audio/alsaaudio.c

   1 /*
   2  * QEMU ALSA audio driver
   3  *
   4  * Copyright (c) 2005 Vassili Karpov (malc)
   5  *
   6  * Permission is hereby granted, free of charge, to any person obtaining a copy
   7  * of this software and associated documentation files (the "Software"), to deal
   8  * in the Software without restriction, including without limitation the rights
   9  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  10  * copies of the Software, and to permit persons to whom the Software is
  11  * furnished to do so, subject to the following conditions:
  12  *
  13  * The above copyright notice and this permission notice shall be included in
  14  * all copies or substantial portions of the Software.
  15  *
  16  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  17  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  18  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
  19  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  20  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  21  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  22  * THE SOFTWARE.
  23  */
  24 #include <alsa/asoundlib.h>
  25 #include "qemu-common.h"
  26 #include "qemu/main-loop.h"
  27 #include "audio.h"
  28
  29 #if QEMU_GNUC_PREREQ(4, 3)
  30 #pragma GCC diagnostic ignored "-Waddress"
  31 #endif
  32
  33 #define AUDIO_CAP "alsa"
  34 #include "audio_int.h"
  35
  36 struct pollhlp {
  37     snd_pcm_t *handle;
  38     struct pollfd *pfds;
  39     int count;
  40     int mask;
  41 };
  42
  43 typedef struct ALSAVoiceOut {
  44     HWVoiceOut hw;
  45     int wpos;
  46     int pending;
  47     void *pcm_buf;
  48     snd_pcm_t *handle;
  49     struct pollhlp pollhlp;
  50 } ALSAVoiceOut;
  51
  52 typedef struct ALSAVoiceIn {
  53     HWVoiceIn hw;
  54     snd_pcm_t *handle;
  55     void *pcm_buf;
  56     struct pollhlp pollhlp;
  57 } ALSAVoiceIn;
  58
  59 static struct {
  60     int size_in_usec_in;
  61     int size_in_usec_out;
  62     const char *pcm_name_in;
  63     const char *pcm_name_out;
  64     unsigned int buffer_size_in;
  65     unsigned int period_size_in;
  66     unsigned int buffer_size_out;
  67     unsigned int period_size_out;
  68     unsigned int threshold;
  69
  70     int buffer_size_in_overridden;
  71     int period_size_in_overridden;
  72
  73     int buffer_size_out_overridden;
  74     int period_size_out_overridden;
  75     int verbose;
  76 } conf = {
  77     .buffer_size_out = 4096,
  78     .period_size_out = 1024,
  79     .pcm_name_out = "default",
  80     .pcm_name_in = "default",
  81 };
  82
  83 struct alsa_params_req {
  84     int freq;
  85     snd_pcm_format_t fmt;
  86     int nchannels;
  87     int size_in_usec;
  88     int override_mask;
  89     unsigned int buffer_size;
  90     unsigned int period_size;
  91 };
  92
  93 struct alsa_params_obt {
  94     int freq;
  95     audfmt_e fmt;
  96     int endianness;
  97     int nchannels;
  98     snd_pcm_uframes_t samples;
  99 };
 100
 101 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
 102 {
 103     va_list ap;
 104
 105     va_start (ap, fmt);
 106     AUD_vlog (AUDIO_CAP, fmt, ap);
 107     va_end (ap);
 108
 109     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 110 }
 111
 112 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
 113     int err,
 114     const char *typ,
 115     const char *fmt,
 116     ...
