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;
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    struct audsettings settings;
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};

static struct {
    struct fixed_settings fixed_out;
    struct fixed_settings fixed_in;
    union {
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        int hertz;
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        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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    { 250 },                    /* 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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struct mixeng_volume nominal_volume = {
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    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_";
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    size_t len, i;
    char *r, *u;
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    if (!s) {
        return NULL;
    }
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    len = strlen (s);
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    r = qemu_malloc (len + sizeof (qemu_prefix));
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    u = r + sizeof (qemu_prefix) - 1;
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    pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
    pstrcat (r, len + sizeof (qemu_prefix), s);
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    for (i = 0; i < len; ++i) {
        u[i] = qemu_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_";
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    size_t preflen, optlen;
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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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        optlen = len + preflen + sizeof (qemu_prefix) + 1;
        optname = qemu_malloc (optlen);
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        pstrcpy (optname, optlen, qemu_prefix);
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        /* copy while upper-casing, including trailing zero */
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        for (i = 0; i <= preflen; ++i) {
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            optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
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        }
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        pstrcat (optname, optlen, "_");
        pstrcat (optname, optlen, opt->name);
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        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 (struct audsettings *as)
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{
    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 (struct audsettings *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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}

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static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *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, struct audsettings *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);
653
            break;
654

655 656 657 658 659 660 661 662 663 664 665 666 667 668
        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;
                }
669
            }
670 671 672 673 674 675 676 677 678 679 680 681
            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);
                }
682

683 684 685
                for (i = 0; i < len << shift; i++) {
                    p[i] = s;
                }
686
            }
687 688 689 690 691 692
            break;

        default:
            AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
                     info->bits);
            break;
693
        }
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    }
}

697 698 699
/*
 * Capture
 */
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static void noop_conv (struct st_sample *dst, const void *src,
                       int samples, struct mixeng_volume *vol)
702 703 704 705 706 707 708 709 710
{
    (void) src;
    (void) dst;
    (void) samples;
    (void) vol;
}

static CaptureVoiceOut *audio_pcm_capture_find_specific (
    AudioState *s,
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    struct audsettings *as
712 713 714 715 716
    )
{
    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)) {
718 719 720 721 722 723
            return cap;
        }
    }
    return NULL;
}

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static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
725
{
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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);
    }
}
735

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static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
{
    if (cap->hw.enabled != enabled) {
        audcnotification_e cmd;
740
        cap->hw.enabled = enabled;
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        cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
        audio_notify_capture (cap, cmd);
743 744 745 746 747 748 749 750 751
    }
}

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) {
753 754 755 756 757
        if (sw->active) {
            enabled = 1;
            break;
        }
    }
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    audio_capture_maybe_changed (cap, enabled);
759 760 761 762
}

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;
770 771 772 773 774 775 776

        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;
786 787 788 789 790 791 792 793 794
    }
}

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;
796
        SWVoiceOut *sw;
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        HWVoiceOut *hw_cap = &cap->hw;
798

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        sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
        if (!sc) {
801
            dolog ("Could not allocate soft capture voice (%zu bytes)\n",
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                   sizeof (*sc));
803 804 805
            return -1;
        }

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        sc->cap = cap;
        sw = &sc->sw;
808
        sw->hw = hw_cap;
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        sw->info = hw->info;
810 811 812 813 814 815 816 817 818 819 820
        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
827
        if (sw->active) {
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            audio_capture_maybe_changed (cap, 1);
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        }
    }
    return 0;
}

834 835 836
/*
 * Hard voice (capture)
 */
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static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
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{
839 840 841 842 843 844 845
    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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    }
847
    return m;
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}

850
int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
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{
852 853 854 855
    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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    }
857
    return live;
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}

860 861 862 863 864 865 866 867 868 869 870 871 872 873 874 875 876
/*
 * 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 {
879
        return hw->samples + rpos;
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    }
}

883
int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
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{
885 886
    HWVoiceIn *hw = sw->hw;
    int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
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    struct st_sample *src, *dst = sw->buf;
888 889 890 891 892 893 894 895 896 897 898 899 900

