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1 // Copyright 2015 The Chromium Authors. All rights reserved. | |
2 // Use of this source code is governed by a BSD-style license that can be | |
3 // found in the LICENSE file. | |
4 | |
5 #include "config.h" | |
6 | |
7 #include "modules/webaudio/IIRProcessor.h" | |
8 | |
9 #include "modules/webaudio/IIRDSPKernel.h" | |
10 | |
11 namespace blink { | |
12 | |
13 IIRProcessor::IIRProcessor(float sampleRate, size_t numberOfChannels, const Vect or<float>& feedforwardCoef, const Vector<float>& feedbackCoef) | |
14 : AudioDSPKernelProcessor(sampleRate, numberOfChannels) | |
15 { | |
16 unsigned feedbackLength = feedbackCoef.size(); | |
17 unsigned feedforwardLength = feedforwardCoef.size(); | |
18 ASSERT(feedbackLength > 0); | |
19 ASSERT(feedforwardLength > 0); | |
20 | |
21 m_feedforward.allocate(feedforwardLength); | |
22 m_feedback.allocate(feedbackLength); | |
23 m_feedforward.copyToRange(feedforwardCoef.data(), 0, feedforwardLength); | |
24 m_feedback.copyToRange(feedbackCoef.data(), 0, feedbackLength); | |
25 | |
26 // Need to scale the feedback and feedforward coefficients appropriately. (I t's up to the caller | |
27 // to ensure feedbackCoef[0] is not 0!) | |
28 ASSERT(feedbackCoef[0]); | |
29 | |
30 if (feedbackCoef[0] != 1) { | |
31 // The provided filter is: | |
32 // | |
33 // a[0]*y(n) + a[1]*y(n-1) + ... = b[0]*x(n) + b[1]*x(n-1) + ... | |
34 // | |
35 // We want the leading coefficient of y(n) to be 1: | |
36 // | |
37 // y(n) + a[1]/a[0]*y(n-1) + ... = b[0]/a[0]*x(n) + b[1]/a[0]*x(n-1) + ... | |
38 // | |
39 // Thus, the feedback and feedforward coefficients need to be scaled by 1/a[0]. | |
40 float scale = feedbackCoef[0]; | |
41 for (unsigned k = 1; k < feedbackLength; ++k) { | |
42 m_feedback[k] /= scale; | |
hongchan
2015/10/12 22:55:56
How did this line pass the presubmit check? It's o
Raymond Toy
2015/10/29 17:10:00
Done. And fixed the following loop too.
| |
43 } | |
44 | |
45 for (unsigned k = 0; k < feedforwardLength; ++k) { | |
46 m_feedforward[k] /= scale; | |
47 } | |
48 | |
49 // The IIRFilter checks to make sure this coefficient is 1, so make it s o. | |
50 m_feedback[0] = 1; | |
51 } | |
52 | |
53 m_responseKernel = adoptPtr(new IIRDSPKernel(this)); | |
54 } | |
55 | |
56 IIRProcessor::~IIRProcessor() | |
57 { | |
58 if (isInitialized()) | |
59 uninitialize(); | |
60 } | |
61 | |
62 PassOwnPtr<AudioDSPKernel> IIRProcessor::createKernel() | |
63 { | |
64 return adoptPtr(new IIRDSPKernel(this)); | |
65 } | |
66 | |
67 void IIRProcessor::process(const AudioBus* source, AudioBus* destination, size_t framesToProcess) | |
68 { | |
69 if (!isInitialized()) { | |
70 destination->zero(); | |
71 return; | |
72 } | |
73 | |
74 // For each channel of our input, process using the corresponding IIRDSPKern el into the output | |
75 // channel. | |
76 for (unsigned i = 0; i < m_kernels.size(); ++i) | |
77 m_kernels[i]->process(source->channel(i)->data(), destination->channel(i )->mutableData(), framesToProcess); | |
78 } | |
79 | |
80 void IIRProcessor::getFrequencyResponse(int nFrequencies, const float* frequency Hz, float* magResponse, float* phaseResponse) | |
81 { | |
82 m_responseKernel->getFrequencyResponse(nFrequencies, frequencyHz, magRespons e, phaseResponse); | |
83 } | |
84 | |
85 } // blink | |
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