/*
------------------------------------------------------------------
This file is part of the Open Ephys GUI
Copyright (C) 2013 Open Ephys
------------------------------------------------------------------
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 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 General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#include "AudioResamplingNode.h"
#include <stdio.h>
AudioResamplingNode::AudioResamplingNode()
: GenericProcessor("Resampling Node"),
sourceBufferSampleRate(40000.0), destBufferSampleRate(44100.0),
ratio(1.0), lastRatio(1.0), destBuffer(0), tempBuffer(0),
destBufferIsTempBuffer(true), isTransmitting(false), destBufferPos(0)
{
settings.numInputs = 2;
settings.numOutputs = 2;
setPlayConfigDetails(getNumInputs(), // number of inputs
getNumOutputs(), // number of outputs
44100.0, // sampleRate
128); // blockSize
filter = new Dsp::SmoothedFilterDesign
<Dsp::RBJ::Design::LowPass, 1> (1024);
if (destBufferIsTempBuffer)
destBufferWidth = 1024;
else
destBufferWidth = 1000;
destBufferTimebaseSecs = 1.0;
destBuffer = new AudioSampleBuffer(16, destBufferWidth);
tempBuffer = new AudioSampleBuffer(16, destBufferWidth);
continuousDataBuffer = new int16[10000];
}
AudioResamplingNode::~AudioResamplingNode()
{
filter = 0;
deleteAndZero(destBuffer);
deleteAndZero(tempBuffer);
}
void AudioResamplingNode::setParameter(int parameterIndex, float newValue)
{
editor->updateParameterButtons(parameterIndex);
switch (parameterIndex)
{
case 0:
destBufferTimebaseSecs = newValue;
break;
case 1:
destBufferWidth = roundToInt(newValue);
}
// reset to zero and clear if timebase or width has changed.
destBufferPos = 0;
destBuffer->clear();
}
void AudioResamplingNode::prepareToPlay(double sampleRate_, int estimatedSamplesPerBlock)
{
std::cout << "AudioResamplingNode preparing to play." << std::endl;
if (destBufferIsTempBuffer)
{
destBufferSampleRate = sampleRate_;
tempBuffer->setSize(getNumInputs(), 4096);
}
else
{
destBufferSampleRate = float(destBufferWidth) / destBufferTimebaseSecs;
destBuffer->setSize(getNumInputs(), destBufferWidth);
}
destBuffer->clear();
tempBuffer->clear();
destBufferPos = 0;
std::cout << "Temp buffer size: " << tempBuffer->getNumChannels() << " x "
<< tempBuffer->getNumSamples() << std::endl;
updateFilter();
}
void AudioResamplingNode::updateFilter()
{
double cutoffFreq = (ratio > 1.0) ? 2 * destBufferSampleRate // downsample
: destBufferSampleRate / 2; // upsample
double sampleFreq = (ratio > 1.0) ? sourceBufferSampleRate // downsample
: destBufferSampleRate; // upsample
Dsp::Params params;
params[0] = sampleFreq; // sample rate
params[1] = cutoffFreq; // cutoff frequency
params[2] = 1.25; //Q //
filter->setParams(params);
}
void AudioResamplingNode::releaseResources()
{
}
void AudioResamplingNode::process(AudioSampleBuffer& buffer,
MidiBuffer& midiMessages,
int& nSamples)
{
int nSamps = nSamples;
int valuesNeeded;
if (destBufferIsTempBuffer)
{
ratio = float(nSamps) / float(buffer.getNumSamples());
valuesNeeded = buffer.getNumSamples();
}
else
{
ratio = sourceBufferSampleRate / destBufferSampleRate;
valuesNeeded = (int) buffer.getNumSamples() / ratio;
//std::cout << std::endl;
//std::cout << "Ratio: " << ratio << std::endl;
//std::cout << "Values needed: " << valuesNeeded << std::endl;
}
if (lastRatio != ratio)
{
updateFilter();
lastRatio = ratio;
}
if (ratio > 1.0001)
{
// pre-apply filter before downsampling
filter->process(nSamps, buffer.getArrayOfChannels());
}
// initialize variables
tempBuffer->clear();
int sourceBufferPos = 0;
