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      1 /*
      2  * QEMU Audio subsystem
      3  *
      4  * Copyright (c) 2007-2008 The Android Open Source Project
      5  * Copyright (c) 2003-2005 Vassili Karpov (malc)
      6  *
      7  * Permission is hereby granted, free of charge, to any person obtaining a copy
      8  * of this software and associated documentation files (the "Software"), to deal
      9  * in the Software without restriction, including without limitation the rights
     10  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
     11  * copies of the Software, and to permit persons to whom the Software is
     12  * furnished to do so, subject to the following conditions:
     13  *
     14  * The above copyright notice and this permission notice shall be included in
     15  * all copies or substantial portions of the Software.
     16  *
     17  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
     18  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
     19  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
     20  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
     21  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
     22  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
     23  * THE SOFTWARE.
     24  */
     25 #include "hw/hw.h"
     26 #include "audio.h"
     27 #include "monitor.h"
     28 #include "qemu-timer.h"
     29 #include "sysemu.h"
     30 
     31 #define AUDIO_CAP "audio"
     32 #include "audio_int.h"
     33 #include "android/utils/system.h"
     34 #include "qemu_debug.h"
     35 #include "android/android.h"
     36 
     37 /* #define DEBUG_PLIVE */
     38 /* #define DEBUG_LIVE */
     39 /* #define DEBUG_OUT */
     40 /* #define DEBUG_CAPTURE */
     41 
     42 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
     43 
     44 static struct audio_driver *drvtab[] = {
     45 #ifdef CONFIG_PULSEAUDIO
     46     &pa_audio_driver,
     47 #endif
     48 #ifdef CONFIG_ESD
     49     &esd_audio_driver,
     50 #endif
     51 #ifdef CONFIG_ALSA
     52     &alsa_audio_driver,
     53 #endif
     54 #ifdef CONFIG_COREAUDIO
     55     &coreaudio_audio_driver,
     56 #endif
     57 #ifdef CONFIG_DSOUND
     58     &dsound_audio_driver,
     59 #endif
     60 #ifdef CONFIG_FMOD
     61     &fmod_audio_driver,
     62 #endif
     63 #ifdef CONFIG_WINAUDIO
     64     &win_audio_driver,
     65 #endif
     66 #ifdef CONFIG_OSS
     67     &oss_audio_driver,
     68 #endif
     69     &no_audio_driver,
     70 #if 0  /* disabled WAV audio for now - until we find a user-friendly way to use it */
     71     &wav_audio_driver
     72 #endif
     73 };
     74 
     75 
     76 int
     77 audio_get_backend_count( int  is_input )
     78 {
     79     int  nn, count = 0;
     80 
     81     for (nn = 0; nn < sizeof(drvtab)/sizeof(drvtab[0]); nn++)
     82     {
     83         if (is_input) {
     84             if ( drvtab[nn]->max_voices_in > 0 )
     85                 count += 1;
     86         } else {
     87             if ( drvtab[nn]->max_voices_out > 0 )
     88                 count += 1;
     89         }
     90     }
     91     return  count;
     92 }
     93 
     94 const char*
     95 audio_get_backend_name( int   is_input, int  index, const char*  *pinfo )
     96 {
     97     int  nn;
     98 
     99     index += 1;
    100     for (nn = 0; nn < sizeof(drvtab)/sizeof(drvtab[0]); nn++)
    101     {
    102         if (is_input) {
    103             if ( drvtab[nn]->max_voices_in > 0 ) {
    104                 if ( --index == 0 ) {
    105                     *pinfo = drvtab[nn]->descr;
    106                     return drvtab[nn]->name;
    107                 }
    108             }
    109         } else {
    110             if ( drvtab[nn]->max_voices_out > 0 ) {
    111                 if ( --index == 0 ) {
    112                     *pinfo = drvtab[nn]->descr;
    113                     return drvtab[nn]->name;
    114                 }
    115             }
    116         }
    117     }
    118     *pinfo = NULL;
    119     return  NULL;
    120 }
    121 
    122 
    123 int
    124 audio_check_backend_name( int  is_input, const char*  name )
    125 {
    126     int  nn;
    127 
    128     for (nn = 0; nn < sizeof(drvtab)/sizeof(drvtab[0]); nn++)
    129     {
    130         if ( !strcmp(drvtab[nn]->name, name) ) {
    131             if (is_input) {
    132                 if (drvtab[nn]->max_voices_in > 0)
    133                     return 1;
    134             } else {
    135                 if (drvtab[nn]->max_voices_out > 0)
    136                     return 1;
    137             }
    138             break;
    139         }
    140     }
    141     return 0;
    142 }
    143 
    144 
    145 struct fixed_settings {
    146     int enabled;
    147     int nb_voices;
    148     int greedy;
    149     struct audsettings settings;
    150 };
    151 
    152 static struct {
    153     struct fixed_settings fixed_out;
    154     struct fixed_settings fixed_in;
    155     union {
    156         int hertz;
    157         int64_t ticks;
    158     } period;
    159     int plive;
    160     int log_to_monitor;
    161     int try_poll_in;
    162     int try_poll_out;
    163 } conf = {
    164     .fixed_out = { /* DAC fixed settings */
    165         .enabled = 1,
    166         .nb_voices = 1,
    167         .greedy = 1,
    168         .settings = {
    169             .freq = 44100,
    170             .nchannels = 2,
    171             .fmt = AUD_FMT_S16,
    172             .endianness =  AUDIO_HOST_ENDIANNESS,
    173         }
    174     },
    175 
    176     .fixed_in = { /* ADC fixed settings */
    177         .enabled = 1,
    178         .nb_voices = 1,
    179         .greedy = 1,
    180         .settings = {
    181             .freq = 44100,
    182             .nchannels = 2,
    183             .fmt = AUD_FMT_S16,
    184             .endianness = AUDIO_HOST_ENDIANNESS,
    185         }
    186     },
    187 
    188     .period = { .hertz = 250 },
    189     .plive = 0,
    190     .log_to_monitor = 0,
    191     .try_poll_in = 1,
    192     .try_poll_out = 1,
    193 };
    194 
    195 static AudioState glob_audio_state;
    196 
    197 struct mixeng_volume nominal_volume = {
    198     .mute = 0,
    199 #ifdef FLOAT_MIXENG
    200     .r = 1.0,
    201     .l = 1.0,
    202 #else
    203     .r = 1ULL << 32,
    204     .l = 1ULL << 32,
    205 #endif
    206 };
    207 
    208 #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
    209 #error No its not
    210 #else
    211 static void audio_print_options (const char *prefix,
    212                                  struct audio_option *opt);
    213 
    214 int audio_bug (const char *funcname, int cond)
    215 {
    216     if (cond) {
    217         static int shown;
    218 
    219         AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
    220         if (!shown) {
    221             struct audio_driver *d;
    222 
    223             shown = 1;
    224             AUD_log (NULL, "Save all your work and restart without audio\n");
    225             AUD_log (NULL, "Please send bug report to av1474 (at) comtv.ru\n");
    226             AUD_log (NULL, "I am sorry\n");
    227             d = glob_audio_state.drv;
    228             if (d) {
    229                 audio_print_options (d->name, d->options);
    230             }
    231         }
    232         AUD_log (NULL, "Context:\n");
    233 
    234 #if defined AUDIO_BREAKPOINT_ON_BUG
    235 #  if defined HOST_I386
    236 #    if defined __GNUC__
    237         __asm__ ("int3");
    238 #    elif defined _MSC_VER
    239         _asm _emit 0xcc;
    240 #    else
    241         abort ();
    242 #    endif
    243 #  else
    244         abort ();
    245 #  endif
    246 #endif
    247     }
    248 
    249     return cond;
    250 }
    251 #endif
    252 
    253 static inline int audio_bits_to_index (int bits)
    254 {
    255     switch (bits) {
    256     case 8:
    257         return 0;
    258 
    259     case 16:
    260         return 1;
    261 
    262     case 32:
    263         return 2;
    264 
    265     default:
    266         audio_bug ("bits_to_index", 1);
    267         AUD_log (NULL, "invalid bits %d\n", bits);
    268         return 0;
    269     }
    270 }
    271 
    272 void *audio_calloc (const char *funcname, int nmemb, size_t size)
    273 {
    274     int cond;
    275     size_t len;
    276 
    277     len = nmemb * size;
    278     cond = !nmemb || !size;
    279     cond |= nmemb < 0;
    280     cond |= len < size;
    281 
    282     if (audio_bug ("audio_calloc", cond)) {
    283         AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
    284                  funcname);
    285         AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
    286         return NULL;
    287     }
