audio.c 48.4 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 {
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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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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);
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    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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        pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
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        pstrcat (r, len + sizeof (qemu_prefix), s);
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        for (i = 0; i < len; ++i) {
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            u[i] = qemu_toupper(u[i]);
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        }
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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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        if (!optname) {
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            dolog ("Could not allocate memory for option name `%s'\n",
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                   opt->name);
            continue;
        }

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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 (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 {
657 658
        switch (info->bits) {
        case 8:
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            memset (buf, 0x80, len << info->shift);
660
            break;
661

662 663 664 665 666 667 668 669 670 671 672 673 674 675
        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;
                }
676
            }
677 678 679 680 681 682 683 684 685 686 687 688
            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);
                }
689

690 691 692
                for (i = 0; i < len << shift; i++) {
                    p[i] = s;
                }
693
            }
694 695 696 697 698 699
            break;

        default:
            AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
                     info->bits);
            break;
700
        }
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    }
}

704 705 706 707 708 709 710 711 712 713 714 715 716 717
/*
 * 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
719 720 721 722 723
    )
{
    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)) {
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            return cap;
        }
    }
    return NULL;
}

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static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
732
{
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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);
    }
}
742

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static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
{
    if (cap->hw.enabled != enabled) {
        audcnotification_e cmd;
747
        cap->hw.enabled = enabled;
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        cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
        audio_notify_capture (cap, cmd);
750 751 752 753 754 755 756 757 758
    }
}

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) {
760 761 762 763 764
        if (sw->active) {
            enabled = 1;
            break;
        }
    }
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    audio_capture_maybe_changed (cap, enabled);
766 767 768 769
}

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;
777 778 779 780 781 782 783

        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;
793 794 795 796 797 798 799 800 801
    }
}

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;
803
        SWVoiceOut *sw;
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        HWVoiceOut *hw_cap = &cap->hw;
805

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        sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
        if (!sc) {
808
            dolog ("Could not allocate soft capture voice (%zu bytes)\n",
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                   sizeof (*sc));
810 811 812
            return -1;
        }

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        sc->cap = cap;
        sw = &sc->sw;
815
        sw->hw = hw_cap;
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        sw->info = hw->info;
817 818 819 820 821 822 823 824 825 826 827
        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
834
        if (sw->active) {
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            audio_capture_maybe_changed (cap, 1);
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        }
    }
    return 0;
}

841 842 843
/*
 * Hard voice (capture)
 */
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static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
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{
846 847 848 849 850 851 852
    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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    }
854
    return m;
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}

857
int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
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{
859 860 861 862
    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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    }
864
    return live;
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}

867 868 869 870 871 872 873 874 875 876 877 878 879 880 881 882 883
/*
 * 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 {
886
        return hw->samples + rpos;
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    }
}

890
int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
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{
892 893
    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;
895 896 897 898 899 900 901 902 903 904 905 906 907

    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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909 910
    swlim = (live * sw->ratio) >> 32;
    swlim = audio_MIN (swlim, samples);
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912 913 914 915 916 917 918
    while (swlim) {
        src = hw->conv_buf + rpos;
        isamp = hw->wpos - rpos;
        /* XXX: <= ? */
        if (isamp <= 0) {
            isamp = hw->samples - rpos;
        }
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920 921 922 923
        if (!isamp) {
            break;
        }
        osamp = swlim;
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925 926
        if (audio_bug (AUDIO_FUNC, osamp < 0)) {
            dolog ("osamp=%d\n", osamp);
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            return 0;
928
        }
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930 931 932 933 934 935 936
        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);
939 940
    sw->total_hw_samples_acquired += total;
    return ret << sw->info.shift;
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}

943 944 945
/*
 * Hard voice (playback)
 */
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static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
947
{
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948 949 950
    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;
961
}
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963 964 965
int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
{
    int smin;
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967 968 969 970
    smin = audio_pcm_hw_find_min_out (hw, nb_live);

    if (!*nb_live) {
        return 0;
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    }
    else {
973 974 975 976 977
        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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        }
979
        return live;
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    }
981 982 983 984 985 986
}

int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
{
    int nb_live;
    int live;
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988 989 990 991
    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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    }
993
    return live;
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}