    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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902 903
    swlim = (live * sw->ratio) >> 32;
    swlim = audio_MIN (swlim, samples);
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905 906 907 908 909 910 911
    while (swlim) {
        src = hw->conv_buf + rpos;
        isamp = hw->wpos - rpos;
        /* XXX: <= ? */
        if (isamp <= 0) {
            isamp = hw->samples - rpos;
        }
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913 914 915 916
        if (!isamp) {
            break;
        }
        osamp = swlim;
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918 919
        if (audio_bug (AUDIO_FUNC, osamp < 0)) {
            dolog ("osamp=%d\n", osamp);
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            return 0;
921
        }
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923 924 925 926 927 928 929
        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);
932 933
    sw->total_hw_samples_acquired += total;
    return ret << sw->info.shift;
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}

936 937 938
/*
 * Hard voice (playback)
 */
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static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
940
{
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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;
954
}
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956 957 958
int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
{
    int smin;
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960 961 962 963
    smin = audio_pcm_hw_find_min_out (hw, nb_live);

    if (!*nb_live) {
        return 0;
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    }
    else {
966 967 968 969 970
        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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        }
972
        return live;
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    }
974 975 976 977 978 979
}

int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
{
    int nb_live;
    int live;
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981 982 983 984
    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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    }
986
    return live;
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}

989 990 991 992
/*
 * Soft voice (playback)
 */
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
B
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{
994 995
    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
    int ret = 0, pos = 0, total = 0;
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997 998 999
    if (!sw) {
        return size;
    }
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1001
    hwsamples = sw->hw->samples;
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1003 1004 1005 1006 1007
    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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1009
    if (live == hwsamples) {
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1010 1011 1012
#ifdef DEBUG_OUT
        dolog ("%s is full %d\n", sw->name, live);
#endif
1013 1014
        return 0;
    }
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1016 1017
    wpos = (sw->hw->rpos + live) % hwsamples;
    samples = size >> sw->info.shift;
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1019 1020 1021 1022 1023 1024 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
    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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1055 1056
        "%s: write size %d ret %d total sw %d\n",
        SW_NAME (sw),
1057 1058
        size >> sw->info.shift,
        ret,
B
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        sw->total_hw_samples_mixed
1060 1061 1062 1063
        );
#endif

    return ret << sw->info.shift;
B
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1064 1065
}

1066 1067
#ifdef DEBUG_AUDIO
static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
B
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1068
{
1069 1070
    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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1071
}
1072
#endif
B
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1074 1075 1076 1077 1078 1079
#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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1080
{
1081
    int bytes;
B
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1082

1083 1084 1085 1086
    if (!sw) {
        /* XXX: Consider options */
        return size;
    }
B
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1087

1088
    if (!sw->hw->enabled) {
B
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1089
        dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
B
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1090 1091 1092
        return 0;
    }

1093 1094 1095 1096 1097 1098 1099
    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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1100

1101 1102 1103
    if (!sw) {
        /* XXX: Consider options */
        return size;
B
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1104
    }
1105 1106

    if (!sw->hw->enabled) {
B
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1107
        dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
1108
        return 0;
B
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1109
    }
1110 1111 1112

    bytes = sw->hw->pcm_ops->read (sw, buf, size);
    return bytes;
B
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1113 1114
}

1115
int AUD_get_buffer_size_out (SWVoiceOut *sw)
B
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1116
{
B
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1117
    return sw->hw->samples << sw->hw->info.shift;
1118 1119 1120 1121 1122
}

void AUD_set_active_out (SWVoiceOut *sw, int on)
{
    HWVoiceOut *hw;
B
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1123

1124
    if (!sw) {
B
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1125
        return;
1126
    }
B
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1127 1128

    hw = sw->hw;
1129 1130
    if (sw->active != on) {
        SWVoiceOut *temp_sw;
B
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1131
        SWVoiceCap *sc;
1132 1133 1134 1135 1136 1137 1138 1139 1140 1141 1142 1143 1144 1145 1146 1147 1148 1149 1150

        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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1152 1153 1154

        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
            sc->sw.active = hw->enabled;
1155
            if (hw->enabled) {
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                audio_capture_maybe_changed (sc->cap, 1);
1157 1158
            }
        }
1159 1160 1161 1162 1163 1164 1165 1166 1167 1168
        sw->active = on;
    }
}

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

    if (!sw) {
        return;
B
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1169 1170
    }

1171
    hw = sw->hw;
B
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1172
    if (sw->active != on) {
1173 1174
        SWVoiceIn *temp_sw;

