[EMSCRIPTEN] more audio fixes, revert to busywait method

config.def.h

@@ -338,8 +338,6 @@ 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

emscripten/RWebAudio.c

@@ -28,8 +28,8 @@ static void *ra_init(const char *device, unsigned rate, unsigned latency)
    (void)device;
    (void)rate;
    void *data = RWebAudioInit(latency);
+   if (data)
       g_settings.audio.out_rate = RWebAudioSampleRate();
-   RARCH_LOG("audio out rate: %u\n", g_settings.audio.out_rate);
    return data;
 }
 

emscripten/RWebAudio.h

@@ -26,4 +26,3 @@ void RWebAudioSetNonblockState(bool state);
 void RWebAudioFree(void);
 size_t RWebAudioWriteAvail(void);
 size_t RWebAudioBufferSize(void);
-int RWebAudioEnoughSpace(void);

emscripten/library_rwebaudio.js

@@ -3,122 +3,135 @@
 var LibraryRWebAudio = {
    $RA__deps: ['$Browser'],
    $RA: {
-    SCRIPTNODE_BUFFER: 1024,
+      BUFFER_SIZE: 256,
 
       context: null,
-    leftBuffer: null,
+      buffers: [],
-    rightBuffer: null,
+      numBuffers: 0,
-    blank: null,
+      bufIndex: 0,
-    scriptNode: null,
+      bufOffset: 0,
-    bufferNode: null,
+      startTime: 0,
-    start: 0,
-    end: 0,
-    size: 0,
-    lastWrite: 0,
       nonblock: false,
 
-    npot: function(n) {
+      setStartTime: function() {
-      n--;
+         if (RA.context.currentTime) {
-      n |= n >> 1;
+            RA.startTime = window['performance']['now']() - RA.context.currentTime * 1000;
-      n |= n >> 2;
+            if (RA.startTime === 0) throw 'startTime is 0';
-      n |= n >> 4;
+            Module["resumeMainLoop"]();
-      n |= n >> 8;
+         } else window['setTimeout'](RA.setStartTime, 0);
-      n |= n >> 16;
-      n++;
-      return n;
       },
 
-    process: function(e) {
+      getCurrentPerfTime: function() {
-      var left = e.outputBuffer.getChannelData(0);
+         if (RA.startTime) return (window['performance']['now']() - RA.startTime) / 1000;
-      var right = e.outputBuffer.getChannelData(1);
+         else throw 'getCurrentPerfTime() called before start time set';
-      var samples1 = RA.size;
+      },
-      var samples2 = 0;
-      samples1 = e.outputBuffer.length > samples1 ? samples1 : e.outputBuffer.length;
 
-      if (samples1 + RA.start > RA.leftBuffer.length) {
+      process: function(queueBuffers) {
-        samples2 = samples1 + RA.start - RA.leftBuffer.length;
+         var currentTime = RA.getCurrentPerfTime();
-        samples1 = samples1 - samples2;
+         for (var i = 0; i < RA.bufIndex; i++) {
+            if (RA.buffers[i].endTime < currentTime) {
+               var buf = RA.buffers.splice(i, 1);
+               RA.buffers[RA.numBuffers - 1] = buf[0];
+               i--;
+               RA.bufIndex--;
+            }
+         }
+      },
+
+      fillBuffer: function(buf, samples) {
+         var count = 0;
+         var leftBuffer = RA.buffers[RA.bufIndex].getChannelData(0);
+         var rightBuffer = RA.buffers[RA.bufIndex].getChannelData(1);
+         while (samples && RA.bufOffset !== RA.BUFFER_SIZE) {
+            leftBuffer[RA.bufOffset] = {{{ makeGetValue('buf', 'count * 8', 'float') }}};
+            rightBuffer[RA.bufOffset] = {{{ makeGetValue('buf', 'count * 8 + 4', 'float') }}};
+            RA.bufOffset++;
+            count++;
+            samples--;
          }
 
-      var remaining = e.outputBuffer.length - (samples1 + samples2);
+         return count;
+      },
 
-      if (samples1) {
+      queueAudio: function() {
-        left.set(RA.leftBuffer.subarray(RA.start, RA.start + samples1), 0);
+         var index = RA.bufIndex;
-        right.set(RA.rightBuffer.subarray(RA.start, RA.start + samples1), 0);
+
+         var startTime;
+         if (RA.bufIndex) startTime = RA.buffers[RA.bufIndex - 1].endTime;
+         else startTime = RA.context.currentTime;
+         RA.buffers[index].endTime = startTime + RA.buffers[index].duration;
+
+         var bufferSource = RA.context.createBufferSource();
+         bufferSource.buffer = RA.buffers[index];
+         bufferSource.connect(RA.context.destination);
+         bufferSource.start(startTime);
+
+         RA.bufIndex++;
+         RA.bufOffset = 0;
+      },
+
+      block: function() {
+         do {
+            RA.process();
+         } while (RA.bufIndex === RA.numBuffers - 1);
       }
+   },
 
-      if (samples2) {
+   RWebAudioInit: function(latency) {
-        left.set(RA.leftBuffer.subarray(0, samples2), samples1);
+      var ac = window['AudioContext'] || window['webkitAudioContext'];
-        right.set(RA.rightBuffer.subarray(0, samples2), samples1);
-      }
 
-      /*if (remaining) {
+      if (!ac) return 0;
-        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.context = new ac();
-      RA.size -= samples1 + samples2;
+
-    }
+      RA.numBuffers = ((latency * RA.context.sampleRate) / (1000 * RA.BUFFER_SIZE))|0;
+      if (RA.numBuffers < 2) RA.numBuffers = 2;
+
+      for (var i = 0; i < RA.numBuffers; i++) RA.buffers[i] = RA.context.createBuffer(2, RA.BUFFER_SIZE, RA.context.sampleRate);
+
+      RA.nonblock = false;
+      RA.startTime = 0;
+      // chrome hack to get currentTime running
+      RA.context.createGain();
+      window['setTimeout'](RA.setStartTime, 0);
+      Module["pauseMainLoop"]();
+      return 1;
    },
    
    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) {
+      RA.process();
       var samples = size / 8;
-    var free = RA.leftBuffer.length - RA.size;
+      var count = 0;
-    if (free < samples)
-      RA.start = (RA.start + free) % RA.leftBuffer.length;
 
-    for (var i = 0; i < samples; i++) {
+      while (samples) {
-      RA.leftBuffer[RA.end] = {{{ makeGetValue('buf', 'i * 8', 'float') }}};
+         var fill = RA.fillBuffer(buf, samples);
-      RA.rightBuffer[RA.end] = {{{ makeGetValue('buf', 'i * 8 + 4', 'float') }}};
+         samples -= fill;
-      RA.end = (RA.end + 1) % RA.leftBuffer.length;
+         count += fill;
+         buf += fill * 8;
+
+         if (RA.bufOffset === RA.BUFFER_SIZE) {
+            if (RA.bufIndex === RA.numBuffers - 1) {
+               if (RA.nonblock) break;
+               else RA.block();
+            }
+            RA.queueAudio();
+         }
       }
 
-    RA.lastWrite = size;
+      return count * 8;
-    RA.size += samples;
-    return size;
    },
 
    RWebAudioStop: function() {
-    RA.scriptNode.onaudioprocess = null;
+      RA.bufIndex = 0;
+      RA.bufOffset = 0;
       return true;
    },
 
    RWebAudioStart: function() {
-    RA.scriptNode.onaudioprocess = RA.process;
       return true;
    },
 
@@ -127,33 +140,18 @@ var LibraryRWebAudio = {
    },
 
    RWebAudioFree: function() {
-    RA.scriptNode.onaudioprocess = null;
+      RA.bufIndex = 0;
-    RA.start = RA.end = RA.size = RA.lastWrite = 0;
+      RA.bufOffset = 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;
+      return RA.numBuffers * RA.BUFFER_SIZE + RA.BUFFER_SIZE;
    },
 
-  RWebAudioEnoughSpace__deps: ['RWebAudioWriteAvail'],
+   RWebAudioWriteAvail: function() {
-  RWebAudioEnoughSpace: function() {
+      RA.process();
-    var guess = RA.lastWrite;
+      return ((RA.numBuffers - RA.bufIndex) * RA.BUFFER_SIZE - RA.bufOffset) * 8;
-    var available = _RWebAudioWriteAvail();
-    if (RA.nonblock) return true;
-    if (!guess) return true;
-    return (guess < available) ? 1 : 0;
    }
 };
 

frontend/frontend_emscripten.c

@@ -56,11 +56,6 @@ static void endloop(void)
 
 static void mainloop(void)
 {
-   if (!RWebAudioEnoughSpace())
-   {
-      return;
-   }
-
    if (g_extern.system.shutdown)
    {
       endloop();