sound/arm/sa11xx-uda1341.c

   1 /*
   2  *  Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
   3  *  Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
   4  *
   5  *   This program is free software; you can redistribute it and/or modify
   6  *   it under the terms of the GNU General Public License.
   7  *
   8  * History:
   9  *
  10  * 2002-03-13   Tomas Kasparek  initial release - based on h3600-uda1341.c from OSS
  11  * 2002-03-20   Tomas Kasparek  playback over ALSA is working
  12  * 2002-03-28   Tomas Kasparek  playback over OSS emulation is working
  13  * 2002-03-29   Tomas Kasparek  basic capture is working (native ALSA)
  14  * 2002-03-29   Tomas Kasparek  capture is working (OSS emulation)
  15  * 2002-04-04   Tomas Kasparek  better rates handling (allow non-standard rates)
  16  * 2003-02-14   Brian Avery     fixed full duplex mode, other updates
  17  * 2003-02-20   Tomas Kasparek  merged updates by Brian (except HAL)
  18  * 2003-04-19   Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel
  19  *                              working suspend and resume
  20  * 2003-04-28   Tomas Kasparek  updated work by Jaroslav to compile it under 2.5.x again
  21  *                              merged HAL layer (patches from Brian)
  22  */
  23
  24 /* $Id: sa11xx-uda1341.c,v 1.27 2005/12/07 09:13:42 cladisch Exp $ */
  25
  26 /***************************************************************************************************
  27 *
  28 * To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
  29 * available in the Alsa doc section on the website
  30 *
  31 * A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
  32 * We are using  SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
  33 * by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
  34 * So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
  35 * transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
  36 * is a mem loc that always decodes to 0's w/ no off chip access.
  37 *
  38 * Some alsa terminology:
  39 *       frame => num_channels * sample_size  e.g stereo 16 bit is 2 * 16 = 32 bytes
  40 *       period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
  41 *             buffer and 4 periods in the runtime structure this means we'll get an int every 256
  42 *             bytes or 4 times per buffer.
  43 *             A number of the sizes are in frames rather than bytes, use frames_to_bytes and
  44 *             bytes_to_frames to convert.  The easiest way to tell the units is to look at the
  45 *             type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
  46 *
  47 *       Notes about the pointer fxn:
  48 *       The pointer fxn needs to return the offset into the dma buffer in frames.
  49 *       Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
  50 *
  51 *       Notes about pause/resume
  52 *       Implementing this would be complicated so it's skipped.  The problem case is:
  53 *       A full duplex connection is going, then play is paused. At this point you need to start xmitting
  54 *       0's to keep the record active which means you cant just freeze the dma and resume it later you'd
  55 *       need to save off the dma info, and restore it properly on a resume.  Yeach!
  56 *
  57 *       Notes about transfer methods:
  58 *       The async write calls fail.  I probably need to implement something else to support them?
