audio.c 48.2 KB
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/*
 * QEMU Audio subsystem
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 *
 * Copyright (c) 2003-2005 Vassili Karpov (malc)
 *
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 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
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#include "hw/hw.h"
#include "audio.h"
#include "console.h"
#include "qemu-timer.h"
#include "sysemu.h"
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#define AUDIO_CAP "audio"
#include "audio_int.h"
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/* #define DEBUG_PLIVE */
/* #define DEBUG_LIVE */
/* #define DEBUG_OUT */
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/* #define DEBUG_CAPTURE */
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#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"

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static struct audio_driver *drvtab[] = {
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    AUDIO_DRIVERS
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    &no_audio_driver,
    &wav_audio_driver
};
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struct fixed_settings {
    int enabled;
    int nb_voices;
    int greedy;
    audsettings_t settings;
};

static struct {
    struct fixed_settings fixed_out;
    struct fixed_settings fixed_in;
    union {
        int hz;
        int64_t ticks;
    } period;
    int plive;
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    int log_to_monitor;
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} conf = {
    {                           /* DAC fixed settings */
        1,                      /* enabled */
        1,                      /* nb_voices */
        1,                      /* greedy */
        {
            44100,              /* freq */
            2,                  /* nchannels */
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            AUD_FMT_S16,        /* fmt */
            AUDIO_HOST_ENDIANNESS
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        }
    },

    {                           /* ADC fixed settings */
        1,                      /* enabled */
        1,                      /* nb_voices */
        1,                      /* greedy */
        {
            44100,              /* freq */
            2,                  /* nchannels */
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            AUD_FMT_S16,        /* fmt */
            AUDIO_HOST_ENDIANNESS
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        }
    },

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    { 0 },                      /* period */
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    0,                          /* plive */
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    0                           /* log_to_monitor */
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};

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static AudioState glob_audio_state;

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volume_t nominal_volume = {
    0,
#ifdef FLOAT_MIXENG
    1.0,
    1.0
#else
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    1ULL << 32,
    1ULL << 32
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#endif
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};

/* http://www.df.lth.se/~john_e/gems/gem002d.html */
/* http://www.multi-platforms.com/Tips/PopCount.htm */
uint32_t popcount (uint32_t u)
{
    u = ((u&0x55555555) + ((u>>1)&0x55555555));
    u = ((u&0x33333333) + ((u>>2)&0x33333333));
    u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
    u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
    u = ( u&0x0000ffff) + (u>>16);
    return u;
}

inline uint32_t lsbindex (uint32_t u)
{
    return popcount ((u&-u)-1);
}

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#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
int audio_bug (const char *funcname, int cond)
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{
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    if (cond) {
        static int shown;

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        AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
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        if (!shown) {
            shown = 1;
            AUD_log (NULL, "Save all your work and restart without audio\n");
            AUD_log (NULL, "Please send bug report to malc@pulsesoft.com\n");
            AUD_log (NULL, "I am sorry\n");
        }
        AUD_log (NULL, "Context:\n");

#if defined AUDIO_BREAKPOINT_ON_BUG
#  if defined HOST_I386
#    if defined __GNUC__
        __asm__ ("int3");
#    elif defined _MSC_VER
        _asm _emit 0xcc;
#    else
        abort ();
#    endif
#  else
        abort ();
#  endif
#endif
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    }

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    return cond;
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}
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#endif
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static inline int audio_bits_to_index (int bits)
{
    switch (bits) {
    case 8:
        return 0;

    case 16:
        return 1;

    case 32:
        return 2;

    default:
        audio_bug ("bits_to_index", 1);
        AUD_log (NULL, "invalid bits %d\n", bits);
        return 0;
    }
}

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void *audio_calloc (const char *funcname, int nmemb, size_t size)
{
    int cond;
    size_t len;

    len = nmemb * size;
    cond = !nmemb || !size;
    cond |= nmemb < 0;
    cond |= len < size;

    if (audio_bug ("audio_calloc", cond)) {
        AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
                 funcname);
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        AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
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        return NULL;
    }

    return qemu_mallocz (len);
}

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static char *audio_alloc_prefix (const char *s)
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{
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    const char qemu_prefix[] = "QEMU_";
    size_t len;
    char *r;
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    if (!s) {
        return NULL;
    }
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    len = strlen (s);
    r = qemu_malloc (len + sizeof (qemu_prefix));
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    if (r) {
        size_t i;
        char *u = r + sizeof (qemu_prefix) - 1;
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        strcpy (r, qemu_prefix);
        strcat (r, s);

        for (i = 0; i < len; ++i) {
            u[i] = toupper (u[i]);
        }
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    }
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    return r;
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}

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static const char *audio_audfmt_to_string (audfmt_e fmt)
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{
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    switch (fmt) {
    case AUD_FMT_U8:
        return "U8";
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    case AUD_FMT_U16:
        return "U16";
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    case AUD_FMT_S8:
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        return "S8";
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    case AUD_FMT_S16:
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        return "S16";
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    case AUD_FMT_U32:
        return "U32";

    case AUD_FMT_S32:
        return "S32";
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    }

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    dolog ("Bogus audfmt %d returning S16\n", fmt);
    return "S16";
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}

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static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
                                        int *defaultp)
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{
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    if (!strcasecmp (s, "u8")) {
        *defaultp = 0;
        return AUD_FMT_U8;
    }
    else if (!strcasecmp (s, "u16")) {
        *defaultp = 0;
        return AUD_FMT_U16;
    }
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    else if (!strcasecmp (s, "u32")) {
        *defaultp = 0;
        return AUD_FMT_U32;
    }
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    else if (!strcasecmp (s, "s8")) {
        *defaultp = 0;
        return AUD_FMT_S8;
    }
    else if (!strcasecmp (s, "s16")) {
        *defaultp = 0;
        return AUD_FMT_S16;
    }
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    else if (!strcasecmp (s, "s32")) {
        *defaultp = 0;
        return AUD_FMT_S32;
    }
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    else {
        dolog ("Bogus audio format `%s' using %s\n",
               s, audio_audfmt_to_string (defval));
        *defaultp = 1;
        return defval;
    }
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}

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static audfmt_e audio_get_conf_fmt (const char *envname,
                                    audfmt_e defval,
                                    int *defaultp)
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{
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    const char *var = getenv (envname);
    if (!var) {
        *defaultp = 1;
        return defval;
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    }
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    return audio_string_to_audfmt (var, defval, defaultp);
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}

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static int audio_get_conf_int (const char *key, int defval, int *defaultp)
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{
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    int val;
    char *strval;
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    strval = getenv (key);
    if (strval) {
        *defaultp = 0;
        val = atoi (strval);
        return val;
    }
    else {
        *defaultp = 1;
        return defval;
    }
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}

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static const char *audio_get_conf_str (const char *key,
                                       const char *defval,
                                       int *defaultp)
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{
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    const char *val = getenv (key);
    if (!val) {
        *defaultp = 1;
        return defval;
    }
    else {
        *defaultp = 0;
        return val;
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    }
}

