qemu/audio/alsaaudio.c
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   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
  25#include "qemu/osdep.h"
  26#include <alsa/asoundlib.h>
  27#include "qemu/main-loop.h"
  28#include "qemu/module.h"
  29#include "audio.h"
  30#include "trace.h"
  31
  32#pragma GCC diagnostic ignored "-Waddress"
  33
  34#define AUDIO_CAP "alsa"
  35#include "audio_int.h"
  36
  37struct pollhlp {
  38    snd_pcm_t *handle;
  39    struct pollfd *pfds;
  40    int count;
  41    int mask;
  42    AudioState *s;
  43};
  44
  45typedef struct ALSAVoiceOut {
  46    HWVoiceOut hw;
  47    snd_pcm_t *handle;
  48    struct pollhlp pollhlp;
  49    Audiodev *dev;
  50} ALSAVoiceOut;
  51
  52typedef struct ALSAVoiceIn {
  53    HWVoiceIn hw;
  54    snd_pcm_t *handle;
  55    struct pollhlp pollhlp;
  56    Audiodev *dev;
  57} ALSAVoiceIn;
  58
  59struct alsa_params_req {
  60    int freq;
  61    snd_pcm_format_t fmt;
  62    int nchannels;
  63};
  64
  65struct alsa_params_obt {
  66    int freq;
  67    AudioFormat fmt;
  68    int endianness;
  69    int nchannels;
  70    snd_pcm_uframes_t samples;
  71};
  72
  73static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
  74{
  75    va_list ap;
  76
  77    va_start (ap, fmt);
  78    AUD_vlog (AUDIO_CAP, fmt, ap);
  79    va_end (ap);
  80
  81    AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
  82}
  83
  84static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
  85    int err,
  86    const char *typ,
  87    const char *fmt,
  88    ...
  89    )
  90{
  91    va_list ap;
  92
  93    AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
  94
  95    va_start (ap, fmt);
  96    AUD_vlog (AUDIO_CAP, fmt, ap);
  97    va_end (ap);
  98
  99    AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 100}
 101
 102static void alsa_fini_poll (struct pollhlp *hlp)
 103{
 104    int i;
 105    struct pollfd *pfds = hlp->pfds;
 106
 107    if (pfds) {
 108        for (i = 0; i < hlp->count; ++i) {
 109            qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 110        }
 111        g_free (pfds);
 112    }
 113    hlp->pfds = NULL;
 114    hlp->count = 0;
 115    hlp->handle = NULL;
 116}
 117
 118static void alsa_anal_close1 (snd_pcm_t **handlep)
 119{
 120    int err = snd_pcm_close (*handlep);
 121    if (err) {
 122        alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 123    }
 124    *handlep = NULL;
 125}
 126
 127static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
 128{
 129    alsa_fini_poll (hlp);
 130    alsa_anal_close1 (handlep);
 131}
 132
 133static int alsa_recover (snd_pcm_t *handle)
 134{
 135    int err = snd_pcm_prepare (handle);
 136    if (err < 0) {
 137        alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 138        return -1;
 139    }
 140    return 0;
 141}
 142
 143static int alsa_resume (snd_pcm_t *handle)
 144{
 145    int err = snd_pcm_resume (handle);
 146    if (err < 0) {
 147        alsa_logerr (err, "Failed to resume handle %p\n", handle);
 148        return -1;
 149    }
 150    return 0;
 151}
 152
 153static void alsa_poll_handler (void *opaque)
 154{
 155    int err, count;
 156    snd_pcm_state_t state;
 157    struct pollhlp *hlp = opaque;
 158    unsigned short revents;
 159
 160    count = poll (hlp->pfds, hlp->count, 0);
 161    if (count < 0) {
 162        dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
 163        return;
 164    }
 165
 166    if (!count) {
 167        return;
 168    }
 169
 170    /* XXX: ALSA example uses initial count, not the one returned by
 171       poll, correct? */
 172    err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
 173                                            hlp->count, &revents);
 174    if (err < 0) {
 175        alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
 176        return;
 177    }
 178
 179    if (!(revents & hlp->mask)) {
 180        trace_alsa_revents(revents);
 181        return;
 182    }
 183
 184    state = snd_pcm_state (hlp->handle);
 185    switch (state) {
 186    case SND_PCM_STATE_SETUP:
 187        alsa_recover (hlp->handle);
 188        break;
 189
 190    case SND_PCM_STATE_XRUN:
 191        alsa_recover (hlp->handle);
 192        break;
 193
 194    case SND_PCM_STATE_SUSPENDED:
 195        alsa_resume (hlp->handle);
 196        break;
 197
 198    case SND_PCM_STATE_PREPARED:
 199        audio_run(hlp->s, "alsa run (prepared)");
 200        break;
 201
 202    case SND_PCM_STATE_RUNNING:
 203        audio_run(hlp->s, "alsa run (running)");
 204        break;
 205
 206    default:
 207        dolog ("Unexpected state %d\n", state);
 208    }
 209}
 210
 211static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
 212{
 213    int i, count, err;
 214    struct pollfd *pfds;
 215
 216    count = snd_pcm_poll_descriptors_count (handle);
 217    if (count <= 0) {
 218        dolog ("Could not initialize poll mode\n"
 219               "Invalid number of poll descriptors %d\n", count);
 220        return -1;
 221    }
 222
 223    pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
 224    if (!pfds) {
 225        dolog ("Could not initialize poll mode\n");
 226        return -1;
 227    }
 228
 229    err = snd_pcm_poll_descriptors (handle, pfds, count);
 230    if (err < 0) {
 231        alsa_logerr (err, "Could not initialize poll mode\n"
 232                     "Could not obtain poll descriptors\n");
 233        g_free (pfds);
 234        return -1;
 235    }
 236
 237    for (i = 0; i < count; ++i) {
 238        if (pfds[i].events & POLLIN) {
 239            qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
 240        }
 241        if (pfds[i].events & POLLOUT) {
 242            trace_alsa_pollout(i, pfds[i].fd);
 243            qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
 244        }
 245        trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
 246
 247    }
 248    hlp->pfds = pfds;
 249    hlp->count = count;
 250    hlp->handle = handle;
 251    hlp->mask = mask;
 252    return 0;
 253}
 254
 255static int alsa_poll_out (HWVoiceOut *hw)
 256{
 257    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 258
 259    return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
 260}
 261
 262static int alsa_poll_in (HWVoiceIn *hw)
 263{
 264    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 265
 266    return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
 267}
 268
 269static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
 270{
 271    switch (fmt) {
 272    case AUDIO_FORMAT_S8:
 273        return SND_PCM_FORMAT_S8;
 274
 275    case AUDIO_FORMAT_U8:
 276        return SND_PCM_FORMAT_U8;
 277
 278    case AUDIO_FORMAT_S16:
 279        if (endianness) {
 280            return SND_PCM_FORMAT_S16_BE;
 281        }
 282        else {
 283            return SND_PCM_FORMAT_S16_LE;
 284        }
 285
 286    case AUDIO_FORMAT_U16:
 287        if (endianness) {
 288            return SND_PCM_FORMAT_U16_BE;
 289        }
 290        else {
 291            return SND_PCM_FORMAT_U16_LE;
 292        }
 293
 294    case AUDIO_FORMAT_S32:
 295        if (endianness) {
 296            return SND_PCM_FORMAT_S32_BE;
 297        }
 298        else {
 299            return SND_PCM_FORMAT_S32_LE;
 300        }
 301
 302    case AUDIO_FORMAT_U32:
 303        if (endianness) {
 304            return SND_PCM_FORMAT_U32_BE;
 305        }
 306        else {
 307            return SND_PCM_FORMAT_U32_LE;
 308        }
 309
 310    default:
 311        dolog ("Internal logic error: Bad audio format %d\n", fmt);
 312#ifdef DEBUG_AUDIO
 313        abort ();
 314#endif
 315        return SND_PCM_FORMAT_U8;
 316    }
 317}
 318
 319static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
 320                           int *endianness)
 321{
 322    switch (alsafmt) {
 323    case SND_PCM_FORMAT_S8:
 324        *endianness = 0;
 325        *fmt = AUDIO_FORMAT_S8;
 326        break;
 327
 328    case SND_PCM_FORMAT_U8:
 329        *endianness = 0;
 330        *fmt = AUDIO_FORMAT_U8;
 331        break;
 332
 333    case SND_PCM_FORMAT_S16_LE:
 334        *endianness = 0;
 335        *fmt = AUDIO_FORMAT_S16;
 336        break;
 337
 338    case SND_PCM_FORMAT_U16_LE:
 339        *endianness = 0;
 340        *fmt = AUDIO_FORMAT_U16;
 341        break;
 342
 343    case SND_PCM_FORMAT_S16_BE:
 344        *endianness = 1;
 345        *fmt = AUDIO_FORMAT_S16;
 346        break;
 347
 348    case SND_PCM_FORMAT_U16_BE:
 349        *endianness = 1;
 350        *fmt = AUDIO_FORMAT_U16;
 351        break;
 352
 353    case SND_PCM_FORMAT_S32_LE:
 354        *endianness = 0;
 355        *fmt = AUDIO_FORMAT_S32;
 356        break;
 357
 358    case SND_PCM_FORMAT_U32_LE:
 359        *endianness = 0;
 360        *fmt = AUDIO_FORMAT_U32;
 361        break;
