sound/arm/omap/omap-alsa.c

   1 /*
   2  * sound/arm/omap-alsa.c
   3  *
   4  * Alsa Driver for OMAP
   5  *
   6  * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
   7  * Written by Daniel Petrini, David Cohen, Anderson Briglia
   8  *            {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
   9  *
  10  * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
  11  *
  12  * Based on sa11xx-uda1341.c,
  13  * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
  14  *
  15  * This program is free software; you can redistribute it and/or modify it
  16  * under the terms of the GNU General Public License as published by the
  17  * Free Software Foundation; either version 2 of the License, or (at your
  18  * option) any later version.
  19  *
  20  * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
  21  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  22  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
  23  * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
  24  * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
  25  * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
  26  * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
  27  * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
  28  * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
  29  * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  30  *
  31  * You should have received a copy of the  GNU General Public License along
  32  * with this program; if not, write  to the Free Software Foundation, Inc.,
  33  * 675 Mass Ave, Cambridge, MA 02139, USA.
  34  *
  35  * History:
  36  *
  37  * 2005-07-29   INdT Kernel Team - Alsa driver for omap osk. Creation of new
  38  *                                 file omap-aic23.c
  39  *
  40  * 2005-12-18   Dirk Behme       - Added L/R Channel Interchange fix as proposed
  41  *                                 by Ajaya Babu
  42  *
  43  */
  44
  45 #include <linux/platform_device.h>
  46 #ifdef CONFIG_PM
  47 #include <linux/pm.h>
  48 #endif
  49 #include <sound/driver.h>
  50 #include <sound/core.h>
  51
  52 #include <asm/arch/omap-alsa.h>
  53 #include "omap-alsa-dma.h"
  54
  55 MODULE_AUTHOR("Mika Laitio, Daniel Petrini, David Cohen, Anderson Briglia - INdT");
  56 MODULE_LICENSE("GPL");
  57 MODULE_DESCRIPTION("OMAP driver for ALSA");
  58 MODULE_ALIAS("omap_alsa_mcbsp.1");
  59
  60 static char *id = NULL;
  61 static struct snd_card_omap_codec       *alsa_codec             = NULL;
  62 static struct omap_alsa_codec_config    *alsa_codec_config      = NULL;
  63
  64 /*
  65  * HW interface start and stop helper functions
  66  */
  67 static int audio_ifc_start(void)
  68 {
  69         omap_mcbsp_start(AUDIO_MCBSP);
  70         return 0;
  71 }
  72
  73 static int audio_ifc_stop(void)
  74 {
  75         omap_mcbsp_stop(AUDIO_MCBSP);
  76         return 0;
  77 }
  78
  79 static void omap_alsa_audio_init(struct snd_card_omap_codec *omap_alsa)
  80 {
  81         /* Setup DMA stuff */
  82         omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].id = "Alsa omap out";
  83         omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id =
  84             SNDRV_PCM_STREAM_PLAYBACK;
  85         omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev =
  86             OMAP_DMA_MCBSP1_TX;
  87         omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].hw_start =
  88             audio_ifc_start;
  89         omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK].hw_stop =
  90             audio_ifc_stop;
  91
  92         omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].id = "Alsa omap in";
  93         omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].stream_id =
  94             SNDRV_PCM_STREAM_CAPTURE;
  95         omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev =
  96             OMAP_DMA_MCBSP1_RX;
  97         omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].hw_start =
  98             audio_ifc_start;
  99         omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE].hw_stop =
 100             audio_ifc_stop;
 101 }
 102
 103 /*
 104  * DMA functions
 105  * Depends on omap-alsa-dma.c functions and (omap) dma.c
 106  *
 107  */
 108 static int audio_dma_request(struct audio_stream *s,
 109                              void (*callback) (void *))
 110 {
 111         int err;
 112         ADEBUG();
 113
 114         err = omap_request_alsa_sound_dma(s->dma_dev, s->id, s, &s->lch);
 115         if (err < 0)
 116                 printk(KERN_ERR "Unable to grab audio dma 0x%x\n", s->dma_dev);
 117         return err;
 118 }
 119
 120 static int audio_dma_free(struct audio_stream *s)
 121 {
 122         int err = 0;
 123         ADEBUG();
 124
 125         err = omap_free_alsa_sound_dma(s, &s->lch);
 126         if (err < 0)
 127                 printk(KERN_ERR "Unable to free audio dma channels!\n");
 128         return err;
 129 }
 130
 131 /*
 132  *  This function should calculate the current position of the dma in the
 133  *  buffer. It will help alsa middle layer to continue update the buffer.
