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 1.18 2004/07/20 15:54:09 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 <linux/config.h>
  63 #include <sound/driver.h>
  64 #include <linux/module.h>
  65 #include <linux/moduleparam.h>
  66 #include <linux/init.h>
  67 #include <linux/errno.h>
  68 #include <linux/ioctl.h>
  69 #include <linux/delay.h>
  70 #include <linux/slab.h>
  71
  72 #ifdef CONFIG_PM
  73 #include <linux/pm.h>
  74 #endif
  75
  76 #include <asm/hardware.h>
  77 #include <asm/arch/h3600.h>
  78 #include <asm/mach-types.h>
  79 #include <asm/dma.h>
  80
  81 #ifdef CONFIG_H3600_HAL
  82 #include <asm/semaphore.h>
  83 #include <asm/uaccess.h>
  84 #include <asm/arch/h3600_hal.h>
  85 #endif
  86
  87 #include <sound/core.h>
  88 #include <sound/pcm.h>
  89 #include <sound/initval.h>
  90
  91 #include <linux/l3/l3.h>
  92
  93 #undef DEBUG_MODE
  94 #undef DEBUG_FUNCTION_NAMES
  95 #include <sound/uda1341.h>
  96
  97 /*
  98  * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
  99  * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
 100  * module for Familiar 0.6.1
 101  */
 102 #ifdef CONFIG_H3600_HAL
 103 #define HH_VERSION 1
 104 #endif
 105
 106 /* {{{ Type definitions */
 107
 108 MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
 109 MODULE_LICENSE("GPL");
 110 MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
 111 MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
 112
 113 static char *id = NULL; /* ID for this card */
 114
 115 module_param(id, charp, 0444);
 116 MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
 117
 118 typedef struct audio_stream {
 119         char *id;               /* identification string */
 120         int stream_id;          /* numeric identification */
 121         dma_device_t dma_dev;   /* device identifier for DMA */
 122 #ifdef HH_VERSION
 123         dmach_t dmach;          /* dma channel identification */
 124 #else
 125         dma_regs_t *dma_regs;   /* points to our DMA registers */
 126 #endif
 127         int active:1;           /* we are using this stream for transfer now */
 128         int period;             /* current transfer period */
 129         int periods;            /* current count of periods registerd in the DMA engine */
 130         int tx_spin;            /* are we recoding - flag used to do DMA trans. for sync */
 131         unsigned int old_offset;
 132         spinlock_t dma_lock;    /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
 133         snd_pcm_substream_t *stream;
 134 }audio_stream_t;
 135
 136 typedef struct snd_card_sa11xx_uda1341 {
 137         snd_card_t *card;
 138         struct l3_client *uda1341;
 139         snd_pcm_t *pcm;
 140         long samplerate;
 141         audio_stream_t s[2];    /* playback & capture */
 142 } sa11xx_uda1341_t;
 143
 144 static struct snd_card_sa11xx_uda1341 *sa11xx_uda1341 = NULL;
 145
 146 static unsigned int rates[] = {
 147         8000,  10666, 10985, 14647,
 148         16000, 21970, 22050, 24000,
 149         29400, 32000, 44100, 48000,
 150 };
 151
 152 static snd_pcm_hw_constraint_list_t hw_constraints_rates = {
 153         .count  = ARRAY_SIZE(rates),
 154         .list   = rates,
 155         .mask   = 0,
 156 };
 157
 158 /* }}} */
 159
 160 /* {{{ Clock and sample rate stuff */
 161
 162 /*
 163  * Stop-gap solution until rest of hh.org HAL stuff is merged.
 164  */
 165 #define GPIO_H3600_CLK_SET0             GPIO_GPIO (12)
 166 #define GPIO_H3600_CLK_SET1             GPIO_GPIO (13)
 167
 168 #ifdef CONFIG_SA1100_H3XXX
 169 #define clr_sa11xx_uda1341_egpio(x)     clr_h3600_egpio(x)
 170 #define set_sa11xx_uda1341_egpio(x)     set_h3600_egpio(x)
 171 #else
 172 #error This driver could serve H3x00 handhelds only!
