* send to the user. The value 'rear' indicates the buffer the DMA is
* currently filling. When 'front' == 'rear' the buffer "ring" is
* empty (we always have an empty available). The 'rear_size' is used
* to track partial offsets into the current buffer. Right now, I just keep
* The DMA running. If the reader can't keep up, the interrupt tosses
* the oldest buffer. We could also shut down the DMA in this case.
*/
static ssize_t sq_read(struct file *file, char *dst, size_t uLeft,
loff_t *ppos)
{
ssize_t uRead, bLeft, bUsed, uUsed;
if (uLeft == 0)
return 0;
if (!read_sq.active)
CS_Record(); /* Kick off the record process. */
uRead = 0;
/* Move what the user requests, depending upon other options.
*/
while (uLeft > 0) {
/* When front == rear, the DMA is not done yet.
*/
while (read_sq.front == read_sq.rear) {
if (NON_BLOCKING(read_sq.open_mode)) {
return uRead > 0 ? uRead : -EAGAIN;
}
SLEEP(read_sq.action_queue);
if (SIGNAL_RECEIVED)
return uRead > 0 ? uRead : -EINTR;
}
/* The amount we move is either what is left in the
* current buffer or what the user wants.
*/
bLeft = read_sq.block_size - read_sq.rear_size;
bUsed = read_sq.rear_size;
uUsed = sound_copy_translate_read(dst, uLeft,
read_sq.buffers[read_sq.front], &bUsed, bLeft);
if (uUsed <= 0)
return uUsed;
dst += uUsed;
uRead += uUsed;
uLeft -= uUsed;
read_sq.rear_size += bUsed;
if (read_sq.rear_size >= read_sq.block_size) {
read_sq.rear_size = 0;
read_sq.front++;
if (read_sq.front >= read_sq.max_active)
read_sq.front = 0;
}
}
return uRead;
}
static int sq_open(struct inode *inode, struct file *file)
{
int rc = 0;
if (file->f_mode & FMODE_WRITE) {
if (sq.busy) {
rc = -EBUSY;
if (NON_BLOCKING(file->f_flags))
goto err_out;
rc = -EINTR;
while (sq.busy) {
SLEEP(sq.open_queue);
if (SIGNAL_RECEIVED)
goto err_out;
}
}
sq.busy = 1; /* Let's play spot-the-race-condition */
if (sq_allocate_buffers()) goto err_out_nobusy;
sq_setup(numBufs, bufSize<<10,sound_buffers);
sq.open_mode = file->f_mode;
}
if (file->f_mode & FMODE_READ) {
if (read_sq.busy) {
rc = -EBUSY;
if (NON_BLOCKING(file->f_flags))
goto err_out;
rc = -EINTR;
while (read_sq.busy) {
SLEEP(read_sq.open_queue);
if (SIGNAL_RECEIVED)
goto err_out;
}
rc = 0;
}
read_sq.busy = 1;
if (sq_allocate_read_buffers()) goto err_out_nobusy;
read_sq_setup(numReadBufs,readbufSize<<10, sound_read_buffers);
read_sq.open_mode = file->f_mode;
}
/* Start up the 4218 by:
* Reset.
* Enable, unreset.
*/
*((volatile uint *)HIOX_CSR4_ADDR) &= ~HIOX_CSR4_RSTAUDIO;
eieio();
*((volatile uint *)HIOX_CSR4_ADDR) |= HIOX_CSR4_ENAUDIO;
mdelay(50);
*((volatile uint *)HIOX_CSR4_ADDR) |= HIOX_CSR4_RSTAUDIO;
/* We need to send the current control word in case someone
* opened /dev/mixer and changed things while we were shut
* down. Chances are good the initialization that follows
* would have done this, but it is still possible it wouldn't.
