renpy-7.4.11-sdk/module/renpysound_core.c

/*
Copyright 2004-2011 Tom Rothamel <pytom@bishoujo.us>

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.
*/

#include "renpysound_core.h"
#include <Python.h>
#include <SDL.h>
#include <SDL_thread.h>
#include <stdio.h>
#include <string.h>
#include <pygame_sdl2/pygame_sdl2.h>

#define MAXVOLUME 16384


SDL_mutex *name_mutex;

#ifdef __EMSCRIPTEN__

#define LOCK_AUDIO() { }
#define UNLOCK_AUDIO() { }

#define LOCK_NAME() { }
#define UNLOCK_NAME() { }

#else

/* These prevent the audio callback from running when held. Use this to
   prevent the audio callback from running while the state of the audio
   system is being changed. */
#define LOCK_AUDIO() { SDL_LockAudio(); }
#define UNLOCK_AUDIO() { SDL_UnlockAudio(); }

/* This is held while the current track is being changed by the audio callback,
   and can also be held to the current track doesn't change while things are
   being processed. */
#define LOCK_NAME() { SDL_LockMutex(name_mutex); }
#define UNLOCK_NAME() { SDL_UnlockMutex(name_mutex); }

#endif

/* Declarations of ffdecode functions. */
struct MediaState;
typedef struct MediaState MediaState;

void media_init(int rate, int status, int equal_mono);

void media_advance_time(void);
void media_sample_surfaces(SDL_Surface *rgb, SDL_Surface *rgba);
MediaState *media_open(SDL_RWops *, const char *);
void media_want_video(MediaState *, int);
void media_start_end(MediaState *, double, double);
void media_start(MediaState *);
void media_pause(MediaState *, int);
void media_close(MediaState *);

int media_read_audio(struct MediaState *is, Uint8 *stream, int len);

int media_video_ready(struct MediaState *ms);
SDL_Surface *media_read_video(struct MediaState *ms);

double media_duration(struct MediaState *ms);
void media_wait_ready(struct MediaState *ms);

/* Min and Max */
#define min(a, b) (((a) < (b)) ? (a) : (b))
#define max(a, b) (((a) > (b)) ? (a) : (b))

/* Various error codes. */
#define SUCCESS 0
#define SDL_ERROR -1
#define SOUND_ERROR -2
#define RPS_ERROR -3

/* This is called with the appropriate error code at the end of a
 * function. */
#define error(err) RPS_error = err
int RPS_error = SUCCESS;
static const char *error_msg = NULL;

/* Have we been initialized? */
static int initialized = 0;

/*
 * This structure represents a channel the system knows about
 * and can play from.
 */
struct Channel {

    /* The currently playing sample, NULL if this sample isn't playing
       anything. */
    struct MediaState *playing;

    /* The name of the playing music. */
    char *playing_name;

    /* The number of ms to take to fade in the playing sample. */
    int playing_fadein;

    /* Is the playing sample tight? */
    int playing_tight;

    /* The start time of the playing sample, in ms. */
    int playing_start_ms;

    /* The queued up sample. */
    struct MediaState *queued;

    /* The name of the queued up sample. */
    char *queued_name;

    /* The number of ms to take to fade in the queued sample. */
    int queued_fadein;

    /* Is the queued sample tight? */
    int queued_tight;

    /* The start time of the queued sample, in ms. */
    int queued_start_ms;

    /* Is this channel paused? */
    int paused;

    /* The volume of the channel. */
    int volume;

    /* The position (in bytes) that this channel has queued to. */
    int pos;

    /*
     * The number of bytes for each step of fade.
     * 0 when no fade is in progress.
     */
    int fade_step_len;

    /* How many bytes we are into the current fade step. */
    int fade_off;

    /* The current fade volume. */
    int fade_vol;

    /* The change in fade_vol for each step. */
    int fade_delta;

    /* The number of bytes in which we'll stop. */
    int stop_bytes;

    /* The event posted to the queue when we finish a track. */
    int event;

    /* The pan being applied to the current channel. */
    float pan_start;
    float pan_end;

    /* The length of the current pan, in samples. */
    unsigned int pan_length;

    /* The number of samples we've finished in the current pan. */
    unsigned int pan_done;


    /* These are used like in pan, above. Unlike the volume parameter,
       the voulme set here is persisted between sessions. */
    float vol2_start;
    float vol2_end;
    unsigned int vol2_length;
    unsigned int vol2_done;