    288 
    289     return qemu_mallocz (len);
    290 }
    291 
    292 static char *audio_alloc_prefix (const char *s)
    293 {
    294     const char qemu_prefix[] = "QEMU_";
    295     size_t len, i;
    296     char *r, *u;
    297 
    298     if (!s) {
    299         return NULL;
    300     }
    301 
    302     len = strlen (s);
    303     r = qemu_malloc (len + sizeof (qemu_prefix));
    304 
    305     u = r + sizeof (qemu_prefix) - 1;
    306 
    307     pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
    308     pstrcat (r, len + sizeof (qemu_prefix), s);
    309 
    310     for (i = 0; i < len; ++i) {
    311         u[i] = qemu_toupper(u[i]);
    312     }
    313 
    314     return r;
    315 }
    316 
    317 static const char *audio_audfmt_to_string (audfmt_e fmt)
    318 {
    319     switch (fmt) {
    320     case AUD_FMT_U8:
    321         return "U8";
    322 
    323     case AUD_FMT_U16:
    324         return "U16";
    325 
    326     case AUD_FMT_S8:
    327         return "S8";
    328 
    329     case AUD_FMT_S16:
    330         return "S16";
    331 
    332     case AUD_FMT_U32:
    333         return "U32";
    334 
    335     case AUD_FMT_S32:
    336         return "S32";
    337     }
    338 
    339     dolog ("Bogus audfmt %d returning S16\n", fmt);
    340     return "S16";
    341 }
    342 
    343 static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
    344                                         int *defaultp)
    345 {
    346     if (!strcasecmp (s, "u8")) {
    347         *defaultp = 0;
    348         return AUD_FMT_U8;
    349     }
    350     else if (!strcasecmp (s, "u16")) {
    351         *defaultp = 0;
    352         return AUD_FMT_U16;
    353     }
    354     else if (!strcasecmp (s, "u32")) {
    355         *defaultp = 0;
    356         return AUD_FMT_U32;
    357     }
    358     else if (!strcasecmp (s, "s8")) {
    359         *defaultp = 0;
    360         return AUD_FMT_S8;
    361     }
    362     else if (!strcasecmp (s, "s16")) {
    363         *defaultp = 0;
    364         return AUD_FMT_S16;
    365     }
    366     else if (!strcasecmp (s, "s32")) {
    367         *defaultp = 0;
    368         return AUD_FMT_S32;
    369     }
    370     else {
    371         dolog ("Bogus audio format `%s' using %s\n",
    372                s, audio_audfmt_to_string (defval));
    373         *defaultp = 1;
    374         return defval;
    375     }
    376 }
    377 
    378 static audfmt_e audio_get_conf_fmt (const char *envname,
    379                                     audfmt_e defval,
    380                                     int *defaultp)
    381 {
    382     const char *var = getenv (envname);
    383     if (!var) {
    384         *defaultp = 1;
    385         return defval;
    386     }
    387     return audio_string_to_audfmt (var, defval, defaultp);
    388 }
    389 
    390 static int audio_get_conf_int (const char *key, int defval, int *defaultp)
    391 {
    392     int val;
    393     char *strval;
    394 
    395     strval = getenv (key);
    396     if (strval) {
    397         *defaultp = 0;
    398         val = atoi (strval);
    399         return val;
    400     }
    401     else {
    402         *defaultp = 1;
    403         return defval;
    404     }
    405 }
    406 
    407 static const char *audio_get_conf_str (const char *key,
    408                                        const char *defval,
    409                                        int *defaultp)
    410 {
    411     const char *val = getenv (key);
    412     if (!val) {
    413         *defaultp = 1;
    414         return defval;
    415     }
    416     else {
    417         *defaultp = 0;
    418         return val;
    419     }
    420 }
    421 
    422 /* defined in android_sdl.c */
    423 extern void  dprintn(const char*  fmt, ...);
    424 extern void  dprintnv(const char*  fmt, va_list  args);
    425 
    426 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
    427 {
    428     if (conf.log_to_monitor) {
    429         if (cap) {
    430             monitor_printf(cur_mon, "%s: ", cap);
    431         }
    432 
    433         monitor_vprintf(cur_mon, fmt, ap);
    434     }
    435     else {
    436         if (!VERBOSE_CHECK(audio))
    437             return;
    438 
    439         if (cap) {
    440             dprintn("%s: ", cap);
    441         }
    442 
    443         dprintnv(fmt, ap);
    444     }
    445 }
    446 
    447 void AUD_log (const char *cap, const char *fmt, ...)
    448 {
    449     va_list ap;
    450 
    451     va_start (ap, fmt);
    452     AUD_vlog (cap, fmt, ap);
    453     va_end (ap);
    454 }
    455 
    456 static void audio_print_options (const char *prefix,
    457                                  struct audio_option *opt)
    458 {
    459     char *uprefix;
    460 
    461     if (!prefix) {
    462         dolog ("No prefix specified\n");
    463         return;
    464     }
    465 
    466     if (!opt) {
    467         dolog ("No options\n");
    468         return;
    469     }
    470 
    471     uprefix = audio_alloc_prefix (prefix);
    472 
    473     for (; opt->name; opt++) {
    474         const char *state = "default";
    475         printf ("  %s_%s: ", uprefix, opt->name);
    476 
    477         if (opt->overriddenp && *opt->overriddenp) {
    478             state = "current";
    479         }
    480 
    481         switch (opt->tag) {
    482         case AUD_OPT_BOOL:
    483             {
    484                 int *intp = opt->valp;
    485                 printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
    486             }
    487             break;
    488 
    489         case AUD_OPT_INT:
    490             {
    491                 int *intp = opt->valp;
    492                 printf ("integer, %s = %d\n", state, *intp);
    493             }
    494             break;
    495 
    496         case AUD_OPT_FMT:
    497             {
    498                 audfmt_e *fmtp = opt->valp;
    499                 printf (
    500                     "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
    501                     state,
    502                     audio_audfmt_to_string (*fmtp)
    503                     );
    504             }
    505             break;
    506 
    507         case AUD_OPT_STR:
    508             {
    509                 const char **strp = opt->valp;
    510                 printf ("string, %s = %s\n",
    511                         state,
    512                         *strp ? *strp : "(not set)");
    513             }
    514             break;
    515 
    516         default:
    517             printf ("???\n");
    518             dolog ("Bad value tag for option %s_%s %d\n",
    519                    uprefix, opt->name, opt->tag);
    520             break;
    521         }
    522         printf ("    %s\n", opt->descr);
    523     }
    524 
    525     qemu_free (uprefix);
    526 }
    527 
    528 static void audio_process_options (const char *prefix,
    529                                    struct audio_option *opt)
    530 {
    531     char *optname;
    532     const char qemu_prefix[] = "QEMU_";
    533     size_t preflen, optlen;
    534 
    535     if (audio_bug (AUDIO_FUNC, !prefix)) {
    536         dolog ("prefix = NULL\n");
    537         return;
    538     }
    539 
    540     if (audio_bug (AUDIO_FUNC, !opt)) {
    541         dolog ("opt = NULL\n");
    542         return;
    543     }
    544 
    545     preflen = strlen (prefix);
    546 
    547     for (; opt->name; opt++) {
    548         size_t len, i;
    549         int def;
    550 
    551         if (!opt->valp) {
    552             dolog ("Option value pointer for `%s' is not set\n",
    553                    opt->name);
    554             continue;
    555         }
    556 
    557         len = strlen (opt->name);
    558         /* len of opt->name + len of prefix + size of qemu_prefix
    559          * (includes trailing zero) + zero + underscore (on behalf of
    560          * sizeof) */
    561         optlen = len + preflen + sizeof (qemu_prefix) + 1;
    562         optname = qemu_malloc (optlen);
    563 
    564         pstrcpy (optname, optlen, qemu_prefix);
    565 
    566         /* copy while upper-casing, including trailing zero */
    567         for (i = 0; i <= preflen; ++i) {
    568             optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
    569         }
    570         pstrcat (optname, optlen, "_");
    571         pstrcat (optname, optlen, opt->name);
    572 
    573         def = 1;
    574         switch (opt->tag) {
    575         case AUD_OPT_BOOL:
    576         case AUD_OPT_INT:
    577             {
    578                 int *intp = opt->valp;
    579                 *intp = audio_get_conf_int (optname, *intp, &def);
    580             }
    581             break;
    582 
    583         case AUD_OPT_FMT:
    584             {
    585                 audfmt_e *fmtp = opt->valp;
    586                 *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
    587             }
    588             break;
    589 
    590         case AUD_OPT_STR:
    591             {
    592                 const char **strp = opt->valp;
    593                 *strp = audio_get_conf_str (optname, *strp, &def);
    594             }
    595             break;
    596 
    597         default:
    598             dolog ("Bad value tag for option `%s' - %d\n",
    599                    optname, opt->tag);
    600             break;
    601         }
    602 
    603         if (!opt->overriddenp) {
    604             opt->overriddenp = &opt->overridden;
    605         }
    606         *opt->overriddenp = !def;
    607         qemu_free (optname);
    608     }
    609 }
    610 
    611 static void audio_print_settings (struct audsettings *as)
    612 {
    613     dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
    614 
    615     switch (as->fmt) {
    616     case AUD_FMT_S8:
    617         AUD_log (NULL, "S8");
    618         break;
    619     case AUD_FMT_U8:
    620         AUD_log (NULL, "U8");
    621         break;
    622     case AUD_FMT_S16:
    623         AUD_log (NULL, "S16");
    624         break;
    625     case AUD_FMT_U16:
    626         AUD_log (NULL, "U16");
    627         break;
    628     case AUD_FMT_S32:
    629         AUD_log (NULL, "S32");
    630         break;
    631     case AUD_FMT_U32:
    632         AUD_log (NULL, "U32");
    633         break;
    634     default:
    635         AUD_log (NULL, "invalid(%d)", as->fmt);
    636         break;
    637     }
    638 
    639     AUD_log (NULL, " endianness=");
    640     switch (as->endianness) {
    641     case 0:
    642         AUD_log (NULL, "little");
    643         break;
    644     case 1:
    645         AUD_log (NULL, "big");
    646         break;
    647     default:
    648         AUD_log (NULL, "invalid");
    649         break;
    650     }
    651     AUD_log (NULL, "\n");
    652 }
    653 
    654 static int audio_validate_settings (struct audsettings *as)
    655 {
    656     int invalid;
    657 
    658     invalid = as->nchannels != 1 && as->nchannels != 2;
    659     invalid |= as->endianness != 0 && as->endianness != 1;
    660 
    661     switch (as->fmt) {
    662     case AUD_FMT_S8:
    663     case AUD_FMT_U8:
    664     case AUD_FMT_S16:
    665     case AUD_FMT_U16:
    666     case AUD_FMT_S32:
    667     case AUD_FMT_U32:
    668         break;
    669     default:
    670         invalid = 1;
    671         break;
    672     }
    673 
    674     invalid |= as->freq <= 0;
    675     return invalid ? -1 : 0;
    676 }
    677 
    678 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
    679 {
    680     int bits = 8, sign = 0;
    681 
    682     switch (as->fmt) {
    683     case AUD_FMT_S8:
    684         sign = 1;
    685     case AUD_FMT_U8:
    686         break;
    687 
    688     case AUD_FMT_S16:
    689         sign = 1;
    690     case AUD_FMT_U16:
    691         bits = 16;
    692         break;
    693 
    694     case AUD_FMT_S32:
    695         sign = 1;
    696     case AUD_FMT_U32:
    697         bits = 32;
    698         break;
    699     }
    700     return info->freq == as->freq
    701         && info->nchannels == as->nchannels
    702         && info->sign == sign
    703         && info->bits == bits
    704         && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
    705 }
    706 
    707 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
    708 {
    709     int bits = 8, sign = 0, shift = 0;
    710 
    711     switch (as->fmt) {
    712     case AUD_FMT_S8:
    713         sign = 1;
    714     case AUD_FMT_U8:
    715         break;
    716 
    717     case AUD_FMT_S16:
    718         sign = 1;
    719     case AUD_FMT_U16:
    720         bits = 16;
    721         shift = 1;
    722         break;
    723 
    724     case AUD_FMT_S32:
    725         sign = 1;
    726     case AUD_FMT_U32:
    727         bits = 32;
    728         shift = 2;
    729         break;
    730     }
    731 
    732     info->freq = as->freq;
    733     info->bits = bits;
    734     info->sign = sign;
    735     info->nchannels = as->nchannels;
    736     info->shift = (as->nchannels == 2) + shift;
    737     info->align = (1 << info->shift) - 1;
    738     info->bytes_per_second = info->freq << info->shift;
    739     info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
    740 }
    741 
    742 void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
    743 {
    744     if (!len) {
    745         return;
    746     }
    747 
    748     if (info->sign) {
    749         memset (buf, 0x00, len << info->shift);
    750     }
    751     else {
    752         switch (info->bits) {
    753         case 8:
    754             memset (buf, 0x80, len << info->shift);
    755             break;
    756 
    757         case 16:
    758             {
    759                 int i;
    760                 uint16_t *p = buf;
    761                 int shift = info->nchannels - 1;
    762                 short s = INT16_MAX;
    763 
    764                 if (info->swap_endianness) {
    765                     s = bswap16 (s);
    766                 }
    767 
    768                 for (i = 0; i < len << shift; i++) {
    769                     p[i] = s;
    770                 }
    771             }
    772             break;
    773 
    774         case 32:
    775             {
    776                 int i;
    777                 uint32_t *p = buf;
    778                 int shift = info->nchannels - 1;
    779                 int32_t s = INT32_MAX;
    780 
    781                 if (info->swap_endianness) {
    782                     s = bswap32 (s);
    783                 }
    784 
    785                 for (i = 0; i < len << shift; i++) {
    786                     p[i] = s;
    787                 }
    788             }
    789             break;
    790 
    791         default:
    792             AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
    793                      info->bits);
    794             break;
    795         }
    796     }
    797 }
    798 
    799 /*
    800  * Capture
    801  */
    802 static void noop_conv (struct st_sample *dst, const void *src,
    803                        int samples, struct mixeng_volume *vol)
    804 {
    805     (void) src;
    806     (void) dst;
    807     (void) samples;
    808     (void) vol;
    809 }
    810 
    811 static CaptureVoiceOut *audio_pcm_capture_find_specific (
    812     struct audsettings *as
    813     )
    814 {
    815     CaptureVoiceOut *cap;
    816     AudioState *s = &glob_audio_state;
    817 
    818     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
    819         if (audio_pcm_info_eq (&cap->hw.info, as)) {
    820             return cap;
    821         }
    822     }
    823     return NULL;
    824 }
    825 
    826 static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
    827 {
    828     struct capture_callback *cb;
    829 
    830 #ifdef DEBUG_CAPTURE
    831     dolog ("notification %d sent\n", cmd);
    832 #endif
    833     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
    834         cb->ops.notify (cb->opaque, cmd);
    835     }
    836 }
    837 
    838 static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
    839 {
    840     if (cap->hw.enabled != enabled) {
    841         audcnotification_e cmd;
    842         cap->hw.enabled = enabled;
    843         cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
    844         audio_notify_capture (cap, cmd);
    845     }
    846 }
    847 
    848 static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
    849 {
    850     HWVoiceOut *hw = &cap->hw;
    851     SWVoiceOut *sw;
    852     int enabled = 0;
    853 
    854     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
    855         if (sw->active) {
    856             enabled = 1;
    857             break;
    858         }
    859     }
    860     audio_capture_maybe_changed (cap, enabled);
    861 }
    862 
    863 static void audio_detach_capture (HWVoiceOut *hw)
    864 {
    865     SWVoiceCap *sc = hw->cap_head.lh_first;
    866 
    867     while (sc) {
    868         SWVoiceCap *sc1 = sc->entries.le_next;
    869         SWVoiceOut *sw = &sc->sw;
    870         CaptureVoiceOut *cap = sc->cap;
    871         int was_active = sw->active;
    872 
    873         if (sw->rate) {
    874             st_rate_stop (sw->rate);
    875             sw->rate = NULL;
    876         }
    877 
    878         QLIST_REMOVE (sw, entries);
    879         QLIST_REMOVE (sc, entries);
    880         qemu_free (sc);
    881         if (was_active) {
    882             /* We have removed soft voice from the capture:
    883                this might have changed the overall status of the capture
    884                since this might have been the only active voice */
    885             audio_recalc_and_notify_capture (cap);
    886         }
    887         sc = sc1;
    888     }
    889 }
    890 
    891 static int audio_attach_capture (HWVoiceOut *hw)
    892 {
    893     AudioState *s = &glob_audio_state;
    894     CaptureVoiceOut *cap;
    895 
    896     audio_detach_capture (hw);
    897     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
    898         SWVoiceCap *sc;
    899         SWVoiceOut *sw;
    900         HWVoiceOut *hw_cap = &cap->hw;
    901 
    902         sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
    903         if (!sc) {
    904             dolog ("Could not allocate soft capture voice (%zu bytes)\n",
    905                    sizeof (*sc));
    906             return -1;
    907         }
    908 
    909         sc->cap = cap;
    910         sw = &sc->sw;
    911         sw->hw = hw_cap;
    912         sw->info = hw->info;
    913         sw->empty = 1;
    914         sw->active = hw->enabled;
    915         sw->conv = noop_conv;
    916         sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
    917         sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
    918         if (!sw->rate) {
    919             dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
    920             qemu_free (sw);
    921             return -1;
    922         }
    923         QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
    924         QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
    925 #ifdef DEBUG_CAPTURE
    926         asprintf (&sw->name, "for %p %d,%d,%d",
    927                   hw, sw->info.freq, sw->info.bits, sw->info.nchannels);
    928         dolog ("Added %s active = %d\n", sw->name, sw->active);
    929 #endif
    930         if (sw->active) {
    931             audio_capture_maybe_changed (cap, 1);
    932         }
    933     }
    934     return 0;
    935 }
    936 
    937 /*
    938  * Hard voice (capture)
    939  */
    940 static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
    941 {
    942     SWVoiceIn *sw;
    943     int m = hw->total_samples_captured;
    944 
    945     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
    946         if (sw->active) {
    947             m = audio_MIN (m, sw->total_hw_samples_acquired);
    948         }
    949     }
    950     return m;
    951 }
    952 
    953 int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
    954 {
    955     int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
    956     if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
    957         dolog ("live=%d hw->samples=%d\n", live, hw->samples);
    958         return 0;
    959     }
    960     return live;
    961 }
    962 
    963 int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
    964                            int live, int pending)
    965 {
    966     int left = hw->samples - pending;
    967     int len = audio_MIN (left, live);
    968     int clipped = 0;
    969 
    970     while (len) {
    971         struct st_sample *src = hw->mix_buf + hw->rpos;
    972         uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
    973         int samples_till_end_of_buf = hw->samples - hw->rpos;
    974         int samples_to_clip = audio_MIN (len, samples_till_end_of_buf);
    975 
    976         hw->clip (dst, src, samples_to_clip);
    977 
    978         hw->rpos = (hw->rpos + samples_to_clip) % hw->samples;
    979         len -= samples_to_clip;
    980         clipped += samples_to_clip;
    981     }
    982     return clipped;
    983 }
    984 
    985 /*
    986  * Soft voice (capture)
    987  */
    988 static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
    989 {
    990     HWVoiceIn *hw = sw->hw;
    991     int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
    992     int rpos;
    993 
    994     if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
    995         dolog ("live=%d hw->samples=%d\n", live, hw->samples);
    996         return 0;
    997     }
    998 
    999     rpos = hw->wpos - live;
   1000     if (rpos >= 0) {
   1001         return rpos;
   1002     }
   1003     else {
   1004         return hw->samples + rpos;
   1005     }
   1006 }
   1007 
   1008 int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
   1009 {
   1010     HWVoiceIn *hw = sw->hw;
   1011     int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
   1012     struct st_sample *src, *dst = sw->buf;
   1013 
   1014     rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
   1015 
   1016     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
   1017     if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
   1018         dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
   1019         return 0;
   1020     }
   1021 
   1022     samples = size >> sw->info.shift;
   1023     if (!live) {
   1024         return 0;
   1025     }
   1026 
   1027     swlim = (live * sw->ratio) >> 32;
   1028     swlim = audio_MIN (swlim, samples);
   1029 
   1030     while (swlim) {
   1031         src = hw->conv_buf + rpos;
   1032         isamp = hw->wpos - rpos;
   1033         /* XXX: <= ? */
   1034         if (isamp <= 0) {
   1035             isamp = hw->samples - rpos;
   1036         }
   1037 
   1038         if (!isamp) {
   1039             break;
   1040         }
   1041         osamp = swlim;
   1042 
   1043         if (audio_bug (AUDIO_FUNC, osamp < 0)) {
   1044             dolog ("osamp=%d\n", osamp);
   1045             return 0;
   1046         }
   1047 
   1048         st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
   1049         swlim -= osamp;
   1050         rpos = (rpos + isamp) % hw->samples;
   1051         dst += osamp;
   1052         ret += osamp;
   1053         total += isamp;
   1054     }
   1055 
   1056     sw->clip (buf, sw->buf, ret);
   1057     sw->total_hw_samples_acquired += total;
   1058     return ret << sw->info.shift;
   1059 }
   1060 
   1061 /*
   1062  * Hard voice (playback)
   1063  */
   1064 static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
   1065 {
   1066     SWVoiceOut *sw;
   1067     int m = INT_MAX;
   1068     int nb_live = 0;
   1069 
   1070     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
   1071         if (sw->active && !sw->empty) {
   1072             m = audio_MIN (m, sw->total_hw_samples_mixed);
   1073             nb_live += 1;
   1074         }
   1075     }
   1076 
   1077     *nb_livep = nb_live;
   1078     return m;
   1079 }
   1080 
   1081 static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
   1082 {
   1083     int smin;
   1084     int nb_live1;
   1085 
   1086     smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
   1087     if (nb_live) {
   1088         *nb_live = nb_live1;
   1089     }
   1090 
   1091     if (nb_live1) {
   1092         int live = smin;
   1093 
   1094         if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
   1095             dolog ("live=%d hw->samples=%d\n", live, hw->samples);
   1096             return 0;
   1097         }
   1098         return live;
   1099     }
   1100     return 0;
   1101 }
   1102 
   1103 /*
   1104  * Soft voice (playback)
   1105  */
   1106 int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
   1107 {
   1108     int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
   1109     int ret = 0, pos = 0, total = 0;
   1110 
   1111     if (!sw) {
   1112         return size;
   1113     }
   1114 
   1115     hwsamples = sw->hw->samples;
   1116 
   1117     live = sw->total_hw_samples_mixed;
   1118     if (audio_bug (AUDIO_FUNC, live < 0 || live > hwsamples)){
   1119         dolog ("live=%d hw->samples=%d\n", live, hwsamples);
   1120         return 0;
   1121     }
   1122 
   1123     if (live == hwsamples) {
   1124 #ifdef DEBUG_OUT
   1125         dolog ("%s is full %d\n", sw->name, live);
   1126 #endif
   1127         return 0;
   1128     }
   1129 
   1130     wpos = (sw->hw->rpos + live) % hwsamples;
   1131     samples = size >> sw->info.shift;
   1132 
   1133     dead = hwsamples - live;
   1134     swlim = ((int64_t) dead << 32) / sw->ratio;
   1135     swlim = audio_MIN (swlim, samples);
   1136     if (swlim) {
   1137         sw->conv (sw->buf, buf, swlim, &sw->vol);
   1138     }
   1139 
   1140     while (swlim) {
   1141         dead = hwsamples - live;
   1142         left = hwsamples - wpos;
   1143         blck = audio_MIN (dead, left);
   1144         if (!blck) {
   1145             break;
   1146         }
   1147         isamp = swlim;
   1148         osamp = blck;
   1149         st_rate_flow_mix (
   1150             sw->rate,
   1151             sw->buf + pos,