 134  *  Its correctness is crucial for good functioning.
 135  */
 136 static u_int audio_get_dma_pos(struct audio_stream *s)
 137 {
 138         snd_pcm_substream_t *substream = s->stream;
 139         snd_pcm_runtime_t *runtime = substream->runtime;
 140         unsigned int offset;
 141         unsigned long flags;
 142         dma_addr_t count;
 143         ADEBUG();
 144
 145         /* this must be called w/ interrupts locked as requested in dma.c */
 146         spin_lock_irqsave(&s->dma_lock, flags);
 147
 148         /* For the current period let's see where we are */
 149         count = omap_get_dma_src_addr_counter(s->lch[s->dma_q_head]);
 150
 151         spin_unlock_irqrestore(&s->dma_lock, flags);
 152
 153         /* Now, the position related to the end of that period */
 154         offset = bytes_to_frames(runtime, s->offset) - bytes_to_frames(runtime, count);
 155
 156         if (offset >= runtime->buffer_size)
 157                 offset = 0;
 158
 159         return offset;
 160 }
 161
 162 /*
 163  * this stops the dma and clears the dma ptrs
 164  */
 165 static void audio_stop_dma(struct audio_stream *s)
 166 {
 167         unsigned long flags;
 168         ADEBUG();
 169
 170         spin_lock_irqsave(&s->dma_lock, flags);
 171         s->active = 0;
 172         s->period = 0;
 173         s->periods = 0;
 174
 175         /* this stops the dma channel and clears the buffer ptrs */
 176         omap_stop_alsa_sound_dma(s);
 177
 178         omap_clear_alsa_sound_dma(s);
 179
 180         spin_unlock_irqrestore(&s->dma_lock, flags);
 181 }
 182
 183 /*
 184  *  Main dma routine, requests dma according where you are in main alsa buffer
 185  */
 186 static void audio_process_dma(struct audio_stream *s)
 187 {
 188         snd_pcm_substream_t *substream = s->stream;
 189         snd_pcm_runtime_t *runtime;
 190         unsigned int dma_size;
 191         unsigned int offset;
 192         int ret;
 193 #ifdef CONFIG_ARCH_OMAP15XX
 194         unsigned long flags;
 195 #endif
 196
 197         ADEBUG();
 198         runtime = substream->runtime;
 199         if (s->active) {
 200                 dma_size = frames_to_bytes(runtime, runtime->period_size);
 201                 offset = dma_size * s->period;
 202                 snd_assert(dma_size <= DMA_BUF_SIZE,);
 203 #ifdef CONFIG_ARCH_OMAP15XX
 204                 /*
 205                  * On omap1510 based devices, we need to call the stop_dma
 206                  * before calling the start_dma or we will not receive the
 207                  * irq from DMA after the first transfered/played buffer.
 208                  * (invocation of callback_omap_alsa_sound_dma() method).