  59 *
  60 ***************************************************************************************************/
  61
  62 #include <sound/driver.h>
  63 #include <linux/module.h>
  64 #include <linux/moduleparam.h>
  65 #include <linux/init.h>
  66 #include <linux/err.h>
  67 #include <linux/platform_device.h>
  68 #include <linux/errno.h>
  69 #include <linux/ioctl.h>
  70 #include <linux/delay.h>
  71 #include <linux/slab.h>
  72
  73 #ifdef CONFIG_PM
  74 #include <linux/pm.h>
  75 #endif
  76
  77 #include <asm/hardware.h>
  78 #include <asm/arch/h3600.h>
  79 #include <asm/mach-types.h>
  80 #include <asm/dma.h>
  81
  82 #include <sound/core.h>
  83 #include <sound/pcm.h>
  84 #include <sound/initval.h>
  85
  86 #include <linux/l3/l3.h>
  87
  88 #undef DEBUG_MODE
  89 #undef DEBUG_FUNCTION_NAMES
  90 #include <sound/uda1341.h>
  91
  92 /*
  93  * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
  94  * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
  95  * module for Familiar 0.6.1
  96  */
  97
  98 /* {{{ Type definitions */
  99
 100 MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
 101 MODULE_LICENSE("GPL");
 102 MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
 103 MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
 104
 105 static char *id;        /* ID for this card */
 106
 107 module_param(id, charp, 0444);
 108 MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
 109
 110 struct audio_stream {
 111         char *id;               /* identification string */
 112         int stream_id;          /* numeric identification */
 113         dma_device_t dma_dev;   /* device identifier for DMA */
 114 #ifdef HH_VERSION
 115         dmach_t dmach;          /* dma channel identification */
 116 #else
 117         dma_regs_t *dma_regs;   /* points to our DMA registers */
 118 #endif
 119         unsigned int active:1;  /* we are using this stream for transfer now */
 120         int period;             /* current transfer period */
 121         int periods;            /* current count of periods registerd in the DMA engine */
 122         int tx_spin;            /* are we recoding - flag used to do DMA trans. for sync */
 123         unsigned int old_offset;
 124         spinlock_t dma_lock;    /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
 125         struct snd_pcm_substream *stream;
 126 };
 127
 128 struct sa11xx_uda1341 {
 129         struct snd_card *card;
 130         struct l3_client *uda1341;
 131         struct snd_pcm *pcm;
 132         long samplerate;
 133         struct audio_stream s[2];       /* playback & capture */
 134 };
 135
 136 static unsigned int rates[] = {
 137         8000,  10666, 10985, 14647,
 138         16000, 21970, 22050, 24000,
 139         29400, 32000, 44100, 48000,
 140 };
 141
 142 static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
 143         .count  = ARRAY_SIZE(rates),
 144         .list   = rates,
 145         .mask   = 0,
 146 };
 147
 148 static struct platform_device *device;
 149
 150 /* }}} */
 151
 152 /* {{{ Clock and sample rate stuff */
 153
 154 /*
 155  * Stop-gap solution until rest of hh.org HAL stuff is merged.
 156  */
 157 #define GPIO_H3600_CLK_SET0             GPIO_GPIO (12)
 158 #define GPIO_H3600_CLK_SET1             GPIO_GPIO (13)
 159
 160 #ifdef CONFIG_SA1100_H3XXX
 161 #define clr_sa11xx_uda1341_egpio(x)     clr_h3600_egpio(x)
 162 #define set_sa11xx_uda1341_egpio(x)     set_h3600_egpio(x)
 163 #else
 164 #error This driver could serve H3x00 handhelds only!
 165 #endif
 166
 167 static void sa11xx_uda1341_set_audio_clock(long val)
 168 {
 169         switch (val) {
 170         case 24000: case 32000: case 48000:     /* 00: 12.288 MHz */
 171                 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
 172                 break;
 173
 174         case 22050: case 29400: case 44100:     /* 01: 11.2896 MHz */
 175                 GPSR = GPIO_H3600_CLK_SET0;
 176                 GPCR = GPIO_H3600_CLK_SET1;
 177                 break;
 178
 179         case 8000: case 10666: case 16000:      /* 10: 4.096 MHz */
 180                 GPCR = GPIO_H3600_CLK_SET0;
 181                 GPSR = GPIO_H3600_CLK_SET1;
 182                 break;
 183
 184         case 10985: case 14647: case 21970:     /* 11: 5.6245 MHz */
 185                 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
 186                 break;
 187         }
 188 }
 189
 190 static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
 191 {
 192         int clk_div = 0;
 193         int clk=0;
 194
 195         /* We don't want to mess with clocks when frames are in flight */
 196         Ser4SSCR0 &= ~SSCR0_SSE;
 197         /* wait for any frame to complete */
 198         udelay(125);
 199
 200         /*
 201          * We have the following clock sources:
 202          * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
 203          * Those can be divided either by 256, 384 or 512.
 204          * This makes up 12 combinations for the following samplerates...