 117     )
 118 {
 119     va_list ap;
 120
 121     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
 122
 123     va_start (ap, fmt);
 124     AUD_vlog (AUDIO_CAP, fmt, ap);
 125     va_end (ap);
 126
 127     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 128 }
 129
 130 static void alsa_fini_poll (struct pollhlp *hlp)
 131 {
 132     int i;
 133     struct pollfd *pfds = hlp->pfds;
 134
 135     if (pfds) {
 136         for (i = 0; i < hlp->count; ++i) {
 137             qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 138         }
 139         g_free (pfds);
 140     }
 141     hlp->pfds = NULL;
 142     hlp->count = 0;
 143     hlp->handle = NULL;
 144 }
 145
 146 static void alsa_anal_close1 (snd_pcm_t **handlep)
 147 {
 148     int err = snd_pcm_close (*handlep);
 149     if (err) {
 150         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 151     }
 152     *handlep = NULL;
 153 }
 154
 155 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
 156 {
 157     alsa_fini_poll (hlp);
 158     alsa_anal_close1 (handlep);
 159 }
 160
 161 static int alsa_recover (snd_pcm_t *handle)
 162 {
 163     int err = snd_pcm_prepare (handle);
 164     if (err < 0) {
 165         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 166         return -1;
 167     }
 168     return 0;
 169 }
 170
 171 static int alsa_resume (snd_pcm_t *handle)
 172 {
 173     int err = snd_pcm_resume (handle);
 174     if (err < 0) {
 175         alsa_logerr (err, "Failed to resume handle %p\n", handle);
 176         return -1;
 177     }
 178     return 0;
 179 }
 180
 181 static void alsa_poll_handler (void *opaque)
 182 {
 183     int err, count;
 184     snd_pcm_state_t state;
 185     struct pollhlp *hlp = opaque;
 186     unsigned short revents;
 187
 188     count = poll (hlp->pfds, hlp->count, 0);
 189     if (count < 0) {
 190         dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
 191         return;
 192     }
 193
 194     if (!count) {
 195         return;
 196     }
 197
 198     /* XXX: ALSA example uses initial count, not the one returned by
 199        poll, correct? */
 200     err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
 201                                             hlp->count, &revents);
 202     if (err < 0) {
 203         alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
 204         return;
 205     }
 206
 207     if (!(revents & hlp->mask)) {
 208         if (conf.verbose) {
 209             dolog ("revents = %d\n", revents);
 210         }
 211         return;
 212     }
 213
 214     state = snd_pcm_state (hlp->handle);
 215     switch (state) {
 216     case SND_PCM_STATE_SETUP:
 217         alsa_recover (hlp->handle);
 218         break;
 219
 220     case SND_PCM_STATE_XRUN:
 221         alsa_recover (hlp->handle);
 222         break;
 223
 224     case SND_PCM_STATE_SUSPENDED:
 225         alsa_resume (hlp->handle);
 226         break;
 227
 228     case SND_PCM_STATE_PREPARED:
 229         audio_run ("alsa run (prepared)");
 230         break;
 231
 232     case SND_PCM_STATE_RUNNING:
 233         audio_run ("alsa run (running)");
 234         break;
 235
 236     default:
 237         dolog ("Unexpected state %d\n", state);
 238     }
 239 }
 240
 241 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
 242 {
 243     int i, count, err;
 244     struct pollfd *pfds;
 245
 246     count = snd_pcm_poll_descriptors_count (handle);
 247     if (count <= 0) {
 248         dolog ("Could not initialize poll mode\n"
 249                "Invalid number of poll descriptors %d\n", count);
 250         return -1;
 251     }
 252
 253     pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
 254     if (!pfds) {
 255         dolog ("Could not initialize poll mode\n");
 256         return -1;
 257     }
 258
 259     err = snd_pcm_poll_descriptors (handle, pfds, count);
 260     if (err < 0) {
 261         alsa_logerr (err, "Could not initialize poll mode\n"
 262                      "Could not obtain poll descriptors\n");
 263         g_free (pfds);
 264         return -1;
 265     }
 266
 267     for (i = 0; i < count; ++i) {
 268         if (pfds[i].events & POLLIN) {
 269             err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler,
 270                                        NULL, hlp);
 271         }
 272         if (pfds[i].events & POLLOUT) {
 273             if (conf.verbose) {
 274                 dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
 275             }
 276             err = qemu_set_fd_handler (pfds[i].fd, NULL,
 277                                        alsa_poll_handler, hlp);
 278         }
 279         if (conf.verbose) {
 280             dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
 281                    pfds[i].events, i, pfds[i].fd, err);
 282         }
 283
 284         if (err) {
 285             dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n",
 286                    pfds[i].events, i, pfds[i].fd, err);