 209                  */
 210                 spin_lock_irqsave(&s->dma_lock, flags);
 211                 omap_stop_alsa_sound_dma(s);
 212                 spin_unlock_irqrestore(&s->dma_lock, flags);
 213 #endif
 214                 ret = omap_start_alsa_sound_dma(s,
 215                                          (dma_addr_t) runtime->dma_area +
 216                                          offset, dma_size);
 217                 if (ret) {
 218                         printk(KERN_ERR
 219                                "audio_process_dma: cannot queue DMA buffer (%i)\n",
 220                                ret);
 221                         return;
 222                 }
 223
 224                 s->period++;
 225                 s->period %= runtime->periods;
 226                 s->periods++;
 227                 s->offset = offset;
 228         }
 229 }
 230
 231 /*
 232  *  This is called when dma IRQ occurs at the end of each transmited block
 233  */
 234 void callback_omap_alsa_sound_dma(void *data)
 235 {
 236         struct audio_stream *s = data;
 237
 238         ADEBUG();
 239         /*
 240          * If we are getting a callback for an active stream then we inform
 241          * the PCM middle layer we've finished a period
 242          */
 243         if (s->active)
 244                 snd_pcm_period_elapsed(s->stream);
 245
 246         spin_lock(&s->dma_lock);
 247         if (s->periods > 0)
 248                 s->periods--;
 249
 250         audio_process_dma(s);
 251         spin_unlock(&s->dma_lock);
 252 }
 253
 254 /*
 255  * Alsa section
 256  * PCM settings and callbacks
 257  */
 258 static int snd_omap_alsa_trigger(snd_pcm_substream_t * substream, int cmd)
 259 {
 260         struct snd_card_omap_codec *chip =
 261             snd_pcm_substream_chip(substream);
 262         int stream_id = substream->pstr->stream;
 263         struct audio_stream *s = &chip->s[stream_id];
 264         int err = 0;
 265
 266         ADEBUG();
 267         /* note local interrupts are already disabled in the midlevel code */
 268         spin_lock(&s->dma_lock);
 269         switch (cmd) {
 270         case SNDRV_PCM_TRIGGER_START:
 271                 /* requested stream startup */
 272                 s->active = 1;
 273                 audio_process_dma(s);
 274                 break;
 275         case SNDRV_PCM_TRIGGER_STOP:
 276                 /* requested stream shutdown */
 277                 audio_stop_dma(s);
 278                 break;
 279         default:
 280                 err = -EINVAL;
 281                 break;
 282         }
 283         spin_unlock(&s->dma_lock);
 284
 285         return err;
 286 }
 287
 288 static int snd_omap_alsa_prepare(snd_pcm_substream_t * substream)
 289 {
 290         struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
 291         snd_pcm_runtime_t *runtime = substream->runtime;
 292         struct audio_stream *s = &chip->s[substream->pstr->stream];
 293
 294         ADEBUG();
 295         /* set requested samplerate */
 296         alsa_codec_config->codec_set_samplerate(runtime->rate);
 297         chip->samplerate = runtime->rate;
 298
 299         s->period = 0;
 300         s->periods = 0;
 301
 302         return 0;
 303 }
 304
 305 static snd_pcm_uframes_t snd_omap_alsa_pointer(snd_pcm_substream_t *substream)
 306 {
 307         struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
 308
 309         ADEBUG();
 310         return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
 311 }
 312
 313 static int snd_card_omap_alsa_open(snd_pcm_substream_t * substream)
 314 {
 315         struct snd_card_omap_codec *chip =
 316             snd_pcm_substream_chip(substream);
 317         snd_pcm_runtime_t *runtime = substream->runtime;
 318         int stream_id = substream->pstr->stream;
 319         int err;
 320
 321         ADEBUG();
 322         chip->s[stream_id].stream = substream;
 323         alsa_codec_config->codec_clock_on();
 324         if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
 325                 runtime->hw = *(alsa_codec_config->snd_omap_alsa_playback);
 326         else
 327                 runtime->hw = *(alsa_codec_config->snd_omap_alsa_capture);
 328
 329         if ((err = snd_pcm_hw_constraint_integer(runtime,
 330                                            SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
 331                 return err;
 332
 333         if ((err = snd_pcm_hw_constraint_list(runtime,
 334                                         0,
 335                                         SNDRV_PCM_HW_PARAM_RATE,
 336                                         alsa_codec_config->hw_constraints_rates)) < 0)
 337                 return err;
 338
 339         return 0;
 340 }
 341
 342 static int snd_card_omap_alsa_close(snd_pcm_substream_t * substream)
 343 {
 344         struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
 345
 346         ADEBUG();
 347         alsa_codec_config->codec_clock_off();
 348         chip->s[substream->pstr->stream].stream = NULL;
