Revert regression in AudioBufferSourceNode.start() with no buffer

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 358 matching lines @@
 
     m_virtualReadIndex = 0;
     m_buffer = buffer;
 }
 
 unsigned AudioBufferSourceNode::numberOfChannels()
 {
     return output(0)->numberOfChannels();
 }
 
-void AudioBufferSourceNode::start(ExceptionState& exceptionState)
-{
-    startPlaying(false, 0, 0, buffer() ? buffer()->duration() : 0, exceptionState);
-}
-
 void AudioBufferSourceNode::start(double when, ExceptionState& exceptionState)
 {
-    startPlaying(false, when, 0, buffer() ? buffer()->duration() : 0, exceptionState);
+    AudioScheduledSourceNode::start(when, exceptionState);
 }
 
 void AudioBufferSourceNode::start(double when, double grainOffset, ExceptionState& exceptionState)
 {
-    startPlaying(true, when, grainOffset, buffer() ? buffer()->duration() : 0, exceptionState);
+    start(when, grainOffset, buffer() ? buffer()->duration() : 0, exceptionState);
 }
 
 void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
 {
-    startPlaying(true, when, grainOffset, grainDuration, exceptionState);
-}
-
-void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
-{
     ASSERT(isMainThread());
 
     if (m_playbackState != UNSCHEDULED_STATE) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "cannot call start more than once.");
         return;
     }
 
     if (!buffer())
         return;
 
-    if (isGrain) {
-        // Do sanity checking of grain parameters versus buffer size.
-        double bufferDuration = buffer()->duration();
+    // Do sanity checking of grain parameters versus buffer size.
+    double bufferDuration = buffer()->duration();
 
-        grainOffset = max(0.0, grainOffset);
-        grainOffset = min(bufferDuration, grainOffset);
-        m_grainOffset = grainOffset;
+    grainOffset = max(0.0, grainOffset);
+    grainOffset = min(bufferDuration, grainOffset);
+    m_grainOffset = grainOffset;
 
-        double maxDuration = bufferDuration - grainOffset;
+    double maxDuration = bufferDuration - grainOffset;
 
-        grainDuration = max(0.0, grainDuration);
-        grainDuration = min(maxDuration, grainDuration);
-        m_grainDuration = grainDuration;
-    }
+    grainDuration = max(0.0, grainDuration);
+    grainDuration = min(maxDuration, grainDuration);
+    m_grainDuration = grainDuration;
 
-    m_isGrain = isGrain;
+    m_isGrain = true;
     m_startTime = when;
 
     // 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& exceptionState)
 {
     // Handle unspecified duration where 0 means the rest of the buffer.
     if (!grainDuration && buffer())
         grainDuration = buffer()->duration();
-    startPlaying(true, when, grainOffset, grainDuration, exceptionState);
+    start(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)