 173 #endif
 174
 175 static void sa11xx_uda1341_set_audio_clock(long val)
 176 {
 177         switch (val) {
 178         case 24000: case 32000: case 48000:     /* 00: 12.288 MHz */
 179                 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
 180                 break;
 181
 182         case 22050: case 29400: case 44100:     /* 01: 11.2896 MHz */
 183                 GPSR = GPIO_H3600_CLK_SET0;
 184                 GPCR = GPIO_H3600_CLK_SET1;
 185                 break;
 186
 187         case 8000: case 10666: case 16000:      /* 10: 4.096 MHz */
 188                 GPCR = GPIO_H3600_CLK_SET0;
 189                 GPSR = GPIO_H3600_CLK_SET1;
 190                 break;
 191
 192         case 10985: case 14647: case 21970:     /* 11: 5.6245 MHz */
 193                 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
 194                 break;
 195         }
 196 }
 197
 198 static void sa11xx_uda1341_set_samplerate(sa11xx_uda1341_t *sa11xx_uda1341, long rate)
 199 {
 200         int clk_div = 0;
 201         int clk=0;
 202
 203         /* We don't want to mess with clocks when frames are in flight */
 204         Ser4SSCR0 &= ~SSCR0_SSE;
 205         /* wait for any frame to complete */
 206         udelay(125);
 207
 208         /*
 209          * We have the following clock sources:
 210          * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
 211          * Those can be divided either by 256, 384 or 512.
 212          * This makes up 12 combinations for the following samplerates...
 213          */
 214         if (rate >= 48000)
 215                 rate = 48000;
 216         else if (rate >= 44100)
 217                 rate = 44100;
 218         else if (rate >= 32000)
 219                 rate = 32000;
 220         else if (rate >= 29400)
 221                 rate = 29400;
 222         else if (rate >= 24000)
 223                 rate = 24000;
 224         else if (rate >= 22050)
 225                 rate = 22050;
 226         else if (rate >= 21970)
 227                 rate = 21970;
 228         else if (rate >= 16000)
 229                 rate = 16000;
 230         else if (rate >= 14647)
 231                 rate = 14647;
 232         else if (rate >= 10985)
 233                 rate = 10985;
 234         else if (rate >= 10666)
 235                 rate = 10666;
 236         else
 237                 rate = 8000;
 238
 239         /* Set the external clock generator */
 240 #ifdef CONFIG_H3600_HAL
 241         h3600_audio_clock(rate);
 242 #else
 243         sa11xx_uda1341_set_audio_clock(rate);
 244 #endif
 245
 246         /* Select the clock divisor */
 247         switch (rate) {
 248         case 8000:
 249         case 10985:
 250         case 22050:
 251         case 24000:
 252                 clk = F512;
 253                 clk_div = SSCR0_SerClkDiv(16);
 254                 break;
 255         case 16000:
 256         case 21970:
 257         case 44100:
 258         case 48000:
 259                 clk = F256;
 260                 clk_div = SSCR0_SerClkDiv(8);
 261                 break;
 262         case 10666:
 263         case 14647:
 264         case 29400:
 265         case 32000:
 266                 clk = F384;
 267                 clk_div = SSCR0_SerClkDiv(12);
 268                 break;
 269         }
 270
 271         /* FMT setting should be moved away when other FMTs are added (FIXME) */
 272         l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
 273
 274         l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
 275         Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
 276         sa11xx_uda1341->samplerate = rate;
 277 }
 278
 279 /* }}} */
 280
 281 /* {{{ HW init and shutdown */
 282
 283 static void sa11xx_uda1341_audio_init(sa11xx_uda1341_t *sa11xx_uda1341)
 284 {
 285         unsigned long flags;
 286
 287         /* Setup DMA stuff */
 288         sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
 289         sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
 290         sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
 291
 292         sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
 293         sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
 294         sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
 295
 296         /* Initialize the UDA1341 internal state */
 297
 298         /* Setup the uarts */
 299         local_irq_save(flags);
 300         GAFR |= (GPIO_SSP_CLK);
 301         GPDR &= ~(GPIO_SSP_CLK);
 302         Ser4SSCR0 = 0;
 303         Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
 304         Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
 305         Ser4SSCR0 |= SSCR0_SSE;
 306         local_irq_restore(flags);
 307
 308         /* Enable the audio power */
 309 #ifdef CONFIG_H3600_HAL
 310         h3600_audio_power(AUDIO_RATE_DEFAULT);
 311 #else
 312         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
 313         set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
 314         set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 315 #endif
 316
 317         /* Wait for the UDA1341 to wake up */
 318         mdelay(1); //FIXME - was removed by Perex - Why?
