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authorWillem Jan Palenstijn2017-02-27 12:32:06 +0100
committerWillem Jan Palenstijn2017-02-28 18:40:04 +0100
commit867511d2d0a5d063e37b07115b8b98f5930d415b (patch)
tree2fd96da5f26b8320b8c3524a2acb20f37c59e61c /audio/softsynth/mt32/srchelper/srctools/src/FIRResampler.cpp
parente2a72393d298a5c338e55b73ae910c9ef4e1b45e (diff)
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MT32: Update Munt to 2.0.3
This update uses upstream commit 777c51cdb4dbb4e02a53c23edea9086f0b600e26. The new SampleRateConverter is added, but not built as we don't use it. Also, building it without source changes will need additional include directories. This update of Munt reduces the stack size, and thus fixes bug #9630.
Diffstat (limited to 'audio/softsynth/mt32/srchelper/srctools/src/FIRResampler.cpp')
-rw-r--r--audio/softsynth/mt32/srchelper/srctools/src/FIRResampler.cpp107
1 files changed, 107 insertions, 0 deletions
diff --git a/audio/softsynth/mt32/srchelper/srctools/src/FIRResampler.cpp b/audio/softsynth/mt32/srchelper/srctools/src/FIRResampler.cpp
new file mode 100644
index 0000000000..2cded0c3d7
--- /dev/null
+++ b/audio/softsynth/mt32/srchelper/srctools/src/FIRResampler.cpp
@@ -0,0 +1,107 @@
+/* Copyright (C) 2015-2017 Sergey V. Mikayev
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 2.1 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <cmath>
+#include <cstring>
+
+#include "FIRResampler.h"
+
+using namespace SRCTools;
+
+FIRResampler::Constants::Constants(const unsigned int upsampleFactor, const double downsampleFactor, const FIRCoefficient kernel[], const unsigned int kernelLength) {
+ usePhaseInterpolation = downsampleFactor != floor(downsampleFactor);
+ FIRCoefficient *kernelCopy = new FIRCoefficient[kernelLength];
+ memcpy(kernelCopy, kernel, kernelLength * sizeof(FIRCoefficient));
+ taps = kernelCopy;
+ numberOfTaps = kernelLength;
+ numberOfPhases = upsampleFactor;
+ phaseIncrement = downsampleFactor;
+ unsigned int minDelayLineLength = static_cast<unsigned int>(ceil(double(kernelLength) / upsampleFactor));
+ unsigned int delayLineLength = 2;
+ while (delayLineLength < minDelayLineLength) delayLineLength <<= 1;
+ delayLineMask = delayLineLength - 1;
+ ringBuffer = new FloatSample[delayLineLength][FIR_INTERPOLATOR_CHANNEL_COUNT];
+ FloatSample *s = *ringBuffer;
+ FloatSample *e = ringBuffer[delayLineLength];
+ while (s < e) *(s++) = 0;
+}
+
+FIRResampler::FIRResampler(const unsigned int upsampleFactor, const double downsampleFactor, const FIRCoefficient kernel[], const unsigned int kernelLength) :
+ constants(upsampleFactor, downsampleFactor, kernel, kernelLength),
+ ringBufferPosition(0),
+ phase(constants.numberOfPhases)
+{}
+
+FIRResampler::~FIRResampler() {
+ delete[] constants.ringBuffer;
+ delete[] constants.taps;
+}
+
+void FIRResampler::process(const FloatSample *&inSamples, unsigned int &inLength, FloatSample *&outSamples, unsigned int &outLength) {
+ while (outLength > 0) {
+ while (needNextInSample()) {
+ if (inLength == 0) return;
+ addInSamples(inSamples);
+ --inLength;
+ }
+ getOutSamplesStereo(outSamples);
+ --outLength;
+ }
+}
+
+unsigned int FIRResampler::estimateInLength(const unsigned int outLength) const {
+ return static_cast<unsigned int>((outLength * constants.phaseIncrement + phase) / constants.numberOfPhases);
+}
+
+bool FIRResampler::needNextInSample() const {
+ return constants.numberOfPhases <= phase;
+}
+
+void FIRResampler::addInSamples(const FloatSample *&inSamples) {
+ ringBufferPosition = (ringBufferPosition - 1) & constants.delayLineMask;
+ for (unsigned int i = 0; i < FIR_INTERPOLATOR_CHANNEL_COUNT; i++) {
+ constants.ringBuffer[ringBufferPosition][i] = *(inSamples++);
+ }
+ phase -= constants.numberOfPhases;
+}
+
+// Optimised for processing stereo interleaved streams
+void FIRResampler::getOutSamplesStereo(FloatSample *&outSamples) {
+ FloatSample leftSample = 0.0;
+ FloatSample rightSample = 0.0;
+ unsigned int delaySampleIx = ringBufferPosition;
+ if (constants.usePhaseInterpolation) {
+ double phaseFraction = phase - floor(phase);
+ unsigned int maxTapIx = phaseFraction == 0 ? constants.numberOfTaps : constants.numberOfTaps - 1;
+ for (unsigned int tapIx = static_cast<unsigned int>(phase); tapIx < maxTapIx; tapIx += constants.numberOfPhases) {
+ FIRCoefficient tap = FIRCoefficient(constants.taps[tapIx] + (constants.taps[tapIx + 1] - constants.taps[tapIx]) * phaseFraction);
+ leftSample += tap * constants.ringBuffer[delaySampleIx][0];
+ rightSample += tap * constants.ringBuffer[delaySampleIx][1];
+ delaySampleIx = (delaySampleIx + 1) & constants.delayLineMask;
+ }
+ } else {
+ // Optimised for rational resampling ratios when phase is always integer
+ for (unsigned int tapIx = static_cast<unsigned int>(phase); tapIx < constants.numberOfTaps; tapIx += constants.numberOfPhases) {
+ FIRCoefficient tap = constants.taps[tapIx];
+ leftSample += tap * constants.ringBuffer[delaySampleIx][0];
+ rightSample += tap * constants.ringBuffer[delaySampleIx][1];
+ delaySampleIx = (delaySampleIx + 1) & constants.delayLineMask;
+ }
+ }
+ *(outSamples++) = leftSample;
+ *(outSamples++) = rightSample;
+ phase += constants.phaseIncrement;
+}