Archipelago » qemu

/* public domain */

#include "qemu-common.h"

#include "audio.h"

#include <pulse/simple.h>

#include <pulse/error.h>

#define AUDIO_CAP "pulseaudio"

#include "audio_int.h"

#include "audio_pt_int.h"

typedef struct {

    HWVoiceOut hw;

    int done;

    int live;

    int decr;

    int rpos;

    pa_simple *s;

    void *pcm_buf;

    struct audio_pt pt;

} PAVoiceOut;

typedef struct {

    HWVoiceIn hw;

    int done;

    int dead;

    int incr;

    int wpos;

    pa_simple *s;

    void *pcm_buf;

    struct audio_pt pt;

} PAVoiceIn;

static struct {

    int samples;

    int divisor;

    char *server;

    char *sink;

    char *source;

} conf = {

    .samples = 1024,

    .divisor = 2,

};

static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)

{

    va_list ap;

    va_start (ap, fmt);

    AUD_vlog (AUDIO_CAP, fmt, ap);

    va_end (ap);

    AUD_log (AUDIO_CAP, "Reason: %s\n", pa_strerror (err));

}

static void *qpa_thread_out (void *arg)

{

    PAVoiceOut *pa = arg;

    HWVoiceOut *hw = &pa->hw;

    int threshold;

    threshold = conf.divisor ? hw->samples / conf.divisor : 0;

    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {

        return NULL;

    }

    for (;;) {

        int decr, to_mix, rpos;

        for (;;) {

            if (pa->done) {

                goto exit;

            }

            if (pa->live > threshold) {

                break;

            }

            if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {

                goto exit;

            }

        }

        decr = to_mix = pa->live;

        rpos = hw->rpos;

        if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {

            return NULL;

        }

        while (to_mix) {

            int error;

            int chunk = audio_MIN (to_mix, hw->samples - rpos);

            struct st_sample *src = hw->mix_buf + rpos;

            hw->clip (pa->pcm_buf, src, chunk);

            if (pa_simple_write (pa->s, pa->pcm_buf,

                                 chunk << hw->info.shift, &error) < 0) {

                qpa_logerr (error, "pa_simple_write failed\n");

                return NULL;

            }

            rpos = (rpos + chunk) % hw->samples;

            to_mix -= chunk;

        }

        if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {

            return NULL;

        }

        pa->live = 0;

        pa->rpos = rpos;

        pa->decr += decr;

    }

 exit:

    audio_pt_unlock (&pa->pt, AUDIO_FUNC);

    return NULL;

}

static int qpa_run_out (HWVoiceOut *hw, int live)

{

    int decr;

    PAVoiceOut *pa = (PAVoiceOut *) hw;

    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {

        return 0;

    }

    decr = audio_MIN (live, pa->decr);

    pa->decr -= decr;

    pa->live = live - decr;

    hw->rpos = pa->rpos;

    if (pa->live > 0) {

        audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);

    }

    else {

        audio_pt_unlock (&pa->pt, AUDIO_FUNC);

    }

    return decr;

}

static int qpa_write (SWVoiceOut *sw, void *buf, int len)

{

    return audio_pcm_sw_write (sw, buf, len);

}

/* capture */

static void *qpa_thread_in (void *arg)

{

    PAVoiceIn *pa = arg;

    HWVoiceIn *hw = &pa->hw;

    int threshold;

    threshold = conf.divisor ? hw->samples / conf.divisor : 0;

    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {

        return NULL;

    }

    for (;;) {

        int incr, to_grab, wpos;

        for (;;) {

            if (pa->done) {

                goto exit;

            }

            if (pa->dead > threshold) {

                break;

            }

            if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {

                goto exit;

            }

        }

        incr = to_grab = pa->dead;

        wpos = hw->wpos;

        if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {

            return NULL;

        }

        while (to_grab) {

            int error;

            int chunk = audio_MIN (to_grab, hw->samples - wpos);

            void *buf = advance (pa->pcm_buf, wpos);

            if (pa_simple_read (pa->s, buf,

                                chunk << hw->info.shift, &error) < 0) {

                qpa_logerr (error, "pa_simple_read failed\n");

                return NULL;

