| File: | root/firefox-clang/media/libsoundtouch/src/RateTransposer.cpp |
| Warning: | line 145, column 9 Value stored to 'count' is never read |
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| 1 | //////////////////////////////////////////////////////////////////////////////// |
| 2 | /// |
| 3 | /// Sample rate transposer. Changes sample rate by using linear interpolation |
| 4 | /// together with anti-alias filtering (first order interpolation with anti- |
| 5 | /// alias filtering should be quite adequate for this application) |
| 6 | /// |
| 7 | /// Author : Copyright (c) Olli Parviainen |
| 8 | /// Author e-mail : oparviai 'at' iki.fi |
| 9 | /// SoundTouch WWW: http://www.surina.net/soundtouch |
| 10 | /// |
| 11 | //////////////////////////////////////////////////////////////////////////////// |
| 12 | // |
| 13 | // License : |
| 14 | // |
| 15 | // SoundTouch audio processing library |
| 16 | // Copyright (c) Olli Parviainen |
| 17 | // |
| 18 | // This library is free software; you can redistribute it and/or |
| 19 | // modify it under the terms of the GNU Lesser General Public |
| 20 | // License as published by the Free Software Foundation; either |
| 21 | // version 2.1 of the License, or (at your option) any later version. |
| 22 | // |
| 23 | // This library is distributed in the hope that it will be useful, |
| 24 | // but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 25 | // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| 26 | // Lesser General Public License for more details. |
| 27 | // |
| 28 | // You should have received a copy of the GNU Lesser General Public |
| 29 | // License along with this library; if not, write to the Free Software |
| 30 | // Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
| 31 | // |
| 32 | //////////////////////////////////////////////////////////////////////////////// |
| 33 | |
| 34 | #include <memory.h> |
| 35 | #include <assert.h> |
| 36 | #include <stdlib.h> |
| 37 | #include <stdio.h> |
| 38 | #include "RateTransposer.h" |
| 39 | #include "InterpolateLinear.h" |
| 40 | #include "InterpolateCubic.h" |
| 41 | #include "InterpolateShannon.h" |
| 42 | #include "AAFilter.h" |
| 43 | |
| 44 | using namespace soundtouch; |
| 45 | |
| 46 | // Define default interpolation algorithm here |
| 47 | TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC; |
| 48 | |
| 49 | |
| 50 | // Constructor |
| 51 | RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer) |
| 52 | { |
| 53 | bUseAAFilter = |
| 54 | #ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER |
| 55 | true; |
| 56 | #else |
| 57 | // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover |
| 58 | false; |
| 59 | #endif |
| 60 | |
| 61 | // Instantiates the anti-alias filter |
| 62 | pAAFilter = new AAFilter(64); |
| 63 | pTransposer = TransposerBase::newInstance(); |
| 64 | clear(); |
| 65 | } |
| 66 | |
| 67 | |
| 68 | RateTransposer::~RateTransposer() |
| 69 | { |
| 70 | delete pAAFilter; |
| 71 | delete pTransposer; |
| 72 | } |
| 73 | |
| 74 | |
| 75 | /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable |
| 76 | void RateTransposer::enableAAFilter(bool newMode) |
| 77 | { |
| 78 | #ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER |
| 79 | // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover |
| 80 | bUseAAFilter = newMode; |
| 81 | clear(); |
| 82 | #endif |
| 83 | } |
| 84 | |
| 85 | |
| 86 | /// Returns nonzero if anti-alias filter is enabled. |
| 87 | bool RateTransposer::isAAFilterEnabled() const |
| 88 | { |
| 89 | return bUseAAFilter; |
| 90 | } |
| 91 | |
| 92 | |
| 93 | AAFilter *RateTransposer::getAAFilter() |
| 94 | { |
| 95 | return pAAFilter; |
| 96 | } |
| 97 | |
| 98 | |
