Optimize busses; add samplerate changing library

Makefile

@@ -84,7 +84,7 @@ ifeq ($(OS), WIN32)
 	EXT = .exe
 else
 	LINKER_FLAGS = $(QFLAGS) -L/usr/local/lib
-	ELIBS =  engine pthread yughc portaudio asound glfw3 c m dl
+	ELIBS =  engine pthread yughc portaudio asound glfw3 c m dl samplerate
 	CLIBS =
 endif
 

source/engine/ffi.c

@@ -120,11 +120,11 @@ duk_ret_t duk_cmd(duk_context *duk) {
          break;
 
        case 14:
-         play_sound(make_sound(duk_to_string(duk, 1)));
+         play_oneshot(make_sound(duk_to_string(duk, 1)));
          break;
 
        case 15:
-
+         music_stop();
          break;
 
        case 16:

source/engine/sound.c

@@ -9,6 +9,8 @@
 #include "music.h"
 #include "stb_vorbis.h"
 
+#include "samplerate.h"
+
 #include "stb_ds.h"
 
 #include "mix.h"
@@ -47,27 +49,53 @@ void new_samplerate(short *in, short *out, int n, int ch, int sr_in, int sr_out)
     */
 }
 
-struct wav change_samplerate(struct wav w, int rate)
+static struct wav change_channels(struct wav w, int ch)
 {
-    //int samples = sizeof(short) * w.ch * w.frames;
+    short *data = w.data;
-    //short *new = malloc(samples);
+    int samples = ch * w.frames;
-    //new_samplerate(w.data, new,
+    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;
+}
 
-    //SDL_AudioStream *stream = SDL_NewAudioStream(AUDIO_S16, w.ch, w.samplerate, AUDIO_S16, w.ch, rate);
+static struct wav change_samplerate(struct wav w, int rate)
-    //SDL_AudioStreamPut(stream, w.data, w.frames*w.ch*sizeof(short));
+{
+    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];
 
-    int oldframes = w.frames;
+    ssrc.data_in = floatdata;
-    w.frames *= (float)rate/w.samplerate;
+    ssrc.data_out = resampled;
-    int samples = sizeof(short) * w.ch * w.frames;
+    ssrc.input_frames = w.frames;
-    w.samplerate = rate;
+    ssrc.output_frames = outframes;
-    short *new = malloc(samples);
+    ssrc.src_ratio = ratio;
-    //SDL_AudioStreamGet(stream, new, samples);
+
+    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 = new;
+    w.data = newdata;
-    //SDL_FreeAudioStream(stream);
+    w.samplerate = rate;
 
     return w;
 }
@@ -165,11 +193,17 @@ struct wav *make_sound(const char *wav)
     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.", wav);
+    if (mwav.samplerate != SAMPLERATE) {
+        YughInfo("Changing samplerate of %s from %d to %d.", wav, mwav.samplerate, 48000);
         mwav = change_samplerate(mwav, 48000);
     }
 
+    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));
@@ -205,6 +239,7 @@ void play_oneshot(struct wav *wav) {
     new->bus = first_free_bus(dsp_filter(new, sound_fillbuf));
     new->playing=1;
     new->loop=0;
+    new->frame = 0;
 }
 
 struct sound *play_sound(struct wav *wav)
@@ -301,9 +336,7 @@ void sound_fillbuf(struct sound *s, short *buf, int n)
 
     short *in = s->data->data;
     for (int i = 0; i < n; i++) {
-        for (int j = 0; j < 2; j++) {
+        for (int j = 0; j < CHANNELS; j++) buf[i*CHANNELS+j] = in[s->frame+j] * gainmult;
-            buf[i*2+j] = in[s->frame+j] * gainmult;
-        }
 
         s->frame++;
         if (s->frame == s->data->frames) {
@@ -313,6 +346,7 @@ void sound_fillbuf(struct sound *s, short *buf, int n)
                 s->frame = 0;
             } else {
                 bus_free(s->bus);
+                return;
             }
         }
     }

source/engine/sound.h

@@ -2,7 +2,6 @@
 #define SOUND_H
 
 struct circbuf;
-struct SDL_AudioStream;
 
 struct Mix_Chunk {
     int i;
@@ -23,16 +22,15 @@ struct soundstream {
 };
 
 struct soundconvstream {
-    // SDL_AudioStream *srconv;
     void *data;
 };
 
 struct soundstream *soundstream_make();
 