 205          */
 206         if (rate >= 48000)
 207                 rate = 48000;
 208         else if (rate >= 44100)
 209                 rate = 44100;
 210         else if (rate >= 32000)
 211                 rate = 32000;
 212         else if (rate >= 29400)
 213                 rate = 29400;
 214         else if (rate >= 24000)
 215                 rate = 24000;
 216         else if (rate >= 22050)
 217                 rate = 22050;
 218         else if (rate >= 21970)
 219                 rate = 21970;
 220         else if (rate >= 16000)
 221                 rate = 16000;
 222         else if (rate >= 14647)
 223                 rate = 14647;
 224         else if (rate >= 10985)
 225                 rate = 10985;
 226         else if (rate >= 10666)
 227                 rate = 10666;
 228         else
 229                 rate = 8000;
 230
 231         /* Set the external clock generator */
 232
 233         sa11xx_uda1341_set_audio_clock(rate);
 234
 235         /* Select the clock divisor */
 236         switch (rate) {
 237         case 8000:
 238         case 10985:
 239         case 22050:
 240         case 24000:
 241                 clk = F512;
 242                 clk_div = SSCR0_SerClkDiv(16);
 243                 break;
 244         case 16000:
 245         case 21970:
 246         case 44100:
 247         case 48000:
 248                 clk = F256;
 249                 clk_div = SSCR0_SerClkDiv(8);
 250                 break;
 251         case 10666:
 252         case 14647:
 253         case 29400:
 254         case 32000:
 255                 clk = F384;
 256                 clk_div = SSCR0_SerClkDiv(12);
 257                 break;
 258         }
 259
 260         /* FMT setting should be moved away when other FMTs are added (FIXME) */
 261         l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
 262
 263         l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
 264         Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
 265         sa11xx_uda1341->samplerate = rate;
 266 }
 267
 268 /* }}} */
 269
 270 /* {{{ HW init and shutdown */
 271
 272 static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
 273 {
 274         unsigned long flags;
 275
 276         /* Setup DMA stuff */
 277         sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
 278         sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
 279         sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
 280
 281         sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
 282         sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
 283         sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
 284
 285         /* Initialize the UDA1341 internal state */
 286
 287         /* Setup the uarts */
 288         local_irq_save(flags);
 289         GAFR |= (GPIO_SSP_CLK);
 290         GPDR &= ~(GPIO_SSP_CLK);
 291         Ser4SSCR0 = 0;
 292         Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
 293         Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
 294         Ser4SSCR0 |= SSCR0_SSE;
 295         local_irq_restore(flags);
 296
 297         /* Enable the audio power */
 298
 299         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
 300         set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
 301         set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 302
 303         /* Wait for the UDA1341 to wake up */
 304         mdelay(1); //FIXME - was removed by Perex - Why?
 305
 306         /* Initialize the UDA1341 internal state */
 307         l3_open(sa11xx_uda1341->uda1341);
 308
 309         /* external clock configuration (after l3_open - regs must be initialized */
 310         sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
 311
 312         /* Wait for the UDA1341 to wake up */
 313         set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
 314         mdelay(1);
 315
 316         /* make the left and right channels unswapped (flip the WS latch) */
 317         Ser4SSDR = 0;
 318
 319         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 320 }
 321
 322 static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
 323 {
 324         /* mute on */
 325         set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 326
 327         /* disable the audio power and all signals leading to the audio chip */
 328         l3_close(sa11xx_uda1341->uda1341);
 329         Ser4SSCR0 = 0;
 330         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
 331
 332         /* power off and mute off */
 333         /* FIXME - is muting off necesary??? */
 334
 335         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
 336         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 337 }
 338
 339 /* }}} */
 340
 341 /* {{{ DMA staff */
 342