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void AUD_vlog (const char *cap, const char *fmt, va_list ap)
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{
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    if (conf.log_to_monitor) {
        if (cap) {
            term_printf ("%s: ", cap);
        }

        term_vprintf (fmt, ap);
    }
    else {
        if (cap) {
            fprintf (stderr, "%s: ", cap);
        }

        vfprintf (stderr, fmt, ap);
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    }
}

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void AUD_log (const char *cap, const char *fmt, ...)
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{
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    va_list ap;

    va_start (ap, fmt);
    AUD_vlog (cap, fmt, ap);
    va_end (ap);
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}

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static void audio_print_options (const char *prefix,
                                 struct audio_option *opt)
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{
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    char *uprefix;

    if (!prefix) {
        dolog ("No prefix specified\n");
        return;
    }

    if (!opt) {
        dolog ("No options\n");
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        return;
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    }
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    uprefix = audio_alloc_prefix (prefix);
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    for (; opt->name; opt++) {
        const char *state = "default";
        printf ("  %s_%s: ", uprefix, opt->name);
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        if (opt->overriddenp && *opt->overriddenp) {
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            state = "current";
        }
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        switch (opt->tag) {
        case AUD_OPT_BOOL:
            {
                int *intp = opt->valp;
                printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
            }
            break;

        case AUD_OPT_INT:
            {
                int *intp = opt->valp;
                printf ("integer, %s = %d\n", state, *intp);
            }
            break;

        case AUD_OPT_FMT:
            {
                audfmt_e *fmtp = opt->valp;
                printf (
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                    "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
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                    state,
                    audio_audfmt_to_string (*fmtp)
                    );
            }
            break;

        case AUD_OPT_STR:
            {
                const char **strp = opt->valp;
                printf ("string, %s = %s\n",
                        state,
                        *strp ? *strp : "(not set)");
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            }
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            break;

        default:
            printf ("???\n");
            dolog ("Bad value tag for option %s_%s %d\n",
                   uprefix, opt->name, opt->tag);
            break;
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        }
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        printf ("    %s\n", opt->descr);
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    }
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    qemu_free (uprefix);
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}

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static void audio_process_options (const char *prefix,
                                   struct audio_option *opt)
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{
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    char *optname;
    const char qemu_prefix[] = "QEMU_";
    size_t preflen;
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    if (audio_bug (AUDIO_FUNC, !prefix)) {
        dolog ("prefix = NULL\n");
        return;
    }
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    if (audio_bug (AUDIO_FUNC, !opt)) {
        dolog ("opt = NULL\n");
        return;
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    }

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    preflen = strlen (prefix);
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    for (; opt->name; opt++) {
        size_t len, i;
        int def;

        if (!opt->valp) {
            dolog ("Option value pointer for `%s' is not set\n",
                   opt->name);
            continue;
        }

        len = strlen (opt->name);
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        /* len of opt->name + len of prefix + size of qemu_prefix
         * (includes trailing zero) + zero + underscore (on behalf of
         * sizeof) */
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        optname = qemu_malloc (len + preflen + sizeof (qemu_prefix) + 1);
        if (!optname) {
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            dolog ("Could not allocate memory for option name `%s'\n",
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                   opt->name);
            continue;
        }

        strcpy (optname, qemu_prefix);
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        /* copy while upper-casing, including trailing zero */
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        for (i = 0; i <= preflen; ++i) {
            optname[i + sizeof (qemu_prefix) - 1] = toupper (prefix[i]);
        }
        strcat (optname, "_");
        strcat (optname, opt->name);

        def = 1;
        switch (opt->tag) {
        case AUD_OPT_BOOL:
        case AUD_OPT_INT:
            {
                int *intp = opt->valp;
                *intp = audio_get_conf_int (optname, *intp, &def);
            }
            break;

        case AUD_OPT_FMT:
            {
                audfmt_e *fmtp = opt->valp;
                *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
            }
            break;

        case AUD_OPT_STR:
            {
                const char **strp = opt->valp;
                *strp = audio_get_conf_str (optname, *strp, &def);
            }
            break;

        default:
            dolog ("Bad value tag for option `%s' - %d\n",
                   optname, opt->tag);
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            break;
        }

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        if (!opt->overriddenp) {
            opt->overriddenp = &opt->overridden;
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        }
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        *opt->overriddenp = !def;
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        qemu_free (optname);
    }
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}

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static void audio_print_settings (audsettings_t *as)
{
    dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);

    switch (as->fmt) {
    case AUD_FMT_S8:
        AUD_log (NULL, "S8");
        break;
    case AUD_FMT_U8:
        AUD_log (NULL, "U8");
        break;
    case AUD_FMT_S16:
        AUD_log (NULL, "S16");
        break;
    case AUD_FMT_U16:
        AUD_log (NULL, "U16");
        break;
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    case AUD_FMT_S32:
        AUD_log (NULL, "S32");
        break;
    case AUD_FMT_U32:
        AUD_log (NULL, "U32");
        break;
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    default:
        AUD_log (NULL, "invalid(%d)", as->fmt);
        break;
    }
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    AUD_log (NULL, " endianness=");
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    switch (as->endianness) {
    case 0:
        AUD_log (NULL, "little");
        break;
    case 1:
        AUD_log (NULL, "big");
        break;
    default:
        AUD_log (NULL, "invalid");
        break;
    }
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    AUD_log (NULL, "\n");
}

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static int audio_validate_settings (audsettings_t *as)
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{
    int invalid;

    invalid = as->nchannels != 1 && as->nchannels != 2;
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    invalid |= as->endianness != 0 && as->endianness != 1;
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    switch (as->fmt) {
    case AUD_FMT_S8:
    case AUD_FMT_U8:
    case AUD_FMT_S16:
    case AUD_FMT_U16:
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    case AUD_FMT_S32:
    case AUD_FMT_U32:
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        break;
    default:
        invalid = 1;
        break;
    }

    invalid |= as->freq <= 0;
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    return invalid ? -1 : 0;
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}

static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
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{
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    int bits = 8, sign = 0;
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    switch (as->fmt) {
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    case AUD_FMT_S8:
        sign = 1;
    case AUD_FMT_U8:
        break;

    case AUD_FMT_S16:
        sign = 1;
    case AUD_FMT_U16:
        bits = 16;
        break;
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    case AUD_FMT_S32:
        sign = 1;
    case AUD_FMT_U32:
        bits = 32;
        break;
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    }
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    return info->freq == as->freq
        && info->nchannels == as->nchannels
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        && info->sign == sign
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        && info->bits == bits
        && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
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}
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void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
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{
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    int bits = 8, sign = 0, shift = 0;
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    switch (as->fmt) {
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    case AUD_FMT_S8:
        sign = 1;
    case AUD_FMT_U8:
        break;

    case AUD_FMT_S16:
        sign = 1;
    case AUD_FMT_U16:
        bits = 16;
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        shift = 1;
        break;

    case AUD_FMT_S32:
        sign = 1;
    case AUD_FMT_U32:
        bits = 32;
        shift = 2;
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        break;
    }