 319
 320         /* Initialize the UDA1341 internal state */
 321         l3_open(sa11xx_uda1341->uda1341);
 322
 323         /* external clock configuration (after l3_open - regs must be initialized */
 324         sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
 325
 326         /* Wait for the UDA1341 to wake up */
 327         set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
 328         mdelay(1);
 329
 330         /* make the left and right channels unswapped (flip the WS latch) */
 331         Ser4SSDR = 0;
 332
 333 #ifdef CONFIG_H3600_HAL
 334         h3600_audio_mute(0);
 335 #else
 336         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 337 #endif
 338 }
 339
 340 static void sa11xx_uda1341_audio_shutdown(sa11xx_uda1341_t *sa11xx_uda1341)
 341 {
 342         /* mute on */
 343 #ifdef CONFIG_H3600_HAL
 344         h3600_audio_mute(1);
 345 #else
 346         set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 347 #endif
 348
 349         /* disable the audio power and all signals leading to the audio chip */
 350         l3_close(sa11xx_uda1341->uda1341);
 351         Ser4SSCR0 = 0;
 352         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
 353
 354         /* power off and mute off */
 355         /* FIXME - is muting off necesary??? */
 356 #ifdef CONFIG_H3600_HAL
 357         h3600_audio_power(0);
 358         h3600_audio_mute(0);
 359 #else
 360         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
 361         clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
 362 #endif
 363 }
 364
 365 /* }}} */
 366
 367 /* {{{ DMA staff */
 368
 369 /*
 370  * these are the address and sizes used to fill the xmit buffer
 371  * so we can get a clock in record only mode
 372  */
 373 #define FORCE_CLOCK_ADDR                (dma_addr_t)FLUSH_BASE_PHYS
 374 #define FORCE_CLOCK_SIZE                4096 // was 2048
 375
 376 // FIXME Why this value exactly - wrote comment
 377 #define DMA_BUF_SIZE    8176    /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
 378
 379 #ifdef HH_VERSION
 380
 381 static int audio_dma_request(audio_stream_t *s, void (*callback)(void *, int))
 382 {
 383         int ret;
 384
 385         ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
 386         if (ret < 0) {
 387                 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
 388                 return ret;
 389         }
 390         sa1100_dma_set_callback(s->dmach, callback);
 391         return 0;
 392 }
 393
 394 static inline void audio_dma_free(audio_stream_t *s)
 395 {
 396         sa1100_free_dma(s->dmach);
 397         s->dmach = -1;
 398 }
 399
 400 #else
 401
 402 static int audio_dma_request(audio_stream_t *s, void (*callback)(void *))
 403 {
 404         int ret;
 405
 406         ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
 407         if (ret < 0)
 408                 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
 409         return ret;
 410 }
 411
 412 static void audio_dma_free(audio_stream_t *s)
 413 {
 414         sa1100_free_dma((s)->dma_regs);
 415         (s)->dma_regs = 0;
 416 }
 417
 418 #endif
 419
 420 static u_int audio_get_dma_pos(audio_stream_t *s)
 421 {
 422         snd_pcm_substream_t * substream = s->stream;
 423         snd_pcm_runtime_t *runtime = substream->runtime;
 424         unsigned int offset;
 425         unsigned long flags;
 426         dma_addr_t addr;
 427
 428         // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
 429         spin_lock_irqsave(&s->dma_lock, flags);
 430 #ifdef HH_VERSION
 431         sa1100_dma_get_current(s->dmach, NULL, &addr);
 432 #else
 433         addr = sa1100_get_dma_pos((s)->dma_regs);
 434 #endif
 435         offset = addr - runtime->dma_addr;
 436         spin_unlock_irqrestore(&s->dma_lock, flags);
 437
 438         offset = bytes_to_frames(runtime,offset);
 439         if (offset >= runtime->buffer_size)
 440                 offset = 0;
