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Pre-process audio in the software mixer

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This should improve performance slightly
This commit is contained in:
Clownacy 2020-04-06 19:52:53 +01:00
parent 5daea02ac6
commit 6456649e11
1 changed files with 7 additions and 6 deletions
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13
src/Backends/Audio/SoftwareMixer.cpp
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@ -14,7 +14,7 @@
struct Mixer_Sound struct Mixer_Sound
{ {
unsigned char *samples; signed char *samples;
size_t frames; size_t frames;
double position; double position;
double advance_delta; double advance_delta;
@ -53,7 +53,7 @@ Mixer_Sound* Mixer_CreateSound(unsigned int frequency, const unsigned char *samp
if (sound == NULL) if (sound == NULL)
return NULL; return NULL;
sound->samples = (unsigned char*)malloc(length + 1); sound->samples = (signed char*)malloc(length + 1);
if (sound->samples == NULL) if (sound->samples == NULL)
{ {
@ -61,7 +61,8 @@ Mixer_Sound* Mixer_CreateSound(unsigned int frequency, const unsigned char *samp
return NULL; 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->frames = length;
sound->playing = false; sound->playing = false;
@ -96,7 +97,7 @@ void Mixer_PlaySound(Mixer_Sound *sound, bool looping)
sound->playing = true; sound->playing = true;
sound->looping = looping; 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) void Mixer_StopSound(Mixer_Sound *sound)
@ -148,8 +149,8 @@ ATTRIBUTE_HOT void Mixer_MixSounds(float *stream, unsigned int frames_total)
const float position_fractional = sound->position - position_integral; const float position_fractional = sound->position - position_integral;
// Get two samples, and normalise them to 0-1 // Get two samples, and normalise them to 0-1
const float sample1 = (sound->samples[position_integral] - 128.0f) / 128.0f; const float sample1 = sound->samples[position_integral] / 128.0f;
const float sample2 = (sound->samples[position_integral + 1] - 128.0f) / 128.0f; const float sample2 = sound->samples[position_integral + 1] / 128.0f;
// Perform linear interpolation // Perform linear interpolation
const float interpolated_sample = sample1 + (sample2 - sample1) * position_fractional; const float interpolated_sample = sample1 + (sample2 - sample1) * position_fractional;