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    info->freq = as->freq;
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    info->bits = bits;
    info->sign = sign;
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    info->nchannels = as->nchannels;
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    info->shift = (as->nchannels == 2) + shift;
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    info->align = (1 << info->shift) - 1;
    info->bytes_per_second = info->freq << info->shift;
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    info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
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}

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void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
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{
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    if (!len) {
        return;
    }

    if (info->sign) {
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        memset (buf, 0x00, len << info->shift);
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    }
    else {
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        switch (info->bits) {
        case 8:
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            memset (buf, 0x80, len << info->shift);
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            break;
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        case 16:
            {
                int i;
                uint16_t *p = buf;
                int shift = info->nchannels - 1;
                short s = INT16_MAX;

                if (info->swap_endianness) {
                    s = bswap16 (s);
                }

                for (i = 0; i < len << shift; i++) {
                    p[i] = s;
                }
675
            }
676 677 678 679 680 681 682 683 684 685 686 687
            break;

        case 32:
            {
                int i;
                uint32_t *p = buf;
                int shift = info->nchannels - 1;
                int32_t s = INT32_MAX;

                if (info->swap_endianness) {
                    s = bswap32 (s);
                }
688

689 690 691
                for (i = 0; i < len << shift; i++) {
                    p[i] = s;
                }
692
            }
693 694 695 696 697 698
            break;

        default:
            AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
                     info->bits);
            break;
699
        }
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    }
}

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/*
 * Capture
 */
static void noop_conv (st_sample_t *dst, const void *src,
                       int samples, volume_t *vol)
{
    (void) src;
    (void) dst;
    (void) samples;
    (void) vol;
}

static CaptureVoiceOut *audio_pcm_capture_find_specific (
    AudioState *s,
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    audsettings_t *as
718 719 720 721 722
    )
{
    CaptureVoiceOut *cap;

    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
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        if (audio_pcm_info_eq (&cap->hw.info, as)) {
724 725 726 727 728 729
            return cap;
        }
    }
    return NULL;
}

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static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
731
{
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    struct capture_callback *cb;

#ifdef DEBUG_CAPTURE
    dolog ("notification %d sent\n", cmd);
#endif
    for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
        cb->ops.notify (cb->opaque, cmd);
    }
}
741

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static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
{
    if (cap->hw.enabled != enabled) {
        audcnotification_e cmd;
746
        cap->hw.enabled = enabled;
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        cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
        audio_notify_capture (cap, cmd);
749 750 751 752 753 754 755 756 757
    }
}

static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
{
    HWVoiceOut *hw = &cap->hw;
    SWVoiceOut *sw;
    int enabled = 0;

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    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
759 760 761 762 763
        if (sw->active) {
            enabled = 1;
            break;
        }
    }
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    audio_capture_maybe_changed (cap, enabled);
765 766 767 768
}

static void audio_detach_capture (HWVoiceOut *hw)
{
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    SWVoiceCap *sc = hw->cap_head.lh_first;

    while (sc) {
        SWVoiceCap *sc1 = sc->entries.le_next;
        SWVoiceOut *sw = &sc->sw;
        CaptureVoiceOut *cap = sc->cap;
        int was_active = sw->active;
776 777 778 779 780 781 782

        if (sw->rate) {
            st_rate_stop (sw->rate);
            sw->rate = NULL;
        }

        LIST_REMOVE (sw, entries);
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        LIST_REMOVE (sc, entries);
        qemu_free (sc);
        if (was_active) {
            /* We have removed soft voice from the capture:
               this might have changed the overall status of the capture
               since this might have been the only active voice */
            audio_recalc_and_notify_capture (cap);
        }
        sc = sc1;
792 793 794 795 796 797 798 799 800
    }
}

static int audio_attach_capture (AudioState *s, HWVoiceOut *hw)
{
    CaptureVoiceOut *cap;

    audio_detach_capture (hw);
    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
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        SWVoiceCap *sc;
802
        SWVoiceOut *sw;
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        HWVoiceOut *hw_cap = &cap->hw;
804

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        sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
        if (!sc) {
807
            dolog ("Could not allocate soft capture voice (%zu bytes)\n",
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                   sizeof (*sc));
809 810 811
            return -1;
        }

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        sc->cap = cap;
        sw = &sc->sw;
814
        sw->hw = hw_cap;
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        sw->info = hw->info;
816 817 818 819 820 821 822 823 824 825 826
        sw->empty = 1;
        sw->active = hw->enabled;
        sw->conv = noop_conv;
        sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
        sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
        if (!sw->rate) {
            dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
            qemu_free (sw);
            return -1;
        }
        LIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
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        LIST_INSERT_HEAD (&hw->cap_head, sc, entries);
#ifdef DEBUG_CAPTURE
        asprintf (&sw->name, "for %p %d,%d,%d",
                  hw, sw->info.freq, sw->info.bits, sw->info.nchannels);
        dolog ("Added %s active = %d\n", sw->name, sw->active);
#endif
833
        if (sw->active) {
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            audio_capture_maybe_changed (cap, 1);
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        }
    }
    return 0;
}

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/*
 * Hard voice (capture)
 */
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static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
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{
845 846 847 848 849 850 851
    SWVoiceIn *sw;
    int m = hw->total_samples_captured;

    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
        if (sw->active) {
            m = audio_MIN (m, sw->total_hw_samples_acquired);
        }
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    }
853
    return m;
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}

856
int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
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{
858 859 860 861
    int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
    if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
        dolog ("live=%d hw->samples=%d\n", live, hw->samples);
        return 0;
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    }
863
    return live;
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}

866 867 868 869 870 871 872 873 874 875 876 877 878 879 880 881 882
/*
 * Soft voice (capture)
 */
static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
{
    HWVoiceIn *hw = sw->hw;
    int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
    int rpos;

    if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
        dolog ("live=%d hw->samples=%d\n", live, hw->samples);
        return 0;
    }

    rpos = hw->wpos - live;
    if (rpos >= 0) {
        return rpos;
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    }
    else {
885
        return hw->samples + rpos;
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    }
}

889
int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
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{
891 892
    HWVoiceIn *hw = sw->hw;
    int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
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    st_sample_t *src, *dst = sw->buf;
894 895 896 897 898 899 900 901 902 903 904 905 906

    rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;

    live = hw->total_samples_captured - sw->total_hw_samples_acquired;
    if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
        dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
        return 0;
    }

    samples = size >> sw->info.shift;
    if (!live) {
        return 0;
    }
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908 909
    swlim = (live * sw->ratio) >> 32;
    swlim = audio_MIN (swlim, samples);
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911 912 913 914 915 916 917
    while (swlim) {
        src = hw->conv_buf + rpos;
        isamp = hw->wpos - rpos;
        /* XXX: <= ? */
        if (isamp <= 0) {
            isamp = hw->samples - rpos;
        }
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919 920 921 922
        if (!isamp) {
            break;
        }
        osamp = swlim;
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924 925
        if (audio_bug (AUDIO_FUNC, osamp < 0)) {
            dolog ("osamp=%d\n", osamp);
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            return 0;
927
        }
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929 930 931 932 933 934 935
        st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
        swlim -= osamp;
        rpos = (rpos + isamp) % hw->samples;
        dst += osamp;
        ret += osamp;
        total += isamp;
    }
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    sw->clip (buf, sw->buf, ret);
938 939
    sw->total_hw_samples_acquired += total;
    return ret << sw->info.shift;
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}

942 943 944
/*
 * Hard voice (playback)
 */
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static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
946
{
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    SWVoiceOut *sw;
    int m = INT_MAX;
    int nb_live = 0;
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    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
        if (sw->active || !sw->empty) {
            m = audio_MIN (m, sw->total_hw_samples_mixed);
            nb_live += 1;
        }
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    }
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    *nb_livep = nb_live;
    return m;
960
}
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962 963 964
int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
{
    int smin;
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966 967 968 969
    smin = audio_pcm_hw_find_min_out (hw, nb_live);

    if (!*nb_live) {
        return 0;
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    }
    else {
972 973 974 975 976
        int live = smin;

        if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
            dolog ("live=%d hw->samples=%d\n", live, hw->samples);
            return 0;
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        }
978
        return live;
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    }
980 981 982 983 984 985
}

int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
{
    int nb_live;
    int live;
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987 988 989 990
    live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
    if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
        dolog ("live=%d hw->samples=%d\n", live, hw->samples);
        return 0;
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    }
992
    return live;
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}

995 996 997 998
/*
 * Soft voice (playback)
 */
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
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{
1000 1001
    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
    int ret = 0, pos = 0, total = 0;
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1003 1004 1005
    if (!sw) {
        return size;
    }
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1007
    hwsamples = sw->hw->samples;
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1009 1010 1011 1012 1013
    live = sw->total_hw_samples_mixed;
    if (audio_bug (AUDIO_FUNC, live < 0 || live > hwsamples)){
        dolog ("live=%d hw->samples=%d\n", live, hwsamples);
        return 0;
    }
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1015
    if (live == hwsamples) {
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1016 1017 1018
#ifdef DEBUG_OUT
        dolog ("%s is full %d\n", sw->name, live);
#endif
1019 1020
        return 0;
    }
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1022 1023
    wpos = (sw->hw->rpos + live) % hwsamples;
    samples = size >> sw->info.shift;
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1025 1026 1027 1028 1029 1030 1031 1032 1033 1034 1035 1036 1037 1038 1039 1040 1041 1042 1043 1044 1045 1046 1047 1048 1049 1050 1051 1052 1053 1054 1055 1056 1057 1058 1059 1060
    dead = hwsamples - live;
    swlim = ((int64_t) dead << 32) / sw->ratio;
    swlim = audio_MIN (swlim, samples);
    if (swlim) {
        sw->conv (sw->buf, buf, swlim, &sw->vol);
    }

    while (swlim) {
        dead = hwsamples - live;
        left = hwsamples - wpos;
        blck = audio_MIN (dead, left);
        if (!blck) {
            break;
        }
        isamp = swlim;
        osamp = blck;
        st_rate_flow_mix (
            sw->rate,
            sw->buf + pos,
            sw->hw->mix_buf + wpos,
            &isamp,
            &osamp
            );
        ret += isamp;
        swlim -= isamp;
        pos += isamp;
        live += osamp;
        wpos = (wpos + osamp) % hwsamples;
        total += osamp;
    }

    sw->total_hw_samples_mixed += total;
    sw->empty = sw->total_hw_samples_mixed == 0;

#ifdef DEBUG_OUT
    dolog (
B
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1061 1062
        "%s: write size %d ret %d total sw %d\n",
        SW_NAME (sw),
1063 1064
        size >> sw->info.shift,
        ret,
B
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        sw->total_hw_samples_mixed
1066 1067 1068 1069
        );
#endif

    return ret << sw->info.shift;
B
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1070 1071
}

1072 1073
#ifdef DEBUG_AUDIO
static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
B
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1074
{
1075 1076
    dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
           cap, info->bits, info->sign, info->freq, info->nchannels);
B
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1077
}
1078
#endif
B
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1080 1081 1082 1083 1084 1085
#define DAC
#include "audio_template.h"
#undef DAC
#include "audio_template.h"

int AUD_write (SWVoiceOut *sw, void *buf, int size)
B
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1086
{
1087
    int bytes;
B
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1088

1089 1090 1091 1092
    if (!sw) {
        /* XXX: Consider options */
        return size;
    }
B
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1093

1094
    if (!sw->hw->enabled) {
B
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1095
        dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
B
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1096 1097 1098
        return 0;
    }

1099 1100 1101 1102 1103 1104 1105
    bytes = sw->hw->pcm_ops->write (sw, buf, size);
    return bytes;
}

int AUD_read (SWVoiceIn *sw, void *buf, int size)
{
    int bytes;
B
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1106

1107 1108 1109
    if (!sw) {
        /* XXX: Consider options */
        return size;
B
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1110
    }
1111 1112

    if (!sw->hw->enabled) {
B
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1113
        dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
1114
        return 0;
B
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1115
    }
1116 1117 1118

    bytes = sw->hw->pcm_ops->read (sw, buf, size);
    return bytes;
B
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1119 1120
}

1121
int AUD_get_buffer_size_out (SWVoiceOut *sw)
B
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{
B
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1123
    return sw->hw->samples << sw->hw->info.shift;
1124 1125 1126 1127 1128
}

void AUD_set_active_out (SWVoiceOut *sw, int on)
{
    HWVoiceOut *hw;
B
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1129

1130
    if (!sw) {
B
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1131
        return;
1132
    }
B
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1133 1134

    hw = sw->hw;
1135 1136
    if (sw->active != on) {
        SWVoiceOut *temp_sw;
B
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1137
        SWVoiceCap *sc;
1138 1139 1140 1141 1142 1143 1144 1145 1146 1147 1148 1149 1150 1151 1152 1153 1154 1155 1156

        if (on) {
            hw->pending_disable = 0;
            if (!hw->enabled) {
                hw->enabled = 1;
                hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
            }
        }
        else {
            if (hw->enabled) {
                int nb_active = 0;

                for (temp_sw = hw->sw_head.lh_first; temp_sw;
                     temp_sw = temp_sw->entries.le_next) {
                    nb_active += temp_sw->active != 0;
                }

                hw->pending_disable = nb_active == 1;
            }
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        }
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        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
            sc->sw.active = hw->enabled;
1161
            if (hw->enabled) {
B
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                audio_capture_maybe_changed (sc->cap, 1);
1163 1164
            }
        }
1165 1166 1167 1168 1169 1170 1171 1172 1173 1174
        sw->active = on;
    }
}

void AUD_set_active_in (SWVoiceIn *sw, int on)
{
    HWVoiceIn *hw;

    if (!sw) {
        return;
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1175 1176
    }