996 997 998 999
/*
 * Soft voice (playback)
 */
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
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{
1001 1002
    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
    int ret = 0, pos = 0, total = 0;
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1004 1005 1006
    if (!sw) {
        return size;
    }
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1008
    hwsamples = sw->hw->samples;
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1010 1011 1012 1013 1014
    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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1016
    if (live == hwsamples) {
B
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1017 1018 1019
#ifdef DEBUG_OUT
        dolog ("%s is full %d\n", sw->name, live);
#endif
1020 1021
        return 0;
    }
B
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1022

1023 1024
    wpos = (sw->hw->rpos + live) % hwsamples;
    samples = size >> sw->info.shift;
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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 1061
    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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1062 1063
        "%s: write size %d ret %d total sw %d\n",
        SW_NAME (sw),
1064 1065
        size >> sw->info.shift,
        ret,
B
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        sw->total_hw_samples_mixed
1067 1068 1069 1070
        );
#endif

    return ret << sw->info.shift;
B
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1071 1072
}

1073 1074
#ifdef DEBUG_AUDIO
static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
B
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1075
{
1076 1077
    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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1078
}
1079
#endif
B
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1081 1082 1083 1084 1085 1086
#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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1087
{
1088
    int bytes;
B
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1089

1090 1091 1092 1093
    if (!sw) {
        /* XXX: Consider options */
        return size;
    }
B
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1094

1095
    if (!sw->hw->enabled) {
B
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1096
        dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
B
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1097 1098 1099
        return 0;
    }

1100 1101 1102 1103 1104 1105 1106
    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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1107

1108 1109 1110
    if (!sw) {
        /* XXX: Consider options */
        return size;
B
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1111
    }
1112 1113

    if (!sw->hw->enabled) {
B
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1114
        dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
1115
        return 0;
B
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1116
    }
1117 1118 1119

    bytes = sw->hw->pcm_ops->read (sw, buf, size);
    return bytes;
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1120 1121
}

1122
int AUD_get_buffer_size_out (SWVoiceOut *sw)
B
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1123
{
B
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1124
    return sw->hw->samples << sw->hw->info.shift;
1125 1126 1127 1128 1129
}

void AUD_set_active_out (SWVoiceOut *sw, int on)
{
    HWVoiceOut *hw;
B
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1130

1131
    if (!sw) {
B
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1132
        return;
1133
    }
B
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1134 1135

    hw = sw->hw;
1136 1137
    if (sw->active != on) {
        SWVoiceOut *temp_sw;
B
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1138
        SWVoiceCap *sc;
1139 1140 1141 1142 1143 1144 1145 1146 1147 1148 1149 1150 1151 1152 1153 1154 1155 1156 1157

        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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1159 1160 1161

        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
            sc->sw.active = hw->enabled;
1162
            if (hw->enabled) {
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                audio_capture_maybe_changed (sc->cap, 1);
1164 1165
            }
        }
1166 1167 1168 1169 1170 1171 1172 1173 1174 1175
        sw->active = on;
    }
}

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

    if (!sw) {
        return;
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1176 1177
    }

1178
    hw = sw->hw;
B
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    if (sw->active != on) {
1180 1181
        SWVoiceIn *temp_sw;

B
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1182 1183 1184
        if (on) {
            if (!hw->enabled) {
                hw->enabled = 1;
1185
                hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
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            }
1187
            sw->total_hw_samples_acquired = hw->total_samples_captured;
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1188 1189
        }
        else {
1190
            if (hw->enabled) {
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                int nb_active = 0;
1192 1193 1194 1195

                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) {
1199 1200
                    hw->enabled = 0;
                    hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
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                }
            }
        }
        sw->active = on;
    }
}

1208 1209 1210 1211 1212 1213 1214 1215 1216 1217 1218 1219 1220 1221 1222
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),
1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242
        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);
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        return 0;
1244 1245 1246 1247 1248
    }

    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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1250
           SW_NAME (sw),
1251
           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
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#endif
1253 1254 1255 1256

