prosperon/source/engine/sound.c

#include "sound.h"
#include "resources.h"
#include <stdlib.h>
#include "log.h"
#include "string.h"
#include "math.h"
#include "limits.h"
#include "time.h"
#include "music.h"
#include "stb_vorbis.h"

#include "samplerate.h"

#include "stb_ds.h"

#include "mix.h"
#include "dsp.h"

#define DR_WAV_IMPLEMENTATION
#include "dr_wav.h"

#define DR_MP3_IMPLEMENTATION
#include "dr_mp3.h"

#include "portaudio.h"

#include "circbuf.h"

#define TSF_IMPLEMENTATION
#include "tsf.h"

#define TML_IMPLEMENTATION
#include "tml.h"

static struct {
    char *key;
    struct wav *value;
} *wavhash = NULL;

static struct wav change_channels(struct wav w, int ch)
{
    short *data = w.data;
    int samples = ch * w.frames;
    short *new = malloc(sizeof(short)*samples);

    if (ch > w.ch) {
        /* Sets all new channels equal to the first one */
        for (int i = 0; i < w.frames; i++) {
            for (int j = 0; j < ch; j++)
                new[i*ch+j] = data[i];
        }
    } else {
        /* Simple method; just use first N channels present in wav */
        for (int i = 0; i < w.frames; i++)
            for (int j = 0; j < ch; j++)
                new[i*ch+j] = data[i*ch+j];
    }

    free (w.data);
    w.data = new;
    return w;
}

static struct wav change_samplerate(struct wav w, int rate)
{
    float ratio = (float)rate/w.samplerate;
    int outframes = w.frames * ratio;
    SRC_DATA ssrc;
    float floatdata[w.frames*w.ch];
    src_short_to_float_array(w.data, floatdata, w.frames*w.ch);
    float resampled[w.ch*outframes];

    ssrc.data_in = floatdata;
    ssrc.data_out = resampled;
    ssrc.input_frames = w.frames;
    ssrc.output_frames = outframes;
    ssrc.src_ratio = ratio;

    src_simple(&ssrc, SRC_SINC_BEST_QUALITY, w.ch);

    short *newdata = malloc(sizeof(short)*outframes*w.ch);
    src_float_to_short_array(resampled, newdata, outframes*w.ch);

    free(w.data);
    w.data = newdata;
    w.samplerate = rate;

    return w;
}

static int patestCallback(const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo, PaStreamCallbackFlags statusFlags, void *userData)
{
    bus_fill_buffers(outputBuffer, framesPerBuffer);

    return 0;
}

void check_pa_err(PaError e)
{
    if (e != paNoError) {
        YughError("PA Error: %s", Pa_GetErrorText(e));
        exit(1);
    }
}

static PaStream *stream_def;

void wav_norm_gain(struct wav *w, double lv)
{
    short tarmax = db2short(lv);
    short max = 0;
    short *s = w->data;
    for (int i = 0; i < w->frames; i++) {
        for (int j = 0; j < w->ch; j++) {
            max = (abs(s[i*w->ch + j]) > max) ? abs(s[i*w->ch + j]) : max;
        }
    }

    float mult = (float)max / tarmax;

    for (int i = 0; i < w->frames; i++) {
        for (int j = 0; j < w->ch; j++) {
            s[i*w->ch + j] *= mult;
        }
    }
}

void print_devices()
{
    int numDevices = Pa_GetDeviceCount();
    const PaDeviceInfo *deviceInfo;

    for (int i = 0; i < numDevices; i++) {
        deviceInfo = Pa_GetDeviceInfo(i);

       YughInfo("Device %i: channels %i, sample rate %f, name %s\n", i, deviceInfo->maxOutputChannels, deviceInfo->defaultSampleRate, deviceInfo->name);
    }
}

void sound_init()
{
    return;
    mixer_init();
     PaError err = Pa_Initialize();
     check_pa_err(err);

    err = Pa_OpenDefaultStream(&stream_def, 0, CHANNELS, paInt16, SAMPLERATE, BUF_FRAMES, patestCallback, NULL);
    check_pa_err(err);

    err = Pa_StartStream(stream_def);
    check_pa_err(err);