1177
    hw = sw->hw;
B
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    if (sw->active != on) {
1179 1180
        SWVoiceIn *temp_sw;

B
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1181 1182 1183
        if (on) {
            if (!hw->enabled) {
                hw->enabled = 1;
1184
                hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
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            }
1186
            sw->total_hw_samples_acquired = hw->total_samples_captured;
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1187 1188
        }
        else {
1189
            if (hw->enabled) {
B
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                int nb_active = 0;
1191 1192 1193 1194

                for (temp_sw = hw->sw_head.lh_first; temp_sw;
                     temp_sw = temp_sw->entries.le_next) {
                    nb_active += temp_sw->active != 0;
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                }

                if (nb_active == 1) {
1198 1199
                    hw->enabled = 0;
                    hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
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                }
            }
        }
        sw->active = on;
    }
}

1207 1208 1209 1210 1211 1212 1213 1214 1215 1216 1217 1218 1219 1220 1221
static int audio_get_avail (SWVoiceIn *sw)
{
    int live;

    if (!sw) {
        return 0;
    }

    live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
    if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
        dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
        return 0;
    }

    ldebug (
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        "%s: get_avail live %d ret %" PRId64 "\n",
B
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        SW_NAME (sw),
1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241
        live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
        );

    return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
}

static int audio_get_free (SWVoiceOut *sw)
{
    int live, dead;

    if (!sw) {
        return 0;
    }

    live = sw->total_hw_samples_mixed;

    if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
        dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
B
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        return 0;
1243 1244 1245 1246 1247
    }

    dead = sw->hw->samples - live;

#ifdef DEBUG_OUT
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    dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
B
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1249
           SW_NAME (sw),
1250
           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
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#endif
1252 1253 1254 1255

    return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
}

1256 1257 1258 1259 1260
static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
{
    int n;

    if (hw->enabled) {
B
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        SWVoiceCap *sc;
1262

B
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        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
            SWVoiceOut *sw = &sc->sw;
1265 1266 1267 1268 1269 1270 1271 1272 1273 1274 1275 1276
            int rpos2 = rpos;

            n = samples;
            while (n) {
                int till_end_of_hw = hw->samples - rpos2;
                int to_write = audio_MIN (till_end_of_hw, n);
                int bytes = to_write << hw->info.shift;
                int written;

                sw->buf = hw->mix_buf + rpos2;
                written = audio_pcm_sw_write (sw, NULL, bytes);
                if (written - bytes) {
B
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1277 1278 1279
                    dolog ("Could not mix %d bytes into a capture "
                           "buffer, mixed %d\n",
                           bytes, written);
1280 1281 1282 1283 1284 1285 1286 1287 1288 1289 1290 1291 1292
                    break;
                }
                n -= to_write;
                rpos2 = (rpos2 + to_write) % hw->samples;
            }
        }
    }

    n = audio_MIN (samples, hw->samples - rpos);
    mixeng_clear (hw->mix_buf + rpos, n);
    mixeng_clear (hw->mix_buf, samples - n);
}

B
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1293
static void audio_run_out (AudioState *s)
1294 1295 1296 1297
{
    HWVoiceOut *hw = NULL;
    SWVoiceOut *sw;

B
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1298
    while ((hw = audio_pcm_hw_find_any_enabled_out (s, hw))) {
1299
        int played;
1300
        int live, free, nb_live, cleanup_required, prev_rpos;
1301 1302 1303 1304 1305

        live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
        if (!nb_live) {
            live = 0;
        }
B
bellard 已提交
1306

1307 1308
        if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
            dolog ("live=%d hw->samples=%d\n", live, hw->samples);
B
bellard 已提交
1309
            continue;
1310 1311 1312
        }

        if (hw->pending_disable && !nb_live) {
B
bellard 已提交
1313
            SWVoiceCap *sc;
1314 1315
#ifdef DEBUG_OUT
            dolog ("Disabling voice\n");
B
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1316
#endif
1317 1318 1319
            hw->enabled = 0;
            hw->pending_disable = 0;
            hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
B
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1320 1321 1322
            for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
                sc->sw.active = 0;
                audio_recalc_and_notify_capture (sc->cap);
1323
            }
1324 1325 1326 1327 1328 1329 1330 1331 1332 1333 1334 1335 1336 1337 1338
            continue;
        }

        if (!live) {
            for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
                if (sw->active) {
                    free = audio_get_free (sw);
                    if (free > 0) {
                        sw->callback.fn (sw->callback.opaque, free);
                    }
                }
            }
            continue;
        }

1339
        prev_rpos = hw->rpos;
1340 1341 1342 1343 1344 1345 1346 1347
        played = hw->pcm_ops->run_out (hw);
        if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
            dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
                   hw->rpos, hw->samples, played);
            hw->rpos = 0;
        }

#ifdef DEBUG_OUT
B
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1348
        dolog ("played=%d\n", played);
B
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1349
#endif
1350 1351 1352

        if (played) {
            hw->ts_helper += played;
1353
            audio_capture_mix_and_clear (hw, prev_rpos, played);
1354 1355
        }

B
bellard 已提交
1356
        cleanup_required = 0;
1357 1358 1359 1360 1361 1362 1363 1364 1365 1366 1367 1368 1369 1370 1371
        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
            if (!sw->active && sw->empty) {
                continue;
            }

            if (audio_bug (AUDIO_FUNC, played > sw->total_hw_samples_mixed)) {
                dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
                       played, sw->total_hw_samples_mixed);
                played = sw->total_hw_samples_mixed;
            }

            sw->total_hw_samples_mixed -= played;

            if (!sw->total_hw_samples_mixed) {
                sw->empty = 1;
B
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1372
                cleanup_required |= !sw->active && !sw->callback.fn;
1373 1374 1375 1376 1377 1378 1379 1380 1381
            }

            if (sw->active) {
                free = audio_get_free (sw);
                if (free > 0) {
                    sw->callback.fn (sw->callback.opaque, free);
                }
            }
        }
B
bellard 已提交
1382 1383

        if (cleanup_required) {
B
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1384 1385 1386 1387 1388
            SWVoiceOut *sw1;

            sw = hw->sw_head.lh_first;
            while (sw) {
                sw1 = sw->entries.le_next;
B
bellard 已提交
1389 1390 1391 1392 1393 1394
                if (!sw->active && !sw->callback.fn) {
#ifdef DEBUG_PLIVE
                    dolog ("Finishing with old voice\n");
#endif
                    audio_close_out (s, sw);
                }
B
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1395
                sw = sw1;
B
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1396 1397
            }
        }
1398 1399 1400
    }
}

