audio.c 48.0 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"
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#include "monitor.h"
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#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 = {
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    .fixed_out = { /* DAC fixed settings */
        .enabled = 1,
        .nb_voices = 1,
        .greedy = 1,
        .settings = {
            .freq = 44100,
            .nchannels = 2,
            .fmt = AUD_FMT_S16,
            .endianness =  AUDIO_HOST_ENDIANNESS,
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        }
    },

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

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    .period = { .hertz = 250 },
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    .plive = 0,
    .log_to_monitor = 0,
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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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    .mute = 0,
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#ifdef FLOAT_MIXENG
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    .r = 1.0,
    .l = 1.0,
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#else
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    .r = 1ULL << 32,
    .l = 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) {
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            monitor_printf(cur_mon, "%s: ", cap);
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        }

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        monitor_vprintf(cur_mon, fmt, ap);
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    }
    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 {
650 651
        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
{
    (void) src;
    (void) dst;
    (void) samples;
    (void) vol;
}

static CaptureVoiceOut *audio_pcm_capture_find_specific (
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    struct audsettings *as
711 712 713
    )
{
    CaptureVoiceOut *cap;
714
    AudioState *s = &glob_audio_state;
715 716

    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
static int audio_attach_capture (HWVoiceOut *hw)
790
{
791
    AudioState *s = &glob_audio_state;
792 793 794 795
    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;
797
        SWVoiceOut *sw;
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        HWVoiceOut *hw_cap = &cap->hw;
799

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        sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
        if (!sc) {
802
            dolog ("Could not allocate soft capture voice (%zu bytes)\n",
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                   sizeof (*sc));
804 805 806
            return -1;
        }

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        sc->cap = cap;
        sw = &sc->sw;
809
        sw->hw = hw_cap;
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        sw->info = hw->info;
811 812 813 814 815 816 817 818 819 820 821
        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
828
        if (sw->active) {
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            audio_capture_maybe_changed (cap, 1);
830 831 832 833 834
        }
    }
    return 0;
}

835 836 837
/*
 * Hard voice (capture)
 */
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static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
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{
840 841 842 843 844 845 846
    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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    }
848
    return m;
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}

851
int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
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{
853 854 855 856
    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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    }
858
    return live;
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}

861 862 863 864 865 866 867 868 869 870 871 872 873 874 875 876 877
/*
 * 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 {
880
        return hw->samples + rpos;
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    }
}

884
int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
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{
886 887
    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;
889 890 891 892 893 894 895 896 897 898 899 900 901

    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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903 904
    swlim = (live * sw->ratio) >> 32;
    swlim = audio_MIN (swlim, samples);
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906 907 908 909 910 911 912
    while (swlim) {
        src = hw->conv_buf + rpos;
        isamp = hw->wpos - rpos;
        /* XXX: <= ? */
        if (isamp <= 0) {
            isamp = hw->samples - rpos;
        }
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914 915 916 917
        if (!isamp) {
            break;
        }
        osamp = swlim;
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919 920
        if (audio_bug (AUDIO_FUNC, osamp < 0)) {
            dolog ("osamp=%d\n", osamp);
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            return 0;
922
        }
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924 925 926 927 928 929 930
        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);
933 934
    sw->total_hw_samples_acquired += total;
    return ret << sw->info.shift;
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}

937 938 939
/*
 * Hard voice (playback)
 */
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static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
941
{
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942 943 944
    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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952 953 954

    *nb_livep = nb_live;
    return m;
955
}
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957 958 959
int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
{
    int smin;
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961 962 963 964
    smin = audio_pcm_hw_find_min_out (hw, nb_live);

    if (!*nb_live) {
        return 0;
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    }
    else {
967 968 969 970 971
        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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        }
973
        return live;
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    }
975 976 977 978 979 980
}

int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
{
    int nb_live;
    int live;
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982 983 984 985
    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;
B
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    }
987
    return live;
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}

990 991 992 993
/*
 * Soft voice (playback)
 */
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
B
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994
{
995 996
    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
    int ret = 0, pos = 0, total = 0;
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998 999 1000
    if (!sw) {
        return size;
    }
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1001

1002
    hwsamples = sw->hw->samples;
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1003

1004 1005 1006 1007 1008
    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;
    }
B
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1009

1010
    if (live == hwsamples) {
B
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1011 1012 1013
#ifdef DEBUG_OUT
        dolog ("%s is full %d\n", sw->name, live);
#endif
1014 1015
        return 0;
    }
B
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1016