 343 /*
 344  * these are the address and sizes used to fill the xmit buffer
 345  * so we can get a clock in record only mode
 346  */
 347 #define FORCE_CLOCK_ADDR                (dma_addr_t)FLUSH_BASE_PHYS
 348 #define FORCE_CLOCK_SIZE                4096 // was 2048
 349
 350 // FIXME Why this value exactly - wrote comment
 351 #define DMA_BUF_SIZE    8176    /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
 352
 353 #ifdef HH_VERSION
 354
 355 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
 356 {
 357         int ret;
 358
 359         ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
 360         if (ret < 0) {
 361                 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
 362                 return ret;
 363         }
 364         sa1100_dma_set_callback(s->dmach, callback);
 365         return 0;
 366 }
 367
 368 static inline void audio_dma_free(struct audio_stream *s)
 369 {
 370         sa1100_free_dma(s->dmach);
 371         s->dmach = -1;
 372 }
 373
 374 #else
 375
 376 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
 377 {
 378         int ret;
 379
 380         ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
 381         if (ret < 0)
 382                 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
 383         return ret;
 384 }
 385
 386 static void audio_dma_free(struct audio_stream *s)
 387 {
 388         sa1100_free_dma(s->dma_regs);
 389         s->dma_regs = 0;
 390 }
 391
 392 #endif
 393
 394 static u_int audio_get_dma_pos(struct audio_stream *s)
 395 {
 396         struct snd_pcm_substream *substream = s->stream;
 397         struct snd_pcm_runtime *runtime = substream->runtime;
 398         unsigned int offset;
 399         unsigned long flags;
 400         dma_addr_t addr;
 401
 402         // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
 403         spin_lock_irqsave(&s->dma_lock, flags);
 404 #ifdef HH_VERSION
 405         sa1100_dma_get_current(s->dmach, NULL, &addr);
 406 #else
 407         addr = sa1100_get_dma_pos((s)->dma_regs);
 408 #endif
 409         offset = addr - runtime->dma_addr;
 410         spin_unlock_irqrestore(&s->dma_lock, flags);
 411
 412         offset = bytes_to_frames(runtime,offset);
 413         if (offset >= runtime->buffer_size)
 414                 offset = 0;
 415
 416         return offset;
 417 }
 418
 419 /*
 420  * this stops the dma and clears the dma ptrs
 421  */
 422 static void audio_stop_dma(struct audio_stream *s)
 423 {
 424         unsigned long flags;
 425
 426         spin_lock_irqsave(&s->dma_lock, flags);
 427         s->active = 0;
 428         s->period = 0;
 429         /* this stops the dma channel and clears the buffer ptrs */
 430 #ifdef HH_VERSION
 431         sa1100_dma_flush_all(s->dmach);
 432 #else
 433         sa1100_clear_dma(s->dma_regs);
 434 #endif
 435         spin_unlock_irqrestore(&s->dma_lock, flags);
 436 }
 437
 438 static void audio_process_dma(struct audio_stream *s)
 439 {
 440         struct snd_pcm_substream *substream = s->stream;
 441         struct snd_pcm_runtime *runtime;
 442         unsigned int dma_size;
 443         unsigned int offset;
 444         int ret;
 445
 446         /* we are requested to process synchronization DMA transfer */
 447         if (s->tx_spin) {
 448                 snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
 449                 /* fill the xmit dma buffers and return */
 450 #ifdef HH_VERSION
 451                 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
 452 #else
 453                 while (1) {
 454                         ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
 455                         if (ret)
 456                                 return;
 457                 }
 458 #endif
 459                 return;
 460         }
 461
 462         /* must be set here - only valid for running streams, not for forced_clock dma fills  */
 463         runtime = substream->runtime;
 464         while (s->active && s->periods < runtime->periods) {
 465                 dma_size = frames_to_bytes(runtime, runtime->period_size);
 466                 if (s->old_offset) {
 467                         /* a little trick, we need resume from old position */
 468                         offset = frames_to_bytes(runtime, s->old_offset - 1);
 469                         s->old_offset = 0;
 470                         s->periods = 0;
 471                         s->period = offset / dma_size;
 472                         offset %= dma_size;
 473                         dma_size = dma_size - offset;
 474                         if (!dma_size)
 475                                 continue;               /* special case */
 476                 } else {
 477                         offset = dma_size * s->period;
 478                         snd_assert(dma_size <= DMA_BUF_SIZE, );
 479                 }
 480 #ifdef HH_VERSION
 481                 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
 482                 if (ret)
 483                         return; //FIXME
 484 #else
 485                 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
 486                 if (ret) {
 487                         printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
 488                         return;
 489                 }
 490 #endif
 491
 492                 s->period++;
 493                 s->period %= runtime->periods;
 494                 s->periods++;
 495         }
 496 }
 497
 498 #ifdef HH_VERSION
 499 static void audio_dma_callback(void *data, int size)
 500 #else
 501 static void audio_dma_callback(void *data)
 502 #endif
 503 {
 504         struct audio_stream *s = data;
 505
 506         /*
 507          * If we are getting a callback for an active stream then we inform
 508          * the PCM middle layer we've finished a period
 509          */
 510         if (s->active)
 511                 snd_pcm_period_elapsed(s->stream);
 512
 513         spin_lock(&s->dma_lock);
 514         if (!s->tx_spin && s->periods > 0)
 515                 s->periods--;
 516         audio_process_dma(s);
 517         spin_unlock(&s->dma_lock);
 518 }
 519
 520 /* }}} */
 521
 522 /* {{{ PCM setting */
 523
 524 /* {{{ trigger & timer */
 525
 526 static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
 527 {
 528         struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
 529         int stream_id = substream->pstr->stream;
 530         struct audio_stream *s = &chip->s[stream_id];
 531         struct audio_stream *s1 = &chip->s[stream_id ^ 1];
 532         int err = 0;
 533
 534         /* note local interrupts are already disabled in the midlevel code */
 535         spin_lock(&s->dma_lock);
 536         switch (cmd) {
 537         case SNDRV_PCM_TRIGGER_START:
 538                 /* now we need to make sure a record only stream has a clock */
 539                 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
 540                         /* we need to force fill the xmit DMA with zeros */
 541                         s1->tx_spin = 1;
 542                         audio_process_dma(s1);
 543                 }
 544                 /* this case is when you were recording then you turn on a
 545                  * playback stream so we stop (also clears it) the dma first,
 546                  * clear the sync flag and then we let it turned on
 547                  */
 548                 else {
 549                         s->tx_spin = 0;
 550                 }
 551
 552                 /* requested stream startup */
 553                 s->active = 1;
 554                 audio_process_dma(s);
 555                 break;
 556         case SNDRV_PCM_TRIGGER_STOP:
 557                 /* requested stream shutdown */
 558                 audio_stop_dma(s);
 559
 560                 /*
 561                  * now we need to make sure a record only stream has a clock
 562                  * so if we're stopping a playback with an active capture
 563                  * we need to turn the 0 fill dma on for the xmit side
 564                  */
 565                 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
 566                         /* we need to force fill the xmit DMA with zeros */
 567                         s->tx_spin = 1;
 568                         audio_process_dma(s);
 569                 }
 570                 /*
 571                  * we killed a capture only stream, so we should also kill
 572                  * the zero fill transmit
 573                  */
 574                 else {
 575                         if (s1->tx_spin) {
 576                                 s1->tx_spin = 0;
 577                                 audio_stop_dma(s1);
 578                         }
 579                 }
 580
 581                 break;
 582         case SNDRV_PCM_TRIGGER_SUSPEND:
 583                 s->active = 0;
 584 #ifdef HH_VERSION
 585                 sa1100_dma_stop(s->dmach);
 586 #else
 587                 //FIXME - DMA API
 588 #endif
 589                 s->old_offset = audio_get_dma_pos(s) + 1;
 590 #ifdef HH_VERSION
 591                 sa1100_dma_flush_all(s->dmach);
 592 #else
 593                 //FIXME - DMA API
 594 #endif
 595                 s->periods = 0;
 596                 break;
 597         case SNDRV_PCM_TRIGGER_RESUME:
 598                 s->active = 1;
 599                 s->tx_spin = 0;
 600                 audio_process_dma(s);
 601                 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
 602                         s1->tx_spin = 1;
 603                         audio_process_dma(s1);
 604                 }
 605                 break;
 606         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 607 #ifdef HH_VERSION
 608                 sa1100_dma_stop(s->dmach);
 609 #else
 610                 //FIXME - DMA API
 611 #endif
 612                 s->active = 0;
 613                 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
 614                         if (s1->active) {
 615                                 s->tx_spin = 1;
 616                                 s->old_offset = audio_get_dma_pos(s) + 1;
 617 #ifdef HH_VERSION
 618                                 sa1100_dma_flush_all(s->dmach);
 619 #else
 620                                 //FIXME - DMA API
 621 #endif
 622                                 audio_process_dma(s);
 623                         }
 624                 } else {
 625                         if (s1->tx_spin) {
 626                                 s1->tx_spin = 0;
 627 #ifdef HH_VERSION
 628                                 sa1100_dma_flush_all(s1->dmach);
 629 #else
 630                                 //FIXME - DMA API
 631 #endif
 632                         }
 633                 }
 634                 break;
 635         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 636                 s->active = 1;
 637                 if (s->old_offset) {
 638                         s->tx_spin = 0;
 639                         audio_process_dma(s);
 640                         break;
 641                 }
 642                 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
 643                         s1->tx_spin = 1;
 644                         audio_process_dma(s1);
 645                 }
 646 #ifdef HH_VERSION
 647                 sa1100_dma_resume(s->dmach);
 648 #else
 649                 //FIXME - DMA API
 650 #endif
 651                 break;
 652         default:
 653                 err = -EINVAL;
 654                 break;
 655         }
 656         spin_unlock(&s->dma_lock);
 657         return err;
 658 }
 659
 660 static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
 661 {
 662         struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
 663         struct snd_pcm_runtime *runtime = substream->runtime;
 664         struct audio_stream *s = &chip->s[substream->pstr->stream];
 665
 666         /* set requested samplerate */
 667         sa11xx_uda1341_set_samplerate(chip, runtime->rate);
 668
 669         /* set requestd format when available */
 670         /* set FMT here !!! FIXME */
 671
 672         s->period = 0;
 673         s->periods = 0;
 674
 675         return 0;
 676 }
 677
 678 static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
 679 {
 680         struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
 681         return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
 682 }
 683
 684 /* }}} */
 685
 686 static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
 687 {
 688         .info                   = (SNDRV_PCM_INFO_INTERLEAVED |
 689                                    SNDRV_PCM_INFO_BLOCK_TRANSFER |
 690                                    SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 691                                    SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
 692         .formats                = SNDRV_PCM_FMTBIT_S16_LE,
 693         .rates                  = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
 694                                    SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
 695                                    SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
 696                                    SNDRV_PCM_RATE_KNOT),
 697         .rate_min               = 8000,
 698         .rate_max               = 48000,
 699         .channels_min           = 2,
 700         .channels_max           = 2,
 701         .buffer_bytes_max       = 64*1024,
 702         .period_bytes_min       = 64,
 703         .period_bytes_max       = DMA_BUF_SIZE,
 704         .periods_min            = 2,
 705         .periods_max            = 255,
 706         .fifo_size              = 0,
 707 };
 708
 709 static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
 710 {
 711         .info                   = (SNDRV_PCM_INFO_INTERLEAVED |
 712                                    SNDRV_PCM_INFO_BLOCK_TRANSFER |
 713                                    SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 714                                    SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
 715         .formats                = SNDRV_PCM_FMTBIT_S16_LE,
 716         .rates                  = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
 717                                    SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