 362
 363    case SND_PCM_FORMAT_S32_BE:
 364        *endianness = 1;
 365        *fmt = AUDIO_FORMAT_S32;
 366        break;
 367
 368    case SND_PCM_FORMAT_U32_BE:
 369        *endianness = 1;
 370        *fmt = AUDIO_FORMAT_U32;
 371        break;
 372
 373    default:
 374        dolog ("Unrecognized audio format %d\n", alsafmt);
 375        return -1;
 376    }
 377
 378    return 0;
 379}
 380
 381static void alsa_dump_info (struct alsa_params_req *req,
 382                            struct alsa_params_obt *obt,
 383                            snd_pcm_format_t obtfmt,
 384                            AudiodevAlsaPerDirectionOptions *apdo)
 385{
 386    dolog("parameter | requested value | obtained value\n");
 387    dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 388    dolog("channels  |      %10d |     %10d\n",
 389          req->nchannels, obt->nchannels);
 390    dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 391    dolog("============================================\n");
 392    dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
 393          apdo->buffer_length, apdo->period_length);
 394    dolog("obtained: samples %ld\n", obt->samples);
 395}
 396
 397static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 398{
 399    int err;
 400    snd_pcm_sw_params_t *sw_params;
 401
 402    snd_pcm_sw_params_alloca (&sw_params);
 403
 404    err = snd_pcm_sw_params_current (handle, sw_params);
 405    if (err < 0) {
 406        dolog ("Could not fully initialize DAC\n");
 407        alsa_logerr (err, "Failed to get current software parameters\n");
 408        return;
 409    }
 410
 411    err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 412    if (err < 0) {
 413        dolog ("Could not fully initialize DAC\n");
 414        alsa_logerr (err, "Failed to set software threshold to %ld\n",
 415                     threshold);
 416        return;
 417    }
 418
 419    err = snd_pcm_sw_params (handle, sw_params);
 420    if (err < 0) {
 421        dolog ("Could not fully initialize DAC\n");
 422        alsa_logerr (err, "Failed to set software parameters\n");
 423        return;
 424    }
 425}
 426
 427static int alsa_open(bool in, struct alsa_params_req *req,
 428                     struct alsa_params_obt *obt, snd_pcm_t **handlep,
 429                     Audiodev *dev)
 430{
 431    AudiodevAlsaOptions *aopts = &dev->u.alsa;
 432    AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
 433    snd_pcm_t *handle;
 434    snd_pcm_hw_params_t *hw_params;
 435    int err;
 436    unsigned int freq, nchannels;
 437    const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
 438    snd_pcm_uframes_t obt_buffer_size;
 439    const char *typ = in ? "ADC" : "DAC";
 440    snd_pcm_format_t obtfmt;
 441
 442    freq = req->freq;
 443    nchannels = req->nchannels;
 444
 445    snd_pcm_hw_params_alloca (&hw_params);
 446
 447    err = snd_pcm_open (
 448        &handle,
 449        pcm_name,
 450        in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 451        SND_PCM_NONBLOCK
 452        );
 453    if (err < 0) {
 454        alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 455        return -1;
 456    }
 457
 458    err = snd_pcm_hw_params_any (handle, hw_params);
 459    if (err < 0) {
 460        alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 461        goto err;
 462    }
 463
 464    err = snd_pcm_hw_params_set_access (
 465        handle,
 466        hw_params,
 467        SND_PCM_ACCESS_RW_INTERLEAVED
 468        );
 469    if (err < 0) {
 470        alsa_logerr2 (err, typ, "Failed to set access type\n");
 471        goto err;
 472    }
 473
 474    err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 475    if (err < 0) {
 476        alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 477    }
 478
 479    err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 480    if (err < 0) {
 481        alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 482        goto err;
 483    }
 484
 485    err = snd_pcm_hw_params_set_channels_near (
 486        handle,
 487        hw_params,
 488        &nchannels
 489        );
 490    if (err < 0) {
 491        alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 492                      req->nchannels);
 493        goto err;
 494    }
 495
 496    if (apdo->buffer_length) {
 497        int dir = 0;
 498        unsigned int btime = apdo->buffer_length;
 499
 500        err = snd_pcm_hw_params_set_buffer_time_near(
 501            handle, hw_params, &btime, &dir);
 502
 503        if (err < 0) {
 504            alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
 505                         apdo->buffer_length);
 506            goto err;
 507        }
 508
 509        if (apdo->has_buffer_length && btime != apdo->buffer_length) {
 510            dolog("Requested buffer time %" PRId32
 511                  " was rejected, using %u\n", apdo->buffer_length, btime);
 512        }
 513    }
 514
 515    if (apdo->period_length) {
 516        int dir = 0;
 517        unsigned int ptime = apdo->period_length;
 518
 519        err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
 520                                                     &dir);
 521
 522        if (err < 0) {
 523            alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
 524                         apdo->period_length);
 525            goto err;
 526        }
 527
 528        if (apdo->has_period_length && ptime != apdo->period_length) {
 529            dolog("Requested period time %" PRId32 " was rejected, using %d\n",
 530                  apdo->period_length, ptime);
 531        }
 532    }
 533
 534    err = snd_pcm_hw_params (handle, hw_params);
 535    if (err < 0) {
 536        alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 537        goto err;
 538    }
 539
 540    err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 541    if (err < 0) {
 542        alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 543        goto err;
 544    }
 545
 546    err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 547    if (err < 0) {
 548        alsa_logerr2 (err, typ, "Failed to get format\n");
 549        goto err;
 550    }
 551
 552    if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 553        dolog ("Invalid format was returned %d\n", obtfmt);
 554        goto err;
 555    }
 556
 557    err = snd_pcm_prepare (handle);
 558    if (err < 0) {
 559        alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 560        goto err;
 561    }
 562
 563    if (!in && aopts->has_threshold && aopts->threshold) {
 564        struct audsettings as = { .freq = freq };
 565        alsa_set_threshold(
 566            handle,
 567            audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
 568                                &as, aopts->threshold));
 569    }
 570
 571    obt->nchannels = nchannels;
 572    obt->freq = freq;
 573    obt->samples = obt_buffer_size;
 574
 575    *handlep = handle;
 576
 577    if (obtfmt != req->fmt ||
 578         obt->nchannels != req->nchannels ||
 579         obt->freq != req->freq) {
 580        dolog ("Audio parameters for %s\n", typ);
 581        alsa_dump_info(req, obt, obtfmt, apdo);
 582    }
 583
 584#ifdef DEBUG
 585    alsa_dump_info(req, obt, obtfmt, pdo);
 586#endif
 587    return 0;
 588
 589 err:
 590    alsa_anal_close1 (&handle);
 591    return -1;
 592}
 593
 594static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 595{
 596    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 597    size_t pos = 0;
 598    size_t len_frames = len / hw->info.bytes_per_frame;
 599
 600    while (len_frames) {
 601        char *src = advance(buf, pos);
 602        snd_pcm_sframes_t written;
 603
 604        written = snd_pcm_writei(alsa->handle, src, len_frames);
 605
 606        if (written <= 0) {
 607            switch (written) {
 608            case 0:
 609                trace_alsa_wrote_zero(len_frames);
 610                return pos;
 611
 612            case -EPIPE:
 613                if (alsa_recover(alsa->handle)) {
 614                    alsa_logerr(written, "Failed to write %zu frames\n",
 615                                len_frames);
 616                    return pos;
 617                }
 618                trace_alsa_xrun_out();
 619                continue;
 620
 621            case -ESTRPIPE:
 622                /*
 623                 * stream is suspended and waiting for an application
 624                 * recovery
 625                 */
 626                if (alsa_resume(alsa->handle)) {
 627                    alsa_logerr(written, "Failed to write %zu frames\n",
 628                                len_frames);
 629                    return pos;
 630                }
 631                trace_alsa_resume_out();
 632                continue;
 633
 634            case -EAGAIN:
 635                return pos;
 636
 637            default:
 638                alsa_logerr(written, "Failed to write %zu frames from %p\n",
 639                            len, src);
 640                return pos;
 641            }
 642        }
 643
 644        pos += written * hw->info.bytes_per_frame;
 645        if (written < len_frames) {
 646            break;
 647        }
 648        len_frames -= written;
 649    }
 650
 651    return pos;
 652}
 653
 654static void alsa_fini_out (HWVoiceOut *hw)
 655{
 656    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 657
 658    ldebug ("alsa_fini\n");
 659    alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 660}
 661
 662static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
 663                         void *drv_opaque)
 664{
 665    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 666    struct alsa_params_req req;
 667    struct alsa_params_obt obt;
 668    snd_pcm_t *handle;
 669    struct audsettings obt_as;
 670    Audiodev *dev = drv_opaque;
 671
 672    req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 673    req.freq = as->freq;
 674    req.nchannels = as->nchannels;
 675
 676    if (alsa_open(0, &req, &obt, &handle, dev)) {
 677        return -1;
 678    }
 679
 680    obt_as.freq = obt.freq;
 681    obt_as.nchannels = obt.nchannels;
 682    obt_as.fmt = obt.fmt;
 683    obt_as.endianness = obt.endianness;
 684
 685    audio_pcm_init_info (&hw->info, &obt_as);
 686    hw->samples = obt.samples;
 687
 688    alsa->pollhlp.s = hw->s;
 689    alsa->handle = handle;
 690    alsa->dev = dev;
 691    return 0;
 692}
 693
 694#define VOICE_CTL_PAUSE 0
 695#define VOICE_CTL_PREPARE 1
 696#define VOICE_CTL_START 2
 697
 698static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 699{
 700    int err;
 701
 702    if (ctl == VOICE_CTL_PAUSE) {
 703        err = snd_pcm_drop (handle);
 704        if (err < 0) {
 705            alsa_logerr (err, "Could not stop %s\n", typ);
 706            return -1;
 707        }
 708    }
 709    else {
 710        err = snd_pcm_prepare (handle);
 711        if (err < 0) {
 712            alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 713            return -1;
 714        }
 715        if (ctl == VOICE_CTL_START) {
 716            err = snd_pcm_start(handle);
 717            if (err < 0) {
 718                alsa_logerr (err, "Could not start handle for %s\n", typ);
 719                return -1;
 720            }
 721        }
 722    }
 723
 724    return 0;
 725}
 726
 727static void alsa_enable_out(HWVoiceOut *hw, bool enable)
 728{
 729    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 730    AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
 731
 732    if (enable) {
 733        bool poll_mode = apdo->try_poll;
 734
 735        ldebug("enabling voice\n");
 736        if (poll_mode && alsa_poll_out(hw)) {
 737            poll_mode = 0;
 738        }
 739        hw->poll_mode = poll_mode;
 740        alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
 741    } else {
 742        ldebug("disabling voice\n");
 743        if (hw->poll_mode) {
 744            hw->poll_mode = 0;
 745            alsa_fini_poll(&alsa->pollhlp);
 746        }
 747        alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
 748    }
 749}
 750
 751static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 752{
 753    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 754    struct alsa_params_req req;
 755    struct alsa_params_obt obt;
 756    snd_pcm_t *handle;
 757    struct audsettings obt_as;
 758    Audiodev *dev = drv_opaque;
 759
 760    req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 761    req.freq = as->freq;
 762    req.nchannels = as->nchannels;
 763
 764    if (alsa_open(1, &req, &obt, &handle, dev)) {
 765        return -1;
 766    }
 767
 768    obt_as.freq = obt.freq;
 769    obt_as.nchannels = obt.nchannels;
 770    obt_as.fmt = obt.fmt;
 771    obt_as.endianness = obt.endianness;
 772
 773    audio_pcm_init_info (&hw->info, &obt_as);
 774    hw->samples = obt.samples;
 775
 776    alsa->pollhlp.s = hw->s;
 777    alsa->handle = handle;
 778    alsa->dev = dev;
 779    return 0;
 780}
 781
 782static void alsa_fini_in (HWVoiceIn *hw)
 783{
 784    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 785
 786    alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 787}
 788
