OpenShot Library | OpenShotAudio 0.2.2
juce_ResamplingAudioSource.cpp
1/*
2 ==============================================================================
3
4 This file is part of the JUCE library.
5 Copyright (c) 2017 - ROLI Ltd.
6
7 JUCE is an open source library subject to commercial or open-source
8 licensing.
9
10 The code included in this file is provided under the terms of the ISC license
11 http://www.isc.org/downloads/software-support-policy/isc-license. Permission
12 To use, copy, modify, and/or distribute this software for any purpose with or
13 without fee is hereby granted provided that the above copyright notice and
14 this permission notice appear in all copies.
15
16 JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
17 EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
18 DISCLAIMED.
19
20 ==============================================================================
21*/
22
23namespace juce
24{
25
27 const bool deleteInputWhenDeleted,
28 const int channels)
29 : input (inputSource, deleteInputWhenDeleted),
30 numChannels (channels)
31{
32 jassert (input != nullptr);
33 zeromem (coefficients, sizeof (coefficients));
34}
35
37
38void ResamplingAudioSource::setResamplingRatio (const double samplesInPerOutputSample)
39{
40 jassert (samplesInPerOutputSample > 0);
41
42 const SpinLock::ScopedLockType sl (ratioLock);
43 ratio = jmax (0.0, samplesInPerOutputSample);
44}
45
46void ResamplingAudioSource::prepareToPlay (int samplesPerBlockExpected, double sampleRate)
47{
48 const SpinLock::ScopedLockType sl (ratioLock);
49
50 auto scaledBlockSize = roundToInt (samplesPerBlockExpected * ratio);
51 input->prepareToPlay (scaledBlockSize, sampleRate * ratio);
52
53 buffer.setSize (numChannels, scaledBlockSize + 32);
54
55 filterStates.calloc (numChannels);
56 srcBuffers.calloc (numChannels);
57 destBuffers.calloc (numChannels);
58 createLowPass (ratio);
59
61}
62
64{
65 const ScopedLock sl (callbackLock);
66
67 buffer.clear();
68 bufferPos = 0;
69 sampsInBuffer = 0;
70 subSampleOffset = 0.0;
71 resetFilters();
72}
73
75{
76 input->releaseResources();
77 buffer.setSize (numChannels, 0);
78}
79
81{
82 const ScopedLock sl (callbackLock);
83
84 double localRatio;
85
86 {
87 const SpinLock::ScopedLockType ratioSl (ratioLock);
88 localRatio = ratio;
89 }
90
91 if (lastRatio != localRatio)
92 {
93 createLowPass (localRatio);
94 lastRatio = localRatio;
95 }
96
97 const int sampsNeeded = roundToInt (info.numSamples * localRatio) + 3;
98
99 int bufferSize = buffer.getNumSamples();
100
101 if (bufferSize < sampsNeeded + 8)
102 {
103 bufferPos %= bufferSize;
104 bufferSize = sampsNeeded + 32;
105 buffer.setSize (buffer.getNumChannels(), bufferSize, true, true);
106 }
107
108 bufferPos %= bufferSize;
109
110 int endOfBufferPos = bufferPos + sampsInBuffer;
111 const int channelsToProcess = jmin (numChannels, info.buffer->getNumChannels());
112
113 while (sampsNeeded > sampsInBuffer)
114 {
115 endOfBufferPos %= bufferSize;
116
117 int numToDo = jmin (sampsNeeded - sampsInBuffer,
118 bufferSize - endOfBufferPos);
119
120 AudioSourceChannelInfo readInfo (&buffer, endOfBufferPos, numToDo);
121 input->getNextAudioBlock (readInfo);
122
123 if (localRatio > 1.0001)
124 {
125 // for down-sampling, pre-apply the filter..
126
127 for (int i = channelsToProcess; --i >= 0;)
128 applyFilter (buffer.getWritePointer (i, endOfBufferPos), numToDo, filterStates[i]);
129 }
130
131 sampsInBuffer += numToDo;
132 endOfBufferPos += numToDo;
133 }
134
135 for (int channel = 0; channel < channelsToProcess; ++channel)
136 {
137 destBuffers[channel] = info.buffer->getWritePointer (channel, info.startSample);
138 srcBuffers[channel] = buffer.getReadPointer (channel);
139 }
140
141 int nextPos = (bufferPos + 1) % bufferSize;
142
143 for (int m = info.numSamples; --m >= 0;)
144 {
145 jassert (sampsInBuffer > 0 && nextPos != endOfBufferPos);
146
147 const float alpha = (float) subSampleOffset;
148
149 for (int channel = 0; channel < channelsToProcess; ++channel)
150 *destBuffers[channel]++ = srcBuffers[channel][bufferPos]
151 + alpha * (srcBuffers[channel][nextPos] - srcBuffers[channel][bufferPos]);
152
153 subSampleOffset += localRatio;
154
155 while (subSampleOffset >= 1.0)
156 {
157 if (++bufferPos >= bufferSize)
158 bufferPos = 0;
159
160 --sampsInBuffer;
161
162 nextPos = (bufferPos + 1) % bufferSize;
163 subSampleOffset -= 1.0;
164 }
165 }
166
167 if (localRatio < 0.9999)
168 {
169 // for up-sampling, apply the filter after transposing..