            }

            hw->conv (hw->conv_buf + wpos, buf, chunk, &nominal_volume);

            wpos = (wpos + chunk) % hw->samples;

            to_grab -= chunk;

        }

        if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {

            return NULL;

        }

        pa->wpos = wpos;

        pa->dead -= incr;

        pa->incr += incr;

    }

 exit:

    audio_pt_unlock (&pa->pt, AUDIO_FUNC);

    return NULL;

}

static int qpa_run_in (HWVoiceIn *hw)

{

    int live, incr, dead;

    PAVoiceIn *pa = (PAVoiceIn *) hw;

    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {

        return 0;

    }

    live = audio_pcm_hw_get_live_in (hw);

    dead = hw->samples - live;

    incr = audio_MIN (dead, pa->incr);

    pa->incr -= incr;

    pa->dead = dead - incr;

    hw->wpos = pa->wpos;

    if (pa->dead > 0) {

        audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);

    }

    else {

        audio_pt_unlock (&pa->pt, AUDIO_FUNC);

    }

    return incr;

}

static int qpa_read (SWVoiceIn *sw, void *buf, int len)

{

    return audio_pcm_sw_read (sw, buf, len);

}

static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)

{

    int format;

    switch (afmt) {

    case AUD_FMT_S8:

    case AUD_FMT_U8:

        format = PA_SAMPLE_U8;

        break;

    case AUD_FMT_S16:

    case AUD_FMT_U16:

        format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;

        break;

    case AUD_FMT_S32:

    case AUD_FMT_U32:

        format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;

        break;

    default:

        dolog ("Internal logic error: Bad audio format %d\n", afmt);

        format = PA_SAMPLE_U8;

        break;

    }

    return format;

}

static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)

{

    switch (fmt) {

    case PA_SAMPLE_U8:

        return AUD_FMT_U8;

    case PA_SAMPLE_S16BE:

        *endianness = 1;

        return AUD_FMT_S16;

    case PA_SAMPLE_S16LE:

        *endianness = 0;

        return AUD_FMT_S16;

    case PA_SAMPLE_S32BE:

        *endianness = 1;

        return AUD_FMT_S32;

    case PA_SAMPLE_S32LE:

        *endianness = 0;

        return AUD_FMT_S32;

    default:

        dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);

        return AUD_FMT_U8;

    }

}

static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)

{

    int error;

    static pa_sample_spec ss;

    struct audsettings obt_as = *as;

    PAVoiceOut *pa = (PAVoiceOut *) hw;

    ss.format = audfmt_to_pa (as->fmt, as->endianness);

    ss.channels = as->nchannels;

    ss.rate = as->freq;

    obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);

    pa->s = pa_simple_new (

        conf.server,

        "qemu",

        PA_STREAM_PLAYBACK,

        conf.sink,

        "pcm.playback",

        &ss,

        NULL,                   /* channel map */

        NULL,                   /* buffering attributes */

        &error

        );

    if (!pa->s) {

        qpa_logerr (error, "pa_simple_new for playback failed\n");

        goto fail1;

    }

    audio_pcm_init_info (&hw->info, &obt_as);

    hw->samples = conf.samples;

    pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);

    if (!pa->pcm_buf) {

        dolog ("Could not allocate buffer (%d bytes)\n",

               hw->samples << hw->info.shift);

        goto fail2;

    }

    if (audio_pt_init (&pa->pt, qpa_thread_out, hw, AUDIO_CAP, AUDIO_FUNC)) {

        goto fail3;

    }

    return 0;

 fail3:

    qemu_free (pa->pcm_buf);

    pa->pcm_buf = NULL;

 fail2:

    pa_simple_free (pa->s);

    pa->s = NULL;

 fail1:

    return -1;

}

static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)

{

    int error;

    static pa_sample_spec ss;

    struct audsettings obt_as = *as;

