Extending supported sample rate from 3.0KHz to 192.0KHz for OfflineAudioContext.

Source/platform/audio/HRTFPanner.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 27 matching lines @@
 
 namespace blink {
 
 // The value of 2 milliseconds is larger than the largest delay which exists in any HRTFKernel from the default HRTFDatabase (0.0136 seconds).
 // We ASSERT the delay values used in process() with this value.
 const double MaxDelayTimeSeconds = 0.002;
 
 const int UninitializedAzimuth = -1;
 const unsigned RenderingQuantum = 128;
 
+const size_t fftSizes[] = { 64, 128, 256, 512, 1024, 2048, 4096 };
+
 HRTFPanner::HRTFPanner(float sampleRate, HRTFDatabaseLoader* databaseLoader)
     : Panner(PanningModelHRTF)
     , m_databaseLoader(databaseLoader)
     , m_sampleRate(sampleRate)
     , m_crossfadeSelection(CrossfadeSelection1)
     , m_azimuthIndex1(UninitializedAzimuth)
     , m_elevation1(0)
     , m_azimuthIndex2(UninitializedAzimuth)
     , m_elevation2(0)
     , m_crossfadeX(0)
@@ skipping 14 matching lines @@
 
 HRTFPanner::~HRTFPanner()
 {
 }
 
 size_t HRTFPanner::fftSizeForSampleRate(float sampleRate)
 {
     // The HRTF impulse responses (loaded as audio resources) are 512 sample-frames @44.1KHz.
     // Currently, we truncate the impulse responses to half this size, but an FFT-size of twice impulse response size is needed (for convolution).
     // So for sample rates around 44.1KHz an FFT size of 512 is good. We double the FFT-size only for sample rates at least double this.
-    ASSERT(sampleRate >= 44100 && sampleRate <= 96000.0);
-    return (sampleRate < 88200.0) ? 512 : 1024;
+    ASSERT(sampleRate >= 3000 && sampleRate <= 192000);

[Raymond Toy, 2014/09/05 05:18:30]

I think you should use AudioBuffer::minAllowedSampleRate() instead of 3000.

[KhNo, 2014/09/05 05:32:39]

Yes, I agree with that. However, current implementation can not call method
module method.(AudioBuffer.cpp is in module/) from plotform/ to module/.
So, now I'm preparing the next patch set of it. I will clean up all sample rate
numeric to use get one method after remove dependancy libmodule and libplatform.

+
+    int index = 6;
+
+    if (sampleRate <= 5512.5)
+        index = 0;
+    else if (sampleRate <= 11025)
+        index = 1;
+    else if (sampleRate <= 22050)
+        index = 2;
+    else if (sampleRate <= 44100)
+        index = 3;
+    else if (sampleRate <= 88200)
+        index = 4;
+    else if (sampleRate <= 176400)
+        index = 5;
+
+    return fftSizes[index];

[Raymond Toy, 2014/09/05 05:18:30]

Since this is so simple, I think you should just delete the fftSizes array and
instead of setting the index, just return the actual FFT size.

[KhNo, 2014/09/05 05:32:39]

Done.

 }
 
 void HRTFPanner::reset()
 {
     m_convolverL1.reset();
     m_convolverR1.reset();
     m_convolverL2.reset();
     m_convolverR2.reset();
     m_delayLineL.reset();
     m_delayLineR.reset();
@@ skipping 210 matching lines @@
 double HRTFPanner::latencyTime() const
 {
     // The latency of a FFTConvolver is also fftSize() / 2, and is in addition to its tailTime of the
     // same value.
     return (fftSize() / 2) / static_cast<double>(sampleRate());
 }
 
 } // namespace blink
 
 #endif // ENABLE(WEB_AUDIO)