Signal exceptions if start/stop is called out-of-order or too many times.

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 10 matching lines @@
  * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "config.h"
 
 #if ENABLE(WEB_AUDIO)
 
 #include "modules/webaudio/AudioBufferSourceNode.h"
 
+#include "bindings/v8/ExceptionMessages.h"
 #include "bindings/v8/ExceptionState.h"
+#include "core/dom/ExceptionCode.h"
 #include "core/page/PageConsole.h"
 #include "platform/audio/AudioUtilities.h"
 #include "modules/webaudio/AudioContext.h"
 #include "modules/webaudio/AudioNodeOutput.h"
 #include "platform/FloatConversion.h"
 #include "wtf/MainThread.h"
 #include "wtf/MathExtras.h"
 #include <algorithm>
 
 using namespace std;
@@ skipping 332 matching lines @@
 
     m_virtualReadIndex = 0;
     m_buffer = buffer;
 }
 
 unsigned AudioBufferSourceNode::numberOfChannels()
 {
     return output(0)->numberOfChannels();
 }
 
-void AudioBufferSourceNode::start(double when)
+void AudioBufferSourceNode::start(ExceptionState& es)
 {
-    startPlaying(false, when, 0, buffer() ? buffer()->duration() : 0);
+    startPlaying(false, 0, 0, buffer() ? buffer()->duration() : 0, es);
 }
 
-void AudioBufferSourceNode::start(double when, double grainOffset)
+void AudioBufferSourceNode::start(double when, ExceptionState& es)
 {
-    startPlaying(true, when, grainOffset, buffer() ? buffer()->duration() : 0);
+    startPlaying(false, when, 0, buffer() ? buffer()->duration() : 0, es);
 }
 
-void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration)
+void AudioBufferSourceNode::start(double when, double grainOffset, ExceptionState& es)
 {
-    startPlaying(true, when, grainOffset, grainDuration);
+    startPlaying(true, when, grainOffset, buffer() ? buffer()->duration() : 0, es);
 }
 
-void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grainOffset, double grainDuration)
+void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration, ExceptionState& es)
+{
+    startPlaying(true, when, grainOffset, grainDuration, es);
+}
+
+void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grainOffset, double grainDuration, ExceptionState& es)
 {
     ASSERT(isMainThread());
 
-    if (m_playbackState != UNSCHEDULED_STATE)
+    if (m_playbackState != UNSCHEDULED_STATE) {
+        es.throwDOMException(
+            InvalidStateError,
+            ExceptionMessages::failedToExecute(
+                "start",
+                nodeTypeName(),
+                "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();
 
         grainOffset = max(0.0, grainOffset);
         grainOffset = min(bufferDuration, grainOffset);
         m_grainOffset = grainOffset;
 
         double maxDuration = bufferDuration - grainOffset;
 
         grainDuration = max(0.0, grainDuration);
         grainDuration = min(maxDuration, grainDuration);
         m_grainDuration = grainDuration;
-
-    } 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)
+void AudioBufferSourceNode::noteGrainOn(double when, double grainOffset, double grainDuration, ExceptionState& es)
 {
     // Handle unspecified duration where 0 means the rest of the buffer.
-    if (!grainDuration)
+    if (!grainDuration && buffer())
         grainDuration = buffer()->duration();
-    startPlaying(true, when, grainOffset, grainDuration);
+    startPlaying(true, when, grainOffset, grainDuration, es);
 }
 
 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)