| 99 | // Sets new target iRate. Normal iRate = 1.0, smaller values represent slower |
| 100 | // iRate, larger faster iRates. |
| 101 | void RateTransposer::setRate(double newRate) |
| 102 | { |
| 103 | double fCutoff; |
| 104 | |
| 105 | pTransposer->setRate(newRate); |
| 106 | |
| 107 | // design a new anti-alias filter |
| 108 | if (newRate > 1.0) |
| 109 | { |
| 110 | fCutoff = 0.5 / newRate; |
| 111 | } |
| 112 | else |
| 113 | { |
| 114 | fCutoff = 0.5 * newRate; |
| 115 | } |
| 116 | pAAFilter->setCutoffFreq(fCutoff); |
| 117 | } |
| 118 | |
| 119 | |
| 120 | // Adds 'nSamples' pcs of samples from the 'samples' memory position into |
| 121 | // the input of the object. |
| 122 | void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples) |
| 123 | { |
| 124 | processSamples(samples, nSamples); |
| 125 | } |
| 126 | |
| 127 | |
| 128 | // Transposes sample rate by applying anti-alias filter to prevent folding. |
| 129 | // Returns amount of samples returned in the "dest" buffer. |
| 130 | // The maximum amount of samples that can be returned at a time is set by |
| 131 | // the 'set_returnBuffer_size' function. |
| 132 | void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples) |
| 133 | { |
| 134 | uint count; |
| 135 | |
| 136 | if (nSamples == 0) return; |
| 137 | |
| 138 | // Store samples to input buffer |
| 139 | inputBuffer.putSamples(src, nSamples); |
| 140 | |
| 141 | // If anti-alias filter is turned off, simply transpose without applying |
| 142 | // the filter |
| 143 | if (bUseAAFilter == false) |
| 144 | { |
| 145 | count = pTransposer->transpose(outputBuffer, inputBuffer); |
Value stored to 'count' is never read | |
| 146 | return; |
| 147 | } |
| 148 | |
| 149 | assert(pAAFilter)(static_cast <bool> (pAAFilter) ? void (0) : __assert_fail ("pAAFilter", __builtin_FILE (), __builtin_LINE (), __extension__ __PRETTY_FUNCTION__)); |
| 150 | |
| 151 | // Transpose with anti-alias filter |
| 152 | if (pTransposer->rate < 1.0f) |
| 153 | { |
| 154 | // If the parameter 'Rate' value is smaller than 1, first transpose |
| 155 | // the samples and then apply the anti-alias filter to remove aliasing. |
| 156 | |
| 157 | // Transpose the samples, store the result to end of "midBuffer" |
| 158 | pTransposer->transpose(midBuffer, inputBuffer); |
| 159 | |
| 160 | // Apply the anti-alias filter for transposed samples in midBuffer |
| 161 | pAAFilter->evaluate(outputBuffer, midBuffer); |
| 162 | } |
| 163 | else |
| 164 | { |
| 165 | // If the parameter 'Rate' value is larger than 1, first apply the |
| 166 | // anti-alias filter to remove high frequencies (prevent them from folding |
| 167 | // over the lover frequencies), then transpose. |
| 168 | |
| 169 | // Apply the anti-alias filter for samples in inputBuffer |
| 170 | pAAFilter->evaluate(midBuffer, inputBuffer); |
| 171 | |
| 172 | // Transpose the AA-filtered samples in "midBuffer" |
| 173 | pTransposer->transpose(outputBuffer, midBuffer); |
| 174 | } |
| 175 | } |
| 176 | |
| 177 | |
| 178 | // Sets the number of channels, 1 = mono, 2 = stereo |
| 179 | void RateTransposer::setChannels(int nChannels) |
| 180 | { |
| 181 | if (!verifyNumberOfChannels(nChannels) || |
| 182 | (pTransposer->numChannels == nChannels)) return; |
| 183 | |
| 184 | pTransposer->setChannels(nChannels); |
| 185 | inputBuffer.setChannels(nChannels); |
| 186 | midBuffer.setChannels(nChannels); |
| 187 | outputBuffer.setChannels(nChannels); |
| 188 | } |
| 189 | |
| 190 | |
| 191 | // Clears all the samples in the object |
| 192 | void RateTransposer::clear() |
| 193 | { |
| 194 | outputBuffer.clear(); |
| 195 | midBuffer.clear(); |
| 196 | inputBuffer.clear(); |
| 197 | pTransposer->resetRegisters(); |
| 198 | |
| 199 | // prefill buffer to avoid losing first samples at beginning of stream |
