Tidy up redundant call chain of CallbackBufferSize

CallbackBufferSize() were scattered around many classes for no specific
reason. This CL cleans them up and contains it to
DefaultAudioDestinationNode.

Bug: 854229
Change-Id: I9116cf8ecd9398108fcdcac89cbbfb8c42de957e
Reviewed-on: https://chromium-review.googlesource.com/1112072
Commit-Queue: Hongchan Choi <hongchan@chromium.org>
Reviewed-by: Raymond Toy <rtoy@chromium.org>
Cr-Commit-Position: refs/heads/master@{#570081}

third_party/blink/renderer/modules/webaudio/audio_destination_node.h

@@ -68,8 +68,6 @@ class AudioDestinationHandler : public AudioHandler, public AudioIOCallback {
   // restart of the context.
   virtual void RestartRendering() = 0;
 
-  // Returns the rendering callback buffer size.
-  size_t CallbackBufferSize() const override = 0;
   virtual double SampleRate() const = 0;
 
  protected:
@@ -84,7 +82,6 @@ class AudioDestinationNode : public AudioNode {
   AudioDestinationHandler& GetAudioDestinationHandler() const;
 
   unsigned long maxChannelCount() const;
-  size_t CallbackBufferSize() const { return Handler().CallbackBufferSize(); }
 
  protected:
   AudioDestinationNode(BaseAudioContext&);

third_party/blink/renderer/modules/webaudio/audio_node.h

@@ -238,11 +238,6 @@ class MODULES_EXPORT AudioHandler : public ThreadSafeRefCounted<AudioHandler> {
   void UpdateChannelCountMode();
   void UpdateChannelInterpretation();
 
-  // Default callbackBufferSize should be the render quantum size
-  virtual size_t CallbackBufferSize() const {
-    return AudioUtilities::kRenderQuantumFrames;
-  }
-
  protected:
   // Inputs and outputs must be created before the AudioHandler is
   // initialized.

third_party/blink/renderer/modules/webaudio/base_audio_context.h

@@ -147,11 +147,6 @@ class MODULES_EXPORT BaseAudioContext
                : ClosedContextSampleRate();
   }
 
-  size_t CallbackBufferSize() const {
-    return destination_node_ ? destination_node_->Handler().CallbackBufferSize()
-                             : 0;
-  }
-
   String state() const;
   AudioContextState ContextState() const { return context_state_; }
   void ThrowExceptionForClosedState(ExceptionState&);

third_party/blink/renderer/modules/webaudio/default_audio_destination_node.cc

@@ -132,7 +132,7 @@ unsigned long DefaultAudioDestinationHandler::MaxChannelCount() const {
   return AudioDestination::MaxChannelCount();
 }
 
-size_t DefaultAudioDestinationHandler::CallbackBufferSize() const {
+size_t DefaultAudioDestinationHandler::GetCallbackBufferSize() const {
   return destination_->CallbackBufferSize();
 }
 

third_party/blink/renderer/modules/webaudio/default_audio_destination_node.h

@@ -55,10 +55,11 @@ class DefaultAudioDestinationHandler final : public AudioDestinationHandler {
   void StopRendering() override;
   void RestartRendering() override;
   unsigned long MaxChannelCount() const override;
-  // Returns the rendering callback buffer size.
-  size_t CallbackBufferSize() const override;
   double SampleRate() const override;
 
+  // Returns a hadrware callback buffer size from audio infra.
+  size_t GetCallbackBufferSize() const;
+
   // Returns a given frames-per-buffer size from audio infra.
   int GetFramesPerBuffer() const;
 
@@ -89,8 +90,6 @@ class DefaultAudioDestinationNode final : public AudioDestinationNode {
   static DefaultAudioDestinationNode* Create(BaseAudioContext*,
                                              const WebAudioLatencyHint&);
 
-  size_t CallbackBufferSize() const { return Handler().CallbackBufferSize(); }
-
  private:
   explicit DefaultAudioDestinationNode(BaseAudioContext&,
                                        const WebAudioLatencyHint&);

third_party/blink/renderer/modules/webaudio/offline_audio_destination_node.cc

@@ -135,12 +135,6 @@ void OfflineAudioDestinationHandler::StopRendering() {
   NOTREACHED();
 }
 
-size_t OfflineAudioDestinationHandler::CallbackBufferSize() const {
-  // The callback buffer size has no meaning for an offline context.
-  NOTREACHED();
-  return 0;
-}
-
 void OfflineAudioDestinationHandler::InitializeOfflineRenderThread(
     AudioBuffer* render_target) {
   DCHECK(IsMainThread());

third_party/blink/renderer/modules/webaudio/offline_audio_destination_node.h

@@ -66,10 +66,6 @@ class OfflineAudioDestinationHandler final : public AudioDestinationHandler {
 
   void RestartRendering() override;
 
