1.1 --- /dev/null Thu Jan 01 00:00:00 1970 +0000
1.2 +++ b/media/libsoundtouch/src/RateTransposer.cpp Wed Dec 31 06:09:35 2014 +0100
1.3 @@ -0,0 +1,302 @@
1.4 +////////////////////////////////////////////////////////////////////////////////
1.5 +///
1.6 +/// Sample rate transposer. Changes sample rate by using linear interpolation
1.7 +/// together with anti-alias filtering (first order interpolation with anti-
1.8 +/// alias filtering should be quite adequate for this application)
1.9 +///
1.10 +/// Author : Copyright (c) Olli Parviainen
1.11 +/// Author e-mail : oparviai 'at' iki.fi
1.12 +/// SoundTouch WWW: http://www.surina.net/soundtouch
1.13 +///
1.14 +////////////////////////////////////////////////////////////////////////////////
1.15 +//
1.16 +// Last changed : $Date: 2014-04-06 10:57:21 -0500 (Sun, 06 Apr 2014) $
1.17 +// File revision : $Revision: 4 $
1.18 +//
1.19 +// $Id: RateTransposer.cpp 195 2014-04-06 15:57:21Z oparviai $
1.20 +//
1.21 +////////////////////////////////////////////////////////////////////////////////
1.22 +//
1.23 +// License :
1.24 +//
1.25 +// SoundTouch audio processing library
1.26 +// Copyright (c) Olli Parviainen
1.27 +//
1.28 +// This library is free software; you can redistribute it and/or
1.29 +// modify it under the terms of the GNU Lesser General Public
1.30 +// License as published by the Free Software Foundation; either
1.31 +// version 2.1 of the License, or (at your option) any later version.
1.32 +//
1.33 +// This library is distributed in the hope that it will be useful,
1.34 +// but WITHOUT ANY WARRANTY; without even the implied warranty of
1.35 +// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
1.36 +// Lesser General Public License for more details.
1.37 +//
1.38 +// You should have received a copy of the GNU Lesser General Public
1.39 +// License along with this library; if not, write to the Free Software
1.40 +// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
1.41 +//
1.42 +////////////////////////////////////////////////////////////////////////////////
1.43 +
1.44 +#include <memory.h>
1.45 +#include <assert.h>
1.46 +#include <stdlib.h>
1.47 +#include <stdio.h>
1.48 +#include "RateTransposer.h"
1.49 +#include "InterpolateLinear.h"
1.50 +#include "InterpolateCubic.h"
1.51 +#include "InterpolateShannon.h"
1.52 +#include "AAFilter.h"
1.53 +
1.54 +using namespace soundtouch;
1.55 +
1.56 +// Define default interpolation algorithm here
1.57 +TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
1.58 +
1.59 +
1.60 +// Constructor
1.61 +RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
1.62 +{
1.63 + bUseAAFilter = true;
1.64 +
1.65 + // Instantiates the anti-alias filter
1.66 + pAAFilter = new AAFilter(64);
1.67 + pTransposer = TransposerBase::newInstance();
1.68 +}
1.69 +
1.70 +
1.71 +
1.72 +RateTransposer::~RateTransposer()
1.73 +{
1.74 + delete pAAFilter;
1.75 + delete pTransposer;
1.76 +}
1.77 +
1.78 +
1.79 +
1.80 +/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
1.81 +void RateTransposer::enableAAFilter(bool newMode)
1.82 +{
1.83 + bUseAAFilter = newMode;
1.84 +}
1.85 +
1.86 +
1.87 +/// Returns nonzero if anti-alias filter is enabled.
1.88 +bool RateTransposer::isAAFilterEnabled() const
1.89 +{
1.90 + return bUseAAFilter;
1.91 +}
1.92 +
1.93 +
1.94 +AAFilter *RateTransposer::getAAFilter()
1.95 +{
1.96 + return pAAFilter;
1.97 +}
1.98 +
1.99 +
1.100 +
1.101 +// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
1.102 +// iRate, larger faster iRates.
1.103 +void RateTransposer::setRate(float newRate)
1.104 +{
1.105 + double fCutoff;
1.106 +
1.107 + pTransposer->setRate(newRate);
1.108 +
1.109 + // design a new anti-alias filter
1.110 + if (newRate > 1.0f)
1.111 + {
1.112 + fCutoff = 0.5f / newRate;
1.113 + }
1.114 + else
1.115 + {
1.116 + fCutoff = 0.5f * newRate;
1.117 + }
1.118 + pAAFilter->setCutoffFreq(fCutoff);
1.119 +}
1.120 +
1.121 +
1.122 +// Adds 'nSamples' pcs of samples from the 'samples' memory position into
1.123 +// the input of the object.
1.124 +void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
1.125 +{
1.126 + processSamples(samples, nSamples);
1.127 +}
1.128 +
1.129 +
1.130 +// Transposes sample rate by applying anti-alias filter to prevent folding.
1.131 +// Returns amount of samples returned in the "dest" buffer.
1.132 +// The maximum amount of samples that can be returned at a time is set by
1.133 +// the 'set_returnBuffer_size' function.
1.134 +void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
1.135 +{
1.136 + uint count;
1.137 +
1.138 + if (nSamples == 0) return;
1.139 +
1.140 + // Store samples to input buffer
1.141 + inputBuffer.putSamples(src, nSamples);
1.142 +
1.143 + // If anti-alias filter is turned off, simply transpose without applying
1.144 + // the filter
1.145 + if (bUseAAFilter == false)
1.146 + {
1.147 + count = pTransposer->transpose(outputBuffer, inputBuffer);
1.148 + return;
1.149 + }
1.150 +
1.151 + assert(pAAFilter);
1.152 +
1.153 + // Transpose with anti-alias filter
1.154 + if (pTransposer->rate < 1.0f)
1.155 + {
1.156 + // If the parameter 'Rate' value is smaller than 1, first transpose
1.157 + // the samples and then apply the anti-alias filter to remove aliasing.