 441
 442         return offset;
 443 }
 444
 445 /*
 446  * this stops the dma and clears the dma ptrs
 447  */
 448 static void audio_stop_dma(audio_stream_t *s)
 449 {
 450         unsigned long flags;
 451
 452         spin_lock_irqsave(&s->dma_lock, flags);
 453         s->active = 0;
 454         s->period = 0;
 455         /* this stops the dma channel and clears the buffer ptrs */
 456 #ifdef HH_VERSION
 457         sa1100_dma_flush_all(s->dmach);
 458 #else
 459         sa1100_clear_dma(s->dma_regs);
 460 #endif
 461         spin_unlock_irqrestore(&s->dma_lock, flags);
 462 }
 463
 464 static void audio_process_dma(audio_stream_t *s)
 465 {
 466         snd_pcm_substream_t *substream = s->stream;
 467         snd_pcm_runtime_t *runtime;
 468         unsigned int dma_size;
 469         unsigned int offset;
 470         int ret;
 471
 472         /* we are requested to process synchronization DMA transfer */
 473         if (s->tx_spin) {
 474                 snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
 475                 /* fill the xmit dma buffers and return */
 476 #ifdef HH_VERSION
 477                 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
 478 #else
 479                 while (1) {
 480                         ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
 481                         if (ret)
 482                                 return;
 483                 }
 484 #endif
 485                 return;
 486         }
 487
 488         /* must be set here - only valid for running streams, not for forced_clock dma fills  */
 489         runtime = substream->runtime;
 490         while (s->active && s->periods < runtime->periods) {
 491                 dma_size = frames_to_bytes(runtime, runtime->period_size);
 492                 if (s->old_offset) {
 493                         /* a little trick, we need resume from old position */
 494                         offset = frames_to_bytes(runtime, s->old_offset - 1);
 495                         s->old_offset = 0;
 496                         s->periods = 0;
 497                         s->period = offset / dma_size;
 498                         offset %= dma_size;
 499                         dma_size = dma_size - offset;
 500                         if (!dma_size)
 501                                 continue;               /* special case */
 502                 } else {
 503                         offset = dma_size * s->period;
 504                         snd_assert(dma_size <= DMA_BUF_SIZE, );
 505                 }
 506 #ifdef HH_VERSION
 507                 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
 508                 if (ret)
 509                         return; //FIXME
 510 #else
 511                 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
 512                 if (ret) {
 513                         printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
 514                         return;
 515                 }
 516 #endif
 517
 518                 s->period++;
 519                 s->period %= runtime->periods;
 520                 s->periods++;
 521         }
 522 }
 523
 524 #ifdef HH_VERSION
 525 static void audio_dma_callback(void *data, int size)
 526 #else
 527 static void audio_dma_callback(void *data)
 528 #endif
 529 {
 530         audio_stream_t *s = data;
 531
 532         /*
 533          * If we are getting a callback for an active stream then we inform
 534          * the PCM middle layer we've finished a period
 535          */
 536         if (s->active)
 537                 snd_pcm_period_elapsed(s->stream);
 538
 539         spin_lock(&s->dma_lock);
 540         if (!s->tx_spin && s->periods > 0)
 541                 s->periods--;
 542         audio_process_dma(s);
 543         spin_unlock(&s->dma_lock);
 544 }
 545
 546 /* }}} */
 547
 548 /* {{{ PCM setting */
 549
 550 /* {{{ trigger & timer */
 551
 552 static int snd_sa11xx_uda1341_trigger(snd_pcm_substream_t * substream, int cmd)