B
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1175 1176 1177
        if (on) {
            if (!hw->enabled) {
                hw->enabled = 1;
1178
                hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
B
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1179
            }
1180
            sw->total_hw_samples_acquired = hw->total_samples_captured;
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1181 1182
        }
        else {
1183
            if (hw->enabled) {
B
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                int nb_active = 0;
1185 1186 1187 1188

                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) {
1192 1193
                    hw->enabled = 0;
                    hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
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                }
            }
        }
        sw->active = on;
    }
}

1201 1202 1203 1204 1205 1206 1207 1208 1209 1210 1211 1212 1213 1214 1215
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),
1218 1219 1220 1221 1222 1223 1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235
        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;
1237 1238 1239 1240 1241
    }

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

#ifdef DEBUG_OUT
B
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    dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
B
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1243
           SW_NAME (sw),
1244
           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
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#endif
1246 1247 1248 1249

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

1250 1251 1252 1253 1254
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;
1256

B
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1257 1258
        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
            SWVoiceOut *sw = &sc->sw;
1259 1260 1261 1262 1263 1264 1265 1266 1267 1268 1269 1270
            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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1271 1272 1273
                    dolog ("Could not mix %d bytes into a capture "
                           "buffer, mixed %d\n",
                           bytes, written);
1274 1275 1276 1277 1278 1279 1280 1281 1282 1283 1284 1285 1286
                    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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1287
static void audio_run_out (AudioState *s)
1288 1289 1290 1291
{
    HWVoiceOut *hw = NULL;
    SWVoiceOut *sw;

B
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1292
    while ((hw = audio_pcm_hw_find_any_enabled_out (s, hw))) {
1293
        int played;
1294
        int live, free, nb_live, cleanup_required, prev_rpos;
1295 1296 1297 1298 1299

        live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
        if (!nb_live) {
            live = 0;
        }
B
bellard 已提交
1300

1301 1302
        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
1303
            continue;
1304 1305 1306
        }

        if (hw->pending_disable && !nb_live) {
B
bellard 已提交
1307
            SWVoiceCap *sc;
1308 1309
#ifdef DEBUG_OUT
            dolog ("Disabling voice\n");
B
bellard 已提交
1310
#endif
1311 1312 1313
            hw->enabled = 0;
            hw->pending_disable = 0;
            hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
B
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1314 1315 1316
            for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
                sc->sw.active = 0;
                audio_recalc_and_notify_capture (sc->cap);
1317
            }
1318 1319 1320 1321 1322 1323 1324 1325 1326 1327 1328 1329 1330 1331 1332
            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;
        }

1333
        prev_rpos = hw->rpos;
1334 1335 1336 1337 1338 1339 1340 1341
        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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1342
        dolog ("played=%d\n", played);
B
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1343
#endif
1344 1345 1346

        if (played) {
            hw->ts_helper += played;
1347
            audio_capture_mix_and_clear (hw, prev_rpos, played);
1348 1349
        }

B
bellard 已提交
1350
        cleanup_required = 0;
1351 1352 1353 1354 1355 1356 1357 1358 1359 1360 1361 1362 1363 1364 1365
        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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1366
                cleanup_required |= !sw->active && !sw->callback.fn;
1367 1368 1369 1370 1371 1372 1373 1374 1375
            }

            if (sw->active) {
                free = audio_get_free (sw);
                if (free > 0) {
                    sw->callback.fn (sw->callback.opaque, free);
                }
            }
        }
B
bellard 已提交
1376 1377

        if (cleanup_required) {
B
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1378 1379 1380 1381 1382
            SWVoiceOut *sw1;

            sw = hw->sw_head.lh_first;
            while (sw) {
                sw1 = sw->entries.le_next;
B
bellard 已提交
1383 1384 1385 1386 1387 1388
                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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1389
                sw = sw1;
B
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1390 1391
            }
        }
1392 1393 1394
    }
}

B
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1395
static void audio_run_in (AudioState *s)
1396 1397 1398
{
    HWVoiceIn *hw = NULL;

B
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1399
    while ((hw = audio_pcm_hw_find_any_enabled_in (s, hw))) {
1400 1401 1402 1403 1404 1405 1406 1407 1408 1409 1410 1411 1412 1413 1414 1415 1416 1417 1418 1419 1420 1421 1422 1423
        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);
                }
            }
        }
    }
}

1424 1425 1426 1427 1428 1429 1430 1431 1432 1433 1434 1435 1436 1437
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);
M
malc 已提交
1438
            struct st_sample *src;
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
            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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1471 1472 1473 1474 1475 1476
static void audio_timer (void *opaque)
{
    AudioState *s = opaque;

    audio_run_out (s);
    audio_run_in (s);
1477
    audio_run_capture (s);
B
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1478 1479 1480 1481

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

1482 1483
static struct audio_option audio_options[] = {
    /* DAC */
B
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1484
    {"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_out.enabled,
1485 1486
     "Use fixed settings for host DAC", NULL, 0},

B
bellard 已提交
1487
    {"DAC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_out.settings.freq,
1488 1489
     "Frequency for fixed host DAC", NULL, 0},

B
bellard 已提交
1490
    {"DAC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_out.settings.fmt,
1491 1492
     "Format for fixed host DAC", NULL, 0},

B
bellard 已提交
1493
    {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_out.settings.nchannels,
1494 1495
     "Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0},

B
bellard 已提交
1496
    {"DAC_VOICES", AUD_OPT_INT, &conf.fixed_out.nb_voices,
1497 1498 1499
     "Number of voices for DAC", NULL, 0},

    /* ADC */
B
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1500
    {"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_in.enabled,
1501 1502
     "Use fixed settings for host ADC", NULL, 0},

B
bellard 已提交
1503 1504
    {"ADC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_in.settings.freq,
     "Frequency for fixed host ADC", NULL, 0},
1505

B
bellard 已提交
1506 1507
    {"ADC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_in.settings.fmt,
     "Format for fixed host ADC", NULL, 0},
1508

B
bellard 已提交
1509
    {"ADC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_in.settings.nchannels,
1510 1511
     "Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0},

B
bellard 已提交
1512
    {"ADC_VOICES", AUD_OPT_INT, &conf.fixed_in.nb_voices,
1513 1514 1515
     "Number of voices for ADC", NULL, 0},

    /* Misc */
M
malc 已提交
1516
    {"TIMER_PERIOD", AUD_OPT_INT, &conf.period.hertz,
B
bellard 已提交
1517
     "Timer period in HZ (0 - use lowest possible)", NULL, 0},
1518

B
bellard 已提交
1519
    {"PLIVE", AUD_OPT_BOOL, &conf.plive,
1520 1521
     "(undocumented)", NULL, 0},

B
bellard 已提交
1522
    {"LOG_TO_MONITOR", AUD_OPT_BOOL, &conf.log_to_monitor,
A
aurel32 已提交
1523
     "print logging messages to monitor instead of stderr", NULL, 0},
B
bellard 已提交
1524

1525
    {NULL, 0, NULL, NULL, NULL, 0}
B
bellard 已提交
1526 1527
};

B
bellard 已提交
1528 1529 1530 1531 1532 1533 1534 1535 1536 1537 1538 1539 1540 1541 1542 1543 1544 1545 1546
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;
    }

}

1547 1548 1549 1550 1551
void AUD_help (void)
{
    size_t i;

    audio_process_options ("AUDIO", audio_options);
1552
    for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1553 1554 1555 1556 1557 1558 1559 1560 1561 1562 1563 1564
        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");

1565
    for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1566 1567 1568 1569 1570
        struct audio_driver *d = drvtab[i];

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

B
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1571 1572
        audio_pp_nb_voices ("playback", d->max_voices_out);
        audio_pp_nb_voices ("capture", d->max_voices_in);
1573 1574 1575 1576 1577 1578 1579 1580 1581 1582 1583 1584 1585 1586 1587 1588

        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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1589
        "  set QEMU_WAV_PATH=c:\\tune.wav\n"
1590 1591 1592 1593 1594 1595 1596 1597 1598
#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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1599
static int audio_driver_init (AudioState *s, struct audio_driver *drv)
B
bellard 已提交
1600
{
1601 1602 1603
    if (drv->options) {
        audio_process_options (drv->name, drv->options);
    }
B
bellard 已提交
1604
    s->drv_opaque = drv->init ();
1605

B
bellard 已提交
1606
    if (s->drv_opaque) {
B
bellard 已提交
1607 1608
        audio_init_nb_voices_out (s, drv);
        audio_init_nb_voices_in (s, drv);
B
bellard 已提交
1609
        s->drv = drv;
1610
        return 0;
B
bellard 已提交
1611 1612
    }
    else {
1613 1614
        dolog ("Could not init `%s' audio driver\n", drv->name);
        return -1;
B
bellard 已提交
1615 1616 1617
    }
}