B
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1401
static void audio_run_in (AudioState *s)
1402 1403 1404
{
    HWVoiceIn *hw = NULL;

B
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1405
    while ((hw = audio_pcm_hw_find_any_enabled_in (s, hw))) {
1406 1407 1408 1409 1410 1411 1412 1413 1414 1415 1416 1417 1418 1419 1420 1421 1422 1423 1424 1425 1426 1427 1428 1429
        SWVoiceIn *sw;
        int captured, min;

        captured = hw->pcm_ops->run_in (hw);

        min = audio_pcm_hw_find_min_in (hw);
        hw->total_samples_captured += captured - min;
        hw->ts_helper += captured;

        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
            sw->total_hw_samples_acquired -= min;

            if (sw->active) {
                int avail;

                avail = audio_get_avail (sw);
                if (avail > 0) {
                    sw->callback.fn (sw->callback.opaque, avail);
                }
            }
        }
    }
}

1430 1431 1432 1433 1434 1435 1436 1437 1438 1439 1440 1441 1442 1443 1444 1445 1446 1447 1448 1449 1450 1451 1452 1453 1454 1455 1456 1457 1458 1459 1460 1461 1462 1463 1464 1465 1466 1467 1468 1469 1470 1471 1472 1473 1474 1475 1476
static void audio_run_capture (AudioState *s)
{
    CaptureVoiceOut *cap;

    for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
        int live, rpos, captured;
        HWVoiceOut *hw = &cap->hw;
        SWVoiceOut *sw;

        captured = live = audio_pcm_hw_get_live_out (hw);
        rpos = hw->rpos;
        while (live) {
            int left = hw->samples - rpos;
            int to_capture = audio_MIN (live, left);
            st_sample_t *src;
            struct capture_callback *cb;

            src = hw->mix_buf + rpos;
            hw->clip (cap->buf, src, to_capture);
            mixeng_clear (src, to_capture);

            for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
                cb->ops.capture (cb->opaque, cap->buf,
                                 to_capture << hw->info.shift);
            }
            rpos = (rpos + to_capture) % hw->samples;
            live -= to_capture;
        }
        hw->rpos = rpos;

        for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
            if (!sw->active && sw->empty) {
                continue;
            }

            if (audio_bug (AUDIO_FUNC, captured > sw->total_hw_samples_mixed)) {
                dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
                       captured, sw->total_hw_samples_mixed);
                captured = sw->total_hw_samples_mixed;
            }

            sw->total_hw_samples_mixed -= captured;
            sw->empty = sw->total_hw_samples_mixed == 0;
        }
    }
}

B
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1477 1478 1479 1480 1481 1482
static void audio_timer (void *opaque)
{
    AudioState *s = opaque;

    audio_run_out (s);
    audio_run_in (s);
1483
    audio_run_capture (s);
B
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1484 1485 1486 1487

    qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
}

1488 1489
static struct audio_option audio_options[] = {
    /* DAC */
B
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1490
    {"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_out.enabled,
1491 1492
     "Use fixed settings for host DAC", NULL, 0},

B
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1493
    {"DAC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_out.settings.freq,
1494 1495
     "Frequency for fixed host DAC", NULL, 0},

B
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1496
    {"DAC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_out.settings.fmt,
1497 1498
     "Format for fixed host DAC", NULL, 0},

B
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1499
    {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_out.settings.nchannels,
1500 1501
     "Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0},

B
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1502
    {"DAC_VOICES", AUD_OPT_INT, &conf.fixed_out.nb_voices,
1503 1504 1505
     "Number of voices for DAC", NULL, 0},

    /* ADC */
B
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1506
    {"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_in.enabled,
1507 1508
     "Use fixed settings for host ADC", NULL, 0},

B
bellard 已提交
1509 1510
    {"ADC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_in.settings.freq,
     "Frequency for fixed host ADC", NULL, 0},
1511

B
bellard 已提交
1512 1513
    {"ADC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_in.settings.fmt,
     "Format for fixed host ADC", NULL, 0},
1514

B
bellard 已提交
1515
    {"ADC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_in.settings.nchannels,
1516 1517
     "Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0},

B
bellard 已提交
1518
    {"ADC_VOICES", AUD_OPT_INT, &conf.fixed_in.nb_voices,
1519 1520 1521
     "Number of voices for ADC", NULL, 0},

    /* Misc */
B
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1522 1523
    {"TIMER_PERIOD", AUD_OPT_INT, &conf.period.hz,
     "Timer period in HZ (0 - use lowest possible)", NULL, 0},
1524

B
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1525
    {"PLIVE", AUD_OPT_BOOL, &conf.plive,
1526 1527
     "(undocumented)", NULL, 0},

B
bellard 已提交
1528
    {"LOG_TO_MONITOR", AUD_OPT_BOOL, &conf.log_to_monitor,
A
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1529
     "print logging messages to monitor instead of stderr", NULL, 0},
B
bellard 已提交
1530

1531
    {NULL, 0, NULL, NULL, NULL, 0}
B
bellard 已提交
1532 1533
};

B
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1534 1535 1536 1537 1538 1539 1540 1541 1542 1543 1544 1545 1546 1547 1548 1549 1550 1551 1552
static void audio_pp_nb_voices (const char *typ, int nb)
{
    switch (nb) {
    case 0:
        printf ("Does not support %s\n", typ);
        break;
    case 1:
        printf ("One %s voice\n", typ);
        break;
    case INT_MAX:
        printf ("Theoretically supports many %s voices\n", typ);
        break;
    default:
        printf ("Theoretically supports upto %d %s voices\n", nb, typ);
        break;
    }

}

1553 1554 1555 1556 1557 1558 1559 1560 1561 1562 1563 1564 1565 1566 1567 1568 1569 1570 1571 1572 1573 1574 1575 1576
void AUD_help (void)
{
    size_t i;

    audio_process_options ("AUDIO", audio_options);
    for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
        struct audio_driver *d = drvtab[i];
        if (d->options) {
            audio_process_options (d->name, d->options);
        }
    }

    printf ("Audio options:\n");
    audio_print_options ("AUDIO", audio_options);
    printf ("\n");

    printf ("Available drivers:\n");

    for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
        struct audio_driver *d = drvtab[i];

        printf ("Name: %s\n", d->name);
        printf ("Description: %s\n", d->descr);

B
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1577 1578
        audio_pp_nb_voices ("playback", d->max_voices_out);
        audio_pp_nb_voices ("capture", d->max_voices_in);
1579 1580 1581 1582 1583 1584 1585 1586 1587 1588 1589 1590 1591 1592 1593 1594

        if (d->options) {
            printf ("Options:\n");
            audio_print_options (d->name, d->options);
        }
        else {
            printf ("No options\n");
        }
        printf ("\n");
    }

    printf (
        "Options are settable through environment variables.\n"
        "Example:\n"
#ifdef _WIN32
        "  set QEMU_AUDIO_DRV=wav\n"
B
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1595
        "  set QEMU_WAV_PATH=c:\\tune.wav\n"
1596 1597 1598 1599 1600 1601 1602 1603 1604
#else
        "  export QEMU_AUDIO_DRV=wav\n"
        "  export QEMU_WAV_PATH=$HOME/tune.wav\n"
        "(for csh replace export with setenv in the above)\n"
#endif
        "  qemu ...\n\n"
        );
}

