new audio core, RWebAudio. Glitchy, but works well, even with requestAnimationFrame callbacks

Makefile.emscripten

@@ -22,6 +22,7 @@ OBJ = frontend/frontend_emscripten.o \
 		screenshot.o \
 		cheats.o \
 		audio/utils.o \
+		emscripten/RWebAudio.o \
 		input/overlay.o \
 		fifo_buffer.o \
 		gfx/scaler/scaler.o \
@@ -39,7 +40,6 @@ OBJ = frontend/frontend_emscripten.o \
 		performance.o
 
 HAVE_OPENGL = 1
-HAVE_AL = 1
 HAVE_RGUI = 1
 HAVE_SDL = 1
 HAVE_SDL_IMAGE = 1
@@ -59,7 +59,7 @@ libretro = libretro_emscripten.bc
 
 LIBS = -lm
 DEFINES = -DHAVE_SCREENSHOTS -DHAVE_NULLAUDIO -DHAVE_BSV_MOVIE -DPACKAGE_VERSION=\"0.9.9.3\"
-LDFLAGS = -L. -s TOTAL_MEMORY=$(MEMORY)
+LDFLAGS = -L. -s TOTAL_MEMORY=$(MEMORY) --js-library emscripten/library_rwebaudio.js
 
 ifeq ($(SCALER_NO_SIMD), 1)
    DEFINES += -DSCALER_NO_SIMD
@@ -90,12 +90,6 @@ ifeq ($(HAVE_OPENGL), 1)
    DEFINES += -DHAVE_OPENGL -DHAVE_OPENGLES -DHAVE_OPENGLES2 -DHAVE_EGL -DHAVE_OVERLAY -DHAVE_GLSL
 endif
 
-ifeq ($(HAVE_AL), 1)
-   OBJ += audio/openal.o
-   DEFINES += -DHAVE_AL
-   LIBS += -lopenal
-endif
-
 ifeq ($(HAVE_ZLIB), 1)
    OBJ += gfx/rpng/rpng.o file_extract.o
    DEFINES += -DHAVE_ZLIB
@@ -165,6 +159,7 @@ clean:
 	rm -f gfx/fonts/*.o
 	rm -f gfx/py_state/*.o
 	rm -f gfx/rpng/*.o
+	rm -f gfx/glsym/*.o
 	rm -f record/*.o
 	rm -f input/*.o
 	rm -f tools/*.o

config.def.h

@@ -63,6 +63,7 @@ enum
    AUDIO_PS3,
    AUDIO_XENON360,
    AUDIO_WII,
+   AUDIO_RWEBAUDIO,
    AUDIO_NULL,
 
    INPUT_ANDROID,
@@ -130,6 +131,8 @@ enum
 #define AUDIO_DEFAULT_DRIVER AUDIO_SL
 #elif defined(HAVE_DSOUND)
 #define AUDIO_DEFAULT_DRIVER AUDIO_DSOUND
+#elif defined(EMSCRIPTEN)
+#define AUDIO_DEFAULT_DRIVER AUDIO_RWEBAUDIO
 #elif defined(HAVE_SDL)
 #define AUDIO_DEFAULT_DRIVER AUDIO_SDL
 #elif defined(HAVE_XAUDIO)
@@ -335,6 +338,8 @@ static const bool rate_control = false;
 // Rate control delta. Defines how much rate_control is allowed to adjust input rate.
 #if defined(__QNX__)
 static const float rate_control_delta = 0.000;
+#elif defined(EMSCRIPTEN)
+static const float rate_control_delta = 0.002;
 #else
 static const float rate_control_delta = 0.005;
 #endif

driver.c

@@ -84,6 +84,9 @@ static const audio_driver_t *audio_drivers[] = {
 #ifdef GEKKO
    &audio_gx,
 #endif
+#ifdef EMSCRIPTEN
+   &audio_rwebaudio,
+#endif
 #ifdef HAVE_NULLAUDIO
    &audio_null,
 #endif

driver.h

@@ -511,6 +511,7 @@ extern const audio_driver_t audio_coreaudio;
 extern const audio_driver_t audio_xenon360;
 extern const audio_driver_t audio_ps3;
 extern const audio_driver_t audio_gx;
+extern const audio_driver_t audio_rwebaudio;
 extern const audio_driver_t audio_null;
 extern const video_driver_t video_gl;
 extern const video_driver_t video_psp1;