| 200 | int prefill = getLatency(); |
| 201 | inputBuffer.addSilent(prefill); |
| 202 | } |
| 203 | |
| 204 | |
| 205 | // Returns nonzero if there aren't any samples available for outputting. |
| 206 | int RateTransposer::isEmpty() const |
| 207 | { |
| 208 | int res; |
| 209 | |
| 210 | res = FIFOProcessor::isEmpty(); |
| 211 | if (res == 0) return 0; |
| 212 | return inputBuffer.isEmpty(); |
| 213 | } |
| 214 | |
| 215 | |
| 216 | /// Return approximate initial input-output latency |
| 217 | int RateTransposer::getLatency() const |
| 218 | { |
| 219 | return pTransposer->getLatency() + |
| 220 | ((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0); |
| 221 | } |
| 222 | |
| 223 | |
| 224 | ////////////////////////////////////////////////////////////////////////////// |
| 225 | // |
| 226 | // TransposerBase - Base class for interpolation |
| 227 | // |
| 228 | |
| 229 | // static function to set interpolation algorithm |
| 230 | void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a) |
| 231 | { |
| 232 | TransposerBase::algorithm = a; |
| 233 | } |
| 234 | |
| 235 | |
| 236 | // Transposes the sample rate of the given samples using linear interpolation. |
| 237 | // Returns the number of samples returned in the "dest" buffer |
| 238 | int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) |
| 239 | { |
| 240 | int numSrcSamples = src.numSamples(); |
| 241 | int sizeDemand = (int)((double)numSrcSamples / rate) + 8; |
| 242 | int numOutput; |
| 243 | SAMPLETYPE *psrc = src.ptrBegin(); |
| 244 | SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand); |
| 245 | |
| 246 | #ifndef USE_MULTICH_ALWAYS |
| 247 | if (numChannels == 1) |
| 248 | { |
| 249 | numOutput = transposeMono(pdest, psrc, numSrcSamples); |
| 250 | } |
| 251 | else if (numChannels == 2) |
| 252 | { |
| 253 | numOutput = transposeStereo(pdest, psrc, numSrcSamples); |
| 254 | } |
| 255 | else |
| 256 | #endif // USE_MULTICH_ALWAYS |
| 257 | { |
| 258 | assert(numChannels > 0)(static_cast <bool> (numChannels > 0) ? void (0) : __assert_fail ("numChannels > 0", __builtin_FILE (), __builtin_LINE (), __extension__ __PRETTY_FUNCTION__)); |
| 259 | numOutput = transposeMulti(pdest, psrc, numSrcSamples); |
| 260 | } |
| 261 | dest.putSamples(numOutput); |
| 262 | src.receiveSamples(numSrcSamples); |
| 263 | return numOutput; |
| 264 | } |
| 265 | |
| 266 | |
| 267 | TransposerBase::TransposerBase() |
| 268 | { |
| 269 | numChannels = 0; |
| 270 | rate = 1.0f; |
| 271 | } |
| 272 | |
| 273 | |
| 274 | TransposerBase::~TransposerBase() |
| 275 | { |
| 276 | } |
| 277 | |
| 278 | |
| 279 | void TransposerBase::setChannels(int channels) |
| 280 | { |
| 281 | numChannels = channels; |
| 282 | resetRegisters(); |
| 283 | } |
| 284 | |
| 285 | |
| 286 | void TransposerBase::setRate(double newRate) |
| 287 | { |
| 288 | rate = newRate; |
| 289 | } |
| 290 | |
| 291 | |
| 292 | // static factory function |
| 293 | TransposerBase *TransposerBase::newInstance() |
| 294 | { |
| 295 | #ifdef SOUNDTOUCH_INTEGER_SAMPLES |
| 296 | // Notice: For integer arithmetic support only linear algorithm (due to simplest calculus) |
| 297 | return ::new InterpolateLinearInteger; |
| 298 | #else |
| 299 | switch (algorithm) |
| 300 | { |
| 301 | case LINEAR: |
| 302 | return new InterpolateLinearFloat; |
| 303 | |
| 304 | case CUBIC: |
| 305 | return new InterpolateCubic; |
| 306 | |
| 307 | case SHANNON: |
| 308 | return new InterpolateShannon; |
| 309 | |
| 310 | default: |
| 311 | assert(false)(static_cast <bool> (false) ? void (0) : __assert_fail ( "false", __builtin_FILE (), __builtin_LINE (), __extension__ __PRETTY_FUNCTION__ )); |
| 312 | return NULL__null; |
| 313 | } |
| 314 | #endif |
| 315 | } |