start/stop method for AudioBufferSourceNodes and OscillatorNodes can take no args.

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 363 matching lines @@
 
     m_virtualReadIndex = 0;
     m_buffer = buffer;
 }
 
 unsigned AudioBufferSourceNode::numberOfChannels()
 {
     return output(0)->numberOfChannels();
 }
 
-void AudioBufferSourceNode::startGrain(double when, double grainOffset)
+void AudioBufferSourceNode::start(double when)
 {
-    // Duration of 0 has special value, meaning calculate based on the entire buffer's duration.
-    startGrain(when, grainOffset, 0);
+    startPlaying(false, when, 0, buffer()->duration());
 }
 
-void AudioBufferSourceNode::startGrain(double when, double grainOffset, double grainDuration)
+void AudioBufferSourceNode::start(double when, double grainOffset)
+{
+    startPlaying(true, when, grainOffset, buffer()->duration());
+}
+
+void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration)
+{
+    startPlaying(true, when, grainOffset, grainDuration);
+}
+
+void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grainOffset, double grainDuration)
 {
     ASSERT(isMainThread());
 
     if (m_playbackState != UNSCHEDULED_STATE)
         return;
 
     if (!buffer())
         return;
 
-    // Do sanity checking of grain parameters versus buffer size.
-    double bufferDuration = buffer()->duration();
+    if (isGrain) {
+        // 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;
 
-    // Handle default/unspecified duration.
-    double maxDuration = bufferDuration - grainOffset;
-    if (!grainDuration)
-        grainDuration = maxDuration;
+        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 = true;
+    } else {
+        // Until crbug.com/306139 is implemented, we initialize m_grainOffset and m_grainDuration
+        // again.
+        m_grainOffset = 0.0;
+        m_grainDuration = DefaultGrainDuration;
+    }
+
+    m_isGrain = isGrain;
     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)
 {
-    startGrain(when, grainOffset, grainDuration);
+    // Handle unspecified duration where 0 means the rest of the buffer.
+    if (!grainDuration)
+        grainDuration = buffer()->duration();
+    startPlaying(true, when, grainOffset, grainDuration);
 }
 
 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)