399 lines
8.5 KiB
C
399 lines
8.5 KiB
C
#include "sound.h"
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#include "resources.h"
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#include <stdlib.h>
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#include "log.h"
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#include "string.h"
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#include "math.h"
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#include "limits.h"
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#include "time.h"
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#include "music.h"
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#include "stb_vorbis.h"
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#include "samplerate.h"
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#include "stb_ds.h"
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#include "mix.h"
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#include "dsp.h"
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#define DR_WAV_IMPLEMENTATION
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#include "dr_wav.h"
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#define DR_MP3_IMPLEMENTATION
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#include "dr_mp3.h"
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#include "portaudio.h"
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#include "circbuf.h"
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#define TSF_IMPLEMENTATION
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#include "tsf.h"
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#define TML_IMPLEMENTATION
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#include "tml.h"
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static struct {
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char *key;
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struct wav *value;
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} *wavhash = NULL;
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const char *audioDriver;
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void new_samplerate(short *in, short *out, int n, int ch, int sr_in, int sr_out)
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{
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/*
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SDL_AudioStream *stream = SDL_NewAudioStream(AUDIO_S16, ch, sr_in, AUDIO_S16, ch, sr_out);
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SDL_AudioStreamPut(stream, in, n * ch * sizeof(short));
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SDL_AudioStreamGet(stream, out, n * ch * sizeof(short));
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SDL_FreeAudioStream(stream);
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*/
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}
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static struct wav change_channels(struct wav w, int ch)
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{
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short *data = w.data;
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int samples = ch * w.frames;
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short *new = malloc(sizeof(short)*samples);
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if (ch > w.ch) {
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/* Sets all new channels equal to the first one */
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for (int i = 0; i < w.frames; i++) {
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for (int j = 0; j < ch; j++)
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new[i*ch+j] = data[i];
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}
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} else {
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/* Simple method; just use first N channels present in wav */
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for (int i = 0; i < w.frames; i++)
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for (int j = 0; j < ch; j++)
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new[i*ch+j] = data[i*ch+j];
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}
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free (w.data);
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w.data = new;
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return w;
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}
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static struct wav change_samplerate(struct wav w, int rate)
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{
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float ratio = (float)rate/w.samplerate;
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int outframes = w.frames * ratio;
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SRC_DATA ssrc;
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float floatdata[w.frames*w.ch];
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src_short_to_float_array(w.data, floatdata, w.frames*w.ch);
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float resampled[w.ch*outframes];
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ssrc.data_in = floatdata;
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ssrc.data_out = resampled;
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ssrc.input_frames = w.frames;
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ssrc.output_frames = outframes;
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ssrc.src_ratio = ratio;
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src_simple(&ssrc, SRC_SINC_BEST_QUALITY, w.ch);
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short *newdata = malloc(sizeof(short)*outframes*w.ch);
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src_float_to_short_array(resampled, newdata, outframes*w.ch);
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free(w.data);
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w.data = newdata;
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w.samplerate = rate;
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return w;
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}
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static int patestCallback(const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo, PaStreamCallbackFlags statusFlags, void *userData)
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{
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short *out = (short*)outputBuffer;
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bus_fill_buffers(outputBuffer, framesPerBuffer);
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return 0;
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}
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void check_pa_err(PaError e)
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{
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if (e != paNoError) {
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YughError("PA Error: %s", Pa_GetErrorText(e));
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exit(1);
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}
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}
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static PaStream *stream_def;
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void wav_norm_gain(struct wav *w, double lv)
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{
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short tarmax = db2short(lv);
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short max = 0;
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short *s = w->data;
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for (int i = 0; i < w->frames; i++) {
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for (int j = 0; j < w->ch; j++) {
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max = (abs(s[i*w->ch + j]) > max) ? abs(s[i*w->ch + j]) : max;
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}
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}
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float mult = (float)max / tarmax;
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for (int i = 0; i < w->frames; i++) {
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for (int j = 0; j < w->ch; j++) {
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s[i*w->ch + j] *= mult;
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}
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}
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}
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void print_devices()
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{
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int numDevices = Pa_GetDeviceCount();
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const PaDeviceInfo *deviceInfo;
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for (int i = 0; i < numDevices; i++) {
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deviceInfo = Pa_GetDeviceInfo(i);
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YughInfo("Device %i: channels %i, sample rate %f, name %s\n", i, deviceInfo->maxOutputChannels, deviceInfo->defaultSampleRate, deviceInfo->name);
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}
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}
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void sound_init()
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{
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PaError err = Pa_Initialize();
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check_pa_err(err);
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/*
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PaStreamParameters outparams;
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outparams.channelCount = 2;
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outparams.device = 19;
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outparams.sampleFormat = paInt16;
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outparams.suggestedLatency = Pa_GetDeviceInfo(outparams.device)->defaultLowOutputLatency;
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outparams.hostApiSpecificStreamInfo = NULL;
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err = Pa_OpenStream(&stream_def, NULL, &outparams, 48000, 4096, paNoFlag, patestCallback, &data);
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*/
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err = Pa_OpenDefaultStream(&stream_def, 0, 2, paInt16, SAMPLERATE, BUF_FRAMES, patestCallback, NULL);
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check_pa_err(err);
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err = Pa_StartStream(stream_def);
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check_pa_err(err);
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}
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void audio_open(const char *device)
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{
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//Mix_OpenAudioDevice(44100, MIX_DEFAULT_FORMAT, 2, 2048, device, 0);
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}
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void audio_close()
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{
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//Mix_CloseAudio();
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}
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struct wav *make_sound(const char *wav)
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{
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int index = shgeti(wavhash, wav);
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if (index != -1) return wavhash[index].value;
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struct wav mwav;
