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 WebCore {
 
 // 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 float coefficientSampleRate = 11025.f;

[Raymond Toy, 2014/07/30 16:49:38]

Where does 11025 come from? I thought the HRTF database was sampled at 44100.

[KhNo, 2014/07/31 16:45:54]

This value is just determined to calculate fftsize actually not related with
HRTF database's samplerate.
It means minimum unit to divide.
floor(sampleRate / coefficientSampleRate) * minimumFFTSize.

Actually, there might be better word and name. Please recommend if you have. 

[Raymond Toy, 2014/07/31 17:25:13]

Might be better to split this into two parts. One constant to give the  sample
rate of the data in the database (to make it explicit).  Then another constant
to produce the correct scaling.

+const size_t minimumFFTSize = 128;
+const size_t maximumFFTSize = 1024;
+
 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 >= 8000 && sampleRate <= 96000.0);
+
+    size_t fftSize = minimumFFTSize;

[Raymond Toy, 2014/07/30 16:49:38]

This initial value seems useless since you give fftSize a new value in the next
line.

[KhNo, 2014/07/31 16:45:54]

Done.

+    fftSize = floor(sampleRate / coefficientSampleRate) * minimumFFTSize;
+    fftSize = max(minimumFFTSize, fftSize);
+    fftSize = min(maximumFFTSize, fftSize);
+    return fftSize;

[Raymond Toy, 2014/07/30 16:49:38]

If we expand the sample rate to 192 kHz, will this still work?

[KhNo, 2014/07/31 16:45:54]

Yes, it can be. kMaxFFTPow2Size is 15 for Android, for others are 24.

[Raymond Toy, 2014/07/31 17:25:13]

One other thing. Don't we need to make fftSize a power of two here? The
sampleRate could be anything in range.

 }
 
 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 WebCore
 
 #endif // ENABLE(WEB_AUDIO)