}

struct wav *make_sound(const char *wav)
{
    int index = shgeti(wavhash, wav);
    if (index != -1) return wavhash[index].value;

    struct wav mwav;
    mwav.data = drwav_open_file_and_read_pcm_frames_s16(wav, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL);


    if (mwav.samplerate != SAMPLERATE) {
        YughInfo("Changing samplerate of %s from %d to %d.", wav, mwav.samplerate, SAMPLERATE);
        mwav = change_samplerate(mwav, SAMPLERATE);
    }

    if (mwav.ch != CHANNELS) {
        YughInfo("Changing channels of %s from %d to %d.", wav, mwav.ch, CHANNELS);
        mwav = change_channels(mwav, CHANNELS);
    }

    mwav.gain = 1.f;

    struct wav *newwav = malloc(sizeof(*newwav));
    *newwav = mwav;

    if (shlen(wavhash) == 0) sh_new_arena(wavhash);

    shput(wavhash, wav, newwav);

    return newwav;
}

void free_sound(const char *wav)
{
    struct wav *w = shget(wavhash, wav);
    if (w == NULL) return;

    free(w->data);
    free(w);
    shdel(wavhash, wav);
}

struct soundstream *soundstream_make()
{
    struct soundstream *new =  malloc(sizeof(*new));
    new->buf = circbuf_make(sizeof(short), BUF_FRAMES*CHANNELS*2);
    return new;
}

void kill_oneshot(struct sound *s)
{
    free(s);
}

void play_oneshot(struct wav *wav) {
    struct sound *new = malloc(sizeof(*new));
    new->data = wav;
    new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
    new->playing=1;
    new->loop=0;
    new->frame = 0;
    new->endcb = kill_oneshot;
}

struct sound *play_sound(struct wav *wav)
{
    struct sound *new = calloc(1, sizeof(*new));
    new->data = wav;

    new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
    new->playing = 1;

    return new;

}

int sound_playing(const struct sound *s)
{
    return s->playing;
}

int sound_paused(const struct sound *s)
{
    return (!s->playing && s->frame < s->data->frames);
}
void sound_pause(struct sound *s)
{
    s->playing = 0;
    bus_free(s->bus);
}

void sound_resume(struct sound *s)
{
    s->playing = 1;
    s->bus = first_free_bus(dsp_filter(s, sound_fillbuf));
}

void sound_stop(struct sound *s)
{
    s->playing = 0;
    s->frame = 0;
    bus_free(s->bus);
}

int sound_finished(const struct sound *s)
{
    return !s->playing && s->frame == s->data->frames;
}

int sound_stopped(const struct sound *s)
{
    return !s->playing && s->frame == 0;
}

struct mp3 make_music(const char *mp3)
{
    drmp3 new;
    if (!drmp3_init_file(&new, mp3, NULL)) {
        YughError("Could not open mp3 file %s.", mp3);
    }

    struct mp3 newmp3 = {};
    return newmp3;
}

void close_audio_device(int device)
{
}

int open_device(const char *adriver)
{
    return 0;
}

void sound_fillbuf(struct sound *s, short *buf, int n)
{
    float gainmult = pct2mult(s->data->gain);

    short *in = s->data->data;
    for (int i = 0; i < n; i++) {
        for (int j = 0; j < CHANNELS; j++) buf[i*CHANNELS+j] = in[s->frame+j] * gainmult;

        s->frame++;
        if (s->frame == s->data->frames) {

            bus_free(s->bus);
            s->bus = NULL;
            s->endcb(s);
            return;
        }
    }
}

void mp3_fillbuf(struct sound *s, short *buf, int n)
{

}



void soundstream_fillbuf(struct soundstream *s, short *buf, int n)
{
    int max = s->buf->write - s->buf->read;
    int lim = (max < n*CHANNELS) ? max : n*CHANNELS;
    for (int i = 0; i < lim; i++) {
        buf[i] = cbuf_shift(s->buf);
    }
}

float short2db(short val)
{
    return 20*log10(abs(val) / SHRT_MAX);
}

short db2short(float db)
{
    return pow(10, db/20.f) * SHRT_MAX;
}

short short_gain(short val, float db)
{
    return (short)(pow(10, db/20.f) * val);
}

float pct2db(float pct)
{
    if (pct <= 0) return -72.f;

    return 10*log2(pct);
}

float pct2mult(float pct)
{
    if (pct <= 0) return 0.f;

    return pow(10, 0.5*log2(pct));
}