 287
 288             while (i--) {
 289                 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 290             }
 291             g_free (pfds);
 292             return -1;
 293         }
 294     }
 295     hlp->pfds = pfds;
 296     hlp->count = count;
 297     hlp->handle = handle;
 298     hlp->mask = mask;
 299     return 0;
 300 }
 301
 302 static int alsa_poll_out (HWVoiceOut *hw)
 303 {
 304     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 305
 306     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
 307 }
 308
 309 static int alsa_poll_in (HWVoiceIn *hw)
 310 {
 311     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 312
 313     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
 314 }
 315
 316 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 317 {
 318     return audio_pcm_sw_write (sw, buf, len);
 319 }
 320
 321 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
 322 {
 323     switch (fmt) {
 324     case AUD_FMT_S8:
 325         return SND_PCM_FORMAT_S8;
 326
 327     case AUD_FMT_U8:
 328         return SND_PCM_FORMAT_U8;
 329
 330     case AUD_FMT_S16:
 331         if (endianness) {
 332             return SND_PCM_FORMAT_S16_BE;
 333         }
 334         else {
 335             return SND_PCM_FORMAT_S16_LE;
 336         }
 337
 338     case AUD_FMT_U16:
 339         if (endianness) {
 340             return SND_PCM_FORMAT_U16_BE;
 341         }
 342         else {
 343             return SND_PCM_FORMAT_U16_LE;
 344         }
 345
 346     case AUD_FMT_S32:
 347         if (endianness) {
 348             return SND_PCM_FORMAT_S32_BE;
 349         }
 350         else {
 351             return SND_PCM_FORMAT_S32_LE;
 352         }
 353
 354     case AUD_FMT_U32:
 355         if (endianness) {
 356             return SND_PCM_FORMAT_U32_BE;
 357         }
 358         else {
 359             return SND_PCM_FORMAT_U32_LE;
 360         }
 361
 362     default:
 363         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 364 #ifdef DEBUG_AUDIO
 365         abort ();
 366 #endif
 367         return SND_PCM_FORMAT_U8;
 368     }
 369 }
 370
 371 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
 372                            int *endianness)
 373 {
 374     switch (alsafmt) {
 375     case SND_PCM_FORMAT_S8:
 376         *endianness = 0;
 377         *fmt = AUD_FMT_S8;
 378         break;
 379
 380     case SND_PCM_FORMAT_U8:
 381         *endianness = 0;
 382         *fmt = AUD_FMT_U8;
 383         break;
 384
 385     case SND_PCM_FORMAT_S16_LE:
 386         *endianness = 0;
 387         *fmt = AUD_FMT_S16;
 388         break;
 389
 390     case SND_PCM_FORMAT_U16_LE:
 391         *endianness = 0;
 392         *fmt = AUD_FMT_U16;
 393         break;
 394
 395     case SND_PCM_FORMAT_S16_BE:
 396         *endianness = 1;
 397         *fmt = AUD_FMT_S16;
 398         break;
 399
 400     case SND_PCM_FORMAT_U16_BE:
 401         *endianness = 1;
 402         *fmt = AUD_FMT_U16;
 403         break;
 404
 405     case SND_PCM_FORMAT_S32_LE:
 406         *endianness = 0;
 407         *fmt = AUD_FMT_S32;
 408         break;
 409
 410     case SND_PCM_FORMAT_U32_LE:
 411         *endianness = 0;
 412         *fmt = AUD_FMT_U32;
 413         break;
 414
 415     case SND_PCM_FORMAT_S32_BE:
 416         *endianness = 1;
 417         *fmt = AUD_FMT_S32;
 418         break;
 419
 420     case SND_PCM_FORMAT_U32_BE:
 421         *endianness = 1;
 422         *fmt = AUD_FMT_U32;
 423         break;
 424
 425     default:
 426         dolog ("Unrecognized audio format %d\n", alsafmt);
 427         return -1;
 428     }
 429
 430     return 0;
 431 }
 432
 433 static void alsa_dump_info (struct alsa_params_req *req,
 434                             struct alsa_params_obt *obt,
 435                             snd_pcm_format_t obtfmt)
 436 {
 437     dolog ("parameter | requested value | obtained value\n");
 438     dolog ("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 439     dolog ("channels  |      %10d |     %10d\n",
 440            req->nchannels, obt->nchannels);
 441     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 442     dolog ("============================================\n");
 443     dolog ("requested: buffer size %d period size %d\n",
 444            req->buffer_size, req->period_size);
 445     dolog ("obtained: samples %ld\n", obt->samples);
 446 }
 447
 448 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 449 {
 450     int err;
 451     snd_pcm_sw_params_t *sw_params;
 452
 453     snd_pcm_sw_params_alloca (&sw_params);
 454
 455     err = snd_pcm_sw_params_current (handle, sw_params);
 456     if (err < 0) {
 457         dolog ("Could not fully initialize DAC\n");
 458         alsa_logerr (err, "Failed to get current software parameters\n");
 459         return;
 460     }
 461
 462     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 463     if (err < 0) {
 464         dolog ("Could not fully initialize DAC\n");