*/
cs4218_ctl_write(cs4218_control);
sound.minDev = iminor(inode) & 0x0f;
sound.soft = sound.dsp;
sound.hard = sound.dsp;
sound_init();
if ((iminor(inode) & 0x0f) == SND_DEV_AUDIO) {
sound_set_speed(8000);
sound_set_stereo(0);
sound_set_format(AFMT_MU_LAW);
}
return nonseekable_open(inode, file);
err_out_nobusy:
if (file->f_mode & FMODE_WRITE) {
sq.busy = 0;
WAKE_UP(sq.open_queue);
}
if (file->f_mode & FMODE_READ) {
read_sq.busy = 0;
WAKE_UP(read_sq.open_queue);
}
err_out:
return rc;
}
static void sq_reset(void)
{
sound_silence();
sq.active = 0;
sq.count = 0;
sq.front = (sq.rear+1) % sq.max_count;
#if 0
init_tdm_buffers();
#endif
}
static int sq_fsync(struct file *filp, struct dentry *dentry)
{
int rc = 0;
sq.syncing = 1;
sq_play(); /* there may be an incomplete frame waiting */
while (sq.active) {
SLEEP(sq.sync_queue);
if (SIGNAL_RECEIVED) {
/* While waiting for audio output to drain, an
* interrupt occurred. Stop audio output immediately
* and clear the queue. */
sq_reset();
rc = -EINTR;
break;
}
}
sq.syncing = 0;
return rc;
}
static int sq_release(struct inode *inode, struct file *file)
{
int rc = 0;
if (sq.busy)
rc = sq_fsync(file, file->f_dentry);
sound.soft = sound.dsp;
sound.hard = sound.dsp;
sound_silence();
sq_release_read_buffers();
sq_release_buffers();
if (file->f_mode & FMODE_READ) {
read_sq.busy = 0;
WAKE_UP(read_sq.open_queue);
}
if (file->f_mode & FMODE_WRITE) {
sq.busy = 0;
WAKE_UP(sq.open_queue);
}
/* Shut down the SMC.
*/
cpmp->cp_smc[1].smc_smcmr &= ~(SMCMR_TEN | SMCMR_REN);
/* Shut down the codec.
*/
*((volatile uint *)HIOX_CSR4_ADDR) |= HIOX_CSR4_RSTAUDIO;
eieio();
*((volatile uint *)HIOX_CSR4_ADDR) &= ~HIOX_CSR4_ENAUDIO;
/* Wake up a process waiting for the queue being released.
* Note: There may be several processes waiting for a call
* to open() returning. */
return rc;
}
static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd,
u_long arg)
{
u_long fmt;
int data;
#if 0
int size, nbufs;
#else
int size;
#endif
switch (cmd) {
case SNDCTL_DSP_RESET:
sq_reset();
return 0;
case SNDCTL_DSP_POST:
case SNDCTL_DSP_SYNC:
return sq_fsync(file, file->f_dentry);
/* ++TeSche: before changing any of these it's
* probably wise to wait until sound playing has
* settled down. */
case SNDCTL_DSP_SPEED:
sq_fsync(file, file->f_dentry);
IOCTL_IN(arg, data);
return IOCTL_OUT(arg, sound_set_speed(data));
case SNDCTL_DSP_STEREO:
sq_fsync(file, file->f_dentry);
IOCTL_IN(arg, data);
return IOCTL_OUT(arg, sound_set_stereo(data));
case SOUND_PCM_WRITE_CHANNELS:
sq_fsync(file, file->f_dentry);
IOCTL_IN(arg, data);
return IOCTL_OUT(arg, sound_set_stereo(data-1)+1);
case SNDCTL_DSP_SETFMT:
sq_fsync(file, file->f_dentry);
IOCTL_IN(arg, data);
return IOCTL_OUT(arg, sound_set_format(data));
case SNDCTL_DSP_GETFMTS:
fmt = 0;
if (sound.trans_write) {
if (sound.trans_write->ct_ulaw)
fmt |= AFMT_MU_LAW;
if (sound.trans_write->ct_alaw)
fmt |= AFMT_A_LAW;
if (sound.trans_write->ct_s8)
fmt |= AFMT_S8;
if (sound.trans_write->ct_u8)
fmt |= AFMT_U8;
if (sound.trans_write->ct_s16be)
fmt |= AFMT_S16_BE;
if (sound.trans_write->ct_u16be)
fmt |= AFMT_U16_BE;
if (sound.trans_write->ct_s16le)
fmt |= AFMT_S16_LE;
if (sound.trans_write->ct_u16le)
fmt |= AFMT_U16_LE;
}
return IOCTL_OUT(arg, fmt);
case SNDCTL_DSP_GETBLKSIZE:
size = sq.block_size
* sound.soft.size * (sound.soft.stereo + 1)
/ (sound.hard.size * (sound.hard.stereo + 1));
return IOCTL_OUT(arg, size);
case SNDCTL_DSP_SUBDIVIDE:
break;
#if 0 /* Sorry can't do this at the moment. The CPM allocated buffers
* long ago that can't be changed.