    /* This is set to 1 if this is a movie channel with dropping, 2 if it's a
     * video channel without dropping. */
    int video;

};

struct Dying {
    struct MediaState *stream;
    struct Dying *next;
};

static struct Dying *dying = NULL;

/*
 * The number of channels the system knows about.
 */
int num_channels = 0;

/*
 * All of the channels that the system knows about.
 */
struct Channel *channels = NULL;

/*
 * The spec of the audio that is playing.
 */
SDL_AudioSpec audio_spec;


static float interpolate_pan(struct Channel *c) {
    float done;

    if (c->pan_done > c->pan_length) {
        c->pan_length = 0;
    }

    if (c->pan_length == 0) {
        return c->pan_end;
    }

    done = 1.0 * c->pan_done / c->pan_length;

    return c->pan_start + done * (c->pan_end - c->pan_start);

}

static float interpolate_vol2(struct Channel *c) {
    float done;

    if (c->vol2_done > c->vol2_length) {
        c->vol2_length = 0;
    }

    if (c->vol2_length == 0) {
        return c->vol2_end;
    }

    done = 1.0 * c->vol2_done / c->vol2_length;

    return c->vol2_start + done * (c->vol2_end - c->vol2_start);
}

static int ms_to_bytes(int ms) {
    return ((long long) ms) * audio_spec.freq * audio_spec.channels * 2 / 1000;
}

static int bytes_to_ms(int bytes) {
    return ((long long) bytes) * 1000 / (audio_spec.freq * audio_spec.channels * 2);
}

static void start_sample(struct Channel* c, int reset_fade) {
    int fade_steps;

    if (!c) return;

    c->pos = 0;

    if (reset_fade) {

        if (c->playing_fadein == 0) {
            c->fade_step_len = 0;
        } else {
            fade_steps = c->volume;
            c->fade_delta = 1;
            c->fade_off = 0;
            c->fade_vol = 0;

            if (fade_steps) {
                c->fade_step_len = ms_to_bytes(c->playing_fadein) / fade_steps;
                c->fade_step_len &= ~0x7; // Even sample.
            } else {
                c->fade_step_len = 0;
            }
        }

        c->stop_bytes = -1;
    }
}

static void free_sample(struct MediaState *ss) {
    media_close(ss);
}

#define MAX_SHORT (32767)
#define MIN_SHORT (-32768)

// Actually mixes the audio.
static void mixaudio(Uint8 *dst, Uint8 *src, int length, int volume) {
    int i;
    short *sdst = (short *) dst;
    short *ssrc = (short *) src;

    for (i = 0; i < length / 2; i++) {
        int sound = *sdst + (volume * *ssrc) / MAXVOLUME;
        if (sound > MAX_SHORT) {
            sound = MAX_SHORT;
        }
        if (sound < MIN_SHORT) {
            sound = MIN_SHORT;
        }

        *sdst++ = (short) sound;
        ssrc++;
    }
}


// Mixes the audio, while performing fading.
static void fade_mixaudio(struct Channel *c,
                          Uint8 *dst, Uint8 *src, int length) {

    while (length) {

        // No fade case.
        if (c->fade_step_len == 0) {
            mixaudio(dst, src, length, c->volume);
            return;
        }

        // Fading, but we have some space left in the current step.
        if (c->fade_off < c->fade_step_len) {
            int l = min(c->fade_step_len - c->fade_off, length);

            mixaudio(dst, src, l, c->fade_vol);

            length -= l;
            dst += l;
            src += l;
            c->fade_off += l;
            continue;
        }

        // Otherwise, we have no space left in the current fade step.
        // Go to the next step.
        c->fade_off = 0;
        c->fade_vol += c->fade_delta;

        // Don't stop on a fadeout.
        if (c->fade_vol <= 0) {
            c->fade_vol = 0;
        }

        // Stop on a fadein.
        if (c->fade_vol >= c->volume) {
            c->fade_vol = c->volume;
            c->fade_step_len = 0;
        }
    }

    return;
}

static void post_event(struct Channel *c) {
    if (! c->event) {
        return;
    }

    SDL_Event e;
    memset(&e, 0, sizeof(e));
    e.type = c->event;
    SDL_PushEvent(&e);
}

/* This handels panning and vol2 manipulations. */
static void pan_audio(struct Channel *c, Uint8 *stream, int length) {
    int i;
    short *sample = (short *) stream;
    length /= 4;

    float pan;
    float vol2;
    int left = 256;
    int right = 256;


    for (i = 0; i < length; i++) {

        if ((i & 0x1f) == 0) {
            pan = interpolate_pan(c);
            vol2 = interpolate_vol2(c);

            // If nothing to do, skip 32 samples.
            if (pan == 0.0 && vol2 == 1.0) {
                i += 31;
                c->pan_done += 32;
                c->vol2_done += 32;
                sample += 32 * 2;
                continue;
            }

            vol2 *= 256.0;

            if (pan < 0) {
                left = (int) vol2;
                right = (int) (vol2 * (1.0 + pan));
            } else {
                left = (int) (vol2 * (1.0 - pan));
                right = (int) vol2;
            }
        }


        *sample = (short) ((*sample * left) >> 8);
        sample++;
        *sample = (short) ((*sample * right) >> 8);
        sample++;

        c->pan_done += 1;
        c->vol2_done += 1;
    }

}

static void callback(void *userdata, Uint8 *stream, int length) {

    int channel = 0;

    memset(stream, 0, length);

    for (channel = 0; channel < num_channels; channel++) {

        int mixed = 0;
        struct Channel *c = &channels[channel];

        if (! c->playing) {
            continue;
        }

        if (c->paused) {
            continue;
        }

        while (mixed < length && c->playing) {
            int mixleft = length - mixed;
            Uint8 buffer[mixleft];
            int bytes;

            // Decode some amount of data.

            bytes = media_read_audio(c->playing, buffer, mixleft);

            // We have some data in the buffer.
            if (c->stop_bytes && bytes) {

                if (c->stop_bytes != -1)
                    bytes = min(c->stop_bytes, bytes);

                pan_audio(c, buffer, bytes);
                fade_mixaudio(c, &stream[mixed], buffer, bytes);

                mixed += bytes;

                if (c->stop_bytes != -1)
                    c->stop_bytes -= bytes;

                c->pos += bytes;

                continue;
            }

            // Otherwise, no data is left in the buffer. Check why,
            // and act accordingly.

            // Skip to the next sample.
            if (c->stop_bytes == 0 || bytes == 0) {

                int old_tight = c->playing_tight;
                struct Dying *d;

                post_event(c);

                LOCK_NAME()

                d = malloc(sizeof(struct Dying));
                d->next = dying;
                d->stream = c->playing;
                dying = d;

                free(c->playing_name);

                c->playing = c->queued;
                c->playing_name = c->queued_name;
                c->playing_fadein = c->queued_fadein;
                c->playing_tight = c->queued_tight;
                c->playing_start_ms = c->queued_start_ms;

                c->queued = NULL;
                c->queued_name = NULL;
                c->queued_fadein = 0;
                c->queued_tight = 0;
                c->queued_start_ms = 0;

                if (c->playing_fadein) {
                    old_tight = 0;
                }

                UNLOCK_NAME()

                start_sample(c, ! old_tight);

                continue;
            }
        }

    }

}

/*
 * Checks that the given channel is in range. Returns 0 if it is,
 * sets an error and returns -1 if it is not. Allocates channels
 * that don't already exist.
 */
static int check_channel(int c) {
    int i;

    if (c < 0) {
        error(RPS_ERROR);
        error_msg = "Channel number out of range.";
        return -1;
    }

    if (c >= num_channels) {
        struct Channel *extended_channels = realloc(channels, sizeof(struct Channel) * (c + 1));
        if (extended_channels == NULL) {
            error(RPS_ERROR);
            error_msg = "Unable to allocate additional channels.";
            return -1;
        }
        channels = extended_channels;

        for (i = num_channels; i <= c; i++) {

        	memset(&channels[i], 0, sizeof(struct Channel));

            channels[i].volume = MAXVOLUME;
            channels[i].paused = 1;
            channels[i].event = 0;
            channels[i].vol2_start = 1.0;
            channels[i].vol2_end = 1.0;
        }

        num_channels = c + 1;
    }

    return 0;
}


/*
 * Loads the provided sample. Returns the sample on success, NULL on
 * failure.
 */
struct MediaState *load_sample(SDL_RWops *rw, const char *ext, double start, double end, int video) {
    struct MediaState *rv;
    rv = media_open(rw, ext);
    if (rv == NULL)
    {
        return NULL;
    }
    media_start_end(rv, start, end);

    if (video) {
    	media_want_video(rv, video);
    }

    media_start(rv);
    return rv;
}


void RPS_play(int channel, SDL_RWops *rw, const char *ext, const char *name, int fadein, int tight, int paused, double start, double end) {