1017 1018
    wpos = (sw->hw->rpos + live) % hwsamples;
    samples = size >> sw->info.shift;
B
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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 1055
    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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1056 1057
        "%s: write size %d ret %d total sw %d\n",
        SW_NAME (sw),
1058 1059
        size >> sw->info.shift,
        ret,
B
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1060
        sw->total_hw_samples_mixed
1061 1062 1063 1064
        );
#endif

    return ret << sw->info.shift;
B
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1065 1066
}

1067 1068
#ifdef DEBUG_AUDIO
static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
B
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1069
{
1070 1071
    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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1072
}
1073
#endif
B
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1074

1075 1076 1077 1078 1079 1080
#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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1081
{
1082
    int bytes;
B
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1083

1084 1085 1086 1087
    if (!sw) {
        /* XXX: Consider options */
        return size;
    }
B
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1088

1089
    if (!sw->hw->enabled) {
B
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1090
        dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
B
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1091 1092 1093
        return 0;
    }

1094 1095 1096 1097 1098 1099 1100
    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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1101

1102 1103 1104
    if (!sw) {
        /* XXX: Consider options */
        return size;
B
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1105
    }
1106 1107

    if (!sw->hw->enabled) {
B
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1108
        dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
1109
        return 0;
B
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1110
    }
1111 1112 1113

    bytes = sw->hw->pcm_ops->read (sw, buf, size);
    return bytes;
B
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1114 1115
}

1116
int AUD_get_buffer_size_out (SWVoiceOut *sw)
B
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1117
{
B
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1118
    return sw->hw->samples << sw->hw->info.shift;
1119 1120 1121 1122 1123
}

void AUD_set_active_out (SWVoiceOut *sw, int on)
{
    HWVoiceOut *hw;
B
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1124

1125
    if (!sw) {
B
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1126
        return;
1127
    }
B
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1128 1129

    hw = sw->hw;
1130
    if (sw->active != on) {
1131
        AudioState *s = &glob_audio_state;
1132
        SWVoiceOut *temp_sw;
B
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1133
        SWVoiceCap *sc;
1134 1135 1136 1137 1138

        if (on) {
            hw->pending_disable = 0;
            if (!hw->enabled) {
                hw->enabled = 1;
1139 1140 1141
                if (s->vm_running) {
                    hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
                }
1142 1143 1144 1145 1146 1147 1148 1149 1150 1151 1152 1153 1154
            }
        }
        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;
            }
B
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        }
B
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1156 1157 1158

        for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
            sc->sw.active = hw->enabled;
1159
            if (hw->enabled) {
B
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1160
                audio_capture_maybe_changed (sc->cap, 1);
1161 1162
            }
        }
1163 1164 1165 1166 1167 1168 1169 1170 1171 1172
        sw->active = on;
    }
}

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

    if (!sw) {
        return;
B
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1173 1174
    }

1175
    hw = sw->hw;
B
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1176
    if (sw->active != on) {
1177
        AudioState *s = &glob_audio_state;
1178 1179
        SWVoiceIn *temp_sw;

B
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1180 1181 1182
        if (on) {
            if (!hw->enabled) {
                hw->enabled = 1;
1183 1184 1185
                if (s->vm_running) {
                    hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
                }
B
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1186
            }
1187
            sw->total_hw_samples_acquired = hw->total_samples_captured;
B
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1188 1189
        }
        else {
1190
            if (hw->enabled) {
B
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1191
                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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1196 1197 1198
                }

                if (nb_active == 1) {
1199 1200
                    hw->enabled = 0;
                    hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
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1201 1202 1203 1204 1205 1206 1207
                }
            }
        }
        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 (
B
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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);
B
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        return 0;
1244 1245 1246 1247 1248
    }

    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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1250
           SW_NAME (sw),
1251
           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
B
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1252
#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) {
B
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1262
        SWVoiceCap *sc;
1263

B
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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
bellard 已提交
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;

1299
    while ((hw = audio_pcm_hw_find_any_enabled_out (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
                if (!sw->active && !sw->callback.fn) {
#ifdef DEBUG_PLIVE
                    dolog ("Finishing with old voice\n");
#endif
1394
                    audio_close_out (sw);
B
bellard 已提交
1395
                }
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;

1406
    while ((hw = audio_pcm_hw_find_any_enabled_in (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
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
1445
            struct st_sample *src;
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
            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
bellard 已提交
1494
    {"DAC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_out.settings.freq,
1495 1496
     "Frequency for fixed host DAC", NULL, 0},

B
bellard 已提交
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
bellard 已提交
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
bellard 已提交
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
void AUD_help (void)
{
    size_t i;

    audio_process_options ("AUDIO", audio_options);
1559
    for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1560 1561 1562 1563 1564 1565 1566 1567 1568 1569 1570 1571
        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");