*/
case SNDCTL_DSP_SETFRAGMENT:
if (sq.count || sq.active || sq.syncing)
return -EINVAL;
IOCTL_IN(arg, size);
nbufs = size >> 16;
if (nbufs < 2 || nbufs > numBufs)
nbufs = numBufs;
size &= 0xffff;
if (size >= 8 && size <= 30) {
size = 1 << size;
size *= sound.hard.size * (sound.hard.stereo + 1);
size /= sound.soft.size * (sound.soft.stereo + 1);
if (size > (bufSize << 10))
size = bufSize << 10;
} else
size = bufSize << 10;
sq_setup(numBufs, size, sound_buffers);
sq.max_active = nbufs;
return 0;
#endif
default:
return mixer_ioctl(inode, file, cmd, arg);
}
return -EINVAL;
}
static struct file_operations sq_fops =
{
.owner = THIS_MODULE,
.llseek = sound_lseek,
.read = sq_read, /* sq_read */
.write = sq_write,
.ioctl = sq_ioctl,
.open = sq_open,
.release = sq_release,
};
static void __init sq_init(void)
{
sq_unit = register_sound_dsp(&sq_fops, -1);
if (sq_unit < 0)
return;
init_waitqueue_head(&sq.action_queue);
init_waitqueue_head(&sq.open_queue);
init_waitqueue_head(&sq.sync_queue);
init_waitqueue_head(&read_sq.action_queue);
init_waitqueue_head(&read_sq.open_queue);
init_waitqueue_head(&read_sq.sync_queue);
sq.busy = 0;
read_sq.busy = 0;
/* whatever you like as startup mode for /dev/dsp,
* (/dev/audio hasn't got a startup mode). note that
* once changed a new open() will *not* restore these!
*/
sound.dsp.format = AFMT_S16_BE;
sound.dsp.stereo = 1;
sound.dsp.size = 16;
/* set minimum rate possible without expanding */
sound.dsp.speed = 8000;
/* before the first open to /dev/dsp this wouldn't be set */
sound.soft = sound.dsp;
sound.hard = sound.dsp;
sound_silence();
}
/*
* /dev/sndstat
*/
/* state.buf should not overflow! */
static int state_open(struct inode *inode, struct file *file)
{
char *buffer = state.buf, *mach = "", cs4218_buf[50];
int len = 0;
if (state.busy)
return -EBUSY;
state.ptr = 0;
state.busy = 1;
sprintf(cs4218_buf, "Crystal CS4218 on TDM, ");
mach = cs4218_buf;
len += sprintf(buffer+len, "%sDMA sound driver:\n", mach);
len += sprintf(buffer+len, "\tsound.format = 0x%x", sound.soft.format);
switch (sound.soft.format) {
case AFMT_MU_LAW:
len += sprintf(buffer+len, " (mu-law)");
break;
case AFMT_A_LAW:
len += sprintf(buffer+len, " (A-law)");
break;
case AFMT_U8:
len += sprintf(buffer+len, " (unsigned 8 bit)");
break;
case AFMT_S8:
len += sprintf(buffer+len, " (signed 8 bit)");
break;
case AFMT_S16_BE:
len += sprintf(buffer+len, " (signed 16 bit big)");
break;
case AFMT_U16_BE:
len += sprintf(buffer+len, " (unsigned 16 bit big)");
break;
case AFMT_S16_LE:
len += sprintf(buffer+len, " (signed 16 bit little)");
break;
case AFMT_U16_LE:
len += sprintf(buffer+len, " (unsigned 16 bit little)");
break;
}
len += sprintf(buffer+len, "\n");
len += sprintf(buffer+len, "\tsound.speed = %dHz (phys. %dHz)\n",
sound.soft.speed, sound.hard.speed);
len += sprintf(buffer+len, "\tsound.stereo = 0x%x (%s)\n",
sound.soft.stereo, sound.soft.stereo ? "stereo" : "mono");
len += sprintf(buffer+len, "\tsq.block_size = %d sq.max_count = %d"
" sq.max_active = %d\n",
sq.block_size, sq.max_count, sq.max_active);
len += sprintf(buffer+len, "\tsq.count = %d sq.rear_size = %d\n", sq.count,
sq.rear_size);
len += sprintf(buffer+len, "\tsq.active = %d sq.syncing = %d\n",
sq.active, sq.syncing);
state.len = len;
return nonseekable_open(inode, file);