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

1257 1258 1259 1260 1261
static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
{
    int n;

    if (hw->enabled) {
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        SWVoiceCap *sc;
1263

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1264 1265
        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
            SWVoiceOut *sw = &sc->sw;
1266 1267 1268 1269 1270 1271 1272 1273 1274 1275 1276 1277
            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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1278 1279 1280
                    dolog ("Could not mix %d bytes into a capture "
                           "buffer, mixed %d\n",
                           bytes, written);
1281 1282 1283 1284 1285 1286 1287 1288 1289 1290 1291 1292 1293
                    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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1294
static void audio_run_out (AudioState *s)
1295 1296 1297 1298
{
    HWVoiceOut *hw = NULL;
    SWVoiceOut *sw;

B
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1299
    while ((hw = audio_pcm_hw_find_any_enabled_out (s, hw))) {
1300
        int played;
1301
        int live, free, nb_live, cleanup_required, prev_rpos;
1302 1303 1304 1305 1306

        live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
        if (!nb_live) {
            live = 0;
        }
B
bellard 已提交
1307

1308 1309
        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
1310
            continue;
1311 1312 1313
        }

        if (hw->pending_disable && !nb_live) {
B
bellard 已提交
1314
            SWVoiceCap *sc;
1315 1316
#ifdef DEBUG_OUT
            dolog ("Disabling voice\n");
B
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1317
#endif
1318 1319 1320
            hw->enabled = 0;
            hw->pending_disable = 0;
            hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
B
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1321 1322 1323
            for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
                sc->sw.active = 0;
                audio_recalc_and_notify_capture (sc->cap);
1324
            }
1325 1326 1327 1328 1329 1330 1331 1332 1333 1334 1335 1336 1337 1338 1339
            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;
        }

1340
        prev_rpos = hw->rpos;
1341 1342 1343 1344 1345 1346 1347 1348
        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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1349
        dolog ("played=%d\n", played);
B
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1350
#endif
1351 1352 1353

        if (played) {
            hw->ts_helper += played;
1354
            audio_capture_mix_and_clear (hw, prev_rpos, played);
1355 1356
        }

B
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1357
        cleanup_required = 0;
1358 1359 1360 1361 1362 1363 1364 1365 1366 1367 1368 1369 1370 1371 1372
        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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1373
                cleanup_required |= !sw->active && !sw->callback.fn;
1374 1375 1376 1377 1378 1379 1380 1381 1382
            }

            if (sw->active) {
                free = audio_get_free (sw);
                if (free > 0) {
                    sw->callback.fn (sw->callback.opaque, free);
                }
            }
        }
B
bellard 已提交
1383 1384

        if (cleanup_required) {
B
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1385 1386 1387 1388 1389
            SWVoiceOut *sw1;

            sw = hw->sw_head.lh_first;
            while (sw) {
                sw1 = sw->entries.le_next;
B
bellard 已提交
1390 1391 1392 1393 1394 1395
                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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1396
                sw = sw1;
B
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1397 1398
            }
        }
1399 1400 1401
    }
}

B
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1402
static void audio_run_in (AudioState *s)
1403 1404 1405
{
    HWVoiceIn *hw = NULL;

B
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1406
    while ((hw = audio_pcm_hw_find_any_enabled_in (s, hw))) {
1407 1408 1409 1410 1411 1412 1413 1414 1415 1416 1417 1418 1419 1420 1421 1422 1423 1424 1425 1426 1427 1428 1429 1430
        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);
                }
            }
        }
    }
}

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 1477
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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1478 1479 1480 1481 1482 1483
static void audio_timer (void *opaque)
{
    AudioState *s = opaque;

    audio_run_out (s);
    audio_run_in (s);
1484
    audio_run_capture (s);
B
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1485 1486 1487 1488

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

1489 1490
static struct audio_option audio_options[] = {
    /* DAC */
B
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1491
    {"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_out.enabled,
1492 1493
     "Use fixed settings for host DAC", NULL, 0},

B
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1494
    {"DAC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_out.settings.freq,
1495 1496
     "Frequency for fixed host DAC", NULL, 0},