-  // Returns the rendering callback buffer size.  This should never be
-  // called.
-  size_t CallbackBufferSize() const override;
-
   double SampleRate() const override { return sample_rate_; }
 
   size_t RenderQuantumFrames() const {

third_party/blink/renderer/modules/webaudio/script_processor_node.cc

@@ -35,6 +35,7 @@
 #include "third_party/blink/renderer/modules/webaudio/audio_node_output.h"
 #include "third_party/blink/renderer/modules/webaudio/audio_processing_event.h"
 #include "third_party/blink/renderer/modules/webaudio/base_audio_context.h"
+#include "third_party/blink/renderer/modules/webaudio/default_audio_destination_node.h"
 #include "third_party/blink/renderer/platform/bindings/exception_state.h"
 #include "third_party/blink/renderer/platform/cross_thread_functional.h"
 #include "third_party/blink/renderer/platform/waitable_event.h"
@@ -373,7 +374,7 @@ static size_t ChooseBufferSize(size_t callback_buffer_size) {
   // Choose a buffer size based on the audio hardware buffer size. Arbitarily
   // make it a power of two that is 4 times greater than the hardware buffer
   // size.
-  // FIXME: What is the best way to choose this?
+  // TODO(crbug.com/855758): What is the best way to choose this?
   size_t buffer_size =
       1 << static_cast<unsigned>(log2(4 * callback_buffer_size) + 0.5);
 
@@ -397,29 +398,29 @@ ScriptProcessorNode* ScriptProcessorNode::Create(
 
 ScriptProcessorNode* ScriptProcessorNode::Create(
     BaseAudioContext& context,
-    size_t buffer_size,
+    size_t requested_buffer_size,
     ExceptionState& exception_state) {
   DCHECK(IsMainThread());
 
   // Default is 2 inputs and 2 outputs.
-  return Create(context, buffer_size, 2, 2, exception_state);
+  return Create(context, requested_buffer_size, 2, 2, exception_state);
 }
 
 ScriptProcessorNode* ScriptProcessorNode::Create(
     BaseAudioContext& context,
-    size_t buffer_size,
+    size_t requested_buffer_size,
     unsigned number_of_input_channels,
     ExceptionState& exception_state) {
   DCHECK(IsMainThread());
 
   // Default is 2 outputs.
-  return Create(context, buffer_size, number_of_input_channels, 2,
+  return Create(context, requested_buffer_size, number_of_input_channels, 2,
                 exception_state);
 }
 
 ScriptProcessorNode* ScriptProcessorNode::Create(
     BaseAudioContext& context,
-    size_t buffer_size,
+    size_t requested_buffer_size,
     unsigned number_of_input_channels,
     unsigned number_of_output_channels,
     ExceptionState& exception_state) {
@@ -455,17 +456,27 @@ ScriptProcessorNode* ScriptProcessorNode::Create(
     return nullptr;
   }
 
-  // Check for valid buffer size.
-  switch (buffer_size) {
+  // Sanitize user-supplied buffer size.
+  size_t buffer_size;
+  switch (requested_buffer_size) {
     case 0:
       // Choose an appropriate size.  For an AudioContext, we need to
       // choose an appropriate size based on the callback buffer size.
       // For OfflineAudioContext, there's no callback buffer size, so
       // just use the minimum valid buffer size.
-      buffer_size =
-          context.HasRealtimeConstraint()
-              ? ChooseBufferSize(context.destination()->CallbackBufferSize())
-              : 256;
+      if (context.HasRealtimeConstraint()) {
+        // TODO(crbug.com/854229): Due to the incompatible constructor between
+        // AudioDestinationNode and DefaultAudioDestinationNode, casting
+        // directly from |destination()| is impossible. This is a temporary
+        // workaround until the refactoring is completed.
+        DefaultAudioDestinationHandler& destination_handler =
+            static_cast<DefaultAudioDestinationHandler&>(
+                context.destination()->GetAudioDestinationHandler());
+        buffer_size =
+            ChooseBufferSize(destination_handler.GetCallbackBufferSize());
+      } else {
+        buffer_size = 256;
+      }
       break;
     case 256:
     case 512:
@@ -474,11 +485,12 @@ ScriptProcessorNode* ScriptProcessorNode::Create(
     case 4096:
     case 8192:
     case 16384:
+      buffer_size = requested_buffer_size;
       break;
     default:
       exception_state.ThrowDOMException(
           DOMExceptionCode::kIndexSizeError,
-          "buffer size (" + String::Number(buffer_size) +
+          "buffer size (" + String::Number(requested_buffer_size) +
               ") must be 0 or a power of two between 256 and 16384.");
       return nullptr;
   }

third_party/blink/renderer/modules/webaudio/script_processor_node.h

@@ -126,14 +126,14 @@ class ScriptProcessorNode final
   // The value chosen must carefully balance between latency and audio quality.
   static ScriptProcessorNode* Create(BaseAudioContext&, ExceptionState&);
   static ScriptProcessorNode* Create(BaseAudioContext&,
-                                     size_t buffer_size,
+                                     size_t requested_buffer_size,
                                      ExceptionState&);
   static ScriptProcessorNode* Create(BaseAudioContext&,
-                                     size_t buffer_size,
+                                     size_t requested_buffer_size,
                                      unsigned number_of_input_channels,
                                      ExceptionState&);
   static ScriptProcessorNode* Create(BaseAudioContext&,
-                                     size_t buffer_size,
+                                     size_t requested_buffer_size,
                                      unsigned number_of_input_channels,
                                      unsigned number_of_output_channels,
                                      ExceptionState&);