 553 {
 554         sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
 555         int stream_id = substream->pstr->stream;
 556         audio_stream_t *s = &chip->s[stream_id];
 557         audio_stream_t *s1 = &chip->s[stream_id ^ 1];
 558         int err = 0;
 559
 560         /* note local interrupts are already disabled in the midlevel code */
 561         spin_lock(&s->dma_lock);
 562         switch (cmd) {
 563         case SNDRV_PCM_TRIGGER_START:
 564                 /* now we need to make sure a record only stream has a clock */
 565                 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
 566                         /* we need to force fill the xmit DMA with zeros */
 567                         s1->tx_spin = 1;
 568                         audio_process_dma(s1);
 569                 }
 570                 /* this case is when you were recording then you turn on a
 571                  * playback stream so we stop (also clears it) the dma first,
 572                  * clear the sync flag and then we let it turned on
 573                  */
 574                 else {
 575                         s->tx_spin = 0;
 576                 }
 577
 578                 /* requested stream startup */
 579                 s->active = 1;
 580                 audio_process_dma(s);
 581                 break;
 582         case SNDRV_PCM_TRIGGER_STOP:
 583                 /* requested stream shutdown */
 584                 audio_stop_dma(s);
 585
 586                 /*
 587                  * now we need to make sure a record only stream has a clock
 588                  * so if we're stopping a playback with an active capture
 589                  * we need to turn the 0 fill dma on for the xmit side
 590                  */
 591                 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
 592                         /* we need to force fill the xmit DMA with zeros */
 593                         s->tx_spin = 1;
 594                         audio_process_dma(s);
 595                 }
 596                 /*
 597                  * we killed a capture only stream, so we should also kill
 598                  * the zero fill transmit
 599                  */
 600                 else {
 601                         if (s1->tx_spin) {
 602                                 s1->tx_spin = 0;
 603                                 audio_stop_dma(s1);
 604                         }
 605                 }
 606
 607                 break;
 608         case SNDRV_PCM_TRIGGER_SUSPEND:
 609                 s->active = 0;
 610 #ifdef HH_VERSION
 611                 sa1100_dma_stop(s->dmach);
 612 #else
 613                 //FIXME - DMA API
 614 #endif
 615                 s->old_offset = audio_get_dma_pos(s) + 1;
 616 #ifdef HH_VERSION
 617                 sa1100_dma_flush_all(s->dmach);
 618 #else
 619                 //FIXME - DMA API
 620 #endif
 621                 s->periods = 0;
 622                 break;
 623         case SNDRV_PCM_TRIGGER_RESUME:
 624                 s->active = 1;
 625                 s->tx_spin = 0;
 626                 audio_process_dma(s);
 627                 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
 628                         s1->tx_spin = 1;
 629                         audio_process_dma(s1);
 630                 }
 631                 break;
 632         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 633 #ifdef HH_VERSION
 634                 sa1100_dma_stop(s->dmach);
 635 #else
 636                 //FIXME - DMA API
 637 #endif
 638                 s->active = 0;
 639                 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
 640                         if (s1->active) {
 641                                 s->tx_spin = 1;
 642                                 s->old_offset = audio_get_dma_pos(s) + 1;
 643 #ifdef HH_VERSION
 644                                 sa1100_dma_flush_all(s->dmach);
 645 #else
 646                                 //FIXME - DMA API
 647 #endif
 648                                 audio_process_dma(s);
 649                         }
 650                 } else {
 651                         if (s1->tx_spin) {
 652                                 s1->tx_spin = 0;
 653 #ifdef HH_VERSION
 654                                 sa1100_dma_flush_all(s1->dmach);
 655 #else
 656                                 //FIXME - DMA API
 657 #endif
 658                         }
 659                 }
 660                 break;
 661         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 662                 s->active = 1;
 663                 if (s->old_offset) {
 664                         s->tx_spin = 0;
 665                         audio_process_dma(s);
 666                         break;
 667                 }
 668                 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
 669                         s1->tx_spin = 1;
 670                         audio_process_dma(s1);
 671                 }
 672 #ifdef HH_VERSION
 673                 sa1100_dma_resume(s->dmach);
 674 #else
 675                 //FIXME - DMA API
 676 #endif
 677                 break;
 678         default:
 679                 err = -EINVAL;
 680                 break;
 681         }
 682         spin_unlock(&s->dma_lock);
 683         return err;
 684 }
 685
 686 static int snd_sa11xx_uda1341_prepare(snd_pcm_substream_t * substream)
 687 {
 688         sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
 689         snd_pcm_runtime_t *runtime = substream->runtime;
 690         audio_stream_t *s = &chip->s[substream->pstr->stream];
 691
 692         /* set requested samplerate */
 693         sa11xx_uda1341_set_samplerate(chip, runtime->rate);
 694
 695         /* set requestd format when available */
 696         /* set FMT here !!! FIXME */
 697
 698         s->period = 0;
 699         s->periods = 0;
 700
 701         return 0;
 702 }
 703
 704 static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(snd_pcm_substream_t * substream)
 705 {
 706         sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
 707         return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
 708 }
 709
 710 /* }}} */
 711
 712 static snd_pcm_hardware_t snd_sa11xx_uda1341_capture =
 713 {
 714         .info                   = (SNDRV_PCM_INFO_INTERLEAVED |
 715                                    SNDRV_PCM_INFO_BLOCK_TRANSFER |
 716                                    SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 717                                    SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
 718         .formats                = SNDRV_PCM_FMTBIT_S16_LE,
 719         .rates                  = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
 720                                    SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
 721                                    SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
 722                                    SNDRV_PCM_RATE_KNOT),
 723         .rate_min               = 8000,
 724         .rate_max               = 48000,
 725         .channels_min           = 2,
 726         .channels_max           = 2,
 727         .buffer_bytes_max       = 64*1024,
 728         .period_bytes_min       = 64,
 729         .period_bytes_max       = DMA_BUF_SIZE,
 730         .periods_min            = 2,
 731         .periods_max            = 255,
 732         .fifo_size              = 0,
 733 };
 734
 735 static snd_pcm_hardware_t snd_sa11xx_uda1341_playback =
 736 {
 737         .info                   = (SNDRV_PCM_INFO_INTERLEAVED |
 738                                    SNDRV_PCM_INFO_BLOCK_TRANSFER |
 739                                    SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 740                                    SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