    PAVoiceIn *pa = (PAVoiceIn *) hw;

    ss.format = audfmt_to_pa (as->fmt, as->endianness);

    ss.channels = as->nchannels;

    ss.rate = as->freq;

    obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);

    pa->s = pa_simple_new (

        conf.server,

        "qemu",

        PA_STREAM_RECORD,

        conf.source,

        "pcm.capture",

        &ss,

        NULL,                   /* channel map */

        NULL,                   /* buffering attributes */

        &error

        );

    if (!pa->s) {

        qpa_logerr (error, "pa_simple_new for capture failed\n");

        goto fail1;

    }

    audio_pcm_init_info (&hw->info, &obt_as);

    hw->samples = conf.samples;

    pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);

    if (!pa->pcm_buf) {

        dolog ("Could not allocate buffer (%d bytes)\n",

               hw->samples << hw->info.shift);

        goto fail2;

    }

    if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) {

        goto fail3;

    }

    return 0;

 fail3:

    qemu_free (pa->pcm_buf);

    pa->pcm_buf = NULL;

 fail2:

    pa_simple_free (pa->s);

    pa->s = NULL;

 fail1:

    return -1;

}

static void qpa_fini_out (HWVoiceOut *hw)

{

    void *ret;

    PAVoiceOut *pa = (PAVoiceOut *) hw;

    audio_pt_lock (&pa->pt, AUDIO_FUNC);

    pa->done = 1;

    audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);

    audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);

    if (pa->s) {

        pa_simple_free (pa->s);

        pa->s = NULL;

    }

    audio_pt_fini (&pa->pt, AUDIO_FUNC);

    qemu_free (pa->pcm_buf);

    pa->pcm_buf = NULL;

}

static void qpa_fini_in (HWVoiceIn *hw)

{

    void *ret;

    PAVoiceIn *pa = (PAVoiceIn *) hw;

    audio_pt_lock (&pa->pt, AUDIO_FUNC);

    pa->done = 1;

    audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);

    audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);

    if (pa->s) {

        pa_simple_free (pa->s);

        pa->s = NULL;

    }

    audio_pt_fini (&pa->pt, AUDIO_FUNC);

    qemu_free (pa->pcm_buf);

    pa->pcm_buf = NULL;

}

static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)

{

    (void) hw;

    (void) cmd;

    return 0;

}

static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)

{

    (void) hw;

    (void) cmd;

    return 0;

}

/* common */

static void *qpa_audio_init (void)

{

    return &conf;

}

static void qpa_audio_fini (void *opaque)

{

    (void) opaque;

}

struct audio_option qpa_options[] = {

    {

        .name  = "SAMPLES",

        .tag   = AUD_OPT_INT,

        .valp  = &conf.samples,

        .descr = "buffer size in samples"

    },

    {

        .name  = "DIVISOR",

        .tag   = AUD_OPT_INT,

        .valp  = &conf.divisor,

        .descr = "threshold divisor"

    },

    {

        .name  = "SERVER",

        .tag   = AUD_OPT_STR,

        .valp  = &conf.server,

        .descr = "server address"

    },

    {

        .name  = "SINK",

        .tag   = AUD_OPT_STR,

        .valp  = &conf.sink,

        .descr = "sink device name"

    },

    {

        .name  = "SOURCE",

        .tag   = AUD_OPT_STR,

        .valp  = &conf.source,

        .descr = "source device name"

    },

    { /* End of list */ }

};

static struct audio_pcm_ops qpa_pcm_ops = {

    .init_out = qpa_init_out,

    .fini_out = qpa_fini_out,

    .run_out  = qpa_run_out,

    .write    = qpa_write,

    .ctl_out  = qpa_ctl_out,

    .init_in  = qpa_init_in,

    .fini_in  = qpa_fini_in,

    .run_in   = qpa_run_in,

    .read     = qpa_read,

    .ctl_in   = qpa_ctl_in

};

struct audio_driver pa_audio_driver = {

    .name           = "pa",

    .descr          = "http://www.pulseaudio.org/",

    .options        = qpa_options,

    .init           = qpa_audio_init,

    .fini           = qpa_audio_fini,

    .pcm_ops        = &qpa_pcm_ops,

    .can_be_default = 1,

    .max_voices_out = INT_MAX,

    .max_voices_in  = INT_MAX,

    .voice_size_out = sizeof (PAVoiceOut),

    .voice_size_in  = sizeof (PAVoiceIn)

};