1572
    for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
1573 1574 1575 1576 1577
        struct audio_driver *d = drvtab[i];

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

B
bellard 已提交
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) {
1614 1615
        audio_init_nb_voices_out (drv);
        audio_init_nb_voices_in (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
    }
}

1625 1626
static void audio_vm_change_state_handler (void *opaque, int running,
                                           int reason)
B
bellard 已提交
1627
{
B
bellard 已提交
1628
    AudioState *s = opaque;
1629 1630
    HWVoiceOut *hwo = NULL;
    HWVoiceIn *hwi = NULL;
B
bellard 已提交
1631
    int op = running ? VOICE_ENABLE : VOICE_DISABLE;
1632

1633
    s->vm_running = running;
1634
    while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
B
bellard 已提交
1635
        hwo->pcm_ops->ctl_out (hwo, op);
1636
    }
B
bellard 已提交
1637

1638
    while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
B
bellard 已提交
1639
        hwi->pcm_ops->ctl_in (hwi, op);
B
bellard 已提交
1640 1641 1642 1643 1644
    }
}

static void audio_atexit (void)
{
B
bellard 已提交
1645
    AudioState *s = &glob_audio_state;
1646 1647 1648
    HWVoiceOut *hwo = NULL;
    HWVoiceIn *hwi = NULL;

1649
    while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
B
bellard 已提交
1650
        SWVoiceCap *sc;
1651

B
bellard 已提交
1652
        hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
1653
        hwo->pcm_ops->fini_out (hwo);
1654

B
bellard 已提交
1655 1656 1657 1658 1659 1660 1661
        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);
            }
1662
        }
1663
    }
B
bellard 已提交
1664

1665
    while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
B
bellard 已提交
1666
        hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
1667
        hwi->pcm_ops->fini_in (hwi);
B
bellard 已提交
1668
    }
B
bellard 已提交
1669 1670 1671 1672

    if (s->drv) {
        s->drv->fini (s->drv_opaque);
    }
B
bellard 已提交
1673 1674 1675 1676
}

static void audio_save (QEMUFile *f, void *opaque)
{
1677 1678
    (void) f;
    (void) opaque;
B
bellard 已提交
1679 1680 1681 1682
}

static int audio_load (QEMUFile *f, void *opaque, int version_id)
{
1683 1684 1685 1686
    (void) f;
    (void) opaque;

    if (version_id != 1) {
B
bellard 已提交
1687
        return -EINVAL;
1688
    }
B
bellard 已提交
1689 1690 1691 1692

    return 0;
}

1693
static void audio_init (void)
B
bellard 已提交
1694
{
1695
    size_t i;
B
bellard 已提交
1696 1697
    int done = 0;
    const char *drvname;
B
bellard 已提交
1698
    AudioState *s = &glob_audio_state;
1699

P
Paul Brook 已提交
1700
    if (s->drv) {
1701
        return;
P
Paul Brook 已提交
1702 1703
    }

B
bellard 已提交
1704 1705
    LIST_INIT (&s->hw_head_out);
    LIST_INIT (&s->hw_head_in);
1706
    LIST_INIT (&s->cap_head);
B
bellard 已提交
1707 1708 1709 1710
    atexit (audio_atexit);

    s->ts = qemu_new_timer (vm_clock, audio_timer, s);
    if (!s->ts) {
P
Paul Brook 已提交
1711
        hw_error("Could not create audio timer\n");
B
bellard 已提交
1712 1713
    }

1714 1715
    audio_process_options ("AUDIO", audio_options);

B
bellard 已提交
1716 1717 1718
    s->nb_hw_voices_out = conf.fixed_out.nb_voices;
    s->nb_hw_voices_in = conf.fixed_in.nb_voices;

1719
    if (s->nb_hw_voices_out <= 0) {
B
bellard 已提交
1720
        dolog ("Bogus number of playback voices %d, setting to 1\n",
1721 1722 1723 1724 1725
               s->nb_hw_voices_out);
        s->nb_hw_voices_out = 1;
    }

    if (s->nb_hw_voices_in <= 0) {
B
bellard 已提交
1726
        dolog ("Bogus number of capture voices %d, setting to 0\n",
1727
               s->nb_hw_voices_in);
B
bellard 已提交
1728
        s->nb_hw_voices_in = 0;
1729
    }
B
bellard 已提交
1730

1731 1732 1733 1734
    {
        int def;
        drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
    }
B
bellard 已提交
1735 1736 1737

    if (drvname) {
        int found = 0;
1738

1739
        for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
B
bellard 已提交
1740
            if (!strcmp (drvname, drvtab[i]->name)) {
B
bellard 已提交
1741
                done = !audio_driver_init (s, drvtab[i]);
B
bellard 已提交
1742 1743 1744 1745
                found = 1;
                break;
            }
        }
1746