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mwav.data = drwav_open_file_and_read_pcm_frames_s16(wav, &mwav.ch, &mwav.samplerate, &mwav.frames, NULL);
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if (mwav.samplerate != SAMPLERATE) {
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YughInfo("Changing samplerate of %s from %d to %d.", wav, mwav.samplerate, 48000);
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mwav = change_samplerate(mwav, 48000);
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}
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if (mwav.ch != CHANNELS) {
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YughInfo("Changing channels of %s from %d to %d.", wav, mwav.ch, CHANNELS);
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//mwav = change_channels(mwav, CHANNELS);
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}
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mwav.gain = 1.f;
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struct wav *newwav = malloc(sizeof(*newwav));
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*newwav = mwav;
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if (shlen(wavhash) == 0) sh_new_arena(wavhash);
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shput(wavhash, wav, newwav);
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return newwav;
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}
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void free_sound(const char *wav)
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{
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struct wav *w = shget(wavhash, wav);
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if (w == NULL) return;
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free(w->data);
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free(w);
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shdel(wavhash, wav);
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}
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struct soundstream *soundstream_make()
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{
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struct soundstream *new = malloc(sizeof(*new));
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new->buf = circbuf_make(sizeof(short), BUF_FRAMES*CHANNELS*2);
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return new;
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}
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void play_oneshot(struct wav *wav) {
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struct sound *new = calloc(1, sizeof(*new));
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new->data = wav;
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new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
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new->playing=1;
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new->loop=0;
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new->frame = 0;
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}
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struct sound *play_sound(struct wav *wav)
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{
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struct sound *new = calloc(1, sizeof(*new));
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new->data = wav;
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new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
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new->playing = 1;
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return new;
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}
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int sound_playing(const struct sound *s)
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{
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return s->playing;
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}
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int sound_paused(const struct sound *s)
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{
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return (!s->playing && s->frame < s->data->frames);
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}
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void sound_pause(struct sound *s)
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{
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s->playing = 0;
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bus_free(s->bus);
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}
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void sound_resume(struct sound *s)
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{
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s->playing = 1;
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s->bus = first_free_bus(dsp_filter(s, sound_fillbuf));
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}
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void sound_stop(struct sound *s)
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{
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s->playing = 0;
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s->frame = 0;
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bus_free(s->bus);
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}
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int sound_finished(const struct sound *s)
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{
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return !s->playing && s->frame == s->data->frames;
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}
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int sound_stopped(const struct sound *s)
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{
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return !s->playing && s->frame == 0;
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}
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struct music make_music(const char *mp3)
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{
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drmp3 new;
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if (!drmp3_init_file(&new, mp3, NULL)) {
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YughError("Could not open mp3 file %s.", mp3);
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}
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}
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void audio_init()
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{
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//audioDriver = SDL_GetAudioDeviceName(0,0);
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}
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void close_audio_device(int device)
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{
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//SDL_CloseAudioDevice(device);
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}
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int open_device(const char *adriver)
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{
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/*
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SDL_AudioSpec audio_spec;
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SDL_memset(&audio_spec, 0, sizeof(audio_spec));
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audio_spec.freq = SAMPLERATE;
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audio_spec.format = AUDIO_F32;
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audio_spec.channels = 2;
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audio_spec.samples = BUF_FRAMES;
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int dev = (int) SDL_OpenAudioDevice(adriver, 0, &audio_spec, NULL, 0);
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SDL_PauseAudioDevice(dev, 0);
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return dev;
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*/
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return 0;
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}
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void sound_fillbuf(struct sound *s, short *buf, int n)
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{
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float gainmult = pct2mult(s->data->gain);
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short *in = s->data->data;
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for (int i = 0; i < n; i++) {
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for (int j = 0; j < CHANNELS; j++) buf[i*CHANNELS+j] = in[s->frame+j] * gainmult;
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s->frame++;
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if (s->frame == s->data->frames) {
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if (s->loop > 0) {
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s->loop--;
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s->frame = 0;
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} else {
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bus_free(s->bus);
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return;
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}
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}
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}
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}
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void mp3_fillbuf(struct sound *s, short *buf, int n)
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{
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}
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void soundstream_fillbuf(struct soundstream *s, short *buf, int n)
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{
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int max = s->buf->write - s->buf->read;
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int lim = (max < n*CHANNELS) ? max : n*CHANNELS;
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for (int i = 0; i < lim; i++) {
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buf[i] = cbuf_shift(&s->buf);
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}
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}
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float short2db(short val)
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{
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return 20*log10(abs((double)val) / SHRT_MAX);
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}
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short db2short(float db)
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{
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return pow(10, db/20.f) * SHRT_MAX;
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}
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short short_gain(short val, float db)
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{
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return (short)(pow(10, db/20.f) * val);
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}
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float pct2db(float pct)
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{
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if (pct <= 0) return -72.f;
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return 10*log2(pct);
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}
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float pct2mult(float pct)
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{
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if (pct <= 0) return 0.f;
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return pow(10, 0.5*log2(pct));
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}
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