 741         .formats                = SNDRV_PCM_FMTBIT_S16_LE,
 742         .rates                  = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
 743                                    SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
 744                                    SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
 745                                    SNDRV_PCM_RATE_KNOT),
 746         .rate_min               = 8000,
 747         .rate_max               = 48000,
 748         .channels_min           = 2,
 749         .channels_max           = 2,
 750         .buffer_bytes_max       = 64*1024,
 751         .period_bytes_min       = 64,
 752         .period_bytes_max       = DMA_BUF_SIZE,
 753         .periods_min            = 2,
 754         .periods_max            = 255,
 755         .fifo_size              = 0,
 756 };
 757
 758 static int snd_card_sa11xx_uda1341_open(snd_pcm_substream_t * substream)
 759 {
 760         sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
 761         snd_pcm_runtime_t *runtime = substream->runtime;
 762         int stream_id = substream->pstr->stream;
 763         int err;
 764
 765         chip->s[stream_id].stream = substream;
 766
 767         if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
 768                 runtime->hw = snd_sa11xx_uda1341_playback;
 769         else
 770                 runtime->hw = snd_sa11xx_uda1341_capture;
 771         if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
 772                 return err;
 773         if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
 774                 return err;
 775
 776         return 0;
 777 }
 778
 779 static int snd_card_sa11xx_uda1341_close(snd_pcm_substream_t * substream)
 780 {
 781         sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
 782
 783         chip->s[substream->pstr->stream].stream = NULL;
 784         return 0;
 785 }
 786
 787 /* {{{ HW params & free */
 788
 789 static int snd_sa11xx_uda1341_hw_params(snd_pcm_substream_t * substream,
 790                                         snd_pcm_hw_params_t * hw_params)
 791 {
 792
 793         return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
 794 }
 795
 796 static int snd_sa11xx_uda1341_hw_free(snd_pcm_substream_t * substream)
 797 {
 798         return snd_pcm_lib_free_pages(substream);
 799 }
 800
 801 /* }}} */
 802
 803 static snd_pcm_ops_t snd_card_sa11xx_uda1341_playback_ops = {
 804         .open                   = snd_card_sa11xx_uda1341_open,
 805         .close                  = snd_card_sa11xx_uda1341_close,
 806         .ioctl                  = snd_pcm_lib_ioctl,
 807         .hw_params              = snd_sa11xx_uda1341_hw_params,
 808         .hw_free                = snd_sa11xx_uda1341_hw_free,
 809         .prepare                = snd_sa11xx_uda1341_prepare,
 810         .trigger                = snd_sa11xx_uda1341_trigger,
 811         .pointer                = snd_sa11xx_uda1341_pointer,
 812 };
 813
 814 static snd_pcm_ops_t snd_card_sa11xx_uda1341_capture_ops = {
 815         .open                   = snd_card_sa11xx_uda1341_open,
 816         .close                  = snd_card_sa11xx_uda1341_close,
 817         .ioctl                  = snd_pcm_lib_ioctl,
 818         .hw_params              = snd_sa11xx_uda1341_hw_params,
 819         .hw_free                = snd_sa11xx_uda1341_hw_free,
 820         .prepare                = snd_sa11xx_uda1341_prepare,
 821         .trigger                = snd_sa11xx_uda1341_trigger,
 822         .pointer                = snd_sa11xx_uda1341_pointer,
 823 };
 824
 825 static int __init snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341_t *sa11xx_uda1341, int device)
 826 {
 827         snd_pcm_t *pcm;
 828         int err;
 829
 830         if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
 831                 return err;
 832
 833         /*
 834          * this sets up our initial buffers and sets the dma_type to isa.
