| 1 | /* |
| 2 | * Copyright (C) 2011 Google Inc. All rights reserved. |
| 3 | * |
| 4 | * Redistribution and use in source and binary forms, with or without |
| 5 | * modification, are permitted provided that the following conditions |
| 6 | * are met: |
| 7 | * |
| 8 | * 1. Redistributions of source code must retain the above copyright |
| 9 | * notice, this list of conditions and the following disclaimer. |
| 10 | * 2. Redistributions in binary form must reproduce the above copyright |
| 11 | * notice, this list of conditions and the following disclaimer in the |
| 12 | * documentation and/or other materials provided with the distribution. |
| 13 | * 3. Neither the name of Apple Inc. ("Apple") nor the names of |
| 14 | * its contributors may be used to endorse or promote products derived |
| 15 | * from this software without specific prior written permission. |
| 16 | * |
| 17 | * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY |
| 18 | * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
| 19 | * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
| 20 | * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
| 21 | * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
| 22 | * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
| 23 | * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND |
| 24 | * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
| 25 | * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF |
| 26 | * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| 27 | */ |
| 28 | |
| 29 | #include "config.h" |
| 30 | |
| 31 | #if ENABLE(WEB_AUDIO) |
| 32 | |
| 33 | #include "DynamicsCompressor.h" |
| 34 | |
| 35 | #include "AudioBus.h" |
| 36 | #include "AudioUtilities.h" |
| 37 | #include <wtf/MathExtras.h> |
| 38 | #include <wtf/StdLibExtras.h> |
| 39 | |
| 40 | namespace WebCore { |
| 41 | |
| 42 | using namespace AudioUtilities; |
| 43 | |
| 44 | DynamicsCompressor::DynamicsCompressor(float sampleRate, unsigned numberOfChannels) |
| 45 | : m_numberOfChannels(numberOfChannels) |
| 46 | , m_sampleRate(sampleRate) |
| 47 | , m_compressor(sampleRate, numberOfChannels) |
| 48 | { |
| 49 | // Uninitialized state - for parameter recalculation. |
| 50 | m_lastFilterStageRatio = -1; |
| 51 | m_lastAnchor = -1; |
| 52 | m_lastFilterStageGain = -1; |
| 53 | |
| 54 | setNumberOfChannels(numberOfChannels); |
| 55 | initializeParameters(); |
| 56 | } |
| 57 | |
| 58 | void DynamicsCompressor::setParameterValue(unsigned parameterID, float value) |
| 59 | { |
| 60 | ASSERT(parameterID < ParamLast); |
| 61 | if (parameterID < ParamLast) |
| 62 | m_parameters[parameterID] = value; |
| 63 | } |
| 64 | |
| 65 | void DynamicsCompressor::initializeParameters() |
| 66 | { |
| 67 | // Initializes compressor to default values. |
| 68 | |
| 69 | m_parameters[ParamThreshold] = -24; // dB |
| 70 | m_parameters[ParamKnee] = 30; // dB |
| 71 | m_parameters[ParamRatio] = 12; // unit-less |
| 72 | m_parameters[ParamAttack] = 0.003f; // seconds |
| 73 | m_parameters[ParamRelease] = 0.250f; // seconds |
| 74 | m_parameters[ParamPreDelay] = 0.006f; // seconds |
| 75 | |
| 76 | // Release zone values 0 -> 1. |
| 77 | m_parameters[ParamReleaseZone1] = 0.09f; |
| 78 | m_parameters[ParamReleaseZone2] = 0.16f; |
| 79 | m_parameters[ParamReleaseZone3] = 0.42f; |
| 80 | m_parameters[ParamReleaseZone4] = 0.98f; |
| 81 | |
| 82 | m_parameters[ParamFilterStageGain] = 4.4f; // dB |
| 83 | m_parameters[ParamFilterStageRatio] = 2; |
| 84 | m_parameters[ParamFilterAnchor] = 15000 / nyquist(); |
| 85 | |
| 86 | m_parameters[ParamPostGain] = 0; // dB |
| 87 | m_parameters[ParamReduction] = 0; // dB |
| 88 | |
| 89 | // Linear crossfade (0 -> 1). |
| 90 | m_parameters[ParamEffectBlend] = 1; |
| 91 | } |
| 92 | |
| 93 | float DynamicsCompressor::parameterValue(unsigned parameterID) |
| 94 | { |
| 95 | ASSERT(parameterID < ParamLast); |
| 96 | return m_parameters[parameterID]; |
| 97 | } |
| 98 | |
| 99 | void DynamicsCompressor::setEmphasisStageParameters(unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */) |
| 100 | { |
| 101 | float gk = 1 - gain / 20; |
