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Merge pull request #120036 from smix8/wav_parsing

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Improve performance of WAV parsing
This commit is contained in:
Thaddeus Crews
2026-06-22 08:29:49 -05:00
parent 26ff15b695 07ed12acd9
commit 9749dbfe76
1 changed files with 44 additions and 19 deletions
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scene/resources/audio_stream_wav.cpp
+44 -19
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@@ -788,18 +788,20 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
ERR_FAIL_COND_V(data.resize(frames * format_channels) != OK, Ref<AudioStreamWAV>());
float *data_ptrw = data.ptrw();
if (compression_code == 1) {
if (format_bits == 8) {
for (int64_t i = 0; i < frames * format_channels; i++) {
// 8 bit samples are UNSIGNED
data.write[i] = int8_t(file->get_8() - 128) / 128.f;
data_ptrw[i] = int8_t(file->get_8() - 128) / 128.f;
}
} else if (format_bits == 16) {
for (int64_t i = 0; i < frames * format_channels; i++) {
//16 bit SIGNED
data.write[i] = int16_t(file->get_16()) / 32768.f;
data_ptrw[i] = int16_t(file->get_16()) / 32768.f;
}
} else {
for (int64_t i = 0; i < frames * format_channels; i++) {
@@ -812,7 +814,7 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
}
s <<= (32 - format_bits);
data.write[i] = (int32_t(s) >> 16) / 32768.f;
data_ptrw[i] = (int32_t(s) >> 16) / 32768.f;
}
}
} else if (compression_code == 3) {
@@ -820,13 +822,13 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
for (int64_t i = 0; i < frames * format_channels; i++) {
//32 bit IEEE Float
data.write[i] = file->get_float();
data_ptrw[i] = file->get_float();
}
} else if (format_bits == 64) {
for (int64_t i = 0; i < frames * format_channels; i++) {
//64 bit IEEE Float
data.write[i] = file->get_double();
data_ptrw[i] = file->get_double();
}
}
}
@@ -943,6 +945,10 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
Vector<float> new_data;
ERR_FAIL_COND_V(new_data.resize(new_data_frames * format_channels) != OK, Ref<AudioStreamWAV>());
const float *data_ptr = data.ptr();
float *new_data_ptrw = new_data.ptrw();
for (int c = 0; c < format_channels; c++) {
float frac = 0.0;
int64_t ipos = 0;
@@ -950,12 +956,12 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
for (int64_t i = 0; i < new_data_frames; i++) {
// Cubic interpolation should be enough.
float y0 = data[MAX(0, ipos - 1) * format_channels + c];
float y1 = data[ipos * format_channels + c];
float y2 = data[MIN(frames - 1, ipos + 1) * format_channels + c];
float y3 = data[MIN(frames - 1, ipos + 2) * format_channels + c];
float y0 = data_ptr[MAX(0, ipos - 1) * format_channels + c];
float y1 = data_ptr[ipos * format_channels + c];
float y2 = data_ptr[MIN(frames - 1, ipos + 1) * format_channels + c];
float y3 = data_ptr[MIN(frames - 1, ipos + 2) * format_channels + c];
new_data.write[i * format_channels + c] = Math::cubic_interpolate(y1, y2, y0, y3, frac);
new_data_ptrw[i * format_channels + c] = Math::cubic_interpolate(y1, y2, y0, y3, frac);
// update position and always keep fractional part within ]0...1]
// in order to avoid 32bit floating point precision errors
@@ -981,8 +987,12 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
if (normalize) {
float max = 0.0;
const float *data_ptr = data.ptr();
float *data_ptrw = data.ptrw();
for (int i = 0; i < data.size(); i++) {
float amp = Math::abs(data[i]);
float amp = Math::abs(data_ptr[i]);
if (amp > max) {
max = amp;
}
@@ -990,8 +1000,9 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
if (max > 0) {
float mult = 1.0 / max;
for (int i = 0; i < data.size(); i++) {
data.write[i] *= mult;
data_ptrw[i] = data_ptr[i] * mult;
}
}
}
@@ -1004,10 +1015,12 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
bool found = false;
float limit = Math::db_to_linear(TRIM_DB_LIMIT);
const float *data_ptr = data.ptr();
for (int64_t i = 0; i < data.size() / format_channels; i++) {
float amp_channel_sum = 0.0;
for (uint16_t j = 0; j < format_channels; j++) {
amp_channel_sum += Math::abs(data[(i * format_channels) + j]);
amp_channel_sum += Math::abs(data_ptr[(i * format_channels) + j]);
}
float amp = Math::abs(amp_channel_sum / (float)format_channels);
@@ -1025,6 +1038,9 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
if (first < last) {
Vector<float> new_data;
ERR_FAIL_COND_V(new_data.resize((last - first) * format_channels) != OK, Ref<AudioStreamWAV>());
float *new_data_ptrw = new_data.ptrw();
for (int64_t i = first; i < last; i++) {
float fade_out_mult = 1.0;
@@ -1033,7 +1049,7 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
}
for (uint16_t j = 0; j < format_channels; j++) {
new_data.write[((i - first) * format_channels) + j] = data[(i * format_channels) + j] * fade_out_mult;
new_data_ptrw[((i - first) * format_channels) + j] = data_ptr[(i * format_channels) + j] * fade_out_mult;
}
}
@@ -1061,8 +1077,12 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
if (force_mono && format_channels == 2) {
Vector<float> new_data;
ERR_FAIL_COND_V(new_data.resize(data.size() / 2) != OK, Ref<AudioStreamWAV>());
const float *data_ptr = data.ptr();
float *new_data_ptrw = new_data.ptrw();
for (int64_t i = 0; i < frames; i++) {
new_data.write[i] = (data[i * 2 + 0] + data[i * 2 + 1]) / 2.0;
new_data_ptrw[i] = (data_ptr[i * 2 + 0] + data_ptr[i * 2 + 1]) / 2.0;
}
data = new_data;
@@ -1090,9 +1110,13 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
ERR_FAIL_COND_V(left.resize(tframes) != OK, Ref<AudioStreamWAV>());
ERR_FAIL_COND_V(right.resize(tframes) != OK, Ref<AudioStreamWAV>());
const float *data_ptr = data.ptr();
float *left_ptrw = left.ptrw();
float *right_ptrw = right.ptrw();
for (int64_t i = 0; i < tframes; i++) {
left.write[i] = data[i * 2 + 0];
right.write[i] = data[i * 2 + 1];
left_ptrw[i] = data_ptr[i * 2 + 0];
right_ptrw[i] = data_ptr[i * 2 + 1];
}
Vector<uint8_t> bleft;
@@ -1128,14 +1152,15 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
ERR_FAIL_COND_V(dst_data.resize(data.size() * (is16 ? 2 : 1)) != OK, Ref<AudioStreamWAV>());
{
uint8_t *w = dst_data.ptrw();
const float *data_ptr = data.ptr();
int ds = data.size();
for (int i = 0; i < ds; i++) {
if (is16) {
int16_t v = CLAMP(data[i] * 32768, -32768, 32767);
int16_t v = CLAMP(data_ptr[i] * 32768, -32768, 32767);
encode_uint16(v, &w[i * 2]);
} else {
int8_t v = CLAMP(data[i] * 128, -128, 127);
int8_t v = CLAMP(data_ptr[i] * 128, -128, 127);
w[i] = v;
}
}