[AUDIO_WORKET] Add support for setting the render quantum size

src/audio_worklet.js

@@ -154,6 +154,10 @@ function createWasmAudioWorkletProcessor() {
       var stackMemoryAligned = (stackMemoryStruct + stackMemoryData + 15) & ~15;
       var structPtr = stackAlloc(stackMemoryAligned);
       var dataPtr = structPtr + (stackMemoryAligned - stackMemoryData);
+#if ASSERTIONS
+      // TODO: look at why stackAlloc isn't tripping the assertions
+      console.assert(stackMemoryAligned <= wwParams.stackSize, `Not enough stack allocated to the AudioWorklet (need ${stackMemoryAligned}, got ${wwParams.stackSize})`);
+#endif
 
       // Copy input audio descriptor structs and data to Wasm (recall, structs
       // first, audio data after). 'inputsPtr' is the start of the C callback's

src/lib/libwebaudio.js

@@ -77,12 +77,7 @@ var LibraryWebAudio = {
 
   // Performs the work of getting the AudioContext's render quantum size.
   $emscriptenGetContextQuantumSize: (contextHandle) => {
-    // TODO: in a future release this will be something like:
-    //   return EmAudio[contextHandle].renderQuantumSize || 128;
-    // It comes two caveats: it needs the hint when generating the context adding to
-    // emscripten_create_audio_context(), and altering the quantum requires a secure
-    // context and fallback implementing. Until then we simply use the 1.0 API value:
-    return 128;
+    return EmAudio[contextHandle].renderQuantumSize || 128;
   },
 
   // emscripten_create_audio_context() does not itself use the
@@ -100,9 +95,15 @@ var LibraryWebAudio = {
 #endif
 #endif
 
+    // AUDIO_CONTEXT_RENDER_SIZE_DEFAULT and AUDIO_CONTEXT_RENDER_SIZE_HARDWARE
+    // into their AudioContextRenderSizeCategory enum, or take the positive int. 
+    function readRenderSizeHint(val) {
+        return (val == 0) ? "default" : ((val < 0) ? "hardware" : val);
+    }
     var opts = options ? {
       latencyHint: UTF8ToString({{{ makeGetValue('options', C_STRUCTS.EmscriptenWebAudioCreateAttributes.latencyHint, '*') }}}) || undefined,
-      sampleRate: {{{ makeGetValue('options', C_STRUCTS.EmscriptenWebAudioCreateAttributes.sampleRate, 'u32') }}} || undefined
+      sampleRate: {{{ makeGetValue('options', C_STRUCTS.EmscriptenWebAudioCreateAttributes.sampleRate, 'u32') }}} || undefined,
+      renderSizeHint: readRenderSizeHint({{{ makeGetValue('options', C_STRUCTS.EmscriptenWebAudioCreateAttributes.renderSizeHint, 'i32') }}})
     } : undefined;
 
 #if WEBAUDIO_DEBUG

src/struct_info.json

@@ -1269,7 +1269,8 @@
         "structs": {
             "EmscriptenWebAudioCreateAttributes": [
               "latencyHint",
-              "sampleRate"
+              "sampleRate",
+              "renderSizeHint"
             ],
             "WebAudioParamDescriptor": [
               "defaultValue",

src/struct_info_generated.json

@@ -700,8 +700,9 @@
             "visibilityState": 4
         },
         "EmscriptenWebAudioCreateAttributes": {
-            "__size__": 8,
+            "__size__": 12,
             "latencyHint": 0,
+            "renderSizeHint": 8,
             "sampleRate": 4
         },
         "EmscriptenWebGLContextAttributes": {

src/struct_info_generated_wasm64.json

@@ -702,6 +702,7 @@
         "EmscriptenWebAudioCreateAttributes": {
             "__size__": 16,
             "latencyHint": 0,
+            "renderSizeHint": 12,
             "sampleRate": 8
         },
         "EmscriptenWebGLContextAttributes": {

system/include/emscripten/webaudio.h

@@ -19,10 +19,16 @@ extern "C" {
 
 typedef int EMSCRIPTEN_WEBAUDIO_T;
 
+// Default render size of 128 frames
+#define AUDIO_CONTEXT_RENDER_SIZE_DEFAULT 0
+// Let the hardware determine the best render size
+#define AUDIO_CONTEXT_RENDER_SIZE_HARDWARE -1
+
 typedef struct EmscriptenWebAudioCreateAttributes
 {
 	const char *latencyHint; // Specify one of "balanced", "interactive" or "playback"
 	uint32_t sampleRate; // E.g. 44100 or 48000
+	int32_t renderSizeHint; // AUDIO_CONTEXT_RENDER_SIZE_* or number of samples
 } EmscriptenWebAudioCreateAttributes;
 
 // Creates a new Web Audio AudioContext, and returns a handle to it.

test/webaudio/audioworklet_params_mixing.c

@@ -9,10 +9,18 @@
 // create variable parameter data sizes, depending on the browser, it's also the
 // ideal to test audio worklets don't corrupt TLS variables.
 
+// Large render size (approx 42ms)
+#define RENDER_SIZE_HINT 2048
+
 // This needs to be big enough for the stereo output, 2x inputs, 2x params and
 // the worker stack. To note that different browsers have different stack size
 // requirement (see notes in process() plus the expansion of the params).
+#ifndef RENDER_SIZE_HINT
 #define AUDIO_STACK_SIZE 6144
+#else
+// float bytes * stereo * ins/outs + extra stack
+#define AUDIO_STACK_SIZE ((RENDER_SIZE_HINT * 4 * 2 * 5) + 1024)
+#endif
 
 // Shared file playback and bootstrap
 #include "audioworklet_test_shared.inc"

test/webaudio/audioworklet_test_shared.inc

@@ -95,8 +95,12 @@ int main(void) {
   assert(workletStack);
 
   // Set at least the latency hint to test the attribute setting
+  // (The render size we take from the calling code if set)
   EmscriptenWebAudioCreateAttributes attrs = {
-    .latencyHint = "balanced"
+    .latencyHint = "balanced",
+#ifdef RENDER_SIZE_HINT
+    .renderSizeHint = RENDER_SIZE_HINT
+#endif
   };
   EMSCRIPTEN_WEBAUDIO_T context = emscripten_create_audio_context(&attrs);
   emscripten_start_wasm_audio_worklet_thread_async(context, workletStack, AUDIO_STACK_SIZE, getStartCallback(), NULL);