izzy/ppsspp
0
0
Fork 0
mirror of https://github.com/izzy2lost/ppsspp.git synced 2026-03-10 12:43:04 -07:00
Code Issues Packages Projects Releases Wiki Activity

Remove the Audio Resampling setting (now always on). Having it off is not useful because it'll never sync up perfectly over time.

Browse Source
This commit is contained in:
Henrik Rydgård
2020-05-17 19:00:15 +02:00
parent 0692b6dca6
commit 90fd2d9e33
4 changed files with 46 additions and 70 deletions
Download Patch File Download Diff File Expand all files Collapse all files

110
Core/HW/StereoResampler.cpp
View File

@@ -181,72 +181,54 @@ unsigned int StereoResampler::Mix(short* samples, unsigned int numSamples, bool
const int INDEX_MASK = (m_maxBufsize * 2 - 1);
lastBufSize_ = (indexR - m_indexW) & INDEX_MASK;
// We force on the audio resampler if the output sample rate doesn't match the input.
if (!g_Config.bAudioResampler && sample_rate == (int)m_input_sample_rate) {
for (currentSample = 0; currentSample < numSamples * 2; currentSample += 2) {
s16 l1 = m_buffer[indexR & INDEX_MASK]; //current
s16 r1 = m_buffer[(indexR + 1) & INDEX_MASK]; //current
samples[currentSample] = l1;
samples[currentSample + 1] = r1;
if (((indexW - indexR) & INDEX_MASK) <= 2) {
// Ran out!
underrunCount_++;
break;
}
indexR += 2;
// Drift prevention mechanism.
float numLeft = (float)(((indexW - indexR) & INDEX_MASK) / 2);
// If we had to discard samples the last frame due to underrun,
// apply an adjustment here. Otherwise we'll overestimate how many
// samples we need.
numLeft -= droppedSamples_;
droppedSamples_ = 0;
// m_numLeftI here becomes a lowpass filtered version of numLeft.
m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1.0f)) / CONTROL_AVG;
// Here we try to keep the buffer size around m_lowwatermark (which is
// really now more like desired_buffer_size) by adjusting the speed.
// Note that the speed of adjustment here does not take the buffer size into
// account. Since this is called once per "output frame", the frame size
// will affect how fast this algorithm reacts, which can't be a good thing.
float offset = (m_numLeftI - (float)m_targetBufsize) * CONTROL_FACTOR;
if (offset > MAX_FREQ_SHIFT) offset = MAX_FREQ_SHIFT;
if (offset < -MAX_FREQ_SHIFT) offset = -MAX_FREQ_SHIFT;
output_sample_rate_ = (float)(m_input_sample_rate + offset);
const u32 ratio = (u32)(65536.0 * output_sample_rate_ / (double)sample_rate);
ratio_ = ratio;
// TODO: consider a higher-quality resampling algorithm.
// TODO: Add a fast path for 1:1.
u32 frac = m_frac;
for (currentSample = 0; currentSample < numSamples * 2; currentSample += 2) {
if (((indexW - indexR) & INDEX_MASK) <= 2) {
// Ran out!
// int missing = numSamples * 2 - currentSample;
// ILOG("Resampler underrun: %d (numSamples: %d, currentSample: %d)", missing, numSamples, currentSample / 2);
underrunCount_++;
break;
}
output_sample_rate_ = (float)sample_rate;
droppedSamples_ = 0;
} else {
// Drift prevention mechanism.
float numLeft = (float)(((indexW - indexR) & INDEX_MASK) / 2);
// If we had to discard samples the last frame due to underrun,
// apply an adjustment here. Otherwise we'll overestimate how many
// samples we need.
numLeft -= droppedSamples_;
droppedSamples_ = 0;
// m_numLeftI here becomes a lowpass filtered version of numLeft.
m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1.0f)) / CONTROL_AVG;
// Here we try to keep the buffer size around m_lowwatermark (which is
// really now more like desired_buffer_size) by adjusting the speed.
// Note that the speed of adjustment here does not take the buffer size into
// account. Since this is called once per "output frame", the frame size
// will affect how fast this algorithm reacts, which can't be a good thing.
float offset = (m_numLeftI - (float)m_targetBufsize) * CONTROL_FACTOR;
if (offset > MAX_FREQ_SHIFT) offset = MAX_FREQ_SHIFT;
if (offset < -MAX_FREQ_SHIFT) offset = -MAX_FREQ_SHIFT;
output_sample_rate_ = (float)(m_input_sample_rate + offset);
const u32 ratio = (u32)(65536.0 * output_sample_rate_ / (double)sample_rate);
ratio_ = ratio;
// TODO: consider a higher-quality resampling algorithm.
// TODO: Add a fast path for 1:1.
u32 frac = m_frac;
for (currentSample = 0; currentSample < numSamples * 2; currentSample += 2) {
if (((indexW - indexR) & INDEX_MASK) <= 2) {
// Ran out!
// int missing = numSamples * 2 - currentSample;
// ILOG("Resampler underrun: %d (numSamples: %d, currentSample: %d)", missing, numSamples, currentSample / 2);
underrunCount_++;
break;
}
u32 indexR2 = indexR + 2; //next sample
s16 l1 = m_buffer[indexR & INDEX_MASK]; //current
s16 r1 = m_buffer[(indexR + 1) & INDEX_MASK]; //current
s16 l2 = m_buffer[indexR2 & INDEX_MASK]; //next
s16 r2 = m_buffer[(indexR2 + 1) & INDEX_MASK]; //next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)frac) >> 16;
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)frac) >> 16;
samples[currentSample] = sampleL;
samples[currentSample + 1] = sampleR;
frac += ratio;
indexR += 2 * (frac >> 16);
frac &= 0xffff;
}
m_frac = frac;
u32 indexR2 = indexR + 2; //next sample
s16 l1 = m_buffer[indexR & INDEX_MASK]; //current
s16 r1 = m_buffer[(indexR + 1) & INDEX_MASK]; //current
s16 l2 = m_buffer[indexR2 & INDEX_MASK]; //next
s16 r2 = m_buffer[(indexR2 + 1) & INDEX_MASK]; //next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)frac) >> 16;
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)frac) >> 16;
samples[currentSample] = sampleL;
samples[currentSample + 1] = sampleR;
frac += ratio;
indexR += 2 * (frac >> 16);
frac &= 0xffff;
}
m_frac = frac;
// Let's not count the underrun padding here.
outputSampleCount_ += currentSample / 2;