emscripten/RWebAudio.c

@@ -0,0 +1,90 @@
+/*  RetroArch - A frontend for libretro.
+ *  Copyright (C) 2010-2013 - Michael Lelli
+ * 
+ *  RetroArch is free software: you can redistribute it and/or modify it under the terms
+ *  of the GNU General Public License as published by the Free Software Found-
+ *  ation, either version 3 of the License, or (at your option) any later version.
+ *
+ *  RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
+ *  without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
+ *  PURPOSE.  See the GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along with RetroArch.
+ *  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "../driver.h"
+#include "../general.h"
+
+#include "RWebAudio.h"
+
+static void ra_free(void *data)
+{
+   RWebAudioFree();
+}
+
+static void *ra_init(const char *device, unsigned rate, unsigned latency)
+{
+   (void)device;
+   (void)rate;
+   void *data = RWebAudioInit(latency);
+   g_settings.audio.out_rate = RWebAudioSampleRate();
+   RARCH_LOG("audio out rate: %u\n", g_settings.audio.out_rate);
+   return data;
+}
+
+static ssize_t ra_write(void *data, const void *buf, size_t size)
+{
+   (void)data;
+   return RWebAudioWrite(buf, size);
+}
+
+static bool ra_stop(void *data)
+{
+   (void)data;
+   return RWebAudioStop();
+}
+
+static void ra_set_nonblock_state(void *data, bool state)
+{
+   (void)data;
+   RWebAudioSetNonblockState(state);
+}
+
+static bool ra_start(void *data)
+{
+   (void)data;
+   return RWebAudioStart();
+}
+
+static bool ra_use_float(void *data)
+{
+   (void)data;
+   return true;
+}
+
+static size_t ra_write_avail(void *data)
+{
+   (void)data;
+   return RWebAudioWriteAvail();
+}
+
+static size_t ra_buffer_size(void *data)
+{
+   (void)data;
+   return RWebAudioBufferSize();
+}
+
+const audio_driver_t audio_rwebaudio = {
+   ra_init,
+   ra_write,
+   ra_stop,
+   ra_start,
+   ra_set_nonblock_state,
+   ra_free,
+   ra_use_float,
+   "rwebaudio",
+   ra_write_avail,
+   ra_buffer_size,
+};
+

emscripten/RWebAudio.h

@@ -0,0 +1,29 @@
+/*  RetroArch - A frontend for libretro.
+ *  Copyright (C) 2010-2013 - Michael Lelli
+ * 
+ *  RetroArch is free software: you can redistribute it and/or modify it under the terms
+ *  of the GNU General Public License as published by the Free Software Found-
+ *  ation, either version 3 of the License, or (at your option) any later version.
+ *
+ *  RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
+ *  without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
+ *  PURPOSE.  See the GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along with RetroArch.
+ *  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdbool.h>
+
+unsigned RWebAudioSampleRate(void);
+void *RWebAudioInit(unsigned latency);
+ssize_t RWebAudioWrite(const void *buf, size_t size);
+bool RWebAudioStop(void);
+bool RWebAudioStart(void);
+void RWebAudioSetNonblockState(bool state);
+void RWebAudioFree(void);
+size_t RWebAudioWriteAvail(void);
+size_t RWebAudioBufferSize(void);
+int RWebAudioEnoughSpace(void);