   1152             sw->hw->mix_buf + wpos,
   1153             &isamp,
   1154             &osamp
   1155             );
   1156         ret += isamp;
   1157         swlim -= isamp;
   1158         pos += isamp;
   1159         live += osamp;
   1160         wpos = (wpos + osamp) % hwsamples;
   1161         total += osamp;
   1162     }
   1163 
   1164     sw->total_hw_samples_mixed += total;
   1165     sw->empty = sw->total_hw_samples_mixed == 0;
   1166 
   1167 #ifdef DEBUG_OUT
   1168     dolog (
   1169         "%s: write size %d ret %d total sw %d\n",
   1170         SW_NAME (sw),
   1171         size >> sw->info.shift,
   1172         ret,
   1173         sw->total_hw_samples_mixed
   1174         );
   1175 #endif
   1176 
   1177     return ret << sw->info.shift;
   1178 }
   1179 
   1180 #ifdef DEBUG_AUDIO
   1181 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
   1182 {
   1183     dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
   1184            cap, info->bits, info->sign, info->freq, info->nchannels);
   1185 }
   1186 #endif
   1187 
   1188 #define DAC
   1189 #include "audio_template.h"
   1190 #undef DAC
   1191 #include "audio_template.h"
   1192 
   1193 /*
   1194  * Timer
   1195  */
   1196 static void audio_timer (void *opaque)
   1197 {
   1198     AudioState *s = opaque;
   1199 #if 0
   1200 #define  MAX_DIFFS  100
   1201     int64_t         now  = qemu_get_clock_ms(vm_clock);
   1202     static int64_t  last = 0;
   1203     static float    diffs[MAX_DIFFS];
   1204     static int      num_diffs;
   1205 
   1206     if (last == 0)
   1207         last = now;
   1208     else {
   1209         diffs[num_diffs] = (float)((now-last)/1e6);  /* last diff in ms */
   1210         if (++num_diffs == MAX_DIFFS) {
   1211             double  min_diff = 1e6, max_diff = -1e6;
   1212             double  all_diff = 0.;
   1213             int     nn;
   1214 
   1215             for (nn = 0; nn < num_diffs; nn++) {
   1216                 if (diffs[nn] < min_diff) min_diff = diffs[nn];
   1217                 if (diffs[nn] > max_diff) max_diff = diffs[nn];
   1218                 all_diff += diffs[nn];
   1219             }
   1220             all_diff *= 1.0/num_diffs;
   1221             printf("audio timer: min_diff=%6.2g max_diff=%6.2g avg_diff=%6.2g samples=%d\n",
   1222                             min_diff, max_diff, all_diff, num_diffs);
   1223             num_diffs = 0;
   1224         }
   1225     }
   1226     last = now;
   1227 #endif
   1228 
   1229     audio_run ("timer");
   1230     qemu_mod_timer (s->ts, qemu_get_clock_ns (vm_clock) + conf.period.ticks);
   1231 }
   1232 
   1233 
   1234 static int audio_is_timer_needed (void)
   1235 {
   1236     HWVoiceIn *hwi = NULL;
   1237     HWVoiceOut *hwo = NULL;
   1238 
   1239     while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
   1240         if (!hwo->poll_mode) return 1;
   1241     }
   1242     while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
   1243         if (!hwi->poll_mode) return 1;
   1244     }
   1245     return 0;
   1246 }
   1247 
   1248 static void audio_reset_timer (void)
   1249 {
   1250     AudioState *s = &glob_audio_state;
   1251 
   1252     if (audio_is_timer_needed ()) {
   1253         qemu_mod_timer (s->ts, qemu_get_clock_ns (vm_clock) + 1);
   1254     }
   1255     else {
   1256         qemu_del_timer (s->ts);
   1257     }
   1258 }
   1259 
   1260 /*
   1261  * Public API
   1262  */
   1263 int AUD_write (SWVoiceOut *sw, void *buf, int size)
   1264 {
   1265     int bytes;
   1266 
   1267     if (!sw) {
   1268         /* XXX: Consider options */
   1269         return size;
   1270     }
   1271 
   1272     if (!sw->hw->enabled) {
   1273         dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
   1274         return 0;
   1275     }
   1276 
   1277     bytes = sw->hw->pcm_ops->write (sw, buf, size);
   1278     return bytes;
   1279 }
   1280 
   1281 int AUD_read (SWVoiceIn *sw, void *buf, int size)
   1282 {
   1283     int bytes;
   1284 
   1285     if (!sw) {
   1286         /* XXX: Consider options */
   1287         return size;
   1288     }
   1289 
   1290     if (!sw->hw->enabled) {
   1291         dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
   1292         return 0;
   1293     }
   1294 
   1295     bytes = sw->hw->pcm_ops->read (sw, buf, size);
   1296     return bytes;
   1297 }
   1298 
   1299 int AUD_get_buffer_size_out (SWVoiceOut *sw)
   1300 {
   1301     return sw->hw->samples << sw->hw->info.shift;
   1302 }
   1303 
   1304 void AUD_set_active_out (SWVoiceOut *sw, int on)
   1305 {
   1306     HWVoiceOut *hw;
   1307 
   1308     if (!sw) {
   1309         return;
   1310     }
   1311 
   1312     hw = sw->hw;
   1313     if (sw->active != on) {
   1314         AudioState *s = &glob_audio_state;
   1315         SWVoiceOut *temp_sw;
   1316         SWVoiceCap *sc;
   1317 
   1318         if (on) {
   1319             hw->pending_disable = 0;
   1320             if (!hw->enabled) {
   1321                 hw->enabled = 1;
   1322                 if (s->vm_running) {
   1323                     hw->pcm_ops->ctl_out (hw, VOICE_ENABLE, conf.try_poll_out);
   1324                     audio_reset_timer ();
   1325                 }
   1326             }
   1327         }
   1328         else {
   1329             if (hw->enabled) {
   1330                 int nb_active = 0;
   1331 
   1332                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
   1333                      temp_sw = temp_sw->entries.le_next) {
   1334                     nb_active += temp_sw->active != 0;
   1335                 }
   1336 
   1337                 hw->pending_disable = nb_active == 1;
   1338             }
   1339         }
   1340 
   1341         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
   1342             sc->sw.active = hw->enabled;
   1343             if (hw->enabled) {
   1344                 audio_capture_maybe_changed (sc->cap, 1);
   1345             }
   1346         }
   1347         sw->active = on;
   1348     }
   1349 }
   1350 
   1351 void AUD_set_active_in (SWVoiceIn *sw, int on)
   1352 {
   1353     HWVoiceIn *hw;
   1354 
   1355     if (!sw) {
   1356         return;
   1357     }
   1358 
   1359     hw = sw->hw;
   1360     if (sw->active != on) {
   1361         AudioState *s = &glob_audio_state;
   1362         SWVoiceIn *temp_sw;
   1363 
   1364         if (on) {
   1365             if (!hw->enabled) {
   1366                 hw->enabled = 1;
   1367                 if (s->vm_running) {
   1368                     hw->pcm_ops->ctl_in (hw, VOICE_ENABLE, conf.try_poll_in);
   1369                 }
   1370             }
   1371             sw->total_hw_samples_acquired = hw->total_samples_captured;
   1372         }
   1373         else {
   1374             if (hw->enabled) {
   1375                 int nb_active = 0;
   1376 
   1377                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
   1378                      temp_sw = temp_sw->entries.le_next) {
   1379                     nb_active += temp_sw->active != 0;
   1380                 }
   1381 
   1382                 if (nb_active == 1) {
   1383                     hw->enabled = 0;
   1384                     hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
   1385                 }
   1386             }
   1387         }
   1388         sw->active = on;
   1389     }
   1390 }
   1391 
   1392 static int audio_get_avail (SWVoiceIn *sw)
   1393 {
   1394     int live;
   1395 
   1396     if (!sw) {
   1397         return 0;
   1398     }
   1399 
   1400     live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
   1401     if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
   1402         dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
   1403         return 0;
   1404     }
   1405 
   1406     ldebug (
   1407         "%s: get_avail live %d ret %" PRId64 "\n",
   1408         SW_NAME (sw),
   1409         live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
   1410         );
   1411 
   1412     return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
   1413 }
   1414 
   1415 static int audio_get_free (SWVoiceOut *sw)
   1416 {
   1417     int live, dead;
   1418 
   1419     if (!sw) {
   1420         return 0;
   1421     }
   1422 
   1423     live = sw->total_hw_samples_mixed;
   1424 
   1425     if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
   1426         dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
   1427         return 0;
   1428     }
   1429 
   1430     dead = sw->hw->samples - live;
   1431 
   1432 #ifdef DEBUG_OUT
   1433     dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
   1434            SW_NAME (sw),
   1435            live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
   1436 #endif
   1437 
   1438     return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
   1439 }
   1440 
