Split "Online" and "Offline" AudioContext processing

Source/modules/webaudio/AudioBufferSourceNode.cpp

 /*
  * Copyright (C) 2010, Google Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
  * are met:
  * 1.  Redistributions of source code must retain the above copyright
  *    notice, this list of conditions and the following disclaimer.
  * 2.  Redistributions in binary form must reproduce the above copyright
  *    notice, this list of conditions and the following disclaimer in the
@@ skipping 12 matching lines @@
  */
 
 #include "config.h"
 #if ENABLE(WEB_AUDIO)
 #include "modules/webaudio/AudioBufferSourceNode.h"
 
 #include "bindings/core/v8/ExceptionMessages.h"
 #include "bindings/core/v8/ExceptionState.h"
 #include "core/dom/ExceptionCode.h"
 #include "core/frame/UseCounter.h"
-#include "modules/webaudio/AudioContext.h"
+#include "modules/webaudio/AbstractAudioContext.h"
 #include "modules/webaudio/AudioNodeOutput.h"
 #include "platform/FloatConversion.h"
 #include "platform/audio/AudioUtilities.h"
 #include "wtf/MainThread.h"
 #include "wtf/MathExtras.h"
 #include <algorithm>
 
 namespace blink {
 
 const double DefaultGrainDuration = 0.020; // 20ms
@@ skipping 296 matching lines @@
     ASSERT(isMainThread());
 
     if (m_buffer) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "Cannot set buffer after it has been already been set");
         return;
     }
 
     // The context must be locked since changing the buffer can re-configure the number of channels that are output.
-    AudioContext::AutoLocker contextLocker(context());
+    AbstractAudioContext::AutoLocker contextLocker(context());
 
     // This synchronizes with process().
     MutexLocker processLocker(m_processLock);
 
     if (buffer) {
         // Do any necesssary re-configuration to the buffer's number of channels.
         unsigned numberOfChannels = buffer->numberOfChannels();
 
         // This should not be possible since AudioBuffers can't be created with too many channels
         // either.
-        if (numberOfChannels > AudioContext::maxNumberOfChannels()) {
+        if (numberOfChannels > AbstractAudioContext::maxNumberOfChannels()) {
             exceptionState.throwDOMException(
                 NotSupportedError,
                 ExceptionMessages::indexOutsideRange(
                     "number of input channels",
                     numberOfChannels,
                     1u,
                     ExceptionMessages::InclusiveBound,
-                    AudioContext::maxNumberOfChannels(),
+                    AbstractAudioContext::maxNumberOfChannels(),
                     ExceptionMessages::InclusiveBound));
             return;
         }
 
         output(0).setNumberOfChannels(numberOfChannels);
 
         m_sourceChannels = adoptArrayPtr(new const float* [numberOfChannels]);
         m_destinationChannels = adoptArrayPtr(new float* [numberOfChannels]);
 
         for (unsigned i = 0; i < numberOfChannels; ++i)
@@ skipping 122 matching lines @@
 
     m_playbackState = SCHEDULED_STATE;
 }
 
 double AudioBufferSourceHandler::computePlaybackRate()
 {
     double dopplerRate = 1;
     if (m_pannerNode)
         dopplerRate = m_pannerNode->dopplerRate();
 
-    // Incorporate buffer's sample-rate versus AudioContext's sample-rate.
+    // Incorporate buffer's sample-rate versus AbstractAudioContext's sample-rate.
     // Normally it's not an issue because buffers are loaded at the
-    // AudioContext's sample-rate, but we can handle it in any case.
+    // AbstractAudioContext's sample-rate, but we can handle it in any case.
     double sampleRateFactor = 1.0;
     if (buffer()) {
         // Use doubles to compute this to full accuracy.
         sampleRateFactor = buffer()->sampleRate() / static_cast<double>(sampleRate());
     }
 
     // Use finalValue() to incorporate changes of AudioParamTimeline and
     // AudioSummingJunction from m_playbackRate AudioParam.
     double basePlaybackRate = m_playbackRate->finalValue();
 
@@ skipping 76 matching lines @@
 }
 
 void AudioBufferSourceHandler::finish()
 {
     clearPannerNode();
     ASSERT(!m_pannerNode);
     AudioScheduledSourceHandler::finish();
 }
 
 // ----------------------------------------------------------------
-AudioBufferSourceNode::AudioBufferSourceNode(AudioContext& context, float sampleRate)
+AudioBufferSourceNode::AudioBufferSourceNode(AbstractAudioContext& context, float sampleRate)
     : AudioScheduledSourceNode(context)
     , m_playbackRate(AudioParam::create(context, 1.0))
     , m_detune(AudioParam::create(context, 0.0))
 {
     setHandler(AudioBufferSourceHandler::create(*this, sampleRate, m_playbackRate->handler(), m_detune->handler()));
 }
 
-AudioBufferSourceNode* AudioBufferSourceNode::create(AudioContext& context, float sampleRate)
+AudioBufferSourceNode* AudioBufferSourceNode::create(AbstractAudioContext& context, float sampleRate)
 {
     return new AudioBufferSourceNode(context, sampleRate);
 }
 
 DEFINE_TRACE(AudioBufferSourceNode)
 {
     visitor->trace(m_playbackRate);
     visitor->trace(m_detune);
     AudioScheduledSourceNode::trace(visitor);
 }
@@ skipping 69 matching lines @@
 }
 
 void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
 {
     audioBufferSourceHandler().start(when, grainOffset, grainDuration, exceptionState);
 }
 
 } // namespace blink
 
 #endif // ENABLE(WEB_AUDIO)