B
bellard 已提交
1747 1748
        if (!found) {
            dolog ("Unknown audio driver `%s'\n", drvname);
1749
            dolog ("Run with -audio-help to list available drivers\n");
B
bellard 已提交
1750 1751 1752 1753
        }
    }

    if (!done) {
1754
        for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
1755
            if (drvtab[i]->can_be_default) {
B
bellard 已提交
1756
                done = !audio_driver_init (s, drvtab[i]);
1757
            }
B
bellard 已提交
1758 1759 1760 1761
        }
    }

    if (!done) {
B
bellard 已提交
1762 1763
        done = !audio_driver_init (s, &no_audio_driver);
        if (!done) {
P
Paul Brook 已提交
1764
            hw_error("Could not initialize audio subsystem\n");
1765 1766
        }
        else {
B
bellard 已提交
1767
            dolog ("warning: Using timer based audio emulation\n");
1768
        }
B
bellard 已提交
1769
    }
1770

P
Paul Brook 已提交
1771
    VMChangeStateEntry *e;
B
bellard 已提交
1772

P
Paul Brook 已提交
1773 1774 1775 1776 1777
    if (conf.period.hertz <= 0) {
        if (conf.period.hertz < 0) {
            dolog ("warning: Timer period is negative - %d "
                   "treating as zero\n",
                   conf.period.hertz);
B
bellard 已提交
1778
        }
P
Paul Brook 已提交
1779 1780 1781
        conf.period.ticks = 1;
    } else {
        conf.period.ticks = ticks_per_sec / conf.period.hertz;
1782
    }
P
Paul Brook 已提交
1783 1784 1785 1786 1787

    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");
1788 1789
    }

B
bellard 已提交
1790 1791 1792
    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);
B
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1793
}
1794

1795 1796 1797 1798 1799 1800 1801 1802 1803 1804 1805 1806 1807 1808 1809
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
    audio_init ();
    card->name = qemu_strdup (name);
    memset (&card->entries, 0, sizeof (card->entries));
    LIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
}

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


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

B
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1820
    if (audio_validate_settings (as)) {
1821 1822
        dolog ("Invalid settings were passed when trying to add capture\n");
        audio_print_settings (as);
B
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1823
        goto err0;
1824 1825 1826 1827 1828 1829 1830 1831 1832 1833 1834
    }

    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;

1835
    cap = audio_pcm_capture_find_specific (as);
1836 1837
    if (cap) {
        LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
B
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1838
        return cap;
1839 1840 1841 1842 1843 1844 1845 1846 1847 1848 1849 1850 1851 1852 1853 1854 1855 1856 1857
    }
    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
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1858
                                    sizeof (struct st_sample));
1859 1860 1861 1862 1863 1864
        if (!hw->mix_buf) {
            dolog ("Could not allocate capture mix buffer (%d samples)\n",
                   hw->samples);
            goto err2;
        }

B
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1865
        audio_pcm_init_info (&hw->info, as);
1866 1867 1868 1869 1870 1871 1872 1873 1874 1875 1876 1877

        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
1878
            [hw->info.swap_endianness]
1879
            [audio_bits_to_index (hw->info.bits)];
1880 1881 1882 1883 1884

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

        hw = NULL;
1885 1886
        while ((hw = audio_pcm_hw_find_any_out (hw))) {
            audio_attach_capture (hw);
1887
        }
B
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1888
        return cap;
1889 1890 1891 1892 1893 1894 1895 1896

    err3:
        qemu_free (cap->hw.mix_buf);
    err2:
        qemu_free (cap);
    err1:
        qemu_free (cb);
    err0:
B
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1897 1898 1899 1900 1901 1902 1903 1904 1905 1906 1907 1908 1909 1910 1911 1912
        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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1913

B
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1914
                while (sw) {
B
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1915
                    SWVoiceCap *sc = (SWVoiceCap *) sw;
B
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1916 1917 1918
#ifdef DEBUG_CAPTURE
                    dolog ("freeing %s\n", sw->name);
#endif
B
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1919

B
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1920 1921 1922 1923 1924 1925
                    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
1926 1927
                    LIST_REMOVE (sc, entries);
                    qemu_free (sc);
B
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1928 1929 1930 1931 1932 1933 1934
                    sw = sw1;
                }
                LIST_REMOVE (cap, entries);
                qemu_free (cap);
            }
            return;
        }
1935 1936
    }
}
1937 1938 1939 1940 1941 1942 1943 1944 1945 1946 1947 1948 1949 1950 1951 1952 1953 1954

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;
    }
}