   1441 static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
   1442 {
   1443     int n;
   1444 
   1445     if (hw->enabled) {
   1446         SWVoiceCap *sc;
   1447 
   1448         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
   1449             SWVoiceOut *sw = &sc->sw;
   1450             int rpos2 = rpos;
   1451 
   1452             n = samples;
   1453             while (n) {
   1454                 int till_end_of_hw = hw->samples - rpos2;
   1455                 int to_write = audio_MIN (till_end_of_hw, n);
   1456                 int bytes = to_write << hw->info.shift;
   1457                 int written;
   1458 
   1459                 sw->buf = hw->mix_buf + rpos2;
   1460                 written = audio_pcm_sw_write (sw, NULL, bytes);
   1461                 if (written - bytes) {
   1462                     dolog ("Could not mix %d bytes into a capture "
   1463                            "buffer, mixed %d\n",
   1464                            bytes, written);
   1465                     break;
   1466                 }
   1467                 n -= to_write;
   1468                 rpos2 = (rpos2 + to_write) % hw->samples;
   1469             }
   1470         }
   1471     }
   1472 
   1473     n = audio_MIN (samples, hw->samples - rpos);
   1474     mixeng_clear (hw->mix_buf + rpos, n);
   1475     mixeng_clear (hw->mix_buf, samples - n);
   1476 }
   1477 
   1478 static void audio_run_out (AudioState *s)
   1479 {
   1480     HWVoiceOut *hw = NULL;
   1481     SWVoiceOut *sw;
   1482 
   1483     while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) {
   1484         int played;
   1485         int live, free, nb_live, cleanup_required, prev_rpos;
   1486 
   1487         live = audio_pcm_hw_get_live_out (hw, &nb_live);
   1488         if (!nb_live) {
   1489             live = 0;
   1490         }
   1491 
   1492         if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
   1493             dolog ("live=%d hw->samples=%d\n", live, hw->samples);
   1494             continue;
   1495         }
   1496 
   1497         if (hw->pending_disable && !nb_live) {
   1498             SWVoiceCap *sc;
   1499 #ifdef DEBUG_OUT
   1500             dolog ("Disabling voice\n");
   1501 #endif
   1502             hw->enabled = 0;
   1503             hw->pending_disable = 0;
   1504             hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
   1505             for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
   1506                 sc->sw.active = 0;
   1507                 audio_recalc_and_notify_capture (sc->cap);
   1508             }
   1509             continue;
   1510         }
   1511 
   1512         if (!live) {
   1513             for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
   1514                 if (sw->active) {
   1515                     free = audio_get_free (sw);
   1516                     if (free > 0) {
   1517                         sw->callback.fn (sw->callback.opaque, free);
   1518                     }
   1519                 }
   1520             }
   1521             continue;
   1522         }
   1523 
   1524         prev_rpos = hw->rpos;
   1525         played = hw->pcm_ops->run_out (hw, live);
   1526         if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
   1527             dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
   1528                    hw->rpos, hw->samples, played);
   1529             hw->rpos = 0;
   1530         }
   1531 
   1532 #ifdef DEBUG_OUT
   1533         dolog ("played=%d\n", played);
   1534 #endif
   1535 
   1536         if (played) {
   1537             hw->ts_helper += played;
   1538             audio_capture_mix_and_clear (hw, prev_rpos, played);
   1539         }
   1540 
   1541         cleanup_required = 0;
   1542         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
   1543             if (!sw->active || sw->empty) {
   1544                 continue;
   1545             }
   1546 
   1547             if (audio_bug (AUDIO_FUNC, played > sw->total_hw_samples_mixed)) {
   1548                 dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
   1549                        played, sw->total_hw_samples_mixed);
   1550                 played = sw->total_hw_samples_mixed;
   1551             }
   1552 
   1553             sw->total_hw_samples_mixed -= played;
   1554 
   1555             if (!sw->total_hw_samples_mixed) {
   1556                 sw->empty = 1;
   1557                 cleanup_required |= !sw->active && !sw->callback.fn;
   1558             }
   1559 
   1560             if (sw->active) {
   1561                 free = audio_get_free (sw);
   1562                 if (free > 0) {
   1563                     sw->callback.fn (sw->callback.opaque, free);
   1564                 }
   1565             }
   1566         }
   1567 
   1568         if (cleanup_required) {
   1569             SWVoiceOut *sw1;
   1570 
   1571             sw = hw->sw_head.lh_first;
   1572             while (sw) {
   1573                 sw1 = sw->entries.le_next;
   1574                 if (!sw->active && !sw->callback.fn) {
   1575 #ifdef DEBUG_PLIVE
   1576                     dolog ("Finishing with old voice\n");
   1577 #endif
   1578                     audio_close_out (sw);
   1579                 }
   1580                 sw = sw1;
   1581             }
   1582         }
   1583     }
   1584 }
   1585 
   1586 static void audio_run_in (AudioState *s)
   1587 {
   1588     HWVoiceIn *hw = NULL;
   1589 
   1590     while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
   1591         SWVoiceIn *sw;
   1592         int captured, min;
   1593 
   1594         captured = hw->pcm_ops->run_in (hw);
   1595 
   1596         min = audio_pcm_hw_find_min_in (hw);
   1597         hw->total_samples_captured += captured - min;
   1598         hw->ts_helper += captured;
   1599 
   1600         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
   1601             sw->total_hw_samples_acquired -= min;
   1602 
   1603             if (sw->active) {
   1604                 int avail;
   1605 
   1606                 avail = audio_get_avail (sw);
   1607                 if (avail > 0) {
   1608                     sw->callback.fn (sw->callback.opaque, avail);
   1609                 }
   1610             }
   1611         }
   1612     }
   1613 }
   1614 
   1615 static void audio_run_capture (AudioState *s)
   1616 {
   1617     CaptureVoiceOut *cap;
   1618 
   1619     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
   1620         int live, rpos, captured;
   1621         HWVoiceOut *hw = &cap->hw;
   1622         SWVoiceOut *sw;
   1623 
   1624         captured = live = audio_pcm_hw_get_live_out (hw, NULL);
   1625         rpos = hw->rpos;
   1626         while (live) {
   1627             int left = hw->samples - rpos;
   1628             int to_capture = audio_MIN (live, left);
   1629             struct st_sample *src;
   1630             struct capture_callback *cb;
   1631 
   1632             src = hw->mix_buf + rpos;
   1633             hw->clip (cap->buf, src, to_capture);
   1634             mixeng_clear (src, to_capture);
   1635 
   1636             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
   1637                 cb->ops.capture (cb->opaque, cap->buf,
   1638                                  to_capture << hw->info.shift);
   1639             }
   1640             rpos = (rpos + to_capture) % hw->samples;
   1641             live -= to_capture;
   1642         }
   1643         hw->rpos = rpos;
   1644 
   1645         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
   1646             if (!sw->active && sw->empty) {
   1647                 continue;
   1648             }
   1649 
   1650             if (audio_bug (AUDIO_FUNC, captured > sw->total_hw_samples_mixed)) {
   1651                 dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
   1652                        captured, sw->total_hw_samples_mixed);
   1653                 captured = sw->total_hw_samples_mixed;
   1654             }
   1655 
   1656             sw->total_hw_samples_mixed -= captured;
   1657             sw->empty = sw->total_hw_samples_mixed == 0;
   1658         }
   1659     }
   1660 }
   1661 
   1662 void audio_run (const char *msg)
   1663 {
   1664     AudioState *s = &glob_audio_state;
   1665 
   1666     audio_run_out (s);
   1667     audio_run_in (s);
   1668     audio_run_capture (s);
   1669 #ifdef DEBUG_POLL
   1670     {
   1671         static double prevtime;
   1672         double currtime;
   1673         struct timeval tv;
   1674 
   1675         if (gettimeofday (&tv, NULL)) {
   1676             perror ("audio_run: gettimeofday");
   1677             return;
   1678         }
   1679 
   1680         currtime = tv.tv_sec + tv.tv_usec * 1e-6;
   1681         dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
   1682         prevtime = currtime;
   1683     }
   1684 #endif
   1685 }
   1686 