 465         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 466                      threshold);
 467         return;
 468     }
 469
 470     err = snd_pcm_sw_params (handle, sw_params);
 471     if (err < 0) {
 472         dolog ("Could not fully initialize DAC\n");
 473         alsa_logerr (err, "Failed to set software parameters\n");
 474         return;
 475     }
 476 }
 477
 478 static int alsa_open (int in, struct alsa_params_req *req,
 479                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 480 {
 481     snd_pcm_t *handle;
 482     snd_pcm_hw_params_t *hw_params;
 483     int err;
 484     int size_in_usec;
 485     unsigned int freq, nchannels;
 486     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 487     snd_pcm_uframes_t obt_buffer_size;
 488     const char *typ = in ? "ADC" : "DAC";
 489     snd_pcm_format_t obtfmt;
 490
 491     freq = req->freq;
 492     nchannels = req->nchannels;
 493     size_in_usec = req->size_in_usec;
 494
 495     snd_pcm_hw_params_alloca (&hw_params);
 496
 497     err = snd_pcm_open (
 498         &handle,
 499         pcm_name,
 500         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 501         SND_PCM_NONBLOCK
 502         );
 503     if (err < 0) {
 504         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 505         return -1;
 506     }
 507
 508     err = snd_pcm_hw_params_any (handle, hw_params);
 509     if (err < 0) {
 510         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 511         goto err;
 512     }
 513
 514     err = snd_pcm_hw_params_set_access (
 515         handle,
 516         hw_params,
 517         SND_PCM_ACCESS_RW_INTERLEAVED
 518         );
 519     if (err < 0) {
 520         alsa_logerr2 (err, typ, "Failed to set access type\n");
 521         goto err;
 522     }
 523
 524     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 525     if (err < 0 && conf.verbose) {
 526         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 527     }
 528
 529     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 530     if (err < 0) {
 531         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 532         goto err;
 533     }
 534
 535     err = snd_pcm_hw_params_set_channels_near (
 536         handle,
 537         hw_params,
 538         &nchannels
 539         );
 540     if (err < 0) {
 541         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 542                       req->nchannels);
 543         goto err;
 544     }
 545
 546     if (nchannels != 1 && nchannels != 2) {
 547         alsa_logerr2 (err, typ,
 548                       "Can not handle obtained number of channels %d\n",
 549                       nchannels);
 550         goto err;
 551     }
 552
 553     if (req->buffer_size) {
 554         unsigned long obt;
 555
 556         if (size_in_usec) {
 557             int dir = 0;
 558             unsigned int btime = req->buffer_size;
 559
 560             err = snd_pcm_hw_params_set_buffer_time_near (
 561                 handle,
 562                 hw_params,
 563                 &btime,
 564                 &dir
 565                 );
 566             obt = btime;
 567         }
 568         else {
 569             snd_pcm_uframes_t bsize = req->buffer_size;
 570
 571             err = snd_pcm_hw_params_set_buffer_size_near (
 572                 handle,
 573                 hw_params,
 574                 &bsize
 575                 );
 576             obt = bsize;
 577         }
 578         if (err < 0) {
 579             alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
 580                           size_in_usec ? "time" : "size", req->buffer_size);
 581             goto err;
 582         }
 583
 584         if ((req->override_mask & 2) && (obt - req->buffer_size))
 585             dolog ("Requested buffer %s %u was rejected, using %lu\n",
 586                    size_in_usec ? "time" : "size", req->buffer_size, obt);
 587     }
 588
 589     if (req->period_size) {
 590         unsigned long obt;
 591
 592         if (size_in_usec) {
 593             int dir = 0;
 594             unsigned int ptime = req->period_size;
 595
 596             err = snd_pcm_hw_params_set_period_time_near (
 597                 handle,
 598                 hw_params,
 599                 &ptime,
 600                 &dir
 601                 );
 602             obt = ptime;
 603         }
 604         else {
 605             int dir = 0;
 606             snd_pcm_uframes_t psize = req->period_size;
 607
 608             err = snd_pcm_hw_params_set_period_size_near (
 609                 handle,
 610                 hw_params,
 611                 &psize,
 612                 &dir
 613                 );
 614             obt = psize;
 615         }
 616
 617         if (err < 0) {
 618             alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
 619                           size_in_usec ? "time" : "size", req->period_size);
 620             goto err;
 621         }
 622
 623         if (((req->override_mask & 1) && (obt - req->period_size)))
 624             dolog ("Requested period %s %u was rejected, using %lu\n",
 625                    size_in_usec ? "time" : "size", req->period_size, obt);
 626     }
 627
 628     err = snd_pcm_hw_params (handle, hw_params);
 629     if (err < 0) {
 630         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 631         goto err;
 632     }
 633
 634     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 635     if (err < 0) {
 636         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 637         goto err;
 638     }
 639
 640     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 641     if (err < 0) {
 642         alsa_logerr2 (err, typ, "Failed to get format\n");
 643         goto err;
 644     }
 645
 646     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 647         dolog ("Invalid format was returned %d\n", obtfmt);
 648         goto err;
 649     }
 650
 651     err = snd_pcm_prepare (handle);
 652     if (err < 0) {
 653         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 654         goto err;
 655     }
 656
 657     if (!in && conf.threshold) {
 658         snd_pcm_uframes_t threshold;
 659         int bytes_per_sec;
 660
 661         bytes_per_sec = freq << (nchannels == 2);
 662
 663         switch (obt->fmt) {
 664         case AUD_FMT_S8:
 665         case AUD_FMT_U8:
 666             break;
 667
 668         case AUD_FMT_S16:
 669         case AUD_FMT_U16:
 670             bytes_per_sec <<= 1;
 671             break;
 672
 673         case AUD_FMT_S32:
 674         case AUD_FMT_U32:
 675             bytes_per_sec <<= 2;
 676             break;
 677         }
 678
 679         threshold = (conf.threshold * bytes_per_sec) / 1000;
 680         alsa_set_threshold (handle, threshold);
 681     }
 682
 683     obt->nchannels = nchannels;
 684     obt->freq = freq;
 685     obt->samples = obt_buffer_size;
 686
 687     *handlep = handle;
 688
 689     if (conf.verbose &&
 690         (obtfmt != req->fmt ||
 691          obt->nchannels != req->nchannels ||
 692          obt->freq != req->freq)) {
 693         dolog ("Audio parameters for %s\n", typ);
 694         alsa_dump_info (req, obt, obtfmt);
 695     }
 696
 697 #ifdef DEBUG
 698     alsa_dump_info (req, obt, obtfmt);
 699 #endif
 700     return 0;
 701
 702  err:
 703     alsa_anal_close1 (&handle);
 704     return -1;
 705 }
 706
 707 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 708 {
 709     snd_pcm_sframes_t avail;
 710
 711     avail = snd_pcm_avail_update (handle);
 712     if (avail < 0) {
 713         if (avail == -EPIPE) {
 714             if (!alsa_recover (handle)) {
 715                 avail = snd_pcm_avail_update (handle);
 716             }
 717         }
 718
 719         if (avail < 0) {
 720             alsa_logerr (avail,
 721                          "Could not obtain number of available frames\n");
 722             return -1;
 723         }
 724     }
 725
 726     return avail;
 727 }
 728
 729 static void alsa_write_pending (ALSAVoiceOut *alsa)
 730 {
 731     HWVoiceOut *hw = &alsa->hw;
 732
 733     while (alsa->pending) {
 734         int left_till_end_samples = hw->samples - alsa->wpos;
 735         int len = audio_MIN (alsa->pending, left_till_end_samples);
 736         char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
 737
 738         while (len) {
 739             snd_pcm_sframes_t written;
 740
 741             written = snd_pcm_writei (alsa->handle, src, len);
 742
 743             if (written <= 0) {
 744                 switch (written) {
 745                 case 0:
 746                     if (conf.verbose) {
 747                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 748                     }
 749                     return;
 750
 751                 case -EPIPE:
 752                     if (alsa_recover (alsa->handle)) {
 753                         alsa_logerr (written, "Failed to write %d frames\n",
 754                                      len);
 755                         return;
 756                     }
 757                     if (conf.verbose) {
 758                         dolog ("Recovering from playback xrun\n");
 759                     }
 760                     continue;
 761
 762                 case -ESTRPIPE:
 763                     /* stream is suspended and waiting for an
 764                        application recovery */
 765                     if (alsa_resume (alsa->handle)) {
 766                         alsa_logerr (written, "Failed to write %d frames\n",
 767                                      len);
 768                         return;
 769                     }
 770                     if (conf.verbose) {
 771                         dolog ("Resuming suspended output stream\n");
 772                     }
 773                     continue;
 774
 775                 case -EAGAIN:
 776                     return;
 777
 778                 default:
 779                     alsa_logerr (written, "Failed to write %d frames from %p\n",
 780                                  len, src);
 781                     return;
 782                 }
 783             }
 784
 785             alsa->wpos = (alsa->wpos + written) % hw->samples;
 786             alsa->pending -= written;
 787             len -= written;
 788         }
 789     }
 790 }
 791
 792 static int alsa_run_out (HWVoiceOut *hw, int live)
 793 {
 794     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 795     int decr;
 796     snd_pcm_sframes_t avail;
 797
 798     avail = alsa_get_avail (alsa->handle);
 799     if (avail < 0) {
 800         dolog ("Could not get number of available playback frames\n");
 801         return 0;
 802     }
 803
 804     decr = audio_MIN (live, avail);
 805     decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
 806     alsa->pending += decr;
 807     alsa_write_pending (alsa);
 808     return decr;
 809 }
 810
 811 static void alsa_fini_out (HWVoiceOut *hw)
 812 {
 813     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 814
 815     ldebug ("alsa_fini\n");
 816     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 817
 818     g_free(alsa->pcm_buf);
 819     alsa->pcm_buf = NULL;
 820 }
 821
 822 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
 823 {
 824     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 825     struct alsa_params_req req;
 826     struct alsa_params_obt obt;
 827     snd_pcm_t *handle;
 828     struct audsettings obt_as;
 829
 830     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 831     req.freq = as->freq;
 832     req.nchannels = as->nchannels;
 833     req.period_size = conf.period_size_out;
 834     req.buffer_size = conf.buffer_size_out;
 835     req.size_in_usec = conf.size_in_usec_out;
 836     req.override_mask =
 837         (conf.period_size_out_overridden ? 1 : 0) |
 838         (conf.buffer_size_out_overridden ? 2 : 0);
 839
 840     if (alsa_open (0, &req, &obt, &handle)) {
 841         return -1;
 842     }
 843
 844     obt_as.freq = obt.freq;
 845     obt_as.nchannels = obt.nchannels;
 846     obt_as.fmt = obt.fmt;
 847     obt_as.endianness = obt.endianness;
 848
 849     audio_pcm_init_info (&hw->info, &obt_as);
 850     hw->samples = obt.samples;
 851
 852     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 853     if (!alsa->pcm_buf) {
 854         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 855                hw->samples, 1 << hw->info.shift);
 856         alsa_anal_close1 (&handle);
 857         return -1;
 858     }
 859
 860     alsa->handle = handle;
 861     return 0;
 862 }
 863
 864 #define VOICE_CTL_PAUSE 0
 865 #define VOICE_CTL_PREPARE 1
 866 #define VOICE_CTL_START 2
 867
 868 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 869 {
 870     int err;
 871
 872     if (ctl == VOICE_CTL_PAUSE) {
 873         err = snd_pcm_drop (handle);
 874         if (err < 0) {
 875             alsa_logerr (err, "Could not stop %s\n", typ);
 876             return -1;
 877         }
 878     }
 879     else {
 880         err = snd_pcm_prepare (handle);
 881         if (err < 0) {
 882             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 883             return -1;
 884         }
 885         if (ctl == VOICE_CTL_START) {
 886             err = snd_pcm_start(handle);
 887             if (err < 0) {
 888                 alsa_logerr (err, "Could not start handle for %s\n", typ);
 889                 return -1;
 890             }
 891         }
 892     }
 893
 894     return 0;
 895 }
 896
 897 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 898 {
 899     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 900
 901     switch (cmd) {
 902     case VOICE_ENABLE:
 903         {
 904             va_list ap;
 905             int poll_mode;
 906
 907             va_start (ap, cmd);
 908             poll_mode = va_arg (ap, int);
 909             va_end (ap);
 910
 911             ldebug ("enabling voice\n");
 912             if (poll_mode && alsa_poll_out (hw)) {
 913                 poll_mode = 0;
 914             }
 915             hw->poll_mode = poll_mode;
 916             return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
 917         }
 918
 919     case VOICE_DISABLE:
 920         ldebug ("disabling voice\n");
 921         if (hw->poll_mode) {
 922             hw->poll_mode = 0;
 923             alsa_fini_poll (&alsa->pollhlp);
 924         }
 925         return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
 926     }
 927
 928     return -1;
 929 }
 930
 931 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
 932 {
 933     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 934     struct alsa_params_req req;
 935     struct alsa_params_obt obt;
 936     snd_pcm_t *handle;
 937     struct audsettings obt_as;
 938
 939     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 940     req.freq = as->freq;
 941     req.nchannels = as->nchannels;
 942     req.period_size = conf.period_size_in;