}
static int state_release(struct inode *inode, struct file *file)
{
state.busy = 0;
return 0;
}
static ssize_t state_read(struct file *file, char *buf, size_t count,
loff_t *ppos)
{
int n = state.len - state.ptr;
if (n > count)
n = count;
if (n <= 0)
return 0;
if (copy_to_user(buf, &state.buf[state.ptr], n))
return -EFAULT;
state.ptr += n;
return n;
}
static struct file_operations state_fops =
{
.owner = THIS_MODULE,
.llseek = sound_lseek,
.read = state_read,
.open = state_open,
.release = state_release,
};
static void __init state_init(void)
{
state_unit = register_sound_special(&state_fops, SND_DEV_STATUS);
if (state_unit < 0)
return;
state.busy = 0;
}
/*** Common stuff ********************************************************/
static long long sound_lseek(struct file *file, long long offset, int orig)
{
return -ESPIPE;
}
/*** Config & Setup **********************************************************/
int __init tdm8xx_sound_init(void)
{
int i, has_sound;
uint dp_offset;
volatile uint *sirp;
volatile cbd_t *bdp;
volatile cpm8xx_t *cp;
volatile smc_t *sp;
volatile smc_uart_t *up;
volatile immap_t *immap;
has_sound = 0;
/* Program the SI/TSA to use TDMa, connected to SMC2, for 4 bytes.
*/
cp = cpmp; /* Get pointer to Communication Processor */
immap = (immap_t *)IMAP_ADDR; /* and to internal registers */
/* Set all TDMa control bits to zero. This enables most features
* we want.
*/
cp->cp_simode &= ~0x00000fff;
/* Enable common receive/transmit clock pins, use IDL format.
* Sync on falling edge, transmit rising clock, receive falling
* clock, delay 1 bit on both Tx and Rx. Common Tx/Rx clocks and
* sync.
* Connect SMC2 to TSA.
*/
cp->cp_simode |= 0x80000141;
/* Configure port A pins for TDMa operation.
* The RPX-Lite (MPC850/823) loses SMC2 when TDM is used.
*/
immap->im_ioport.iop_papar |= 0x01c0; /* Enable TDMa functions */
immap->im_ioport.iop_padir |= 0x00c0; /* Enable TDMa Tx/Rx */
immap->im_ioport.iop_padir &= ~0x0100; /* Enable L1RCLKa */
immap->im_ioport.iop_pcpar |= 0x0800; /* Enable L1RSYNCa */
immap->im_ioport.iop_pcdir &= ~0x0800;
/* Initialize the SI TDM routing table. We use TDMa only.
* The receive table and transmit table each have only one
* entry, to capture/send four bytes after each frame pulse.
* The 16-bit ram entry is 0000 0001 1000 1111. (SMC2)
*/
cp->cp_sigmr = 0;
sirp = (uint *)cp->cp_siram;
*sirp = 0x018f0000; /* Receive entry */
sirp += 64;
*sirp = 0x018f0000; /* Tramsmit entry */
/* Enable single TDMa routing.
*/
cp->cp_sigmr = 0x04;
/* Initialize the SMC for transparent operation.
*/
sp = &cpmp->cp_smc[1];
up = (smc_uart_t *)&cp->cp_dparam[PROFF_SMC2];
/* We need to allocate a transmit and receive buffer
* descriptors from dual port ram.
*/
dp_addr = cpm_dpalloc(sizeof(cbd_t) * numReadBufs, 8);
/* Set the physical address of the host memory
* buffers in the buffer descriptors, and the
* virtual address for us to work with.
*/
bdp = (cbd_t *)&cp->cp_dpmem[dp_addr];
up->smc_rbase = dp_offset;
rx_cur = rx_base = (cbd_t *)bdp;
for (i=0; i<(numReadBufs-1); i++) {
bdp->cbd_bufaddr = 0;
bdp->cbd_datlen = 0;
bdp->cbd_sc = BD_SC_EMPTY | BD_SC_INTRPT;
bdp++;
}
bdp->cbd_bufaddr = 0;
bdp->cbd_datlen = 0;
bdp->cbd_sc = BD_SC_WRAP | BD_SC_EMPTY | BD_SC_INTRPT;
/* Now, do the same for the transmit buffers.