| 102 | float f1 = normalizedFrequency * gk; |
| 103 | float f2 = normalizedFrequency / gk; |
| 104 | float r1 = expf(-f1 * piFloat); |
| 105 | float r2 = expf(-f2 * piFloat); |
| 106 | |
| 107 | ASSERT(m_numberOfChannels == m_preFilterPacks.size()); |
| 108 | |
| 109 | for (unsigned i = 0; i < m_numberOfChannels; ++i) { |
| 110 | // Set pre-filter zero and pole to create an emphasis filter. |
| 111 | ZeroPole& preFilter = m_preFilterPacks[i]->filters[stageIndex]; |
| 112 | preFilter.setZero(r1); |
| 113 | preFilter.setPole(r2); |
| 114 | |
| 115 | // Set post-filter with zero and pole reversed to create the de-emphasis filter. |
| 116 | // If there were no compressor kernel in between, they would cancel each other out (allpass filter). |
| 117 | ZeroPole& postFilter = m_postFilterPacks[i]->filters[stageIndex]; |
| 118 | postFilter.setZero(r2); |
| 119 | postFilter.setPole(r1); |
| 120 | } |
| 121 | } |
| 122 | |
| 123 | void DynamicsCompressor::setEmphasisParameters(float gain, float anchorFreq, float filterStageRatio) |
| 124 | { |
| 125 | setEmphasisStageParameters(0, gain, anchorFreq); |
| 126 | setEmphasisStageParameters(1, gain, anchorFreq / filterStageRatio); |
| 127 | setEmphasisStageParameters(2, gain, anchorFreq / (filterStageRatio * filterStageRatio)); |
| 128 | setEmphasisStageParameters(3, gain, anchorFreq / (filterStageRatio * filterStageRatio * filterStageRatio)); |
| 129 | } |
| 130 | |
| 131 | void DynamicsCompressor::process(const AudioBus* sourceBus, AudioBus* destinationBus, unsigned framesToProcess) |
| 132 | { |
| 133 | // Though numberOfChannels is retrived from destinationBus, we still name it numberOfChannels instead of numberOfDestinationChannels. |
| 134 | // It's because we internally match sourceChannels's size to destinationBus by channel up/down mix. Thus we need numberOfChannels |
| 135 | // to do the loop work for both m_sourceChannels and m_destinationChannels. |
| 136 | |
| 137 | unsigned numberOfChannels = destinationBus->numberOfChannels(); |
| 138 | unsigned numberOfSourceChannels = sourceBus->numberOfChannels(); |
| 139 | |
| 140 | ASSERT(numberOfChannels == m_numberOfChannels && numberOfSourceChannels); |
| 141 | |
| 142 | if (numberOfChannels != m_numberOfChannels || !numberOfSourceChannels) { |
| 143 | destinationBus->zero(); |
| 144 | return; |
| 145 | } |
| 146 | |
| 147 | switch (numberOfChannels) { |
| 148 | case 2: // stereo |
| 149 | m_sourceChannels[0] = sourceBus->channel(0)->data(); |
| 150 | |
| 151 | if (numberOfSourceChannels > 1) |
| 152 | m_sourceChannels[1] = sourceBus->channel(1)->data(); |
| 153 | else |
| 154 | // Simply duplicate mono channel input data to right channel for stereo processing. |
| 155 | m_sourceChannels[1] = m_sourceChannels[0]; |
| 156 | |
| 157 | break; |
| 158 | default: |
| 159 | // FIXME : support other number of channels. |
| 160 | ASSERT_NOT_REACHED(); |
| 161 | destinationBus->zero(); |
| 162 | return; |
| 163 | } |
| 164 | |
| 165 | for (unsigned i = 0; i < numberOfChannels; ++i) |
| 166 | m_destinationChannels[i] = destinationBus->channel(i)->mutableData(); |
| 167 | |
| 168 | float filterStageGain = parameterValue(ParamFilterStageGain); |
| 169 | float filterStageRatio = parameterValue(ParamFilterStageRatio); |
| 170 | float anchor = parameterValue(ParamFilterAnchor); |
| 171 | |
| 172 | if (filterStageGain != m_lastFilterStageGain || filterStageRatio != m_lastFilterStageRatio || anchor != m_lastAnchor) { |
| 173 | m_lastFilterStageGain = filterStageGain; |
| 174 | m_lastFilterStageRatio = filterStageRatio; |
| 175 | m_lastAnchor = anchor; |
| 176 | |
| 177 | setEmphasisParameters(filterStageGain, anchor, filterStageRatio); |
| 178 | } |
| 179 | |
| 180 | // Apply pre-emphasis filter. |
| 181 | // Note that the final three stages are computed in-place in the destination buffer. |
| 182 | for (unsigned i = 0; i < numberOfChannels; ++i) { |
| 183 | const float* sourceData = m_sourceChannels[i]; |
| 184 | float* destinationData = m_destinationChannels[i]; |
| 185 | ZeroPole* preFilters = m_preFilterPacks[i]->filters; |
| 186 | |