170 for (int i = channelsToProcess; --i >= 0;)
171 applyFilter (info.buffer->getWritePointer (i, info.startSample), info.numSamples, filterStates[i]);
172 }
173 else if (localRatio <= 1.0001 && info.numSamples > 0)
174 {
175 // if the filter's not currently being applied, keep it stoked with the last couple of samples to avoid discontinuities
176 for (int i = channelsToProcess; --i >= 0;)
177 {
178 const float* const endOfBuffer = info.buffer->getReadPointer (i, info.startSample + info.numSamples - 1);
179 FilterState& fs = filterStates[i];
180
181 if (info.numSamples > 1)
182 {
183 fs.y2 = fs.x2 = *(endOfBuffer - 1);
184 }
185 else
186 {
187 fs.y2 = fs.y1;
188 fs.x2 = fs.x1;
189 }
190
191 fs.y1 = fs.x1 = *endOfBuffer;
192 }
193 }
194
195 jassert (sampsInBuffer >= 0);
196}
197
198void ResamplingAudioSource::createLowPass (const double frequencyRatio)
199{
200 const double proportionalRate = (frequencyRatio > 1.0) ? 0.5 / frequencyRatio
201 : 0.5 * frequencyRatio;
202
203 const double n = 1.0 / std::tan (MathConstants<double>::pi * jmax (0.001, proportionalRate));
204 const double nSquared = n * n;
205 const double c1 = 1.0 / (1.0 + MathConstants<double>::sqrt2 * n + nSquared);
206
207 setFilterCoefficients (c1,
208 c1 * 2.0f,
209 c1,
210 1.0,
211 c1 * 2.0 * (1.0 - nSquared),
212 c1 * (1.0 - MathConstants<double>::sqrt2 * n + nSquared));
213}
214
215void ResamplingAudioSource::setFilterCoefficients (double c1, double c2, double c3, double c4, double c5, double c6)
216{
217 const double a = 1.0 / c4;
218
219 c1 *= a;
220 c2 *= a;
221 c3 *= a;
222 c5 *= a;
223 c6 *= a;
224
225 coefficients[0] = c1;
226 coefficients[1] = c2;
227 coefficients[2] = c3;
228 coefficients[3] = c4;
229 coefficients[4] = c5;
230 coefficients[5] = c6;
231}
232
233void ResamplingAudioSource::resetFilters()
234{
235 if (filterStates != nullptr)
236 filterStates.clear ((size_t) numChannels);
237}
238
239void ResamplingAudioSource::applyFilter (float* samples, int num, FilterState& fs)
240{
241 while (--num >= 0)
242 {
243 const double in = *samples;
244
245 double out = coefficients[0] * in
246 + coefficients[1] * fs.x1
247 + coefficients[2] * fs.x2
248 - coefficients[4] * fs.y1
249 - coefficients[5] * fs.y2;
250
251 #if JUCE_INTEL
252 if (! (out < -1.0e-8 || out > 1.0e-8))
253 out = 0;
254 #endif
255
256 fs.x2 = fs.x1;
257 fs.x1 = in;
258 fs.y2 = fs.y1;
259 fs.y1 = out;
260
261 *samples++ = (float) out;
262 }
263}
264
265} // namespace juce
void setSize(int newNumChannels, int newNumSamples, bool keepExistingContent=false, bool clearExtraSpace=false, bool avoidReallocating=false)
Changes the buffer's size or number of channels.
Type * getWritePointer(int channelNumber) noexcept
Returns a writeable pointer to one of the buffer's channels.
int getNumChannels() const noexcept
Returns the number of channels of audio data that this buffer contains.
int getNumSamples() const noexcept
Returns the number of samples allocated in each of the buffer's channels.
void clear() noexcept
Clears all the samples in all channels.
const Type * getReadPointer(int channelNumber) const noexcept
Returns a pointer to an array of read-only samples in one of the buffer's channels.
Base class for objects that can produce a continuous stream of audio.
Automatically locks and unlocks a mutex object.
void clear(SizeType numElements) noexcept
This fills the block with zeros, up to the number of elements specified.
void calloc(SizeType newNumElements, const size_t elementSize=sizeof(ElementType))
Allocates a specified amount of memory and clears it.
void releaseResources() override
Allows the source to release anything it no longer needs after playback has stopped.
void getNextAudioBlock(const AudioSourceChannelInfo &) override
Called repeatedly to fetch subsequent blocks of audio data.
void prepareToPlay(int samplesPerBlockExpected, double sampleRate) override
Tells the source to prepare for playing.
void flushBuffers()
Clears any buffers and filters that the resampler is using.
~ResamplingAudioSource() override
Destructor.
void setResamplingRatio(double samplesInPerOutputSample)
Changes the resampling ratio.
ResamplingAudioSource(AudioSource *inputSource, bool deleteInputWhenDeleted, int numChannels=2)
Creates a ResamplingAudioSource for a given input source.
Used by AudioSource::getNextAudioBlock().
int numSamples
The number of samples in the buffer which the callback is expected to fill with data.
AudioBuffer< float > * buffer
The destination buffer to fill with audio data.
int startSample
The first sample in the buffer from which the callback is expected to write data.
Commonly used mathematical constants.