*/
dp_offset = cpm_dpalloc(sizeof(cbd_t) * numBufs, 8);
bdp = (cbd_t *)&cp->cp_dpmem[dp_addr];
up->smc_tbase = dp_offset;
tx_cur = tx_base = (cbd_t *)bdp;
for (i=0; i<(numBufs-1); i++) {
bdp->cbd_bufaddr = 0;
bdp->cbd_datlen = 0;
bdp->cbd_sc = BD_SC_INTRPT;
bdp++;
}
bdp->cbd_bufaddr = 0;
bdp->cbd_datlen = 0;
bdp->cbd_sc = (BD_SC_WRAP | BD_SC_INTRPT);
/* Set transparent SMC mode.
* A few things are specific to our application. The codec interface
* is MSB first, hence the REVD selection. The CD/CTS pulse are
* used by the TSA to indicate the frame start to the SMC.
*/
up->smc_rfcr = SCC_EB;
up->smc_tfcr = SCC_EB;
up->smc_mrblr = readbufSize * 1024;
/* Set 16-bit reversed data, transparent mode.
*/
sp->smc_smcmr = smcr_mk_clen(15) |
SMCMR_SM_TRANS | SMCMR_REVD | SMCMR_BS;
/* Enable and clear events.
* Because of FIFO delays, all we need is the receive interrupt
* and we can process both the current receive and current
* transmit interrupt within a few microseconds of the transmit.
*/
sp->smc_smce = 0xff;
sp->smc_smcm = SMCM_TXE | SMCM_TX | SMCM_RX;
/* Send the CPM an initialize command.
*/
cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2,
CPM_CR_INIT_TRX) | CPM_CR_FLG;
while (cp->cp_cpcr & CPM_CR_FLG);
sound.mach = mach_cs4218;
has_sound = 1;
/* Initialize beep stuff */
orig_mksound = kd_mksound;
kd_mksound = cs_mksound;
beep_buf = (short *) kmalloc(BEEP_BUFLEN * 4, GFP_KERNEL);
if (beep_buf == NULL)
printk(KERN_WARNING "dmasound: no memory for "
"beep buffer\n");
if (!has_sound)
return -ENODEV;
/* Initialize the software SPI.
*/
sw_spi_init();
/* Set up sound queue, /dev/audio and /dev/dsp. */
/* Set default settings. */
sq_init();
/* Set up /dev/sndstat. */
state_init();
/* Set up /dev/mixer. */
mixer_init();
if (!sound.mach.irqinit()) {
printk(KERN_ERR "DMA sound driver: Interrupt initialization failed\n");
return -ENODEV;
}
#ifdef MODULE
irq_installed = 1;
#endif
printk(KERN_INFO "DMA sound driver installed, using %d buffers of %dk.\n",
numBufs, bufSize);
return 0;
}
/* Due to FIFOs and bit delays, the transmit interrupt occurs a few
* microseconds ahead of the receive interrupt.
* When we get an interrupt, we service the transmit first, then
* check for a receive to prevent the overhead of returning through
* the interrupt handler only to get back here right away during
* full duplex operation.
*/
static void
cs4218_intr(void *dev_id, struct pt_regs *regs)
{
volatile smc_t *sp;
volatile cpm8xx_t *cp;
sp = &cpmp->cp_smc[1];
if (sp->smc_smce & SCCM_TX) {
sp->smc_smce = SCCM_TX;
cs4218_tdm_tx_intr((void *)sp);
}
if (sp->smc_smce & SCCM_RX) {
sp->smc_smce = SCCM_RX;
cs4218_tdm_rx_intr((void *)sp);
}
if (sp->smc_smce & SCCM_TXE) {
/* Transmit underrun. This happens with the application
* didn't keep up sending buffers. We tell the SMC to
* restart, which will cause it to poll the current (next)
* BD. If the user supplied data since this occurred,
* we just start running again. If they didn't, the SMC
* will poll the descriptor until data is placed there.