 349
 350         return 0;
 351 }
 352
 353 /* HW params & free */
 354 static int snd_omap_alsa_hw_params(snd_pcm_substream_t * substream,
 355                                     snd_pcm_hw_params_t * hw_params)
 356 {
 357         return snd_pcm_lib_malloc_pages(substream,
 358                                         params_buffer_bytes(hw_params));
 359 }
 360
 361 static int snd_omap_alsa_hw_free(snd_pcm_substream_t * substream)
 362 {
 363         return snd_pcm_lib_free_pages(substream);
 364 }
 365
 366 /* pcm operations */
 367 static snd_pcm_ops_t snd_card_omap_alsa_playback_ops = {
 368         .open =         snd_card_omap_alsa_open,
 369         .close =        snd_card_omap_alsa_close,
 370         .ioctl =        snd_pcm_lib_ioctl,
 371         .hw_params =    snd_omap_alsa_hw_params,
 372         .hw_free =      snd_omap_alsa_hw_free,
 373         .prepare =      snd_omap_alsa_prepare,
 374         .trigger =      snd_omap_alsa_trigger,
 375         .pointer =      snd_omap_alsa_pointer,
 376 };
 377
 378 static snd_pcm_ops_t snd_card_omap_alsa_capture_ops = {
 379         .open =         snd_card_omap_alsa_open,
 380         .close =        snd_card_omap_alsa_close,
 381         .ioctl =        snd_pcm_lib_ioctl,
 382         .hw_params =    snd_omap_alsa_hw_params,
 383         .hw_free =      snd_omap_alsa_hw_free,
 384         .prepare =      snd_omap_alsa_prepare,
 385         .trigger =      snd_omap_alsa_trigger,
 386         .pointer =      snd_omap_alsa_pointer,
 387 };
 388
 389 /*
 390  *  Alsa init and exit section
 391  *
 392  *  Inits pcm alsa structures, allocate the alsa buffer, suspend, resume
 393  */
 394 static int __init snd_card_omap_alsa_pcm(struct snd_card_omap_codec *omap_alsa,
 395                                         int device)
 396 {
 397         snd_pcm_t *pcm;
 398         int err;
 399
 400         ADEBUG();
 401         if ((err = snd_pcm_new(omap_alsa->card, "OMAP PCM", device, 1, 1, &pcm)) < 0)
 402                 return err;
 403
 404         /* sets up initial buffer with continuous allocation */
 405         snd_pcm_lib_preallocate_pages_for_all(pcm,
 406                                               SNDRV_DMA_TYPE_CONTINUOUS,
 407                                               snd_dma_continuous_data
 408                                               (GFP_KERNEL),
 409                                               128 * 1024, 128 * 1024);
 410
 411         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
 412                         &snd_card_omap_alsa_playback_ops);
 413         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
 414                         &snd_card_omap_alsa_capture_ops);
 415         pcm->private_data = omap_alsa;
 416         pcm->info_flags = 0;
 417         strcpy(pcm->name, "omap alsa pcm");
 418
 419         omap_alsa_audio_init(omap_alsa);
 420
 421         /* setup DMA controller */
 422         audio_dma_request(&omap_alsa->s[SNDRV_PCM_STREAM_PLAYBACK],
 423                           callback_omap_alsa_sound_dma);
 424         audio_dma_request(&omap_alsa->s[SNDRV_PCM_STREAM_CAPTURE],
 425                           callback_omap_alsa_sound_dma);
 426
 427         omap_alsa->pcm = pcm;
 428
 429         return 0;
 430 }
 431
 432
 433 #ifdef CONFIG_PM
 434 /*
 435  * Driver suspend/resume - calls alsa functions. Some hints from aaci.c
 436  */
 437 int snd_omap_alsa_suspend(struct platform_device *pdev, pm_message_t state)
 438 {
 439         struct snd_card_omap_codec *chip;
 440         snd_card_t *card = platform_get_drvdata(pdev);
 441
 442         if (card->power_state != SNDRV_CTL_POWER_D3hot) {
 443                 chip = card->private_data;
 444                 if (chip->card->power_state != SNDRV_CTL_POWER_D3hot) {
 445                         snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
 446                         snd_pcm_suspend_all(chip->pcm);
 447                         /* Mutes and turn clock off */
 448                         alsa_codec_config->codec_clock_off();
 449                         snd_omap_suspend_mixer();
 450                 }
 451         }
 452         return 0;
 453 }
 454
 455 int snd_omap_alsa_resume(struct platform_device *pdev)
 456 {
 457         struct snd_card_omap_codec *chip;
 458         snd_card_t *card = platform_get_drvdata(pdev);
 459
 460         if (card->power_state != SNDRV_CTL_POWER_D0) {
 461                 chip = card->private_data;
 462                 if (chip->card->power_state != SNDRV_CTL_POWER_D0) {
 463                         snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
 464                         alsa_codec_config->codec_clock_on();
 465                         snd_omap_resume_mixer();
 466                 }
 467         }
 468         return 0;
 469 }
 470
 471 #endif  /* CONFIG_PM */
 472
 473 void snd_omap_alsa_free(snd_card_t * card)
 474 {
 475         struct snd_card_omap_codec *chip = card->private_data;
 476         ADEBUG();
 477
 478         /*
 479          * Turn off codec after it is done.