B
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1497
    {"DAC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_out.settings.fmt,
1498 1499
     "Format for fixed host DAC", NULL, 0},

B
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1500
    {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_out.settings.nchannels,
1501 1502
     "Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0},

B
bellard 已提交
1503
    {"DAC_VOICES", AUD_OPT_INT, &conf.fixed_out.nb_voices,
1504 1505 1506
     "Number of voices for DAC", NULL, 0},

    /* ADC */
B
bellard 已提交
1507
    {"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_in.enabled,
1508 1509
     "Use fixed settings for host ADC", NULL, 0},

B
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1510 1511
    {"ADC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_in.settings.freq,
     "Frequency for fixed host ADC", NULL, 0},
1512

B
bellard 已提交
1513 1514
    {"ADC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_in.settings.fmt,
     "Format for fixed host ADC", NULL, 0},
1515

B
bellard 已提交
1516
    {"ADC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_in.settings.nchannels,
1517 1518
     "Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0},

B
bellard 已提交
1519
    {"ADC_VOICES", AUD_OPT_INT, &conf.fixed_in.nb_voices,
1520 1521 1522
     "Number of voices for ADC", NULL, 0},

    /* Misc */
M
malc 已提交
1523
    {"TIMER_PERIOD", AUD_OPT_INT, &conf.period.hertz,
B
bellard 已提交
1524
     "Timer period in HZ (0 - use lowest possible)", NULL, 0},
1525

B
bellard 已提交
1526
    {"PLIVE", AUD_OPT_BOOL, &conf.plive,
1527 1528
     "(undocumented)", NULL, 0},

B
bellard 已提交
1529
    {"LOG_TO_MONITOR", AUD_OPT_BOOL, &conf.log_to_monitor,
A
aurel32 已提交
1530
     "print logging messages to monitor instead of stderr", NULL, 0},
B
bellard 已提交
1531

1532
    {NULL, 0, NULL, NULL, NULL, 0}
B
bellard 已提交
1533 1534
};

B
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1535 1536 1537 1538 1539 1540 1541 1542 1543 1544 1545 1546 1547 1548 1549 1550 1551 1552 1553
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;
    }

}

1554 1555 1556 1557 1558 1559 1560 1561 1562 1563 1564 1565 1566 1567 1568 1569 1570 1571 1572 1573 1574 1575 1576 1577
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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1578 1579
        audio_pp_nb_voices ("playback", d->max_voices_out);
        audio_pp_nb_voices ("capture", d->max_voices_in);
1580 1581 1582 1583 1584 1585 1586 1587 1588 1589 1590 1591 1592 1593 1594 1595

        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
bellard 已提交
1596
        "  set QEMU_WAV_PATH=c:\\tune.wav\n"
1597 1598 1599 1600 1601 1602 1603 1604 1605
#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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1606
static int audio_driver_init (AudioState *s, struct audio_driver *drv)
B
bellard 已提交
1607
{
1608 1609 1610
    if (drv->options) {
        audio_process_options (drv->name, drv->options);
    }
B
bellard 已提交
1611
    s->drv_opaque = drv->init ();
1612

B
bellard 已提交
1613
    if (s->drv_opaque) {
B
bellard 已提交
1614 1615
        audio_init_nb_voices_out (s, drv);
        audio_init_nb_voices_in (s, drv);
B
bellard 已提交
1616
        s->drv = drv;
1617
        return 0;
B
bellard 已提交
1618 1619
    }
    else {
1620 1621
        dolog ("Could not init `%s' audio driver\n", drv->name);
        return -1;
B
bellard 已提交
1622 1623 1624
    }
}

B
bellard 已提交
1625
static void audio_vm_change_state_handler (void *opaque, int running)
B
bellard 已提交
1626
{
B
bellard 已提交
1627
    AudioState *s = opaque;
1628 1629
    HWVoiceOut *hwo = NULL;
    HWVoiceIn *hwi = NULL;
B
bellard 已提交
1630
    int op = running ? VOICE_ENABLE : VOICE_DISABLE;
1631

B
bellard 已提交
1632 1633
    while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
        hwo->pcm_ops->ctl_out (hwo, op);
1634
    }
B
bellard 已提交
1635

B
bellard 已提交
1636 1637
    while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
        hwi->pcm_ops->ctl_in (hwi, op);
B
bellard 已提交
1638 1639 1640 1641 1642
    }
}

static void audio_atexit (void)
{
B
bellard 已提交
1643
    AudioState *s = &glob_audio_state;
1644 1645 1646
    HWVoiceOut *hwo = NULL;
    HWVoiceIn *hwi = NULL;

B
bellard 已提交
1647
    while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
B
bellard 已提交
1648
        SWVoiceCap *sc;
1649

B
bellard 已提交
1650
        hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
1651
        hwo->pcm_ops->fini_out (hwo);
1652

B
bellard 已提交
1653 1654 1655 1656 1657 1658 1659
        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);
            }
1660
        }
1661
    }
B
bellard 已提交
1662

B
bellard 已提交
1663 1664
    while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
        hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
1665
        hwi->pcm_ops->fini_in (hwi);
B
bellard 已提交
1666
    }
B
bellard 已提交
1667 1668 1669 1670

    if (s->drv) {
        s->drv->fini (s->drv_opaque);
    }
B
bellard 已提交
1671 1672 1673 1674
}

static void audio_save (QEMUFile *f, void *opaque)
{
1675 1676
    (void) f;
    (void) opaque;
B
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1677 1678 1679 1680
}

static int audio_load (QEMUFile *f, void *opaque, int version_id)
{
1681 1682 1683 1684
    (void) f;
    (void) opaque;

    if (version_id != 1) {
B
bellard 已提交
1685
        return -EINVAL;
1686
    }
B
bellard 已提交
1687 1688 1689 1690

    return 0;
}

B
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1691 1692 1693 1694 1695 1696 1697 1698 1699 1700 1701 1702 1703 1704 1705 1706
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
1707
{
1708
    size_t i;
B
bellard 已提交
1709 1710
    int done = 0;
    const char *drvname;
B
bellard 已提交
1711
    AudioState *s = &glob_audio_state;
1712

B
bellard 已提交
1713 1714
    LIST_INIT (&s->hw_head_out);
    LIST_INIT (&s->hw_head_in);
1715
    LIST_INIT (&s->cap_head);
B
bellard 已提交
1716 1717 1718 1719 1720 1721 1722 1723
    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;
    }

1724 1725
    audio_process_options ("AUDIO", audio_options);

B
bellard 已提交
1726 1727 1728
    s->nb_hw_voices_out = conf.fixed_out.nb_voices;
    s->nb_hw_voices_in = conf.fixed_in.nb_voices;

1729
    if (s->nb_hw_voices_out <= 0) {
B
bellard 已提交
1730
        dolog ("Bogus number of playback voices %d, setting to 1\n",
1731 1732 1733 1734 1735
               s->nb_hw_voices_out);
        s->nb_hw_voices_out = 1;
    }

    if (s->nb_hw_voices_in <= 0) {
B
bellard 已提交
1736
        dolog ("Bogus number of capture voices %d, setting to 0\n",
1737
               s->nb_hw_voices_in);
B
bellard 已提交
1738
        s->nb_hw_voices_in = 0;
1739
    }
B
bellard 已提交
1740

1741 1742 1743 1744
    {
        int def;
        drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
    }
B
bellard 已提交
1745 1746 1747

    if (drvname) {
        int found = 0;
1748

B
bellard 已提交
1749 1750
        for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
            if (!strcmp (drvname, drvtab[i]->name)) {
B
bellard 已提交
1751
                done = !audio_driver_init (s, drvtab[i]);
B
bellard 已提交
1752 1753 1754 1755
                found = 1;
                break;
            }
        }
1756

B
bellard 已提交
1757 1758
        if (!found) {
            dolog ("Unknown audio driver `%s'\n", drvname);
1759
            dolog ("Run with -audio-help to list available drivers\n");
B
bellard 已提交
1760 1761 1762 1763 1764
        }
    }

    if (!done) {
        for (i = 0; !done && i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
1765
            if (drvtab[i]->can_be_default) {
B
bellard 已提交
1766
                done = !audio_driver_init (s, drvtab[i]);
1767
            }
B
bellard 已提交
1768 1769 1770 1771
        }
    }

    if (!done) {
B
bellard 已提交
1772 1773 1774
        done = !audio_driver_init (s, &no_audio_driver);
        if (!done) {
            dolog ("Could not initialize audio subsystem\n");
1775 1776
        }
        else {
B
bellard 已提交
1777
            dolog ("warning: Using timer based audio emulation\n");
1778
        }
B
bellard 已提交
1779
    }
1780

B
bellard 已提交
1781
    if (done) {
B
bellard 已提交
1782 1783
        VMChangeStateEntry *e;

M
malc 已提交
1784 1785
        if (conf.period.hertz <= 0) {
            if (conf.period.hertz < 0) {
B
bellard 已提交
1786 1787
                dolog ("warning: Timer period is negative - %d "
                       "treating as zero\n",
M
malc 已提交
1788
                       conf.period.hertz);
B
bellard 已提交
1789 1790
            }
            conf.period.ticks = 1;
1791
        }
B
bellard 已提交
1792
        else {
M
malc 已提交
1793
            conf.period.ticks = ticks_per_sec / conf.period.hertz;
B
bellard 已提交
1794 1795
        }

B
bellard 已提交
1796 1797 1798 1799 1800
        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");
        }
1801 1802
    }
    else {
B
bellard 已提交
1803 1804
        qemu_del_timer (s->ts);
        return NULL;
1805 1806
    }

B
bellard 已提交
1807 1808 1809 1810
    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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1811
}
1812

B
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1813
CaptureVoiceOut *AUD_add_capture (
1814 1815 1816 1817 1818 1819 1820 1821 1822 1823 1824 1825 1826 1827
    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
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1828
    if (audio_validate_settings (as)) {
1829 1830
        dolog ("Invalid settings were passed when trying to add capture\n");
        audio_print_settings (as);
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        goto err0;
1832 1833 1834 1835 1836 1837 1838 1839 1840 1841 1842
    }

    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
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1843
    cap = audio_pcm_capture_find_specific (s, as);
1844 1845
    if (cap) {
        LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
B
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1846
        return cap;
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 1872
    }
    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
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1873
        audio_pcm_init_info (&hw->info, as);
1874 1875 1876 1877 1878 1879 1880 1881 1882 1883 1884 1885

        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
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1886
            [hw->info.swap_endianness]
1887
            [audio_bits_to_index (hw->info.bits)];
1888 1889 1890 1891 1892 1893 1894 1895

        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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1896
        return cap;
1897 1898 1899 1900 1901 1902 1903 1904

    err3:
        qemu_free (cap->hw.mix_buf);
    err2:
        qemu_free (cap);
    err1:
        qemu_free (cb);
    err0:
B
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1905 1906 1907 1908 1909 1910 1911 1912 1913 1914 1915 1916 1917 1918 1919 1920
        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
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1921

B
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1922
                while (sw) {
B
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1923
                    SWVoiceCap *sc = (SWVoiceCap *) sw;
B
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1924 1925 1926
#ifdef DEBUG_CAPTURE
                    dolog ("freeing %s\n", sw->name);
#endif
B
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1927

B
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1928 1929 1930 1931 1932 1933
                    sw1 = sw->entries.le_next;
                    if (sw->rate) {
                        st_rate_stop (sw->rate);
                        sw->rate = NULL;
                    }
                    LIST_REMOVE (sw, entries);
B
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1934 1935
                    LIST_REMOVE (sc, entries);
                    qemu_free (sc);
B
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1936 1937 1938 1939 1940 1941 1942
                    sw = sw1;
                }
                LIST_REMOVE (cap, entries);
                qemu_free (cap);
            }
            return;
        }
1943 1944
    }
}
1945 1946 1947 1948 1949 1950 1951 1952 1953 1954 1955 1956 1957 1958 1959 1960 1961 1962

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;
    }
}