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Pre-process audio in the software mixer

This should improve performance slightly

--- a/src/Backends/Audio/SoftwareMixer.cpp

+++ b/src/Backends/Audio/SoftwareMixer.cpp

@@ -14,7 +14,7 @@

 struct Mixer_Sound

 {

-	unsigned char *samples;

+	signed char *samples;

 	size_t frames;

 	double position;

 	double advance_delta;

@@ -53,7 +53,7 @@

 	if (sound == NULL)

 		return NULL;

-	sound->samples = (unsigned char*)malloc(length + 1);

+	sound->samples = (signed char*)malloc(length + 1);

 	if (sound->samples == NULL)

 	{

@@ -61,7 +61,8 @@

 		return NULL;

 	}

-	memcpy(sound->samples, samples, length);

+	for (size_t i = 0; i < length; ++i)

+		sound->samples[i] = samples[i] - 0x80;	// Convert from unsigned 8-bit PCM to signed

 	sound->frames = length;

 	sound->playing = false;

@@ -96,7 +97,7 @@

 	sound->playing = true;

 	sound->looping = looping;

-	sound->samples[sound->frames] = looping ? sound->samples[0] : 0x80;	// For the linear interpolator

+	sound->samples[sound->frames] = looping ? sound->samples[0] : 0;	// For the linear interpolator

 }

 void Mixer_StopSound(Mixer_Sound *sound)

@@ -148,8 +149,8 @@

 				const float position_fractional = sound->position - position_integral;

 				// Get two samples, and normalise them to 0-1

-				const float sample1 = (sound->samples[position_integral] - 128.0f) / 128.0f;

-				const float sample2 = (sound->samples[position_integral + 1] - 128.0f) / 128.0f;

+				const float sample1 = sound->samples[position_integral] / 128.0f;

+				const float sample2 = sound->samples[position_integral + 1] / 128.0f;

 				// Perform linear interpolation

 				const float interpolated_sample = sample1 + (sample2 - sample1) * position_fractional;