 480          * Can't do it immediately, since it may still have
 481          * buffered data.
 482          */
 483         schedule_timeout_interruptible(2);
 484
 485         omap_mcbsp_stop(AUDIO_MCBSP);
 486         omap_mcbsp_free(AUDIO_MCBSP);
 487
 488         audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
 489         audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
 490 }
 491
 492 /* module init & exit */
 493
 494 /*
 495  * Inits alsa soudcard structure.
 496  * Called by the probe method in codec after function pointers has been set.
 497  */
 498 int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_codec_config *config)
 499 {
 500         int err = 0;
 501         int def_rate;
 502         snd_card_t *card;
 503
 504         ADEBUG();
 505         alsa_codec_config       = config;
 506
 507         alsa_codec_config->codec_clock_setup();
 508         alsa_codec_config->codec_clock_on();
 509
 510         omap_mcbsp_request(AUDIO_MCBSP);
 511         omap_mcbsp_stop(AUDIO_MCBSP);
 512         omap_mcbsp_config(AUDIO_MCBSP, alsa_codec_config->mcbsp_regs_alsa);
 513         omap_mcbsp_start(AUDIO_MCBSP);
 514
 515         if (alsa_codec_config && alsa_codec_config->codec_configure_dev)
 516                 alsa_codec_config->codec_configure_dev();
 517
 518         alsa_codec_config->codec_clock_off();
 519
 520         /* register the soundcard */
 521         card = snd_card_new(-1, id, THIS_MODULE, sizeof(alsa_codec));
 522         if (card == NULL)
 523                 goto nodev1;
 524
 525         alsa_codec = kcalloc(1, sizeof(*alsa_codec), GFP_KERNEL);
 526         if (alsa_codec == NULL)
 527                 goto nodev2;
 528
 529         card->private_data = (void *)alsa_codec;
 530         card->private_free = snd_omap_alsa_free;
 531
 532         alsa_codec->card        = card;
 533         def_rate                = alsa_codec_config->get_default_samplerate();
 534         alsa_codec->samplerate  = def_rate;
 535
 536         spin_lock_init(&alsa_codec->s[0].dma_lock);
 537         spin_lock_init(&alsa_codec->s[1].dma_lock);
 538
 539         /* mixer */
 540         if ((err = snd_omap_mixer(alsa_codec)) < 0)
 541                 goto nodev3;
 542
 543         /* PCM */
 544         if ((err = snd_card_omap_alsa_pcm(alsa_codec, 0)) < 0)
 545                 goto nodev3;
 546
 547         strcpy(card->driver, "OMAP_ALSA");
 548         strcpy(card->shortname, alsa_codec_config->name);
 549         sprintf(card->longname, alsa_codec_config->name);
 550
 551         snd_omap_init_mixer();
 552         snd_card_set_dev(card, &pdev->dev);
 553
 554         if ((err = snd_card_register(card)) == 0) {
 555                 printk(KERN_INFO "audio support initialized\n");
 556                 platform_set_drvdata(pdev, card);
 557                 return 0;
 558         }
 559
 560 nodev3:
 561         kfree(alsa_codec);
 562 nodev2:
 563         snd_card_free(card);
 564 nodev1:
 565         omap_mcbsp_stop(AUDIO_MCBSP);
 566         omap_mcbsp_free(AUDIO_MCBSP);
 567
 568         return err;
 569 }
 570
 571 int snd_omap_alsa_remove(struct platform_device *pdev)
 572 {
 573         snd_card_t *card = platform_get_drvdata(pdev);
 574         struct snd_card_omap_codec *chip = card->private_data;
 575
 576         snd_card_free(card);
 577
 578         alsa_codec = NULL;
 579         card->private_data = NULL;
 580         kfree(chip);
 581
 582         platform_set_drvdata(pdev, NULL);
 583
 584         return 0;
 585 }