*/
sp->smc_smce = SCCM_TXE;
cp = cpmp; /* Get pointer to Communication Processor */
cp->cp_cpcr = mk_cr_cmd(CPM_CR_CH_SMC2,
CPM_CR_RESTART_TX) | CPM_CR_FLG;
while (cp->cp_cpcr & CPM_CR_FLG);
}
}
#define MAXARGS 8 /* Should be sufficient for now */
void __init dmasound_setup(char *str, int *ints)
{
/* check the bootstrap parameter for "dmasound=" */
switch (ints[0]) {
case 3:
if ((ints[3] < 0) || (ints[3] > MAX_CATCH_RADIUS))
printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius);
else
catchRadius = ints[3];
/* fall through */
case 2:
if (ints[1] < MIN_BUFFERS)
printk("dmasound_setup: invalid number of buffers, using default = %d\n", numBufs);
else
numBufs = ints[1];
if (ints[2] < MIN_BUFSIZE || ints[2] > MAX_BUFSIZE)
printk("dmasound_setup: invalid buffer size, using default = %d\n", bufSize);
else
bufSize = ints[2];
break;
case 0:
break;
default:
printk("dmasound_setup: invalid number of arguments\n");
}
}
/* Software SPI functions.
* These are on Port B.
*/
#define PB_SPICLK ((uint)0x00000002)
#define PB_SPIMOSI ((uint)0x00000004)
#define PB_SPIMISO ((uint)0x00000008)
static
void sw_spi_init(void)
{
volatile cpm8xx_t *cp;
volatile uint *hcsr4;
hcsr4 = (volatile uint *)HIOX_CSR4_ADDR;
cp = cpmp; /* Get pointer to Communication Processor */
*hcsr4 &= ~HIOX_CSR4_AUDSPISEL; /* Disable SPI select */
/* Make these Port B signals general purpose I/O.
* First, make sure the clock is low.
*/
cp->cp_pbdat &= ~PB_SPICLK;
cp->cp_pbpar &= ~(PB_SPICLK | PB_SPIMOSI | PB_SPIMISO);
/* Clock and Master Output are outputs.
*/
cp->cp_pbdir |= (PB_SPICLK | PB_SPIMOSI);
/* Master Input.
*/
cp->cp_pbdir &= ~PB_SPIMISO;
}
/* Write the CS4218 control word out the SPI port. While the
* the control word is going out, the status word is arriving.
*/
static
uint cs4218_ctl_write(uint ctlreg)
{
uint status;
sw_spi_io((u_char *)&ctlreg, (u_char *)&status, 4);
/* Shadow the control register.....I guess we could do
* the same for the status, but for now we just return it
* and let the caller decide.
*/
cs4218_control = ctlreg;
return status;
}
static
void sw_spi_io(u_char *obuf, u_char *ibuf, uint bcnt)
{
int bits, i;
u_char outbyte, inbyte;
volatile cpm8xx_t *cp;
volatile uint *hcsr4;
hcsr4 = (volatile uint *)HIOX_CSR4_ADDR;
cp = cpmp; /* Get pointer to Communication Processor */
/* The timing on the bus is pretty slow. Code inefficiency
* and eieio() is our friend here :-).
*/
cp->cp_pbdat &= ~PB_SPICLK;
*hcsr4 |= HIOX_CSR4_AUDSPISEL; /* Enable SPI select */
eieio();
/* Clock in/out the bytes. Data is valid on the falling edge
* of the clock. Data is MSB first.
*/
for (i=0; i<bcnt; i++) {
outbyte = *obuf++;
inbyte = 0;
for (bits=0; bits<8; bits++) {
eieio();
cp->cp_pbdat |= PB_SPICLK;
eieio();
if (outbyte & 0x80)
cp->cp_pbdat |= PB_SPIMOSI;
else
cp->cp_pbdat &= ~PB_SPIMOSI;
eieio();
cp->cp_pbdat &= ~PB_SPICLK;
eieio();
outbyte <<= 1;
inbyte <<= 1;
if (cp->cp_pbdat & PB_SPIMISO)
inbyte |= 1;
}
*ibuf++ = inbyte;
}
*hcsr4 &= ~HIOX_CSR4_AUDSPISEL; /* Disable SPI select */
eieio();
}
void cleanup_module(void)
{
if (irq_installed) {
sound_silence();
#ifdef MODULE
sound.mach.irqcleanup();
#endif
}
sq_release_read_buffers();
sq_release_buffers();
if (mixer_unit >= 0)
unregister_sound_mixer(mixer_unit);
if (state_unit >= 0)
unregister_sound_special(state_unit);
if (sq_unit >= 0)
unregister_sound_dsp(sq_unit);
}
module_init(tdm8xx_sound_init);
module_exit(cleanup_module);
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