1618 1619
static void audio_vm_change_state_handler (void *opaque, int running,
                                           int reason)
B
bellard 已提交
1620
{
B
bellard 已提交
1621
    AudioState *s = opaque;
1622 1623
    HWVoiceOut *hwo = NULL;
    HWVoiceIn *hwi = NULL;
B
bellard 已提交
1624
    int op = running ? VOICE_ENABLE : VOICE_DISABLE;
1625

B
bellard 已提交
1626 1627
    while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
        hwo->pcm_ops->ctl_out (hwo, op);
1628
    }
B
bellard 已提交
1629

B
bellard 已提交
1630 1631
    while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
        hwi->pcm_ops->ctl_in (hwi, op);
B
bellard 已提交
1632 1633 1634 1635 1636
    }
}

static void audio_atexit (void)
{
B
bellard 已提交
1637
    AudioState *s = &glob_audio_state;
1638 1639 1640
    HWVoiceOut *hwo = NULL;
    HWVoiceIn *hwi = NULL;

B
bellard 已提交
1641
    while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
B
bellard 已提交
1642
        SWVoiceCap *sc;
1643

B
bellard 已提交
1644
        hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
1645
        hwo->pcm_ops->fini_out (hwo);
1646

B
bellard 已提交
1647 1648 1649 1650 1651 1652 1653
        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);
            }
1654
        }
1655
    }
B
bellard 已提交
1656

B
bellard 已提交
1657 1658
    while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
        hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
1659
        hwi->pcm_ops->fini_in (hwi);
B
bellard 已提交
1660
    }
B
bellard 已提交
1661 1662 1663 1664

    if (s->drv) {
        s->drv->fini (s->drv_opaque);
    }
B
bellard 已提交
1665 1666 1667 1668
}

static void audio_save (QEMUFile *f, void *opaque)
{
1669 1670
    (void) f;
    (void) opaque;
B
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1671 1672 1673 1674
}

static int audio_load (QEMUFile *f, void *opaque, int version_id)
{
1675 1676 1677 1678
    (void) f;
    (void) opaque;

    if (version_id != 1) {
B
bellard 已提交
1679
        return -EINVAL;
1680
    }
B
bellard 已提交
1681 1682 1683 1684

    return 0;
}

B
bellard 已提交
1685 1686 1687 1688 1689 1690 1691 1692 1693 1694 1695 1696 1697 1698 1699 1700
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
1701
{
1702
    size_t i;
B
bellard 已提交
1703 1704
    int done = 0;
    const char *drvname;
B
bellard 已提交
1705
    AudioState *s = &glob_audio_state;
1706

B
bellard 已提交
1707 1708
    LIST_INIT (&s->hw_head_out);
    LIST_INIT (&s->hw_head_in);
1709
    LIST_INIT (&s->cap_head);
B
bellard 已提交
1710 1711 1712 1713 1714 1715 1716 1717
    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;
    }

1718 1719
    audio_process_options ("AUDIO", audio_options);

B
bellard 已提交
1720 1721 1722
    s->nb_hw_voices_out = conf.fixed_out.nb_voices;
    s->nb_hw_voices_in = conf.fixed_in.nb_voices;

1723
    if (s->nb_hw_voices_out <= 0) {
B
bellard 已提交
1724
        dolog ("Bogus number of playback voices %d, setting to 1\n",
1725 1726 1727 1728 1729
               s->nb_hw_voices_out);
        s->nb_hw_voices_out = 1;
    }

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

1735 1736 1737 1738
    {
        int def;
        drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
    }
B
bellard 已提交
1739 1740 1741

    if (drvname) {
        int found = 0;
1742

1743
        for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
B
bellard 已提交
1744
            if (!strcmp (drvname, drvtab[i]->name)) {
B
bellard 已提交
1745
                done = !audio_driver_init (s, drvtab[i]);
B
bellard 已提交
1746 1747 1748 1749
                found = 1;
                break;
            }
        }
1750