   1687 static struct audio_option audio_options[] = {
   1688     /* DAC */
   1689     {
   1690         .name  = "DAC_FIXED_SETTINGS",
   1691         .tag   = AUD_OPT_BOOL,
   1692         .valp  = &conf.fixed_out.enabled,
   1693         .descr = "Use fixed settings for host DAC"
   1694     },
   1695     {
   1696         .name  = "DAC_FIXED_FREQ",
   1697         .tag   = AUD_OPT_INT,
   1698         .valp  = &conf.fixed_out.settings.freq,
   1699         .descr = "Frequency for fixed host DAC"
   1700     },
   1701     {
   1702         .name  = "DAC_FIXED_FMT",
   1703         .tag   = AUD_OPT_FMT,
   1704         .valp  = &conf.fixed_out.settings.fmt,
   1705         .descr = "Format for fixed host DAC"
   1706     },
   1707     {
   1708         .name  = "DAC_FIXED_CHANNELS",
   1709         .tag   = AUD_OPT_INT,
   1710         .valp  = &conf.fixed_out.settings.nchannels,
   1711         .descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)"
   1712     },
   1713     {
   1714         .name  = "DAC_VOICES",
   1715         .tag   = AUD_OPT_INT,
   1716         .valp  = &conf.fixed_out.nb_voices,
   1717         .descr = "Number of voices for DAC"
   1718     },
   1719     {
   1720         .name  = "DAC_TRY_POLL",
   1721         .tag   = AUD_OPT_BOOL,
   1722         .valp  = &conf.try_poll_out,
   1723         .descr = "Attempt using poll mode for DAC"
   1724     },
   1725     /* ADC */
   1726     {
   1727         .name  = "ADC_FIXED_SETTINGS",
   1728         .tag   = AUD_OPT_BOOL,
   1729         .valp  = &conf.fixed_in.enabled,
   1730         .descr = "Use fixed settings for host ADC"
   1731     },
   1732     {
   1733         .name  = "ADC_FIXED_FREQ",
   1734         .tag   = AUD_OPT_INT,
   1735         .valp  = &conf.fixed_in.settings.freq,
   1736         .descr = "Frequency for fixed host ADC"
   1737     },
   1738     {
   1739         .name  = "ADC_FIXED_FMT",
   1740         .tag   = AUD_OPT_FMT,
   1741         .valp  = &conf.fixed_in.settings.fmt,
   1742         .descr = "Format for fixed host ADC"
   1743     },
   1744     {
   1745         .name  = "ADC_FIXED_CHANNELS",
   1746         .tag   = AUD_OPT_INT,
   1747         .valp  = &conf.fixed_in.settings.nchannels,
   1748         .descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)"
   1749     },
   1750     {
   1751         .name  = "ADC_VOICES",
   1752         .tag   = AUD_OPT_INT,
   1753         .valp  = &conf.fixed_in.nb_voices,
   1754         .descr = "Number of voices for ADC"
   1755     },
   1756     {
   1757         .name  = "ADC_TRY_POLL",
   1758         .tag   = AUD_OPT_BOOL,
   1759         .valp  = &conf.try_poll_in,
   1760         .descr = "Attempt using poll mode for ADC"
   1761     },
   1762     /* Misc */
   1763     {
   1764         .name  = "TIMER_PERIOD",
   1765         .tag   = AUD_OPT_INT,
   1766         .valp  = &conf.period.hertz,
   1767         .descr = "Timer period in HZ (0 - use lowest possible)"
   1768     },
   1769     {
   1770         .name  = "PLIVE",
   1771         .tag   = AUD_OPT_BOOL,
   1772         .valp  = &conf.plive,
   1773         .descr = "(undocumented)"
   1774     },
   1775     {
   1776         .name  = "LOG_TO_MONITOR",
   1777         .tag   = AUD_OPT_BOOL,
   1778         .valp  = &conf.log_to_monitor,
   1779         .descr = "Print logging messages to monitor instead of stderr"
   1780     },
   1781     { /* End of list */ }
   1782 };
   1783 
   1784 static void audio_pp_nb_voices (const char *typ, int nb)
   1785 {
   1786     switch (nb) {
   1787     case 0:
   1788         printf ("Does not support %s\n", typ);
   1789         break;
   1790     case 1:
   1791         printf ("One %s voice\n", typ);
   1792         break;
   1793     case INT_MAX:
   1794         printf ("Theoretically supports many %s voices\n", typ);
   1795         break;
   1796     default:
   1797         printf ("Theoretically supports upto %d %s voices\n", nb, typ);
   1798         break;
   1799     }
   1800 
   1801 }
   1802 
   1803 void AUD_help (void)
   1804 {
   1805     size_t i;
   1806 
   1807     audio_process_options ("AUDIO", audio_options);
   1808     for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
   1809         struct audio_driver *d = drvtab[i];
   1810         if (d->options) {
   1811             audio_process_options (d->name, d->options);
   1812         }
   1813     }
   1814 
   1815     printf ("Audio options:\n");
   1816     audio_print_options ("AUDIO", audio_options);
   1817     printf ("\n");
   1818 
   1819     printf ("Available drivers:\n");
   1820 
   1821     for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
   1822         struct audio_driver *d = drvtab[i];
   1823 
   1824         printf ("Name: %s\n", d->name);
   1825         printf ("Description: %s\n", d->descr);
   1826 
   1827         audio_pp_nb_voices ("playback", d->max_voices_out);
   1828         audio_pp_nb_voices ("capture", d->max_voices_in);
   1829 
   1830         if (d->options) {
   1831             printf ("Options:\n");
   1832             audio_print_options (d->name, d->options);
   1833         }
   1834         else {
   1835             printf ("No options\n");
   1836         }
   1837         printf ("\n");
   1838     }
   1839 
   1840     printf (
   1841         "Options are settable through environment variables.\n"
   1842         "Example:\n"
   1843 #ifdef _WIN32
   1844         "  set QEMU_AUDIO_DRV=wav\n"
   1845         "  set QEMU_WAV_PATH=c:\\tune.wav\n"
   1846 #else
   1847         "  export QEMU_AUDIO_DRV=wav\n"
   1848         "  export QEMU_WAV_PATH=$HOME/tune.wav\n"
   1849         "(for csh replace export with setenv in the above)\n"
   1850 #endif
   1851         "  qemu ...\n\n"
   1852         );
   1853 }
   1854 
   1855 static int audio_driver_init (AudioState *s, struct audio_driver *drv)
   1856 {
   1857     if (drv->options) {
   1858         audio_process_options (drv->name, drv->options);
   1859     }
   1860     s->drv_opaque = drv->init ();
   1861 
   1862     if (s->drv_opaque) {
   1863         audio_init_nb_voices_out (drv);
   1864         audio_init_nb_voices_in (drv);
   1865         s->drv = drv;
   1866         return 0;
   1867     }
   1868     else {
   1869         dolog ("Could not init `%s' audio driver\n", drv->name);
   1870         return -1;
   1871     }
   1872 }
   1873 
   1874 static void audio_vm_change_state_handler (void *opaque, int running,
   1875                                            int reason)
   1876 {
   1877     AudioState *s = opaque;
   1878     HWVoiceOut *hwo = NULL;
   1879     HWVoiceIn *hwi = NULL;
   1880     int op = running ? VOICE_ENABLE : VOICE_DISABLE;
   1881 
   1882     s->vm_running = running;
   1883     while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
   1884         hwo->pcm_ops->ctl_out (hwo, op, conf.try_poll_out);
   1885     }
   1886 
   1887     while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
   1888         hwi->pcm_ops->ctl_in (hwi, op, conf.try_poll_in);
   1889     }
   1890     audio_reset_timer ();
   1891 }
   1892 
   1893 static int initialized;
   1894 
   1895 static void audio_atexit (void)
   1896 {
   1897     AudioState *s = &glob_audio_state;
   1898     HWVoiceOut *hwo = NULL;
   1899     HWVoiceIn *hwi = NULL;
   1900 
   1901     if (!initialized) return;
   1902     initialized = 0;
   1903 
   1904     while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
   1905         SWVoiceCap *sc;
   1906 
   1907         hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
   1908         hwo->pcm_ops->fini_out (hwo);
   1909 
   1910         for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
   1911             CaptureVoiceOut *cap = sc->cap;
   1912             struct capture_callback *cb;
   1913 
   1914             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
   1915                 cb->ops.destroy (cb->opaque);
   1916             }
   1917         }
   1918     }
   1919 
   1920     while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
   1921         hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
   1922         hwi->pcm_ops->fini_in (hwi);
   1923     }
   1924 
   1925     if (s->drv) {
   1926         s->drv->fini (s->drv_opaque);
   1927     }
   1928 }
   1929 
   1930 static void audio_save (QEMUFile *f, void *opaque)
   1931 {
   1932     (void) f;
   1933     (void) opaque;
   1934 }
   1935 
   1936 static int audio_load (QEMUFile *f, void *opaque, int version_id)
   1937 {
   1938     (void) f;
   1939     (void) opaque;
   1940 
   1941     if (version_id != 1) {
   1942         return -EINVAL;
   1943     }
   1944 
   1945     return 0;
   1946 }
   1947 
   1948 
   1949 
   1950 static void audio_init (void)
   1951 {
   1952     size_t i;
   1953     int done = 0;
   1954     const char *drvname;
   1955     VMChangeStateEntry *e;
   1956     AudioState *s = &glob_audio_state;
