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2822 2823 2824 2825 2826 2827 2828 2829 2830 2831 2832 2833 2834 | /* 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, 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, 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 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); |