int sourceBufferSize = buffer.getNumSamples();
float subSampleOffset = 0.0;
int nextPos = (sourceBufferPos + 1) % sourceBufferSize;
int tempBufferPos;
//int totalSamples = 0;
// code modified from "juce_ResamplingAudioSource.cpp":
for (tempBufferPos = 0; tempBufferPos < valuesNeeded; tempBufferPos++)
{
float gain = 1.0;
float alpha = (float) subSampleOffset;
float invAlpha = 1.0f - alpha;
for (int channel = 0; channel < buffer.getNumChannels(); ++channel)
{
tempBuffer->addFrom(channel, // destChannel
tempBufferPos, // destSampleOffset
buffer, // source
channel, // sourceChannel
sourceBufferPos,// sourceSampleOffset
1, // number of samples
invAlpha*gain); // gain to apply to source
tempBuffer->addFrom(channel, // destChannel
tempBufferPos, // destSampleOffset
buffer, // source
channel, // sourceChannel
nextPos, // sourceSampleOffset
1, // number of samples
alpha*gain); // gain to apply to source
}
subSampleOffset += ratio;
while (subSampleOffset >= 1.0)
{
if (++sourceBufferPos >= sourceBufferSize)
sourceBufferPos = 0;
nextPos = (sourceBufferPos + 1) % sourceBufferSize;
subSampleOffset -= 1.0;
}
}
// std::cout << sourceBufferPos << " " << tempBufferPos << std::endl;
if (ratio < 0.9999)
{
filter->process(tempBufferPos, tempBuffer->getArrayOfChannels());
// apply the filter after upsampling
///////filter->process (totalSamples, buffer.getArrayOfChannels());
}
else if (ratio <= 1.0001)
{
// no resampling is being applied, no need to filter, BUT...
// keep the filter stoked with samples to avoid discontinuities
}
if (destBufferIsTempBuffer)
{
// copy the temp buffer into the original buffer
buffer = AudioSampleBuffer(tempBuffer->getArrayOfChannels(), 2, tempBufferPos);//buffer.getNumSamples());
}
else
{
//std::cout << "Copying into dest buffer..." << std::endl;
// copy the temp buffer into the destination buffer
int pos = 0;
while (*tempBuffer->getSampleData(0,pos) != 0)
pos++;
int spaceAvailable = destBufferWidth - destBufferPos;
int blockSize1 = (spaceAvailable > pos) ? pos : spaceAvailable;
int blockSize2 = (spaceAvailable > pos) ? 0 : (pos - spaceAvailable);
for (int channel = 0; channel < destBuffer->getNumChannels(); channel++)
{
// copy first block
destBuffer->copyFrom(channel, //destChannel
destBufferPos, //destStartSample
*tempBuffer, //source
channel, //sourceChannel
0, //sourceStartSample
blockSize1 //numSamples
);
// copy second block
destBuffer->copyFrom(channel, //destChannel
0, //destStartSample
*tempBuffer, //source
channel, //sourceChannel
blockSize1, //sourceStartSample
blockSize2 //numSamples
);
}
//destBufferPos = (spaceAvailable > tempBufferPos) ? destBufferPos
destBufferPos += pos;
destBufferPos %= destBufferWidth;
//std::cout << "Temp buffer position: " << tempBufferPos << std::endl;
//std::cout << "Resampling node value:" << *destBuffer->getSampleData(0,0) << std::endl;
}
}
void AudioResamplingNode::writeContinuousBuffer(float* data, int nSamples, int channel)
{
// find file and write samples to disk
timestamp = timer.getHighResolutionTicks();
AudioDataConverters::convertFloatToInt16BE(data, continuousDataBuffer, nSamples);
//int16 samps = nSamples;
fwrite(×tamp, // ptr
8, // size of each element
1, // count
file); // ptr to FILE object
fwrite(&nSamples, // ptr
sizeof(nSamples), // size of each element
1, // count
file); // ptr to FILE object
fwrite(continuousDataBuffer, // ptr
2, // size of each element
nSamples, // count
file); // ptr to FILE object
// FIXME: check that return value of each fwrite() equals "count";
// otherwise, there was an error.
}