 789static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
 790{
 791    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 792    size_t pos = 0;
 793
 794    while (len) {
 795        void *dst = advance(buf, pos);
 796        snd_pcm_sframes_t nread;
 797
 798        nread = snd_pcm_readi(
 799            alsa->handle, dst, len / hw->info.bytes_per_frame);
 800
 801        if (nread <= 0) {
 802            switch (nread) {
 803            case 0:
 804                trace_alsa_read_zero(len);
 805                return pos;;
 806
 807            case -EPIPE:
 808                if (alsa_recover(alsa->handle)) {
 809                    alsa_logerr(nread, "Failed to read %zu frames\n", len);
 810                    return pos;
 811                }
 812                trace_alsa_xrun_in();
 813                continue;
 814
 815            case -EAGAIN:
 816                return pos;
 817
 818            default:
 819                alsa_logerr(nread, "Failed to read %zu frames to %p\n",
 820                            len, dst);
 821                return pos;;
 822            }
 823        }
 824
 825        pos += nread * hw->info.bytes_per_frame;
 826        len -= nread * hw->info.bytes_per_frame;
 827    }
 828
 829    return pos;
 830}
 831
 832static void alsa_enable_in(HWVoiceIn *hw, bool enable)
 833{
 834    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 835    AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
 836
 837    if (enable) {
 838        bool poll_mode = apdo->try_poll;
 839
 840        ldebug("enabling voice\n");
 841        if (poll_mode && alsa_poll_in(hw)) {
 842            poll_mode = 0;
 843        }
 844        hw->poll_mode = poll_mode;
 845
 846        alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
 847    } else {
 848        ldebug ("disabling voice\n");
 849        if (hw->poll_mode) {
 850            hw->poll_mode = 0;
 851            alsa_fini_poll(&alsa->pollhlp);
 852        }
 853        alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
 854    }
 855}
 856
 857static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
 858{
 859    if (!apdo->has_try_poll) {
 860        apdo->try_poll = true;
 861        apdo->has_try_poll = true;
 862    }
 863}
 864
 865static void *alsa_audio_init(Audiodev *dev)
 866{
 867    AudiodevAlsaOptions *aopts;
 868    assert(dev->driver == AUDIODEV_DRIVER_ALSA);
 869
 870    aopts = &dev->u.alsa;
 871    alsa_init_per_direction(aopts->in);
 872    alsa_init_per_direction(aopts->out);
 873
 874    /*
 875     * need to define them, as otherwise alsa produces no sound
 876     * doesn't set has_* so alsa_open can identify it wasn't set by the user
 877     */
 878    if (!dev->u.alsa.out->has_period_length) {
 879        /* 1024 frames assuming 44100Hz */
 880        dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
 881    }
 882    if (!dev->u.alsa.out->has_buffer_length) {
 883        /* 4096 frames assuming 44100Hz */
 884        dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
 885    }
 886
 887    /*
 888     * OptsVisitor sets unspecified optional fields to zero, but do not depend
 889     * on it...
 890     */
 891    if (!dev->u.alsa.in->has_period_length) {
 892        dev->u.alsa.in->period_length = 0;
 893    }
 894    if (!dev->u.alsa.in->has_buffer_length) {
 895        dev->u.alsa.in->buffer_length = 0;
 896    }
 897
 898    return dev;
 899}
 900
 901static void alsa_audio_fini (void *opaque)
 902{
 903}
 904
 905static struct audio_pcm_ops alsa_pcm_ops = {
 906    .init_out = alsa_init_out,
 907    .fini_out = alsa_fini_out,
 908    .write    = alsa_write,
 909    .enable_out = alsa_enable_out,
 910
 911    .init_in  = alsa_init_in,
 912    .fini_in  = alsa_fini_in,
 913    .read     = alsa_read,
 914    .enable_in = alsa_enable_in,
 915};
 916
 917static struct audio_driver alsa_audio_driver = {
 918    .name           = "alsa",
 919    .descr          = "ALSA http://www.alsa-project.org",
 920    .init           = alsa_audio_init,
 921    .fini           = alsa_audio_fini,
 922    .pcm_ops        = &alsa_pcm_ops,
 923    .can_be_default = 1,
 924    .max_voices_out = INT_MAX,
 925    .max_voices_in  = INT_MAX,
 926    .voice_size_out = sizeof (ALSAVoiceOut),
 927    .voice_size_in  = sizeof (ALSAVoiceIn)
 928};
 929
 930static void register_audio_alsa(void)
 931{
 932    audio_driver_register(&alsa_audio_driver);
 933}
 934type_init(register_audio_alsa);
 935