| 187 | preFilters[0].process(sourceData, destinationData, framesToProcess); |
| 188 | preFilters[1].process(destinationData, destinationData, framesToProcess); |
| 189 | preFilters[2].process(destinationData, destinationData, framesToProcess); |
| 190 | preFilters[3].process(destinationData, destinationData, framesToProcess); |
| 191 | } |
| 192 | |
| 193 | float dbThreshold = parameterValue(ParamThreshold); |
| 194 | float dbKnee = parameterValue(ParamKnee); |
| 195 | float ratio = parameterValue(ParamRatio); |
| 196 | float attackTime = parameterValue(ParamAttack); |
| 197 | float releaseTime = parameterValue(ParamRelease); |
| 198 | float preDelayTime = parameterValue(ParamPreDelay); |
| 199 | |
| 200 | // This is effectively a master volume on the compressed signal (pre-blending). |
| 201 | float dbPostGain = parameterValue(ParamPostGain); |
| 202 | |
| 203 | // Linear blending value from dry to completely processed (0 -> 1) |
| 204 | // 0 means the signal is completely unprocessed. |
| 205 | // 1 mixes in only the compressed signal. |
| 206 | float effectBlend = parameterValue(ParamEffectBlend); |
| 207 | |
| 208 | float releaseZone1 = parameterValue(ParamReleaseZone1); |
| 209 | float releaseZone2 = parameterValue(ParamReleaseZone2); |
| 210 | float releaseZone3 = parameterValue(ParamReleaseZone3); |
| 211 | float releaseZone4 = parameterValue(ParamReleaseZone4); |
| 212 | |
| 213 | // Apply compression to the pre-filtered signal. |
| 214 | // The processing is performed in place. |
| 215 | m_compressor.process(m_destinationChannels.get(), |
| 216 | m_destinationChannels.get(), |
| 217 | numberOfChannels, |
| 218 | framesToProcess, |
| 219 | |
| 220 | dbThreshold, |
| 221 | dbKnee, |
| 222 | ratio, |
| 223 | attackTime, |
| 224 | releaseTime, |
| 225 | preDelayTime, |
| 226 | dbPostGain, |
| 227 | effectBlend, |
| 228 | |
| 229 | releaseZone1, |
| 230 | releaseZone2, |
| 231 | releaseZone3, |
| 232 | releaseZone4 |
| 233 | ); |
| 234 | |
| 235 | // Update the compression amount. |
| 236 | setParameterValue(ParamReduction, m_compressor.meteringGain()); |
| 237 | |
| 238 | // Apply de-emphasis filter. |
| 239 | for (unsigned i = 0; i < numberOfChannels; ++i) { |
| 240 | float* destinationData = m_destinationChannels[i]; |
| 241 | ZeroPole* postFilters = m_postFilterPacks[i]->filters; |
| 242 | |
| 243 | postFilters[0].process(destinationData, destinationData, framesToProcess); |
| 244 | postFilters[1].process(destinationData, destinationData, framesToProcess); |
| 245 | postFilters[2].process(destinationData, destinationData, framesToProcess); |
| 246 | postFilters[3].process(destinationData, destinationData, framesToProcess); |
| 247 | } |
| 248 | } |
| 249 | |
| 250 | void DynamicsCompressor::reset() |
| 251 | { |
| 252 | m_lastFilterStageRatio = -1; // for recalc |
| 253 | m_lastAnchor = -1; |
| 254 | m_lastFilterStageGain = -1; |
| 255 | |
| 256 | for (unsigned channel = 0; channel < m_numberOfChannels; ++channel) { |
| 257 | for (unsigned stageIndex = 0; stageIndex < 4; ++stageIndex) { |
| 258 | m_preFilterPacks[channel]->filters[stageIndex].reset(); |
| 259 | m_postFilterPacks[channel]->filters[stageIndex].reset(); |
| 260 | } |
| 261 | } |
| 262 | |
| 263 | m_compressor.reset(); |
| 264 | } |
| 265 | |
| 266 | void DynamicsCompressor::setNumberOfChannels(unsigned numberOfChannels) |
| 267 | { |
| 268 | if (m_preFilterPacks.size() == numberOfChannels) |
| 269 | return; |
| 270 | |
| 271 | m_preFilterPacks.clear(); |
| 272 | m_postFilterPacks.clear(); |
| 273 | for (unsigned i = 0; i < numberOfChannels; ++i) { |
| 274 | m_preFilterPacks.append(std::make_unique<ZeroPoleFilterPack4>()); |
| 275 | m_postFilterPacks.append(std::make_unique<ZeroPoleFilterPack4>()); |
| 276 | } |
| 277 | |
| 278 | m_sourceChannels = makeUniqueArray<const float*>(numberOfChannels); |
| 279 | m_destinationChannels = makeUniqueArray<float*>(numberOfChannels); |
| 280 | |
| 281 | m_compressor.setNumberOfChannels(numberOfChannels); |
| 282 | m_numberOfChannels = numberOfChannels; |
| 283 | } |
| 284 | |
| 285 | } // namespace WebCore |
| 286 | |
| 287 | #endif // ENABLE(WEB_AUDIO) |
| 288 | |