emscripten/library_rwebaudio.js

@@ -0,0 +1,161 @@
+//"use strict";
+
+var LibraryRWebAudio = {
+  $RA__deps: ['$Browser'],
+  $RA: {
+    SCRIPTNODE_BUFFER: 1024,
+
+    context: null,
+    leftBuffer: null,
+    rightBuffer: null,
+    blank: null,
+    scriptNode: null,
+    bufferNode: null,
+    start: 0,
+    end: 0,
+    size: 0,
+    lastWrite: 0,
+    nonblock: false,
+    
+    npot: function(n) {
+      n--;
+      n |= n >> 1;
+      n |= n >> 2;
+      n |= n >> 4;
+      n |= n >> 8;
+      n |= n >> 16;
+      n++;
+      return n;
+    },
+
+    process: function(e) {
+      var left = e.outputBuffer.getChannelData(0);
+      var right = e.outputBuffer.getChannelData(1);
+      var samples1 = RA.size;
+      var samples2 = 0;
+      samples1 = e.outputBuffer.length > samples1 ? samples1 : e.outputBuffer.length;
+
+      if (samples1 + RA.start > RA.leftBuffer.length) {
+        samples2 = samples1 + RA.start - RA.leftBuffer.length;
+        samples1 = samples1 - samples2;
+      }
+
+      var remaining = e.outputBuffer.length - (samples1 + samples2);
+
+      if (samples1) {
+        left.set(RA.leftBuffer.subarray(RA.start, RA.start + samples1), 0);
+        right.set(RA.rightBuffer.subarray(RA.start, RA.start + samples1), 0);
+      }
+
+      if (samples2) {
+        left.set(RA.leftBuffer.subarray(0, samples2), samples1);
+        right.set(RA.rightBuffer.subarray(0, samples2), samples1);
+      }
+
+      /*if (remaining) {
+        left.set(RA.blank.subarray(0, remaining), samples1 + samples2);
+        right.set(RA.blank.subarray(0, remaining), samples1 + samples2);
+      }*/
+
+      RA.start = (RA.start + samples1 + samples2) % RA.leftBuffer.length;
+      RA.size -= samples1 + samples2;
+    }
+  },
+  
+  RWebAudioSampleRate: function() {
+    return RA.context.sampleRate;
+  },
+
+  RWebAudioInit: function(latency) {
+    var ac = window['AudioContext'] || window['webkitAudioContext'];
+    var bufferSize;
+
+    if (!ac) return 0;
+
+    RA.context = new ac();
+    // account for script processor overhead
+    latency -= 32;
+    // because we have to guess on how many samples the core will send when
+    // returning early, we double the buffer size to account for times when it
+    // sends more than we expect it to without losing samples
+    bufferSize = RA.npot(RA.context.sampleRate * latency / 1000) * 2;
+    RA.leftBuffer = new Float32Array(bufferSize);
+    RA.rightBuffer = new Float32Array(bufferSize);
+    RA.blank = new Float32Array(RA.SCRIPTNODE_BUFFER);
+    RA.bufferNode = RA.context.createBufferSource();
+    RA.bufferNode.buffer = RA.context.createBuffer(2, RA.SCRIPTNODE_BUFFER, RA.context.sampleRate);
+    RA.bufferNode.loop = true;
+    RA.scriptNode = RA.context.createScriptProcessor(RA.SCRIPTNODE_BUFFER, 2, 2);
+    RA.scriptNode.onaudioprocess = RA.process;
+    RA.bufferNode.connect(RA.scriptNode);
+    RA.scriptNode.connect(RA.context.destination);
+    RA.bufferNode.start(0);
+    RA.start = RA.end = RA.size = 0;
+    RA.nonblock = false;
+    return 1;
+  },
+
+  RWebAudioWrite: function (buf, size) {
+    var samples = size / 8;
+    var free = RA.leftBuffer.length - RA.size;
+    if (free < samples)
+      RA.start = (RA.start + free) % RA.leftBuffer.length;
+
+    for (var i = 0; i < samples; i++) {
+      RA.leftBuffer[RA.end] = {{{ makeGetValue('buf', 'i * 8', 'float') }}};
+      RA.rightBuffer[RA.end] = {{{ makeGetValue('buf', 'i * 8 + 4', 'float') }}};
+      RA.end = (RA.end + 1) % RA.leftBuffer.length;
+    }
+
+    RA.lastWrite = size;
+    RA.size += samples;
+    return size;
+  },
+
+  RWebAudioStop: function() {
+    RA.scriptNode.onaudioprocess = null;
+    return true;
+  },
+
+  RWebAudioStart: function() {
+    RA.scriptNode.onaudioprocess = RA.process;
+    return true;
+  },
+
+  RWebAudioSetNonblockState: function(state) {
+    RA.nonblock = state;
+  },
+
+  RWebAudioFree: function() {
+    RA.scriptNode.onaudioprocess = null;
+    RA.start = RA.end = RA.size = RA.lastWrite = 0;
+    return;
+  },
+
+  RWebAudioWriteAvail: function() {
+    var free  = (RA.leftBuffer.length / 2) - RA.size;
+    // 4 byte samples, 2 channels
+    free *= 8;
+
+    if (free < 0)
+      return 0;
+    else
+      return free;
+  },
+
+  RWebAudioBufferSize: function() {
+    return RA.leftBuffer.length / 2;
+  },
+  
+  RWebAudioEnoughSpace__deps: ['RWebAudioWriteAvail'],
+  RWebAudioEnoughSpace: function() {
+    var guess = RA.lastWrite;
+    var available = _RWebAudioWriteAvail();
+    if (RA.nonblock) return true;
+    if (!guess) return true;
+    return (guess < available) ? 1 : 0;
+  }
+};
+
+autoAddDeps(LibraryRWebAudio, '$RA');
+mergeInto(LibraryManager.library, LibraryRWebAudio);

frontend/frontend_emscripten.c

@@ -18,6 +18,7 @@
 #include "../general.h"
 #include "../conf/config_file.h"
 #include "../file.h"
+#include "../emscripten/RWebAudio.h"
 
 #ifdef HAVE_RGUI
 #include "../frontend/menu/rgui.h"
@@ -55,6 +56,11 @@ static void endloop(void)
 
 static void mainloop(void)
 {
+   if (!RWebAudioEnoughSpace())
+   {
+      return;
+   }
+
    if (g_extern.system.shutdown)
    {
       endloop();

settings.c

@@ -67,6 +67,8 @@ const char *config_get_default_audio(void)
          return "ps3";
       case AUDIO_WII:
          return "gx";
+      case AUDIO_RWEBAUDIO:
+         return "rwebaudio";
       case AUDIO_NULL:
          return "null";
       default: