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diff --git a/util/audio.cpp b/util/audio.cpp
index 99872f93..fa9ad256 100644
--- a/util/audio.cpp
+++ b/util/audio.cpp
@@ -1,169 +1,176 @@
#include <string.h>
#include "audio.h"
#include "debug.h"
namespace Util{
#ifdef USE_SDL1
AudioConverter::AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
Encoding outputEncoding, int outputChannels, int outputFrequency){
SDL_BuildAudioCVT(&conversion, inputEncoding, inputChannels, inputFrequency,
outputEncoding, outputChannels, outputFrequency);
}
AudioConverter::~AudioConverter(){
}
int AudioConverter::convertedLength(int length){
return length;
}
int AudioConverter::convert(void * input, int length){
if (conversion.needed){
conversion.buf = (Uint8*) input;
conversion.len = length;
/* then convert to whatever the real output wants */
SDL_ConvertAudio(&conversion);
return conversion.len_cvt;
} else {
return length;
}
}
#else
AudioConverter::AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
- Encoding outputEncoding, int outputChannels, int outputFrequency){
+ Encoding outputEncoding, int outputChannels, int outputFrequency):
+buffer(NULL),
+bufferSize(0){
input.bytes = inputEncoding;
input.channels = inputChannels;
input.frequency = inputFrequency;
output.bytes = outputEncoding;
output.channels = outputChannels;
output.frequency = outputFrequency;
sizeRatio = (double) byteSize(output) * output.frequency / ((double) byteSize(input) * input.frequency);
}
int AudioConverter::byteSize(const Format & what){
return encodingBytes(what.bytes) * what.channels;
}
/* how many bytes an encoding takes up */
int AudioConverter::encodingBytes(Encoding what){
switch (what){
case Signed16: return 2;
case Float32: return 4;
}
return 1;
}
int AudioConverter::convertedLength(int length){
return length * sizeRatio;
}
double CubicInterpolate(double y0,double y1,
double y2,double y3,
double mu){
double a0,a1,a2,a3,mu2;
mu2 = mu*mu;
a0 = y3 - y2 - y0 + y1;
a1 = y0 - y1 - a0;
a2 = y2 - y0;
a3 = y1;
return (a0*mu*mu2+a1*mu2+a2*mu+a3);
}
template <class Size>
Size clamp(double input){
Size top = (1 << (sizeof(Size) * 8 - 1)) - 1;
Size bottom = -(1 << (sizeof(Size) * 8 - 1));
if (input > top){
return top;
}
if (input < bottom){
return bottom;
}
return input;
}
template <>
float clamp(double input){
return input;
}
template <class Size>
void doConvertRate(Size * input, Size * buffer, int length, int outputLength, double ratio, int channels){
for (int sample = 0; sample < outputLength; sample += 1){
double inputSample = sample / ratio;
for (int channel = 0; channel < channels; channel += 1){
int sample0 = ((int) inputSample - 1) * channels + channel;
int sample1 = ((int) inputSample + 0) * channels + channel;
int sample2 = ((int) inputSample + 1) * channels + channel;
int sample3 = ((int) inputSample + 2) * channels + channel;
if (sample0 < 0){
sample0 = sample1;
}
if (sample2 >= length * channels){
sample2 = sample1;
}
if (sample3 >= length * channels){
sample3 = sample2;
}
buffer[sample * channels + channel] = clamp<Size>(CubicInterpolate(input[sample0], input[sample1], input[sample2], input[sample3], inputSample - (int) inputSample));
// Global::debug(0) << "Input[" << sample << "] " << channel << ": " << input[sample1] << " Output: " << buffer[sample * 2 + channel] << std::endl;
}
}
}
int AudioConverter::convert(void * input, int length){
/* no conversion needed */
if (this->input == this->output){
return length;
}
int total = convertedLength(length);
/* make sure we get an even number of samples */
if (total % byteSize(output) != 0){
total -= total % byteSize(output);
}
- char * buffer = new char[total];
+ /* cache the buffer for future use */
+ if (total > bufferSize){
+ delete[] buffer;
+ bufferSize = total;
+ buffer = new char[bufferSize];
+ }
if (this->input.channels == output.channels &&
this->input.bytes == output.bytes){
switch (this->input.bytes){
case Signed16: doConvertRate<signed short>((signed short*) input, (signed short*) buffer, length / sizeof(signed short) / this->input.channels, total / sizeof(signed short) / output.channels, sizeRatio, output.channels); break;
case Float32: doConvertRate<float>((float*) input, (float*) buffer, length / sizeof(float) / this->input.channels, total / sizeof(float) / output.channels, sizeRatio, output.channels);
}
}
memcpy(input, buffer, total);
- delete[] buffer;
return total;
}
AudioConverter::~AudioConverter(){
+ delete[] buffer;
}
bool AudioConverter::Format::operator==(const AudioConverter::Format & him) const {
return this->bytes == him.bytes &&
this->channels == him.channels &&
this->frequency == him.frequency;
}
#endif
}
diff --git a/util/audio.h b/util/audio.h
index 02aaa935..585adcb8 100644
--- a/util/audio.h
+++ b/util/audio.h
@@ -1,61 +1,63 @@
#ifndef _paintown_audio_h
#define _paintown_audio_h
#ifdef USE_SDL
#include <SDL.h>
#endif
/* Deals with audio conversion between any source format and any destination format.
* A format consists of
* byte encoding (8/16/32 bit, signed/unsigned, floating point/integer)
* number of channels (mono/stereo)
* frequency (22050hz, 44100hz, arbitrary hz)
*/
namespace Util{
/* endianness is always native */
enum Encoding{
Signed16,
Float32
};
class AudioConverter{
public:
AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
Encoding outputEncoding, int outputChannels, int outputFrequency);
/* given some input length, return how long the converted output will be */
int convertedLength(int length);
/* convert the audio, put the output in the same buffer passed in -- 'input'
* and returns the number of converted samples.
* 'length' is the number of input samples in *bytes*
* 'input' should be large enough to hold convertedLength(length) samples
*/
int convert(void * input, int length);
virtual ~AudioConverter();
protected:
#ifdef USE_SDL
SDL_AudioCVT conversion;
#endif
struct Format{
Encoding bytes;
int channels;
int frequency;
bool operator==(const Format & him) const;
};
int byteSize(const Format & what);
int encodingBytes(Encoding what);
Format input, output;
double sizeRatio;
+ char * buffer;
+ int bufferSize;
};
}
#endif

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