   1957 
   1958     if (s->drv) {
   1959         return;
   1960     }
   1961 
   1962     QLIST_INIT (&s->hw_head_out);
   1963     QLIST_INIT (&s->hw_head_in);
   1964     QLIST_INIT (&s->cap_head);
   1965     atexit (audio_atexit);
   1966 
   1967     s->ts = qemu_new_timer_ns (vm_clock, audio_timer, s);
   1968     if (!s->ts) {
   1969         dolog ("Could not create audio timer\n");
   1970         return;
   1971     }
   1972 
   1973     audio_process_options ("AUDIO", audio_options);
   1974 
   1975     s->nb_hw_voices_out = conf.fixed_out.nb_voices;
   1976     s->nb_hw_voices_in = conf.fixed_in.nb_voices;
   1977 
   1978     if (s->nb_hw_voices_out <= 0) {
   1979         dolog ("Bogus number of playback voices %d, setting to 1\n",
   1980                s->nb_hw_voices_out);
   1981         s->nb_hw_voices_out = 1;
   1982     }
   1983 
   1984     if (s->nb_hw_voices_in <= 0) {
   1985         dolog ("Bogus number of capture voices %d, setting to 0\n",
   1986                s->nb_hw_voices_in);
   1987         s->nb_hw_voices_in = 0;
   1988     }
   1989 
   1990     {
   1991         int def;
   1992         drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
   1993     }
   1994 
   1995     if (drvname) {
   1996         int found = 0;
   1997 
   1998         for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
   1999             if (!strcmp (drvname, drvtab[i]->name)) {
   2000                 done = !audio_driver_init (s, drvtab[i]);
   2001                 found = 1;
   2002                 break;
   2003             }
   2004         }
   2005 
   2006         if (!found) {
   2007             dolog ("Unknown audio driver `%s'\n", drvname);
   2008             dolog ("Run with -audio-help to list available drivers\n");
   2009         }
   2010     }
   2011 
   2012     if (!done) {
   2013         for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
   2014             if (drvtab[i]->can_be_default) {
   2015                 done = !audio_driver_init (s, drvtab[i]);
   2016             }
   2017         }
   2018     }
   2019 
   2020     if (!done) {
   2021         done = !audio_driver_init (s, &no_audio_driver);
   2022         if (!done) {
   2023             hw_error("Could not initialize audio subsystem\n");
   2024         }
   2025         else {
   2026             dolog ("warning: Using timer based audio emulation\n");
   2027         }
   2028     }
   2029 
   2030     if (conf.period.hertz <= 0) {
   2031         if (conf.period.hertz < 0) {
   2032             dolog ("warning: Timer period is negative - %d "
   2033                    "treating as zero\n",
   2034                    conf.period.hertz);
   2035         }
   2036         conf.period.ticks = 1;
   2037     } else {
   2038         conf.period.ticks =
   2039             muldiv64 (1, get_ticks_per_sec (), conf.period.hertz);
   2040     }
   2041 
   2042     e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
   2043     if (!e) {
   2044         dolog ("warning: Could not register change state handler\n"
   2045                "(Audio can continue looping even after stopping the VM)\n");
   2046     }
   2047     initialized = 1;
   2048 
   2049     QLIST_INIT (&s->card_head);
   2050     register_savevm ("audio", 0, 1, audio_save, audio_load, s);
   2051     audio_reset_timer();
   2052 }
   2053 
   2054 void AUD_register_card (const char *name, QEMUSoundCard *card)
   2055 {
   2056     audio_init ();
   2057     card->name = qemu_strdup (name);
   2058     memset (&card->entries, 0, sizeof (card->entries));
   2059     QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
   2060 }
   2061 
   2062 void AUD_remove_card (QEMUSoundCard *card)
   2063 {
   2064     QLIST_REMOVE (card, entries);
   2065     qemu_free (card->name);
   2066 }
   2067 
   2068 
   2069 CaptureVoiceOut *AUD_add_capture (
   2070     struct audsettings *as,
   2071     struct audio_capture_ops *ops,
   2072     void *cb_opaque
   2073     )
   2074 {
   2075     AudioState *s = &glob_audio_state;
   2076     CaptureVoiceOut *cap;
   2077     struct capture_callback *cb;
   2078 
   2079     if (audio_validate_settings (as)) {
   2080         dolog ("Invalid settings were passed when trying to add capture\n");
   2081         audio_print_settings (as);
   2082         goto err0;
   2083     }
   2084 
   2085     cb = audio_calloc (AUDIO_FUNC, 1, sizeof (*cb));
   2086     if (!cb) {
   2087         dolog ("Could not allocate capture callback information, size %zu\n",
   2088                sizeof (*cb));
   2089         goto err0;
   2090     }
   2091     cb->ops = *ops;
   2092     cb->opaque = cb_opaque;
   2093 
   2094     cap = audio_pcm_capture_find_specific (as);
   2095     if (cap) {
   2096         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
   2097         return cap;
   2098     }
   2099     else {
   2100         HWVoiceOut *hw;
   2101         CaptureVoiceOut *cap;
   2102 
   2103         cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap));
   2104         if (!cap) {
   2105             dolog ("Could not allocate capture voice, size %zu\n",
   2106                    sizeof (*cap));
   2107             goto err1;
   2108         }
   2109 
   2110         hw = &cap->hw;
   2111         QLIST_INIT (&hw->sw_head);
   2112         QLIST_INIT (&cap->cb_head);
   2113 
   2114         /* XXX find a more elegant way */
   2115         hw->samples = 4096 * 4;
   2116         hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
   2117                                     sizeof (struct st_sample));
   2118         if (!hw->mix_buf) {
   2119             dolog ("Could not allocate capture mix buffer (%d samples)\n",
   2120                    hw->samples);
   2121             goto err2;
   2122         }
   2123 
   2124         audio_pcm_init_info (&hw->info, as);
   2125 
   2126         cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
   2127         if (!cap->buf) {
   2128             dolog ("Could not allocate capture buffer "
   2129                    "(%d samples, each %d bytes)\n",
   2130                    hw->samples, 1 << hw->info.shift);
   2131             goto err3;
   2132         }
   2133 
   2134         hw->clip = mixeng_clip
   2135             [hw->info.nchannels == 2]
   2136             [hw->info.sign]
   2137             [hw->info.swap_endianness]
   2138             [audio_bits_to_index (hw->info.bits)];
   2139 
   2140         QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
   2141         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
   2142 
   2143         hw = NULL;
   2144         while ((hw = audio_pcm_hw_find_any_out (hw))) {
   2145             audio_attach_capture (hw);
   2146         }
   2147         return cap;
   2148 
   2149     err3:
   2150         qemu_free (cap->hw.mix_buf);
   2151     err2:
   2152         qemu_free (cap);
   2153     err1:
   2154         qemu_free (cb);
   2155     err0:
   2156         return NULL;
   2157     }
   2158 }
   2159 
   2160 void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
   2161 {
   2162     struct capture_callback *cb;
   2163 
   2164     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
   2165         if (cb->opaque == cb_opaque) {
   2166             cb->ops.destroy (cb_opaque);
   2167             QLIST_REMOVE (cb, entries);
   2168             qemu_free (cb);
   2169 
   2170             if (!cap->cb_head.lh_first) {
   2171                 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
   2172 
   2173                 while (sw) {
   2174                     SWVoiceCap *sc = (SWVoiceCap *) sw;
   2175 #ifdef DEBUG_CAPTURE
   2176                     dolog ("freeing %s\n", sw->name);
   2177 #endif
   2178 
   2179                     sw1 = sw->entries.le_next;
   2180                     if (sw->rate) {
   2181                         st_rate_stop (sw->rate);
   2182                         sw->rate = NULL;
   2183                     }
   2184                     QLIST_REMOVE (sw, entries);
   2185                     QLIST_REMOVE (sc, entries);
   2186                     qemu_free (sc);
   2187                     sw = sw1;
   2188                 }
   2189                 QLIST_REMOVE (cap, entries);
   2190                 qemu_free (cap);
   2191             }
   2192             return;
   2193         }
   2194     }
   2195 }
   2196 
   2197 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
   2198 {
   2199     if (sw) {
   2200         sw->vol.mute = mute;
   2201         sw->vol.l = nominal_volume.l * lvol / 255;
   2202         sw->vol.r = nominal_volume.r * rvol / 255;
   2203     }
   2204 }
   2205 
   2206 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
   2207 {
   2208     if (sw) {
   2209         sw->vol.mute = mute;
   2210         sw->vol.l = nominal_volume.l * lvol / 255;
   2211         sw->vol.r = nominal_volume.r * rvol / 255;
   2212     }
   2213 }
   2214