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android_kernel_samsung_msm8976/arch/ppc/8xx_io/cs4218_tdm.c
Arjan van de Ven 5dfe4c964a [PATCH] mark struct file_operations const 2 
Many struct file_operations in the kernel can be "const".  Marking them const
moves these to the .rodata section, which avoids false sharing with potential
dirty data.  In addition it'll catch accidental writes at compile time to
these shared resources.

[akpm@osdl.org: sparc64 fix]
Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2007-02-12 09:48:44 -08:00

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/* 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/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_param(catchRadius, int, 0);
module_param(numBufs, int, 0);
module_param(bufSize, int, 0);
module_param(numreadBufs, int, 0);
module_param(readbufSize, int, 0);
#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, gfp_t 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);
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, gfp_t 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);
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 DEFINE_TIMER(beep_timer, 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 const 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_path.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_path.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_path.dentry);
IOCTL_IN(arg, data);
return IOCTL_OUT(arg, sound_set_speed(data));
case SNDCTL_DSP_STEREO:
sq_fsync(file, file->f_path.dentry);
IOCTL_IN(arg, data);
return IOCTL_OUT(arg, sound_set_stereo(data));
case SOUND_PCM_WRITE_CHANNELS:
sq_fsync(file, file->f_path.dentry);
IOCTL_IN(arg, data);
return IOCTL_OUT(arg, sound_set_stereo(data-1)+1);
case SNDCTL_DSP_SETFMT:
sq_fsync(file, file->f_path.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 const 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 const 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 = 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)
{
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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