1.158 +
1.159 + // Transpose the samples, store the result to end of "midBuffer"
1.160 + pTransposer->transpose(midBuffer, inputBuffer);
1.161 +
1.162 + // Apply the anti-alias filter for transposed samples in midBuffer
1.163 + pAAFilter->evaluate(outputBuffer, midBuffer);
1.164 + }
1.165 + else
1.166 + {
1.167 + // If the parameter 'Rate' value is larger than 1, first apply the
1.168 + // anti-alias filter to remove high frequencies (prevent them from folding
1.169 + // over the lover frequencies), then transpose.
1.170 +
1.171 + // Apply the anti-alias filter for samples in inputBuffer
1.172 + pAAFilter->evaluate(midBuffer, inputBuffer);
1.173 +
1.174 + // Transpose the AA-filtered samples in "midBuffer"
1.175 + pTransposer->transpose(outputBuffer, midBuffer);
1.176 + }
1.177 +}
1.178 +
1.179 +
1.180 +// Sets the number of channels, 1 = mono, 2 = stereo
1.181 +void RateTransposer::setChannels(int nChannels)
1.182 +{
1.183 + assert(nChannels > 0);
1.184 +
1.185 + if (pTransposer->numChannels == nChannels) return;
1.186 + pTransposer->setChannels(nChannels);
1.187 +
1.188 + inputBuffer.setChannels(nChannels);
1.189 + midBuffer.setChannels(nChannels);
1.190 + outputBuffer.setChannels(nChannels);
1.191 +}
1.192 +
1.193 +
1.194 +// Clears all the samples in the object
1.195 +void RateTransposer::clear()
1.196 +{
1.197 + outputBuffer.clear();
1.198 + midBuffer.clear();
1.199 + inputBuffer.clear();
1.200 +}
1.201 +
1.202 +
1.203 +// Returns nonzero if there aren't any samples available for outputting.
1.204 +int RateTransposer::isEmpty() const
1.205 +{
1.206 + int res;
1.207 +
1.208 + res = FIFOProcessor::isEmpty();
1.209 + if (res == 0) return 0;
1.210 + return inputBuffer.isEmpty();
1.211 +}
1.212 +
1.213 +
1.214 +//////////////////////////////////////////////////////////////////////////////
1.215 +//
1.216 +// TransposerBase - Base class for interpolation
1.217 +//
1.218 +
1.219 +// static function to set interpolation algorithm
1.220 +void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
1.221 +{
1.222 + TransposerBase::algorithm = a;
1.223 +}
1.224 +
1.225 +
1.226 +// Transposes the sample rate of the given samples using linear interpolation.
1.227 +// Returns the number of samples returned in the "dest" buffer
1.228 +int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
1.229 +{
1.230 + int numSrcSamples = src.numSamples();
1.231 + int sizeDemand = (int)((float)numSrcSamples / rate) + 8;
1.232 + int numOutput;
1.233 + SAMPLETYPE *psrc = src.ptrBegin();
1.234 + SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
1.235 +
1.236 +#ifndef USE_MULTICH_ALWAYS
1.237 + if (numChannels == 1)
1.238 + {
1.239 + numOutput = transposeMono(pdest, psrc, numSrcSamples);
1.240 + }
1.241 + else if (numChannels == 2)
1.242 + {
1.243 + numOutput = transposeStereo(pdest, psrc, numSrcSamples);
1.244 + }
1.245 + else
1.246 +#endif // USE_MULTICH_ALWAYS
1.247 + {
1.248 + assert(numChannels > 0);
1.249 + numOutput = transposeMulti(pdest, psrc, numSrcSamples);
1.250 + }
1.251 + dest.putSamples(numOutput);
1.252 + src.receiveSamples(numSrcSamples);
1.253 + return numOutput;
1.254 +}
1.255 +
1.256 +
1.257 +TransposerBase::TransposerBase()
1.258 +{
1.259 + numChannels = 0;
1.260 + rate = 1.0f;
1.261 +}
1.262 +
1.263 +
1.264 +TransposerBase::~TransposerBase()
1.265 +{
1.266 +}
1.267 +
1.268 +
1.269 +void TransposerBase::setChannels(int channels)
1.270 +{
1.271 + numChannels = channels;
1.272 + resetRegisters();
1.273 +}
1.274 +
1.275 +
1.276 +void TransposerBase::setRate(float newRate)
1.277 +{
1.278 + rate = newRate;
1.279 +}
1.280 +
1.281 +
1.282 +// static factory function
1.283 +TransposerBase *TransposerBase::newInstance()
1.284 +{
1.285 +#ifdef SOUNDTOUCH_INTEGER_SAMPLES
1.286 + // Notice: For integer arithmetics support only linear algorithm (due to simplest calculus)
1.287 + return ::new InterpolateLinearInteger;
1.288 +#else
1.289 + switch (algorithm)
1.290 + {
1.291 + case LINEAR:
1.292 + return new InterpolateLinearFloat;
1.293 +
1.294 + case CUBIC:
1.295 + return new InterpolateCubic;
1.296 +
1.297 + case SHANNON:
1.298 + return new InterpolateShannon;
1.299 +
1.300 + default:
1.301 + assert(false);
1.302 + return NULL;
1.303 + }
1.304 +#endif
1.305 +}
