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-rw-r--r--arch/ppc/8xx_io/cs4218_tdm.c2836
1 files changed, 2836 insertions, 0 deletions
diff --git a/arch/ppc/8xx_io/cs4218_tdm.c b/arch/ppc/8xx_io/cs4218_tdm.c
new file mode 100644
index 00000000000..89fe0ceeaa4
--- /dev/null
+++ b/arch/ppc/8xx_io/cs4218_tdm.c
@@ -0,0 +1,2836 @@
+
+/* 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);
+
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-17 16:09:40 +0000