-/* A playing sound;
+/* A playing sound */
 struct sound {
-    int loop;
+    int loop; /* How many times to loop */
-    unsigned int frame;
+    unsigned int frame; /* Pointing to the current frame on the wav */
     int playing;
     float gain;
 

source/engine/sound/dsp.c

@@ -160,6 +160,7 @@ struct dsp_filter dsp_filter(void *data, void (*filter)(void *data, short *out,
     struct dsp_filter new;
     new.data = data;
     new.filter = filter;
+    new.inputs = 0;
     return new;
 }
 

source/engine/sound/mix.c

@@ -4,10 +4,12 @@
 #include "sound.h"
 #include "dsp.h"
 #include <string.h>
+#include "log.h"
 
 static struct bus bus[256];
 static int first = 0; /* First bus available */
-static int first_on = -1;
+//static struct bus *first_on = NULL;
+static int first_on = -1; /* First bus to fill buffer with */
 short mastermix[BUF_FRAMES*CHANNELS];
 
 void mixer_init() {
@@ -22,46 +24,44 @@ void mixer_init() {
 
 struct bus *first_free_bus(struct dsp_filter in) {
     if (first == -1) return NULL;
-    struct bus *ret = bus[first];
+    int ret = first;
-    first = ret->next;
+    first = bus[ret].next;
-    ret->on = 1;
-    ret->in = in;
 
-    ret->next = first_on;
+    bus[ret].on = 1;
-    first_on = ret->id;
+    bus[ret].in = in;
 
-    return ret;
+    if (first_on != -1)  bus[first_on].prev = ret;
+
+    bus[ret].next = first_on;
+    bus[ret].prev = -1;
+    first_on = ret;
+
+    return &bus[ret];
 }
 
-void bus_free(struct bus *bus)
+void bus_free(struct bus *b)
 {
-    bus->next = first;
+    if (first_on == b->id) first_on = b->next;
-    bus->on = 0;
+    if (b->next != -1) bus[b->next].prev = b->prev;
-    first = bus->id;
+    if (b->prev != -1) bus[b->prev].next = b->next;
+
+    b->next = first;
+    first = b->id;
+    b->on = 0;
 }
 
 void bus_fill_buffers(short *master, int n) {
-    int curbus = first_one
+    int curbus = first_on;
+    if (curbus == -1) return;
     memset(master, 0, BUF_FRAMES*CHANNELS*sizeof(short));
-    while (bus[curbus].next != -1) {
+
+    while (curbus != -1) {
+        int nextbus = bus[curbus].next; /* Save this in case busses get changed during fill */
         dsp_run(bus[curbus].in, bus[curbus].buf, BUF_FRAMES);
         for (int i = 0; i < BUF_FRAMES*CHANNELS; i++)
             master[i] += bus[curbus].buf[i];
 
-        curbus = bus[curbus].next;
+        curbus = nextbus;
     }
 
-    /*
-    for (int i = 0; i < 256; i++) {
-        if (bus[i].on != 1) continue;
-        dsp_run(bus[i].in, bus[i].buf, BUF_FRAMES);
-    }
-
-        for (int j = 0; j < 256; j++) {
-            if (!bus[j].on) continue;
-            for (int i = 0; i < BUF_FRAMES*CHANNELS; i++) {
-                master[i] += bus[j].buf[i];
-            }
-        }
-            */
 }

source/engine/sound/music.c

@@ -61,6 +61,8 @@ void dsp_midi_fillbuf(struct dsp_midi_song *song, void *out, int n)
     dsp_pan(&music_pan, out, n);
 }
 
+struct bus *musicbus;
+
 void play_song(const char *midi, const char *sf)
 {
     gsong.midi = tml_load_filename(midi);
@@ -85,7 +87,7 @@ void play_song(const char *midi, const char *sf)
 
     cursong.data = &gsong;
     cursong.filter = dsp_midi_fillbuf;
-    first_free_bus(cursong);
+    musicbus = first_free_bus(cursong);
 }
 
 
@@ -96,7 +98,7 @@ void music_play()
 
 void music_stop()
 {
-
+    bus_free(musicbus);
 }
 
 void music_volume()

source/engine/sound/music.h

@@ -14,6 +14,7 @@ struct dsp_midi_song {
 extern float music_pan;
 
 void play_song(const char *midi, const char *sf);
+void music_stop();
 void dsp_midi_fillbuf(struct dsp_midi_song *song, void *out, int n);
 
 #endif