 718                                    SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
 719                                    SNDRV_PCM_RATE_KNOT),
 720         .rate_min               = 8000,
 721         .rate_max               = 48000,
 722         .channels_min           = 2,
 723         .channels_max           = 2,
 724         .buffer_bytes_max       = 64*1024,
 725         .period_bytes_min       = 64,
 726         .period_bytes_max       = DMA_BUF_SIZE,
 727         .periods_min            = 2,
 728         .periods_max            = 255,
 729         .fifo_size              = 0,
 730 };
 731
 732 static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
 733 {
 734         struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
 735         struct snd_pcm_runtime *runtime = substream->runtime;
 736         int stream_id = substream->pstr->stream;
 737         int err;
 738
 739         chip->s[stream_id].stream = substream;
 740
 741         if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
 742                 runtime->hw = snd_sa11xx_uda1341_playback;
 743         else
 744                 runtime->hw = snd_sa11xx_uda1341_capture;
 745         if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
 746                 return err;
 747         if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
 748                 return err;
 749
 750         return 0;
 751 }
 752
 753 static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
 754 {
 755         struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
 756
 757         chip->s[substream->pstr->stream].stream = NULL;
 758         return 0;
 759 }
 760
 761 /* {{{ HW params & free */
 762
 763 static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
 764                                         struct snd_pcm_hw_params *hw_params)
 765 {
 766
 767         return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
 768 }
 769
 770 static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
 771 {
 772         return snd_pcm_lib_free_pages(substream);
 773 }
 774
 775 /* }}} */
 776
 777 static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
 778         .open                   = snd_card_sa11xx_uda1341_open,
 779         .close                  = snd_card_sa11xx_uda1341_close,
 780         .ioctl                  = snd_pcm_lib_ioctl,
 781         .hw_params              = snd_sa11xx_uda1341_hw_params,
 782         .hw_free                = snd_sa11xx_uda1341_hw_free,
 783         .prepare                = snd_sa11xx_uda1341_prepare,
 784         .trigger                = snd_sa11xx_uda1341_trigger,
 785         .pointer                = snd_sa11xx_uda1341_pointer,
 786 };
 787
 788 static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
 789         .open                   = snd_card_sa11xx_uda1341_open,
 790         .close                  = snd_card_sa11xx_uda1341_close,
 791         .ioctl                  = snd_pcm_lib_ioctl,
 792         .hw_params              = snd_sa11xx_uda1341_hw_params,
 793         .hw_free                = snd_sa11xx_uda1341_hw_free,
 794         .prepare                = snd_sa11xx_uda1341_prepare,
 795         .trigger                = snd_sa11xx_uda1341_trigger,
 796         .pointer                = snd_sa11xx_uda1341_pointer,
 797 };
 798
 799 static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
 800 {
 801         struct snd_pcm *pcm;
 802         int err;
 803
 804         if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
 805                 return err;
 806
 807         /*
 808          * this sets up our initial buffers and sets the dma_type to isa.
 809          * isa works but I'm not sure why (or if) it's the right choice
 810          * this may be too large, trying it for now
 811          */
 812         snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 813                                               snd_dma_isa_data(),
 814                                               64*1024, 64*1024);
 815
 816         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
 817         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
 818         pcm->private_data = sa11xx_uda1341;
 819         pcm->info_flags = 0;
 820         strcpy(pcm->name, "UDA1341 PCM");
 821
 822         sa11xx_uda1341_audio_init(sa11xx_uda1341);
 823
 824         /* setup DMA controller */
 825         audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
 826         audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
 827
 828         sa11xx_uda1341->pcm = pcm;
 829
 830         return 0;
 831 }
 832
 833 /* }}} */
 834
 835 /* {{{ module init & exit */
 836
 837 #ifdef CONFIG_PM
 838
 839 static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
 840                                       pm_message_t state)
 841 {