B
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1605
static int audio_driver_init (AudioState *s, struct audio_driver *drv)
B
bellard 已提交
1606
{
1607 1608 1609
    if (drv->options) {
        audio_process_options (drv->name, drv->options);
    }
B
bellard 已提交
1610
    s->drv_opaque = drv->init ();
1611

B
bellard 已提交
1612
    if (s->drv_opaque) {
B
bellard 已提交
1613 1614
        audio_init_nb_voices_out (s, drv);
        audio_init_nb_voices_in (s, drv);
B
bellard 已提交
1615
        s->drv = drv;
1616
        return 0;
B
bellard 已提交
1617 1618
    }
    else {
1619 1620
        dolog ("Could not init `%s' audio driver\n", drv->name);
        return -1;
B
bellard 已提交
1621 1622 1623
    }
}

B
bellard 已提交
1624
static void audio_vm_change_state_handler (void *opaque, int running)
B
bellard 已提交
1625
{
B
bellard 已提交
1626
    AudioState *s = opaque;
1627 1628
    HWVoiceOut *hwo = NULL;
    HWVoiceIn *hwi = NULL;
B
bellard 已提交
1629
    int op = running ? VOICE_ENABLE : VOICE_DISABLE;
1630

B
bellard 已提交
1631 1632
    while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
        hwo->pcm_ops->ctl_out (hwo, op);
1633
    }
B
bellard 已提交
1634

B
bellard 已提交
1635 1636
    while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
        hwi->pcm_ops->ctl_in (hwi, op);
B
bellard 已提交
1637 1638 1639 1640 1641
    }
}

static void audio_atexit (void)
{
B
bellard 已提交
1642
    AudioState *s = &glob_audio_state;
1643 1644 1645
    HWVoiceOut *hwo = NULL;
    HWVoiceIn *hwi = NULL;

B
bellard 已提交
1646
    while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
B
bellard 已提交
1647
        SWVoiceCap *sc;
1648

B
bellard 已提交
1649
        hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
1650
        hwo->pcm_ops->fini_out (hwo);
1651

B
bellard 已提交
1652 1653 1654 1655 1656 1657 1658
        for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
            CaptureVoiceOut *cap = sc->cap;
            struct capture_callback *cb;

            for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
                cb->ops.destroy (cb->opaque);
            }
1659
        }
1660
    }
B
bellard 已提交
1661

B
bellard 已提交
1662 1663
    while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
        hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
1664
        hwi->pcm_ops->fini_in (hwi);
B
bellard 已提交
1665
    }
B
bellard 已提交
1666 1667 1668 1669

    if (s->drv) {
        s->drv->fini (s->drv_opaque);
    }
B
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1670 1671 1672 1673
}

static void audio_save (QEMUFile *f, void *opaque)
{
1674 1675
    (void) f;
    (void) opaque;
B
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1676 1677 1678 1679
}

static int audio_load (QEMUFile *f, void *opaque, int version_id)
{
1680 1681 1682 1683
    (void) f;
    (void) opaque;

    if (version_id != 1) {
B
bellard 已提交
1684
        return -EINVAL;
1685
    }
B
bellard 已提交
1686 1687 1688 1689

    return 0;
}

B
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1690 1691 1692 1693 1694 1695 1696 1697 1698 1699 1700 1701 1702 1703 1704 1705
void AUD_register_card (AudioState *s, const char *name, QEMUSoundCard *card)
{
    card->audio = s;
    card->name = qemu_strdup (name);
    memset (&card->entries, 0, sizeof (card->entries));
    LIST_INSERT_HEAD (&s->card_head, card, entries);
}

void AUD_remove_card (QEMUSoundCard *card)
{
    LIST_REMOVE (card, entries);
    card->audio = NULL;
    qemu_free (card->name);
}

AudioState *AUD_init (void)
B
bellard 已提交
1706
{
1707
    size_t i;
B
bellard 已提交
1708 1709
    int done = 0;
    const char *drvname;
B
bellard 已提交
1710
    AudioState *s = &glob_audio_state;
1711

B
bellard 已提交
1712 1713
    LIST_INIT (&s->hw_head_out);
    LIST_INIT (&s->hw_head_in);
1714
    LIST_INIT (&s->cap_head);
B
bellard 已提交
1715 1716 1717 1718 1719 1720 1721 1722
    atexit (audio_atexit);

    s->ts = qemu_new_timer (vm_clock, audio_timer, s);
    if (!s->ts) {
        dolog ("Could not create audio timer\n");
        return NULL;
    }

1723 1724
    audio_process_options ("AUDIO", audio_options);

B
bellard 已提交
1725 1726 1727
    s->nb_hw_voices_out = conf.fixed_out.nb_voices;
    s->nb_hw_voices_in = conf.fixed_in.nb_voices;

1728
    if (s->nb_hw_voices_out <= 0) {
B
bellard 已提交
1729
        dolog ("Bogus number of playback voices %d, setting to 1\n",
1730 1731 1732 1733 1734
               s->nb_hw_voices_out);
        s->nb_hw_voices_out = 1;
    }

    if (s->nb_hw_voices_in <= 0) {
B
bellard 已提交
1735
        dolog ("Bogus number of capture voices %d, setting to 0\n",
1736
               s->nb_hw_voices_in);
B
bellard 已提交
1737
        s->nb_hw_voices_in = 0;
1738
    }
B
bellard 已提交
1739

1740 1741 1742 1743
    {
        int def;
        drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
    }
B
bellard 已提交
1744 1745 1746

    if (drvname) {
        int found = 0;
1747

B
bellard 已提交
1748 1749
        for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
            if (!strcmp (drvname, drvtab[i]->name)) {
B
bellard 已提交
1750
                done = !audio_driver_init (s, drvtab[i]);
B
bellard 已提交
1751 1752 1753 1754
                found = 1;
                break;
            }
        }
1755

B
bellard 已提交
1756 1757
        if (!found) {
            dolog ("Unknown audio driver `%s'\n", drvname);
1758
            dolog ("Run with -audio-help to list available drivers\n");
B
bellard 已提交
1759 1760 1761 1762 1763
        }
    }

    if (!done) {
        for (i = 0; !done && i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
1764
            if (drvtab[i]->can_be_default) {
B
bellard 已提交
1765
                done = !audio_driver_init (s, drvtab[i]);
1766
            }
B
bellard 已提交
1767 1768 1769 1770
        }
    }

    if (!done) {
B
bellard 已提交
1771 1772 1773
        done = !audio_driver_init (s, &no_audio_driver);
        if (!done) {
            dolog ("Could not initialize audio subsystem\n");
1774 1775
        }
        else {
B
bellard 已提交
1776
            dolog ("warning: Using timer based audio emulation\n");
1777
        }
B
bellard 已提交
1778
    }
1779