    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    LOCK_AUDIO();

    /* Free playing and queued samples. */
    if (c->playing) {
        free_sample(c->playing);
        c->playing = NULL;
        free(c->playing_name);
        c->playing_name = NULL;
        c->playing_tight = 0;
        c->playing_start_ms = 0;
    }

    if (c->queued) {
        free_sample(c->queued);
        c->queued = NULL;
        free(c->queued_name);
        c->queued_name = NULL;
        c->queued_tight = 0;
        c->queued_start_ms = 0;
    }

    /* Allocate playing sample. */

    c->playing = load_sample(rw, ext, start, end, c->video);

    if (! c->playing) {
        UNLOCK_AUDIO();
        error(SOUND_ERROR);
        return;
    }

    c->playing_name = strdup(name);
    c->playing_fadein = fadein;
    c->playing_tight = tight;
    c->playing_start_ms = (int) (start * 1000);

    c->paused = paused;

    start_sample(c, 1);

    UNLOCK_AUDIO();

    error(SUCCESS);
}

void RPS_queue(int channel, SDL_RWops *rw, const char *ext, const char *name, int fadein, int tight, double start, double end) {

    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    /* If we're not playing, then we should play instead of queue. */
    if (!c->playing) {
        RPS_play(channel, rw, ext, name, fadein, tight, 0, start, end);
        return;
    }

    LOCK_AUDIO();

    /* Free queued sample. */

    if (c->queued) {
        free_sample(c->queued);
        c->queued = NULL;
        free(c->queued_name);
        c->queued_name = NULL;
        c->queued_tight = 0;
    }

    /* Allocate queued sample. */
    c->queued = load_sample(rw, ext, start, end, c->video);

    if (! c->queued) {
        UNLOCK_AUDIO();

        error(SOUND_ERROR);
        return;
    }

    c->queued_name = strdup(name);
    c->queued_fadein = fadein;
    c->queued_tight = tight;

    c->queued_start_ms = (int) (start * 1000);

    UNLOCK_AUDIO();

    error(SUCCESS);
}


/*
 * Stops all music from playing, freeing the data used by the
 * music.
 */
void RPS_stop(int channel) {

    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    LOCK_AUDIO();

    if (c->playing) {
        post_event(c);
    }

    /* Free playing and queued samples. */
    if (c->playing) {
        free_sample(c->playing);
        c->playing = NULL;
        free(c->playing_name);
        c->playing_name = NULL;
        c->playing_start_ms = 0;
    }

    if (c->queued) {
        free_sample(c->queued);
        c->queued = NULL;
        free(c->queued_name);
        c->queued_name = NULL;
        c->queued_start_ms = 0;
    }

    UNLOCK_AUDIO();

    error(SUCCESS);
}

/*
 * This dequeues the queued sound from the supplied channel, if
 * such a sound is queued. This does nothing to the playing
 * sound.
 *
 * This does nothing if the playing sound is tight, ever_tight is
 * false.
 */
void RPS_dequeue(int channel, int even_tight) {

    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    LOCK_AUDIO();

    if (c->queued && (! c->playing_tight || even_tight)) {
        free_sample(c->queued);
        c->queued = NULL;
        free(c->queued_name);
        c->queued_name = NULL;
    } else {
        c->queued_tight = 0;
    }

    c->queued_start_ms = 0;

    UNLOCK_AUDIO();

    error(SUCCESS);
}

/*
 * Returns the queue depth of the current channel. This is 0 if we're
 * stopped, 1 if there's something playing but nothing queued, and 2
 * if there's both something playing and something queued.
 */
int RPS_queue_depth(int channel) {
    int rv = 0;

    struct Channel *c;

    if (check_channel(channel)) {
        return 0;
    }

    c = &channels[channel];

    LOCK_NAME();

    if (c->playing) rv++;
    if (c->queued) rv++;