 842         struct snd_card *card = platform_get_drvdata(devptr);
 843         struct sa11xx_uda1341 *chip = card->private_data;
 844
 845         snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 846         snd_pcm_suspend_all(chip->pcm);
 847 #ifdef HH_VERSION
 848         sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
 849         sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
 850 #else
 851         //FIXME
 852 #endif
 853         l3_command(chip->uda1341, CMD_SUSPEND, NULL);
 854         sa11xx_uda1341_audio_shutdown(chip);
 855
 856         return 0;
 857 }
 858
 859 static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
 860 {
 861         struct snd_card *card = platform_get_drvdata(devptr);
 862         struct sa11xx_uda1341 *chip = card->private_data;
 863
 864         sa11xx_uda1341_audio_init(chip);
 865         l3_command(chip->uda1341, CMD_RESUME, NULL);
 866 #ifdef HH_VERSION
 867         sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
 868         sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
 869 #else
 870         //FIXME
 871 #endif
 872         snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 873         return 0;
 874 }
 875 #endif /* COMFIG_PM */
 876
 877 void snd_sa11xx_uda1341_free(struct snd_card *card)
 878 {
 879         struct sa11xx_uda1341 *chip = card->private_data;
 880
 881         audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
 882         audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
 883 }
 884
 885 static int __init sa11xx_uda1341_probe(struct platform_device *devptr)
 886 {
 887         int err;
 888         struct snd_card *card;
 889         struct sa11xx_uda1341 *chip;
 890
 891         /* register the soundcard */
 892         card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
 893         if (card == NULL)
 894                 return -ENOMEM;
 895
 896         chip = card->private_data;
 897         spin_lock_init(&chip->s[0].dma_lock);
 898         spin_lock_init(&chip->s[1].dma_lock);
 899
 900         card->private_free = snd_sa11xx_uda1341_free;
 901         chip->card = card;
 902         chip->samplerate = AUDIO_RATE_DEFAULT;
 903
 904         // mixer
 905         if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
 906                 goto nodev;
 907
 908         // PCM
 909         if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
 910                 goto nodev;
 911
 912         strcpy(card->driver, "UDA1341");
 913         strcpy(card->shortname, "H3600 UDA1341TS");
 914         sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
 915
 916         snd_card_set_dev(card, &devptr->dev);
 917
 918         if ((err = snd_card_register(card)) == 0) {
 919                 printk( KERN_INFO "iPAQ audio support initialized\n" );
 920                 platform_set_drvdata(devptr, card);
 921                 return 0;
 922         }
 923
 924  nodev:
 925         snd_card_free(card);
 926         return err;
 927 }
 928
 929 static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
 930 {
 931         snd_card_free(platform_get_drvdata(devptr));
 932         platform_set_drvdata(devptr, NULL);
 933         return 0;
 934 }
 935
 936 #define SA11XX_UDA1341_DRIVER   "sa11xx_uda1341"
 937
 938 static struct platform_driver sa11xx_uda1341_driver = {
 939         .probe          = sa11xx_uda1341_probe,
 940         .remove         = __devexit_p(sa11xx_uda1341_remove),
 941 #ifdef CONFIG_PM
 942         .suspend        = snd_sa11xx_uda1341_suspend,
 943         .resume         = snd_sa11xx_uda1341_resume,
 944 #endif
 945         .driver         = {
 946                 .name   = SA11XX_UDA1341_DRIVER,
 947         },
 948 };
 949
 950 static int __init sa11xx_uda1341_init(void)
 951 {
 952         int err;
 953
 954         if (!machine_is_h3xxx())
 955                 return -ENODEV;
 956         if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
 957                 return err;
 958         device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
 959         if (!IS_ERR(device)) {
 960                 if (platform_get_drvdata(device))
 961                         return 0;
 962                 platform_device_unregister(device);
 963                 err = -ENODEV;
 964         } else
 965                 err = PTR_ERR(device);
 966         platform_driver_unregister(&sa11xx_uda1341_driver);
 967         return err;
 968 }
 969
 970 static void __exit sa11xx_uda1341_exit(void)
 971 {
 972         platform_device_unregister(device);
 973         platform_driver_unregister(&sa11xx_uda1341_driver);
 974 }
 975
 976 module_init(sa11xx_uda1341_init);
 977 module_exit(sa11xx_uda1341_exit);
 978
 979 /* }}} */
 980
 981 /*
 982  * Local variables:
 983  * indent-tabs-mode: t
 984  * End:
 985  */