Rename es => exceptionState in other than bindings/

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 320 matching lines @@
     return true;
 }
 
 
 void AudioBufferSourceNode::reset()
 {
     m_virtualReadIndex = 0;
     m_lastGain = gain()->value();
 }
 
-void AudioBufferSourceNode::setBuffer(AudioBuffer* buffer, ExceptionState& es)
+void AudioBufferSourceNode::setBuffer(AudioBuffer* buffer, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
     // FIXME: It does not look like we should throw if the buffer is null as
     // the attribute is nullable in the specification.
     if (!buffer) {
-        es.throwTypeError("buffer cannot be null");
+        exceptionState.throwTypeError("buffer cannot be null");
         return;
     }
 
     // The context must be locked since changing the buffer can re-configure the number of channels that are output.
     AudioContext::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();
 
         if (numberOfChannels > AudioContext::maxNumberOfChannels()) {
-            es.throwTypeError("number of input channels (" + String::number(numberOfChannels)
+            exceptionState.throwTypeError("number of input channels (" + String::number(numberOfChannels)
                 + ") exceeds maximum ("
                 + String::number(AudioContext::maxNumberOfChannels()) + ").");
             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)
             m_sourceChannels[i] = buffer->getChannelData(i)->data();
     }
 
     m_virtualReadIndex = 0;
     m_buffer = buffer;
 }
 
 unsigned AudioBufferSourceNode::numberOfChannels()
 {
     return output(0)->numberOfChannels();
 }
 
-void AudioBufferSourceNode::start(ExceptionState& es)
+void AudioBufferSourceNode::start(ExceptionState& exceptionState)
 {
-    startPlaying(false, 0, 0, buffer() ? buffer()->duration() : 0, es);
+    startPlaying(false, 0, 0, buffer() ? buffer()->duration() : 0, exceptionState);
 }
 
-void AudioBufferSourceNode::start(double when, ExceptionState& es)
+void AudioBufferSourceNode::start(double when, ExceptionState& exceptionState)
 {
-    startPlaying(false, when, 0, buffer() ? buffer()->duration() : 0, es);
+    startPlaying(false, when, 0, buffer() ? buffer()->duration() : 0, exceptionState);
 }
 
-void AudioBufferSourceNode::start(double when, double grainOffset, ExceptionState& es)
+void AudioBufferSourceNode::start(double when, double grainOffset, ExceptionState& exceptionState)
 {
-    startPlaying(true, when, grainOffset, buffer() ? buffer()->duration() : 0, es);
+    startPlaying(true, when, grainOffset, buffer() ? buffer()->duration() : 0, exceptionState);
 }
 
-void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration, ExceptionState& es)
+void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
 {
-    startPlaying(true, when, grainOffset, grainDuration, es);
+    startPlaying(true, when, grainOffset, grainDuration, exceptionState);
 }
 
-void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grainOffset, double grainDuration, ExceptionState& es)
+void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
 
     if (m_playbackState != UNSCHEDULED_STATE) {
-        es.throwDOMException(
+        exceptionState.throwDOMException(
             InvalidStateError,
             ExceptionMessages::failedToExecute(
                 "start",
                 nodeTypeName(),
                 "cannot call start more than once."));
         return;
     }
 
     if (!buffer())
         return;
@@ skipping 18 matching lines @@
 
     // We call timeToSampleFrame here since at playbackRate == 1 we don't want to go through linear interpolation
     // at a sub-sample position since it will degrade the quality.
     // When aligned to the sample-frame the playback will be identical to the PCM data stored in the buffer.
     // Since playbackRate == 1 is very common, it's worth considering quality.
     m_virtualReadIndex = AudioUtilities::timeToSampleFrame(m_grainOffset, buffer()->sampleRate());
 
     m_playbackState = SCHEDULED_STATE;
 }
 
-void AudioBufferSourceNode::noteGrainOn(double when, double grainOffset, double grainDuration, ExceptionState& es)
+void AudioBufferSourceNode::noteGrainOn(double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
 {
     // Handle unspecified duration where 0 means the rest of the buffer.
     if (!grainDuration && buffer())
         grainDuration = buffer()->duration();
-    startPlaying(true, when, grainOffset, grainDuration, es);
+    startPlaying(true, when, grainOffset, grainDuration, exceptionState);
 }
 
 double AudioBufferSourceNode::totalPitchRate()
 {
     double dopplerRate = 1.0;
     if (m_pannerNode)
         dopplerRate = m_pannerNode->dopplerRate();
 
     // Incorporate buffer's sample-rate versus AudioContext'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.
@@ skipping 47 matching lines @@
 void AudioBufferSourceNode::finish()
 {
     clearPannerNode();
     ASSERT(!m_pannerNode);
     AudioScheduledSourceNode::finish();
 }
 
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)