audio.c

xen虚拟机源代码安装包

        strcat (optname, opt->name);

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

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

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

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

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

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;
    default:
        AUD_log (NULL, "invalid(%d)", as->fmt);
        break;
    }

    AUD_log (NULL, " endianness=");
    switch (as->endianness) {
    case 0:
        AUD_log (NULL, "little");
        break;
    case 1:
        AUD_log (NULL, "big");
        break;
    default:
        AUD_log (NULL, "invalid");
        break;
    }
    AUD_log (NULL, "\n");
}

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

    invalid = as->nchannels != 1 && as->nchannels != 2;
    invalid |= as->endianness != 0 && as->endianness != 1;

    switch (as->fmt) {
    case AUD_FMT_S8:
    case AUD_FMT_U8:
    case AUD_FMT_S16:
    case AUD_FMT_U16:
    case AUD_FMT_S32:
    case AUD_FMT_U32:
        break;
    default:
        invalid = 1;
        break;
    }

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

static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
{
    int bits = 8, sign = 0;

    switch (as->fmt) {
    case AUD_FMT_S8:
        sign = 1;
    case AUD_FMT_U8:
        break;

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

    case AUD_FMT_S32:
        sign = 1;
    case AUD_FMT_U32:
        bits = 32;
        break;
    }
    return info->freq == as->freq
        && info->nchannels == as->nchannels
        && info->sign == sign
        && info->bits == bits
        && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
}

void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
{
    int bits = 8, sign = 0, shift = 0;

    switch (as->fmt) {
    case AUD_FMT_S8:
        sign = 1;
    case AUD_FMT_U8:
        break;

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

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

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

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

    if (info->sign) {
        memset (buf, 0x00, len << info->shift);
    }
    else {
        switch (info->bits) {
        case 8:
            memset (buf, 0x80, len << info->shift);
            break;

        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;
                }
            }
            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);
                }

                for (i = 0; i < len << shift; i++) {
                    p[i] = s;
                }
            }
            break;

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

/*
 * 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,
    audsettings_t *as
    )
{
    CaptureVoiceOut *cap;

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

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

static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
{
    if (cap->hw.enabled != enabled) {
        audcnotification_e cmd;
        cap->hw.enabled = enabled;
        cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
        audio_notify_capture (cap, cmd);
    }
}

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

    for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
        if (sw->active) {
            enabled = 1;
            break;
        }
    }
    audio_capture_maybe_changed (cap, enabled);
}

static void audio_detach_capture (HWVoiceOut *hw)
{
    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;

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

        LIST_REMOVE (sw, entries);
        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;
    }
}

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) {
        SWVoiceCap *sc;
        SWVoiceOut *sw;
        HWVoiceOut *hw_cap = &cap->hw;

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

        sc->cap = cap;
        sw = &sc->sw;
        sw->hw = hw_cap;
        sw->info = hw->info;
        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);
        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
        if (sw->active) {
            audio_capture_maybe_changed (cap, 1);
        }
    }
    return 0;
}

/*
 * Hard voice (capture)
 */
static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
    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);
        }
    }
    return m;
}

int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
{
    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;
    }
    return live;
}

/*
 * 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;
    }
    else {
        return hw->samples + rpos;
    }
}

int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
{
    HWVoiceIn *hw = sw->hw;
    int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
    st_sample_t *src, *dst = sw->buf;

    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;
    }

    swlim = (live * sw->ratio) >> 32;
    swlim = audio_MIN (swlim, samples);

    while (swlim) {
        src = hw->conv_buf + rpos;
        isamp = hw->wpos - rpos;
        /* XXX: <= ? */
        if (isamp <= 0) {
            isamp = hw->samples - rpos;
        }

        if (!isamp) {
            break;
        }
        osamp = swlim;

        if (audio_bug (AUDIO_FUNC, osamp < 0)) {
            dolog ("osamp=%d\n", osamp);
            return 0;
        }

        st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
        swlim -= osamp;
        rpos = (rpos + isamp) % hw->samples;
        dst += osamp;
        ret += osamp;
        total += isamp;
    }

    sw->clip (buf, sw->buf, ret);
    sw->total_hw_samples_acquired += total;
    return ret << sw->info.shift;
}

/*
 * Hard voice (playback)
 */
static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
{
    SWVoiceOut *sw;
    int m = INT_MAX;
    int nb_live = 0;

    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;
        }
    }

    *nb_livep = nb_live;
    return m;
}

int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
{
    int smin;

    smin = audio_pcm_hw_find_min_out (hw, nb_live);

    if (!*nb_live) {
        return 0;
    }
    else {
        int live = smin;

        if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {