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Remove a per-audio-frame allocation in sceAudio

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This commit is contained in:
Henrik Rydgård 2022-09-01 11:54:48 +02:00
parent c9048c3748
commit 7bced814ce
1 changed files with 8 additions and 11 deletions
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19
Core/HLE/__sceAudio.cpp
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@ -64,8 +64,8 @@ int srcFrequency = 0;
const int hwSampleRate = 44100;
int hwBlockSize = 64;
int hostAttemptBlockSize = 512;
const int hwBlockSize = 64;
const int hostAttemptBlockSize = 512;
static int audioIntervalCycles;
static int audioHostIntervalCycles;
@ -110,8 +110,6 @@ void __AudioInit() {
chanQueueMaxSizeFactor = 2;
chanQueueMinSizeFactor = 1;
hwBlockSize = 64;
hostAttemptBlockSize = 512;
__AudioCPUMHzChange();
@ -338,7 +336,8 @@ void __AudioUpdate(bool resetRecording) {
// to the CPU. Much better to throttle the frame rate on frame display and just throw away audio
// if the buffer somehow gets full.
bool firstChannel = true;
std::vector<int16_t> srcBuffer;
const int16_t srcBufferSize = hwBlockSize * 2;
int16_t srcBuffer[srcBufferSize];
for (u32 i = 0; i < PSP_AUDIO_CHANNEL_MAX + 1; i++) {
if (!chans[i].reserved)
@ -351,7 +350,7 @@ void __AudioUpdate(bool resetRecording) {
}
bool needsResample = i == PSP_AUDIO_CHANNEL_SRC && srcFrequency != 0 && srcFrequency != mixFrequency;
size_t sz = needsResample ? (hwBlockSize * 2 * srcFrequency) / mixFrequency : hwBlockSize * 2;
size_t sz = needsResample ? (srcBufferSize * srcFrequency) / mixFrequency : srcBufferSize;
if (sz > chanSampleQueues[i].size()) {
ERROR_LOG(SCEAUDIO, "Channel %i buffer underrun at %i of %i", i, (int)chanSampleQueues[i].size() / 2, (int)sz / 2);
}
@ -372,13 +371,11 @@ void __AudioUpdate(bool resetRecording) {
return buf1[sz1 - 1];
};
srcBuffer.resize(hwBlockSize * 2);
// TODO: This is terrible, since it's doing it by small chunk and discarding frac.
const uint32_t ratio = (uint32_t)(65536.0 * srcFrequency / (double)mixFrequency);
uint32_t frac = 0;
size_t readIndex = 0;
for (size_t outIndex = 0; readIndex < sz && outIndex < srcBuffer.size(); outIndex += 2) {
for (size_t outIndex = 0; readIndex < sz && outIndex < srcBufferSize; outIndex += 2) {
size_t readIndex2 = readIndex + 2;
int16_t l1 = read(readIndex);
int16_t r1 = read(readIndex + 1);
@ -393,8 +390,8 @@ void __AudioUpdate(bool resetRecording) {
frac &= 0xffff;
}
buf1 = srcBuffer.data();
sz1 = srcBuffer.size();
buf1 = srcBuffer;
sz1 = srcBufferSize;
buf2 = nullptr;
sz2 = 0;
}