 943     req.buffer_size = conf.buffer_size_in;
 944     req.size_in_usec = conf.size_in_usec_in;
 945     req.override_mask =
 946         (conf.period_size_in_overridden ? 1 : 0) |
 947         (conf.buffer_size_in_overridden ? 2 : 0);
 948
 949     if (alsa_open (1, &req, &obt, &handle)) {
 950         return -1;
 951     }
 952
 953     obt_as.freq = obt.freq;
 954     obt_as.nchannels = obt.nchannels;
 955     obt_as.fmt = obt.fmt;
 956     obt_as.endianness = obt.endianness;
 957
 958     audio_pcm_init_info (&hw->info, &obt_as);
 959     hw->samples = obt.samples;
 960
 961     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 962     if (!alsa->pcm_buf) {
 963         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 964                hw->samples, 1 << hw->info.shift);
 965         alsa_anal_close1 (&handle);
 966         return -1;
 967     }
 968
 969     alsa->handle = handle;
 970     return 0;
 971 }
 972
 973 static void alsa_fini_in (HWVoiceIn *hw)
 974 {
 975     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 976
 977     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 978
 979     g_free(alsa->pcm_buf);
 980     alsa->pcm_buf = NULL;
 981 }
 982
 983 static int alsa_run_in (HWVoiceIn *hw)
 984 {
 985     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 986     int hwshift = hw->info.shift;
 987     int i;
 988     int live = audio_pcm_hw_get_live_in (hw);
 989     int dead = hw->samples - live;
 990     int decr;
 991     struct {
 992         int add;
 993         int len;
 994     } bufs[2] = {
 995         { .add = hw->wpos, .len = 0 },
 996         { .add = 0,        .len = 0 }
 997     };
 998     snd_pcm_sframes_t avail;
 999     snd_pcm_uframes_t read_samples = 0;
1000
1001     if (!dead) {
1002         return 0;
1003     }
1004
1005     avail = alsa_get_avail (alsa->handle);
1006     if (avail < 0) {
1007         dolog ("Could not get number of captured frames\n");
1008         return 0;
1009     }
1010
1011     if (!avail) {
1012         snd_pcm_state_t state;
1013
1014         state = snd_pcm_state (alsa->handle);
1015         switch (state) {
1016         case SND_PCM_STATE_PREPARED:
1017             avail = hw->samples;
1018             break;
1019         case SND_PCM_STATE_SUSPENDED:
1020             /* stream is suspended and waiting for an application recovery */
1021             if (alsa_resume (alsa->handle)) {
1022                 dolog ("Failed to resume suspended input stream\n");
1023                 return 0;
1024             }
1025             if (conf.verbose) {
1026                 dolog ("Resuming suspended input stream\n");
1027             }
1028             break;
1029         default:
1030             if (conf.verbose) {
1031                 dolog ("No frames available and ALSA state is %d\n", state);
1032             }
1033             return 0;
1034         }
1035     }
1036
1037     decr = audio_MIN (dead, avail);
1038     if (!decr) {
1039         return 0;
1040     }
1041
1042     if (hw->wpos + decr > hw->samples) {
1043         bufs[0].len = (hw->samples - hw->wpos);
1044         bufs[1].len = (decr - (hw->samples - hw->wpos));
1045     }
1046     else {
1047         bufs[0].len = decr;
1048     }
1049
1050     for (i = 0; i < 2; ++i) {
1051         void *src;
1052         struct st_sample *dst;
1053         snd_pcm_sframes_t nread;
1054         snd_pcm_uframes_t len;
1055
1056         len = bufs[i].len;
1057
1058         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
1059         dst = hw->conv_buf + bufs[i].add;
1060
1061         while (len) {
1062             nread = snd_pcm_readi (alsa->handle, src, len);
1063
1064             if (nread <= 0) {
1065                 switch (nread) {
1066                 case 0:
1067                     if (conf.verbose) {
1068                         dolog ("Failed to read %ld frames (read zero)\n", len);
1069                     }
1070                     goto exit;
1071
1072                 case -EPIPE:
1073                     if (alsa_recover (alsa->handle)) {
1074                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
1075                         goto exit;
1076                     }
1077                     if (conf.verbose) {
1078                         dolog ("Recovering from capture xrun\n");
1079                     }
1080                     continue;
1081
1082                 case -EAGAIN:
1083                     goto exit;
1084
1085                 default:
1086                     alsa_logerr (
1087                         nread,
1088                         "Failed to read %ld frames from %p\n",
1089                         len,
1090                         src
1091                         );
1092                     goto exit;
1093                 }
1094             }
1095
1096             hw->conv (dst, src, nread);
1097
1098             src = advance (src, nread << hwshift);
1099             dst += nread;
1100
1101             read_samples += nread;