    UNLOCK_NAME();

    error(SUCCESS);

    return rv;
}

PyObject *RPS_playing_name(int channel) {
    PyObject *rv;
    struct Channel *c;

    if (check_channel(channel)) {
        Py_INCREF(Py_None);
        return Py_None;
    }

    c = &channels[channel];

    LOCK_NAME();

    if (c->playing_name) {
        rv = PyBytes_FromString(c->playing_name);
    } else {
        Py_INCREF(Py_None);
        rv = Py_None;
    }

    UNLOCK_NAME();

    error(SUCCESS);

    return rv;
}

/*
 * Causes the given channel to fadeout playing after a specified
 * number of milliseconds. The playing sound stops once the
 * fadeout finishes (a queued sound may then start at full volume).
 */
void RPS_fadeout(int channel, int ms) {

    int fade_steps;
    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    LOCK_AUDIO();

    if (ms == 0) {
        c->stop_bytes = 0;
        UNLOCK_AUDIO();

        error(SUCCESS);
        return;
    }

    fade_steps = c->volume;
    c->fade_delta = -1;
    c->fade_off = 0;
    c->fade_vol = c->volume;

    if (fade_steps) {
        c->fade_step_len = ms_to_bytes(ms) / fade_steps;
        c->fade_step_len &= ~0x7; // Even sample.
    } else {
        c->fade_step_len = 0;
    }

    c->stop_bytes = ms_to_bytes(ms);
    c->queued_tight = 0;

    if (!c->queued) {
        c->playing_tight = 0;
    }

    UNLOCK_AUDIO();

    error(SUCCESS);
}

/*
 * Sets the pause flag on the given channel 0 = unpaused, 1 = paused.
 */
void RPS_pause(int channel, int pause) {

    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    c->paused = pause;

    if (c->playing) {
        media_pause(c->playing, pause);
    }

    error(SUCCESS);

}

void RPS_unpause_all_at_start(void) {

    int i;

    /* Since media_wait_ready can block, we need to release the GIL. */
    Py_BEGIN_ALLOW_THREADS

    for (i = 0; i < num_channels; i++) {
        if (channels[i].playing && channels[i].paused && channels[i].pos == 0) {
            media_wait_ready(channels[i].playing);
        }
    }

    Py_END_ALLOW_THREADS

    for (i = 0; i < num_channels; i++) {
        if (channels[i].playing && channels[i].pos == 0) {
            channels[i].paused = 0;
            media_pause(channels[i].playing, 0);
        }
    }

    error(SUCCESS);

}

/*
 * Returns the position of the given channel, in ms.
 */
int RPS_get_pos(int channel) {
    int rv;
    struct Channel *c;

    if (check_channel(channel)) {
        return -1;
    }

    c = &channels[channel];

    LOCK_NAME();

    if (c->playing) {
        rv = bytes_to_ms(c->pos) + c->playing_start_ms;
    } else {
        rv = -1;
    }

    UNLOCK_NAME();

    error(SUCCESS);
    return rv;
}

/*
 * Returns the duration of the file playing on the given channel, in
 * seconds.
 */
double RPS_get_duration(int channel) {
    double rv;
    struct Channel *c;

    if (check_channel(channel)) {
        return 0.0;
    }

    c = &channels[channel];

    LOCK_NAME();

    if (c->playing) {
        rv = media_duration(c->playing);
    } else {
        rv = 0.0;
    }

    UNLOCK_NAME();

    error(SUCCESS);
    return rv;
}

/*
 * Sets an event that is queued up when the track on the given channel
 * ends due to natural termination or a forced stop.
 */
void RPS_set_endevent(int channel, int event) {
    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    c->event = event;

    error(SUCCESS);
}

/*
 * This sets the natural volume of the channel. (This may not take
 * effect immediately if a fade is going on.)
 */
void RPS_set_volume(int channel, float volume) {
    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];
    c->volume = (int) (volume * MAXVOLUME);

    error(SUCCESS);
}


float RPS_get_volume(int channel) {

    float rv;

    struct Channel *c;

    if (check_channel(channel)) {
        return 0.0;
    }

    c = &channels[channel];

    rv = 1.0 * c->volume / MAXVOLUME;

    error(SUCCESS);
    return rv;
}

/*
 * This sets the pan of the channel... independent volumes for the
 * left and right channels.
 */
void RPS_set_pan(int channel, float pan, float delay) {
    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    LOCK_AUDIO();

    c->pan_start = interpolate_pan(c);
    c->pan_end = pan;
    c->pan_length = (int) (audio_spec.freq * delay);
    c->pan_done = 0;

    UNLOCK_AUDIO();

    error(SUCCESS);
}

/*
 * This sets the secondary volume of the channel.
 */
void RPS_set_secondary_volume(int channel, float vol2, float delay) {
    struct Channel *c;

    if (check_channel(channel)) {
        return;
    }

    c = &channels[channel];

    LOCK_AUDIO();

    c->vol2_start = interpolate_vol2(c);
    c->vol2_end = vol2;
    c->vol2_length = (int) (audio_spec.freq * delay);
    c->vol2_done = 0;

    UNLOCK_AUDIO();

    error(SUCCESS);
}

PyObject *RPS_read_video(int channel) {
    struct Channel *c;
    SDL_Surface *surf = NULL;

    if (check_channel(channel)) {
        Py_INCREF(Py_None);
    	return Py_None;
    }

    c = &channels[channel];

    if (c->playing) {
        surf = media_read_video(c->playing);
    }

    error(SUCCESS);

    if (surf) {
    	return PySurface_New(surf);
    } else {
        Py_INCREF(Py_None);
    	return Py_None;
    }

}

int RPS_video_ready(int channel) {
    struct Channel *c;
    int rv;

    if (check_channel(channel)) {
    	return 1;
    }

    c = &channels[channel];

    if (c->playing) {
    	rv = media_video_ready(c->playing);
    } else {
    	rv = 1;
    }

    error(SUCCESS);

    return rv;
}

/**
 * Marks channel as a video channel.
 */
void RPS_set_video(int channel, int video) {
	struct Channel *c;

	if (check_channel(channel)) {
    	return;
    }

    c = &channels[channel];

    c->video = video;
}


/*
 * Initializes the sound to the given frequencies, channels, and
 * sample buffer size.
 */
void RPS_init(int freq, int stereo, int samples, int status, int equal_mono) {

    if (initialized) {
        return;
    }

    name_mutex = SDL_CreateMutex();

#ifndef __EMSCRIPTEN__
    PyEval_InitThreads();
#endif

    import_pygame_sdl2();

    if (SDL_Init(SDL_INIT_AUDIO)) {
        error(SDL_ERROR);
        return;
    }

    audio_spec.freq = freq;
    audio_spec.format = AUDIO_S16SYS;
    audio_spec.channels = stereo;
    audio_spec.samples = samples;
    audio_spec.callback = callback;
    audio_spec.userdata = NULL;

    if (SDL_OpenAudio(&audio_spec, NULL)) {
        error(SDL_ERROR);
        return;
    }

    media_init(audio_spec.freq, status, equal_mono);

    SDL_PauseAudio(0);

    initialized = 1;

    error(SUCCESS);
}

void RPS_quit() {

    if (! initialized) {
        return;
    }

    int i;

    LOCK_AUDIO();
    SDL_PauseAudio(1);
    UNLOCK_AUDIO();

    for (i = 0; i < num_channels; i++) {
        RPS_stop(i);
    }

    SDL_CloseAudio();

    num_channels = 0;
    initialized = 0;
    error(SUCCESS);
}

/* This must be called frequently, to take care of deallocating dead
 * streams. */
void RPS_periodic() {

    LOCK_NAME();
    struct Dying *d = dying;
    dying = NULL;
    UNLOCK_NAME();

    while (d) {
        media_close(d->stream);
        struct Dying *next_d = d->next;
        free(d);
        d = next_d;
    }

}

void RPS_advance_time(void) {
	media_advance_time();
}

void RPS_sample_surfaces(PyObject *rgb, PyObject *rgba) {
    import_pygame_sdl2();

    media_sample_surfaces(
			PySurface_AsSurface(rgb),
			PySurface_AsSurface(rgba)
		);

}

/*
 * Returns the error message string if an error has occured, or
 * NULL if no error has happened.
 */
char *RPS_get_error() {
    switch(RPS_error) {
    case 0:
        return (char *) "";
    case SDL_ERROR:
        return (char *) SDL_GetError();
    case SOUND_ERROR:
        return (char *) "Some sort of codec error.";
    case RPS_ERROR:
        return (char *) error_msg;
    default:
        return (char *) "Error getting error.";
    }
}