 835          * isa works but I'm not sure why (or if) it's the right choice
 836          * this may be too large, trying it for now
 837          */
 838         snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_ISA,
 839                                               snd_pcm_dma_flags(0),
 840                                               64*1024, 64*1024);
 841
 842         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
 843         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
 844         pcm->private_data = sa11xx_uda1341;
 845         pcm->info_flags = 0;
 846         strcpy(pcm->name, "UDA1341 PCM");
 847
 848         sa11xx_uda1341_audio_init(sa11xx_uda1341);
 849
 850         /* setup DMA controller */
 851         audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
 852         audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
 853
 854         sa11xx_uda1341->pcm = pcm;
 855
 856         return 0;
 857 }
 858
 859 /* }}} */
 860
 861 /* {{{ module init & exit */
 862
 863 #ifdef CONFIG_PM
 864
 865 static int snd_sa11xx_uda1341_suspend(snd_card_t *card, unsigned int state)
 866 {
 867         sa11xx_uda1341_t *chip = card->pm_private_data;
 868
 869         snd_pcm_suspend_all(chip->pcm);
 870 #ifdef HH_VERSION
 871         sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
 872         sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
 873 #else
 874         //FIXME
 875 #endif
 876         l3_command(chip->uda1341, CMD_SUSPEND, NULL);
 877         sa11xx_uda1341_audio_shutdown(chip);
 878         snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 879         return 0;
 880 }
 881
 882 static int snd_sa11xx_uda1341_resume(snd_card_t *card, unsigned int state)
 883 {
 884         sa11xx_uda1341_t *chip = card->pm_private_data;
 885
 886         sa11xx_uda1341_audio_init(chip);
 887         l3_command(chip->uda1341, CMD_RESUME, NULL);
 888 #ifdef HH_VERSION
 889         sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
 890         sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
 891 #else
 892         //FIXME
 893 #endif
 894         snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 895         return 0;
 896 }
 897 #endif /* COMFIG_PM */
 898
 899 void snd_sa11xx_uda1341_free(snd_card_t *card)
 900 {
 901         sa11xx_uda1341_t *chip = card->private_data;
 902
 903         audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
 904         audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
 905         sa11xx_uda1341 = NULL;
 906         card->private_data = NULL;
 907         kfree(chip);
 908 }
 909
 910 static int __init sa11xx_uda1341_init(void)
 911 {
 912         int err;
 913         snd_card_t *card;
 914
 915         if (!machine_is_h3xxx())
 916                 return -ENODEV;
 917
 918         /* register the soundcard */
 919         card = snd_card_new(-1, id, THIS_MODULE, sizeof(sa11xx_uda1341_t));
 920         if (card == NULL)
 921                 return -ENOMEM;
 922
 923         sa11xx_uda1341 = kcalloc(1, sizeof(*sa11xx_uda1341), GFP_KERNEL);
 924         if (sa11xx_uda1341 == NULL)
 925                 return -ENOMEM;
 926         spin_lock_init(&chip->s[0].dma_lock);
 927         spin_lock_init(&chip->s[1].dma_lock);
 928
 929         card->private_data = (void *)sa11xx_uda1341;
 930         card->private_free = snd_sa11xx_uda1341_free;
 931
 932         sa11xx_uda1341->card = card;
 933         sa11xx_uda1341->samplerate = AUDIO_RATE_DEFAULT;
 934
 935         // mixer
 936         if ((err = snd_chip_uda1341_mixer_new(sa11xx_uda1341->card, &sa11xx_uda1341->uda1341)))
 937                 goto nodev;
 938
 939         // PCM
 940         if ((err = snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341, 0)) < 0)
 941                 goto nodev;
 942
 943         snd_card_set_dev_pm_callback(card, PM_SYS_DEV,
 944                                      snd_sa11xx_uda1341_suspend, snd_sa11_uda1341_resume,
 945                                      sa11xx_uda1341);
 946
 947         strcpy(card->driver, "UDA1341");
 948         strcpy(card->shortname, "H3600 UDA1341TS");
 949         sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
 950
 951         if ((err = snd_card_register(card)) == 0) {
 952                 printk( KERN_INFO "iPAQ audio support initialized\n" );
 953                 return 0;
 954         }
 955
 956  nodev:
 957         snd_card_free(card);
 958         return err;
 959 }
 960
 961 static void __exit sa11xx_uda1341_exit(void)
 962 {
 963         snd_card_free(sa11xx_uda1341->card);
 964 }
 965
 966 module_init(sa11xx_uda1341_init);
 967 module_exit(sa11xx_uda1341_exit);
 968
 969 /* }}} */
 970
 971 /*
 972  * Local variables:
 973  * indent-tabs-mode: t
 974  * End:
 975  */