1102             len -= nread;
1103         }
1104     }
1105
1106  exit:
1107     hw->wpos = (hw->wpos + read_samples) % hw->samples;
1108     return read_samples;
1109 }
1110
1111 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
1112 {
1113     return audio_pcm_sw_read (sw, buf, size);
1114 }
1115
1116 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
1117 {
1118     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
1119
1120     switch (cmd) {
1121     case VOICE_ENABLE:
1122         {
1123             va_list ap;
1124             int poll_mode;
1125
1126             va_start (ap, cmd);
1127             poll_mode = va_arg (ap, int);
1128             va_end (ap);
1129
1130             ldebug ("enabling voice\n");
1131             if (poll_mode && alsa_poll_in (hw)) {
1132                 poll_mode = 0;
1133             }
1134             hw->poll_mode = poll_mode;
1135
1136             return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
1137         }
1138
1139     case VOICE_DISABLE:
1140         ldebug ("disabling voice\n");
1141         if (hw->poll_mode) {
1142             hw->poll_mode = 0;
1143             alsa_fini_poll (&alsa->pollhlp);
1144         }
1145         return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
1146     }
1147
1148     return -1;
1149 }
1150
1151 static void *alsa_audio_init (void)
1152 {
1153     return &conf;
1154 }
1155
1156 static void alsa_audio_fini (void *opaque)
1157 {
1158     (void) opaque;
1159 }
1160
1161 static struct audio_option alsa_options[] = {
1162     {
1163         .name        = "DAC_SIZE_IN_USEC",
1164         .tag         = AUD_OPT_BOOL,
1165         .valp        = &conf.size_in_usec_out,
1166         .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"
1167     },
1168     {
1169         .name        = "DAC_PERIOD_SIZE",
1170         .tag         = AUD_OPT_INT,
1171         .valp        = &conf.period_size_out,
1172         .descr       = "DAC period size (0 to go with system default)",
1173         .overriddenp = &conf.period_size_out_overridden
1174     },
1175     {
1176         .name        = "DAC_BUFFER_SIZE",
1177         .tag         = AUD_OPT_INT,
1178         .valp        = &conf.buffer_size_out,
1179         .descr       = "DAC buffer size (0 to go with system default)",
1180         .overriddenp = &conf.buffer_size_out_overridden
1181     },
1182     {
1183         .name        = "ADC_SIZE_IN_USEC",
1184         .tag         = AUD_OPT_BOOL,
1185         .valp        = &conf.size_in_usec_in,
1186         .descr       =
1187         "ADC period/buffer size in microseconds (otherwise in frames)"
1188     },
1189     {
1190         .name        = "ADC_PERIOD_SIZE",
1191         .tag         = AUD_OPT_INT,
1192         .valp        = &conf.period_size_in,
1193         .descr       = "ADC period size (0 to go with system default)",
1194         .overriddenp = &conf.period_size_in_overridden
1195     },
1196     {
1197         .name        = "ADC_BUFFER_SIZE",
1198         .tag         = AUD_OPT_INT,
1199         .valp        = &conf.buffer_size_in,
1200         .descr       = "ADC buffer size (0 to go with system default)",
1201         .overriddenp = &conf.buffer_size_in_overridden
1202     },
1203     {
1204         .name        = "THRESHOLD",
1205         .tag         = AUD_OPT_INT,
1206         .valp        = &conf.threshold,
1207         .descr       = "(undocumented)"
1208     },
1209     {
1210         .name        = "DAC_DEV",
1211         .tag         = AUD_OPT_STR,
1212         .valp        = &conf.pcm_name_out,
1213         .descr       = "DAC device name (for instance dmix)"
1214     },
1215     {
1216         .name        = "ADC_DEV",
1217         .tag         = AUD_OPT_STR,
1218         .valp        = &conf.pcm_name_in,
1219         .descr       = "ADC device name"
1220     },
1221     {
1222         .name        = "VERBOSE",
1223         .tag         = AUD_OPT_BOOL,
1224         .valp        = &conf.verbose,
1225         .descr       = "Behave in a more verbose way"
1226     },
1227     { /* End of list */ }
1228 };
1229
1230 static struct audio_pcm_ops alsa_pcm_ops = {
1231     .init_out = alsa_init_out,
1232     .fini_out = alsa_fini_out,
1233     .run_out  = alsa_run_out,
1234     .write    = alsa_write,
1235     .ctl_out  = alsa_ctl_out,
1236
1237     .init_in  = alsa_init_in,
1238     .fini_in  = alsa_fini_in,
1239     .run_in   = alsa_run_in,
1240     .read     = alsa_read,
1241     .ctl_in   = alsa_ctl_in,
1242 };
1243
1244 struct audio_driver alsa_audio_driver = {
1245     .name           = "alsa",
1246     .descr          = "ALSA http://www.alsa-project.org",
1247     .options        = alsa_options,
1248     .init           = alsa_audio_init,
1249     .fini           = alsa_audio_fini,
1250     .pcm_ops        = &alsa_pcm_ops,
1251     .can_be_default = 1,
1252     .max_voices_out = INT_MAX,
1253     .max_voices_in  = INT_MAX,
1254     .voice_size_out = sizeof (ALSAVoiceOut),
1255     .voice_size_in  = sizeof (ALSAVoiceIn)
1256 };