B
bellard 已提交
1780
    if (done) {
B
bellard 已提交
1781 1782
        VMChangeStateEntry *e;

B
bellard 已提交
1783 1784 1785 1786 1787 1788 1789
        if (conf.period.hz <= 0) {
            if (conf.period.hz < 0) {
                dolog ("warning: Timer period is negative - %d "
                       "treating as zero\n",
                       conf.period.hz);
            }
            conf.period.ticks = 1;
1790
        }
B
bellard 已提交
1791 1792 1793 1794
        else {
            conf.period.ticks = ticks_per_sec / conf.period.hz;
        }

B
bellard 已提交
1795 1796 1797 1798 1799
        e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
        if (!e) {
            dolog ("warning: Could not register change state handler\n"
                   "(Audio can continue looping even after stopping the VM)\n");
        }
1800 1801
    }
    else {
B
bellard 已提交
1802 1803
        qemu_del_timer (s->ts);
        return NULL;
1804 1805
    }

B
bellard 已提交
1806 1807 1808 1809
    LIST_INIT (&s->card_head);
    register_savevm ("audio", 0, 1, audio_save, audio_load, s);
    qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
    return s;
B
bellard 已提交
1810
}
1811

B
bellard 已提交
1812
CaptureVoiceOut *AUD_add_capture (
1813 1814 1815 1816 1817 1818 1819 1820 1821 1822 1823 1824 1825 1826
    AudioState *s,
    audsettings_t *as,
    struct audio_capture_ops *ops,
    void *cb_opaque
    )
{
    CaptureVoiceOut *cap;
    struct capture_callback *cb;

    if (!s) {
        /* XXX suppress */
        s = &glob_audio_state;
    }

B
bellard 已提交
1827
    if (audio_validate_settings (as)) {
1828 1829
        dolog ("Invalid settings were passed when trying to add capture\n");
        audio_print_settings (as);
B
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1830
        goto err0;
1831 1832 1833 1834 1835 1836 1837 1838 1839 1840 1841
    }

    cb = audio_calloc (AUDIO_FUNC, 1, sizeof (*cb));
    if (!cb) {
        dolog ("Could not allocate capture callback information, size %zu\n",
               sizeof (*cb));
        goto err0;
    }
    cb->ops = *ops;
    cb->opaque = cb_opaque;

B
bellard 已提交
1842
    cap = audio_pcm_capture_find_specific (s, as);
1843 1844
    if (cap) {
        LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
B
bellard 已提交
1845
        return cap;
1846 1847 1848 1849 1850 1851 1852 1853 1854 1855 1856 1857 1858 1859 1860 1861 1862 1863 1864 1865 1866 1867 1868 1869 1870 1871
    }
    else {
        HWVoiceOut *hw;
        CaptureVoiceOut *cap;

        cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap));
        if (!cap) {
            dolog ("Could not allocate capture voice, size %zu\n",
                   sizeof (*cap));
            goto err1;
        }

        hw = &cap->hw;
        LIST_INIT (&hw->sw_head);
        LIST_INIT (&cap->cb_head);

        /* XXX find a more elegant way */
        hw->samples = 4096 * 4;
        hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
                                    sizeof (st_sample_t));
        if (!hw->mix_buf) {
            dolog ("Could not allocate capture mix buffer (%d samples)\n",
                   hw->samples);
            goto err2;
        }

B
bellard 已提交
1872
        audio_pcm_init_info (&hw->info, as);
1873 1874 1875 1876 1877 1878 1879 1880 1881 1882 1883 1884

        cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
        if (!cap->buf) {
            dolog ("Could not allocate capture buffer "
                   "(%d samples, each %d bytes)\n",
                   hw->samples, 1 << hw->info.shift);
            goto err3;
        }

        hw->clip = mixeng_clip
            [hw->info.nchannels == 2]
            [hw->info.sign]
B
bellard 已提交
1885
            [hw->info.swap_endianness]
1886
            [audio_bits_to_index (hw->info.bits)];
1887 1888 1889 1890 1891 1892 1893 1894

        LIST_INSERT_HEAD (&s->cap_head, cap, entries);
        LIST_INSERT_HEAD (&cap->cb_head, cb, entries);

        hw = NULL;
        while ((hw = audio_pcm_hw_find_any_out (s, hw))) {
            audio_attach_capture (s, hw);
        }
B
bellard 已提交
1895
        return cap;
1896 1897 1898 1899 1900 1901 1902 1903

    err3:
        qemu_free (cap->hw.mix_buf);
    err2:
        qemu_free (cap);
    err1:
        qemu_free (cb);
    err0:
B
bellard 已提交
1904 1905 1906 1907 1908 1909 1910 1911 1912 1913 1914 1915 1916 1917 1918 1919
        return NULL;
    }
}

void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
{
    struct capture_callback *cb;

    for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
        if (cb->opaque == cb_opaque) {
            cb->ops.destroy (cb_opaque);
            LIST_REMOVE (cb, entries);
            qemu_free (cb);

            if (!cap->cb_head.lh_first) {
                SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
B
bellard 已提交
1920

B
bellard 已提交
1921
                while (sw) {
B
bellard 已提交
1922
                    SWVoiceCap *sc = (SWVoiceCap *) sw;
B
bellard 已提交
1923 1924 1925
#ifdef DEBUG_CAPTURE
                    dolog ("freeing %s\n", sw->name);
#endif
B
bellard 已提交
1926

B
bellard 已提交
1927 1928 1929 1930 1931 1932
                    sw1 = sw->entries.le_next;
                    if (sw->rate) {
                        st_rate_stop (sw->rate);
                        sw->rate = NULL;
                    }
                    LIST_REMOVE (sw, entries);
B
bellard 已提交
1933 1934
                    LIST_REMOVE (sc, entries);
                    qemu_free (sc);
B
bellard 已提交
1935 1936 1937 1938 1939 1940 1941
                    sw = sw1;
                }
                LIST_REMOVE (cap, entries);
                qemu_free (cap);
            }
            return;
        }
1942 1943
    }
}
1944 1945 1946 1947 1948 1949 1950 1951 1952 1953 1954 1955 1956 1957 1958 1959 1960 1961

void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
{
    if (sw) {
        sw->vol.mute = mute;
        sw->vol.l = nominal_volume.l * lvol / 255;
        sw->vol.r = nominal_volume.r * rvol / 255;
    }
}

void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
{
    if (sw) {
        sw->vol.mute = mute;
        sw->vol.l = nominal_volume.l * lvol / 255;
        sw->vol.r = nominal_volume.r * rvol / 255;
    }
}