B
bellard 已提交
1751 1752
        if (!found) {
            dolog ("Unknown audio driver `%s'\n", drvname);
1753
            dolog ("Run with -audio-help to list available drivers\n");
B
bellard 已提交
1754 1755 1756 1757
        }
    }

    if (!done) {
1758
        for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
1759
            if (drvtab[i]->can_be_default) {
B
bellard 已提交
1760
                done = !audio_driver_init (s, drvtab[i]);
1761
            }
B
bellard 已提交
1762 1763 1764 1765
        }
    }

    if (!done) {
B
bellard 已提交
1766 1767 1768
        done = !audio_driver_init (s, &no_audio_driver);
        if (!done) {
            dolog ("Could not initialize audio subsystem\n");
1769 1770
        }
        else {
B
bellard 已提交
1771
            dolog ("warning: Using timer based audio emulation\n");
1772
        }
B
bellard 已提交
1773
    }
1774

B
bellard 已提交
1775
    if (done) {
B
bellard 已提交
1776 1777
        VMChangeStateEntry *e;

M
malc 已提交
1778 1779
        if (conf.period.hertz <= 0) {
            if (conf.period.hertz < 0) {
B
bellard 已提交
1780 1781
                dolog ("warning: Timer period is negative - %d "
                       "treating as zero\n",
M
malc 已提交
1782
                       conf.period.hertz);
B
bellard 已提交
1783 1784
            }
            conf.period.ticks = 1;
1785
        }
B
bellard 已提交
1786
        else {
M
malc 已提交
1787
            conf.period.ticks = ticks_per_sec / conf.period.hertz;
B
bellard 已提交
1788 1789
        }

B
bellard 已提交
1790 1791 1792 1793 1794
        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");
        }
1795 1796
    }
    else {
B
bellard 已提交
1797 1798
        qemu_del_timer (s->ts);
        return NULL;
1799 1800
    }

B
bellard 已提交
1801 1802 1803 1804
    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
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1805
}
1806

B
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1807
CaptureVoiceOut *AUD_add_capture (
1808
    AudioState *s,
M
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1809
    struct audsettings *as,
1810 1811 1812 1813 1814 1815 1816 1817 1818 1819 1820 1821
    struct audio_capture_ops *ops,
    void *cb_opaque
    )
{
    CaptureVoiceOut *cap;
    struct capture_callback *cb;

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

B
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1822
    if (audio_validate_settings (as)) {
1823 1824
        dolog ("Invalid settings were passed when trying to add capture\n");
        audio_print_settings (as);
B
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1825
        goto err0;
1826 1827 1828 1829 1830 1831 1832 1833 1834 1835 1836
    }

    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
1837
    cap = audio_pcm_capture_find_specific (s, as);
1838 1839
    if (cap) {
        LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
B
bellard 已提交
1840
        return cap;
1841 1842 1843 1844 1845 1846 1847 1848 1849 1850 1851 1852 1853 1854 1855 1856 1857 1858 1859
    }
    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,
M
malc 已提交
1860
                                    sizeof (struct st_sample));
1861 1862 1863 1864 1865 1866
        if (!hw->mix_buf) {
            dolog ("Could not allocate capture mix buffer (%d samples)\n",
                   hw->samples);
            goto err2;
        }

B
bellard 已提交
1867
        audio_pcm_init_info (&hw->info, as);
1868 1869 1870 1871 1872 1873 1874 1875 1876 1877 1878 1879

        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
1880
            [hw->info.swap_endianness]
1881
            [audio_bits_to_index (hw->info.bits)];
1882 1883 1884 1885 1886 1887 1888 1889

        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
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1890
        return cap;
1891 1892 1893 1894 1895 1896 1897 1898

    err3:
        qemu_free (cap->hw.mix_buf);
    err2:
        qemu_free (cap);
    err1:
        qemu_free (cb);
    err0:
B
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1899 1900 1901 1902 1903 1904 1905 1906 1907 1908 1909 1910 1911 1912 1913 1914
        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
1915

B
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1916
                while (sw) {
B
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1917
                    SWVoiceCap *sc = (SWVoiceCap *) sw;
B
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1918 1919 1920
#ifdef DEBUG_CAPTURE
                    dolog ("freeing %s\n", sw->name);
#endif
B
bellard 已提交
1921

B
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1922 1923 1924 1925 1926 1927
                    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
1928 1929
                    LIST_REMOVE (sc, entries);
                    qemu_free (sc);
B
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1930 1931 1932 1933 1934 1935 1936
                    sw = sw1;
                }
                LIST_REMOVE (cap, entries);
                qemu_free (cap);
            }
            return;
        }
1937 1938
    }
}
1939 1940 1941 1942 1943 1944 1945 1946 1947 1948 1949 1950 1951 1952 1953 1954 1955 1956

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;
    }
}