AudioBufferSourceNode loop duration should be the actual duration

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 386 matching lines @@
 }
 
 void AudioBufferSourceNode::clampGrainParameters(const AudioBuffer* buffer)
 {
     ASSERT(buffer);
 
     // We have a buffer so we can clip the offset and duration to lie within the buffer.
     double bufferDuration = buffer->duration();
 
     m_grainOffset = clampTo(m_grainOffset, 0.0, bufferDuration);
-    m_grainDuration = clampTo(m_grainDuration, 0.0, bufferDuration - m_grainOffset);
+
+    if (loop()) {
+        // We're looping a grain with a grain duration specified. Schedule the loop to stop after
+        // grainDuration seconds after starting, possibly running the loop multiple times if
+        // grainDuration is larger than the buffer duration. The net effect is as if the user called
+        // stop(when + grainDuration).
+        m_grainDuration = clampTo(m_grainDuration, 0.0, std::numeric_limits<double>::infinity());
+        m_endTime = m_startTime + m_grainDuration;
+    } else {
+        m_grainDuration = clampTo(m_grainDuration, 0.0, bufferDuration - m_grainOffset);
+    }
 
     // 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());
 }
 
 void AudioBufferSourceNode::start(double when, ExceptionState& exceptionState)
 {
@@ skipping 34 matching lines @@
         exceptionState.throwDOMException(
             InvalidStateError,
             "Duration must be a finite non-negative number: " + String::number(grainDuration));
         return;
     }
 
     m_isGrain = true;
     m_grainOffset = grainOffset;
     m_grainDuration = grainDuration;
 
+    m_startTime = when;
+
     if (buffer())
         clampGrainParameters(buffer());
 
-    m_startTime = when;
     m_playbackState = SCHEDULED_STATE;
 }
 
 double AudioBufferSourceNode::totalPitchRate()
 {
     double dopplerRate = 1.0;
     if (m_pannerNode)
         dopplerRate = m_pannerNode->dopplerRate();
 
     // Incorporate buffer's sample-rate versus AudioContext's sample-rate.
@@ skipping 56 matching lines @@
 {
     visitor->trace(m_buffer);
     visitor->trace(m_playbackRate);
     visitor->trace(m_pannerNode);
     AudioScheduledSourceNode::trace(visitor);
 }
 
 } // namespace blink
 
 #endif // ENABLE(WEB_AUDIO)