// numFrames is number of stereo frames.
// This is called from *outside* the emulator thread.
int __AudioMix(short *outstereo, int numFrames)
{
// TODO: if mixFrequency != the actual output frequency, resample!
lock_guard guard(section);
int underrun = -1;
s16 sampleL = 0;
s16 sampleR = 0;
bool anythingToPlay = false;
for (int i = 0; i < numFrames; i++) {
if (outAudioQueue.size() >= 2) {
sampleL = outAudioQueue.pop_front();
sampleR = outAudioQueue.pop_front();
outstereo[i * 2 + 0] = sampleL;
outstereo[i * 2 + 1] = sampleR;
anythingToPlay = true;
} else {
if (underrun == -1) underrun = i;
outstereo[i * 2 + 0] = sampleL; // repeat last sample, can reduce clicking
outstereo[i * 2 + 1] = sampleR; // repeat last sample, can reduce clicking
}
}
if (anythingToPlay && underrun >= 0) {
DEBUG_LOG(HLE, "Audio out buffer UNDERRUN at %i of %i", underrun, numFrames);
} else {
// DEBUG_LOG(HLE, "No underrun, mixed %i samples fine", numFrames);
}
return underrun >= 0 ? underrun : numFrames;
}
void __AudioDoState(PointerWrap &p) {
auto s = p.Section("sceAudio", 1);
if (!s)
return;
p.Do(eventAudioUpdate);
CoreTiming::RestoreRegisterEvent(eventAudioUpdate, "AudioUpdate", &hleAudioUpdate);
p.Do(eventHostAudioUpdate);
CoreTiming::RestoreRegisterEvent(eventHostAudioUpdate, "AudioUpdateHost", &hleHostAudioUpdate);
p.Do(mixFrequency);
if (s >= 2) {
resampler.DoState(p);
} else {
// Only to preserve the previous file format. Might cause a slight audio glitch on upgrades?
FixedSizeQueue<s16, 512 * 16> outAudioQueue;
outAudioQueue.DoState(p);
resampler.Clear();
}
int chanCount = ARRAY_SIZE(chans);
p.Do(chanCount);
if (chanCount != ARRAY_SIZE(chans))
{
ERROR_LOG(SCEAUDIO, "Savestate failure: different number of audio channels.");
return;
}
for (int i = 0; i < chanCount; ++i)
chans[i].DoState(p);
__AudioCPUMHzChange();
}
// Mix samples from the various audio channels into a single sample queue.
// This single sample queue is where __AudioMix should read from. If the sample queue is full, we should
// just sleep the main emulator thread a little.
void __AudioUpdate()
{
// Audio throttle doesn't really work on the PSP since the mixing intervals are so closely tied
// to the CPU. Much better to throttle the frame rate on frame display and just throw away audio
// if the buffer somehow gets full.
s32 mixBuffer[hwBlockSize * 2];
memset(mixBuffer, 0, sizeof(mixBuffer));
for (int i = 0; i < PSP_AUDIO_CHANNEL_MAX; i++)
{
if (!chans[i].reserved)
continue;
if (!chans[i].sampleQueue.size()) {
// ERROR_LOG(HLE, "No queued samples, skipping channel %i", i);
continue;
}
for (int s = 0; s < hwBlockSize; s++)
{
if (chans[i].sampleQueue.size() >= 2)
{
s16 sampleL = chans[i].sampleQueue.pop_front();
s16 sampleR = chans[i].sampleQueue.pop_front();
mixBuffer[s * 2 + 0] += sampleL;
mixBuffer[s * 2 + 1] += sampleR;
}
else
{
ERROR_LOG(HLE, "Channel %i buffer underrun at %i of %i", i, s, hwBlockSize);
break;
}
}
if (chans[i].sampleQueue.size() < chans[i].sampleCount * 2 * chanQueueMinSizeFactor)
{
// Ask the thread to send more samples until next time, queue is being drained.
if (chans[i].waitingThread) {
SceUID waitingThread = chans[i].waitingThread;
chans[i].waitingThread = 0;
// DEBUG_LOG(HLE, "Woke thread %i for some buffer filling", waitingThread);
__KernelResumeThreadFromWait(waitingThread, chans[i].sampleCount);
}
}
}
if (g_Config.bEnableSound) {
section.lock();
if (outAudioQueue.room() >= hwBlockSize * 2) {
// Push the mixed samples onto the output audio queue.
for (int i = 0; i < hwBlockSize; i++) {
s32 sampleL = mixBuffer[i * 2 + 0] >> 2; // TODO - what factor?
s32 sampleR = mixBuffer[i * 2 + 1] >> 2;
outAudioQueue.push((s16)sampleL);
outAudioQueue.push((s16)sampleR);
}
} else {
// Mix samples from the various audio channels into a single sample queue.
// This single sample queue is where __AudioMix should read from. If the sample queue is full, we should
// just sleep the main emulator thread a little.
void __AudioUpdate()
{
// Audio throttle doesn't really work on the PSP since the mixing intervals are so closely tied
// to the CPU. Much better to throttle the frame rate on frame display and just throw away audio
// if the buffer somehow gets full.
s32 mixBuffer[hwBlockSize * 2];
memset(mixBuffer, 0, sizeof(mixBuffer));
for (u32 i = 0; i < PSP_AUDIO_CHANNEL_MAX + 1; i++)
{
if (!chans[i].reserved)
continue;
__AudioWakeThreads(chans[i], hwBlockSize);
if (!chans[i].sampleQueue.size()) {
// ERROR_LOG(HLE, "No queued samples, skipping channel %i", i);
continue;
}
for (int s = 0; s < hwBlockSize; s++)
{
if (chans[i].sampleQueue.size() >= 2)
{
s16 sampleL = chans[i].sampleQueue.pop_front();
s16 sampleR = chans[i].sampleQueue.pop_front();
mixBuffer[s * 2 + 0] += sampleL;
mixBuffer[s * 2 + 1] += sampleR;
}
else
{
ERROR_LOG(HLE, "Channel %i buffer underrun at %i of %i", i, s, hwBlockSize);
break;
}
}
}
if (g_Config.bEnableSound) {
section.lock();
if (outAudioQueue.room() >= hwBlockSize * 2) {
// Push the mixed samples onto the output audio queue.
for (int i = 0; i < hwBlockSize; i++) {
s16 sampleL = clamp_s16(mixBuffer[i * 2 + 0]);
s16 sampleR = clamp_s16(mixBuffer[i * 2 + 1]);
outAudioQueue.push((s16)sampleL);
outAudioQueue.push((s16)sampleR);
}
} else {
// This happens quite a lot. There's still something slightly off
// about the amount of audio we produce.
DEBUG_LOG(HLE, "Audio outbuffer overrun! room = %i / %i", outAudioQueue.room(), (u32)outAudioQueue.capacity());
}
section.unlock();
}
}
// numFrames is number of stereo frames.
// This is called from *outside* the emulator thread.
int __AudioMix(short *outstereo, int numFrames)
{
// TODO: if mixFrequency != the actual output frequency, resample!
int underrun = -1;
s16 sampleL = 0;
s16 sampleR = 0;
const s16 *buf1 = 0, *buf2 = 0;
size_t sz1, sz2;
{
lock_guard guard(section);
outAudioQueue.popPointers(numFrames * 2, &buf1, &sz1, &buf2, &sz2);
memcpy(outstereo, buf1, sz1 * sizeof(s16));
if (buf2) {
memcpy(outstereo + sz1, buf2, sz2 * sizeof(s16));
}
}
int remains = (int)(numFrames * 2 - sz1 - sz2);
if (remains > 0)
memset(outstereo + numFrames * 2 - remains, 0, remains);
if (sz1 + sz2 < (size_t)numFrames) {
underrun = (int)(sz1 + sz2) / 2;
VERBOSE_LOG(SCEAUDIO, "Audio out buffer UNDERRUN at %i of %i", underrun, numFrames);
}
return underrun >= 0 ? underrun : numFrames;
}
void __AudioDoState(PointerWrap &p) {
p.Do(eventAudioUpdate);
CoreTiming::RestoreRegisterEvent(eventAudioUpdate, "AudioUpdate", &hleAudioUpdate);
p.Do(eventHostAudioUpdate);
CoreTiming::RestoreRegisterEvent(eventHostAudioUpdate, "AudioUpdateHost", &hleHostAudioUpdate);
p.Do(mixFrequency);
{
lock_guard guard(section);
outAudioQueue.DoState(p);
}
int chanCount = ARRAY_SIZE(chans);
p.Do(chanCount);
if (chanCount != ARRAY_SIZE(chans))
{
ERROR_LOG(SCEAUDIO, "Savestate failure: different number of audio channels.");
return;
}
for (int i = 0; i < chanCount; ++i)
chans[i].DoState(p);
p.DoMarker("sceAudio");
}
int main(){
FixedSizeQueue currentBuffer;
int currentHigh;
char dummyChar;
std::ifstream dataFile;
//open file
dataFile.open("data/problem7Data.txt");
if(dataFile.is_open()){
//get first summation of 5 numbers
for (int i=0; i<5; i++) {
dataFile>>dummyChar;
//convert char to integer value
currentBuffer.insert((dummyChar - '0') % 48);
}
currentHigh=currentBuffer.product();
/*continue adding 1 number, and checking summation against
* current high until end of file*/
while (!(dataFile.eof())) {
dataFile>>dummyChar;
currentBuffer.insert((dummyChar - '0') % 48);
if ((currentBuffer.product()) > (currentHigh)) {
currentHigh=currentBuffer.product();
}
}
dataFile.close();
}else{
// Mix samples from the various audio channels into a single sample queue.
// This single sample queue is where __AudioMix should read from. If the sample queue is full, we should
// just sleep the main emulator thread a little.
void __AudioUpdate() {
// Audio throttle doesn't really work on the PSP since the mixing intervals are so closely tied
// 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;
for (u32 i = 0; i < PSP_AUDIO_CHANNEL_MAX + 1; i++) {
if (!chans[i].reserved)
continue;
__AudioWakeThreads(chans[i], 0, hwBlockSize);
if (!chans[i].sampleQueue.size()) {
continue;
}
if (hwBlockSize * 2 > chans[i].sampleQueue.size()) {
ERROR_LOG(SCEAUDIO, "Channel %i buffer underrun at %i of %i", i, (int)chans[i].sampleQueue.size() / 2, hwBlockSize);
}
const s16 *buf1 = 0, *buf2 = 0;
size_t sz1, sz2;
chans[i].sampleQueue.popPointers(hwBlockSize * 2, &buf1, &sz1, &buf2, &sz2);
if (firstChannel) {
for (size_t s = 0; s < sz1; s++)
mixBuffer[s] = buf1[s];
if (buf2) {
for (size_t s = 0; s < sz2; s++)
mixBuffer[s + sz1] = buf2[s];
}
firstChannel = false;
} else {
for (size_t s = 0; s < sz1; s++)
mixBuffer[s] += buf1[s];
if (buf2) {
for (size_t s = 0; s < sz2; s++)
mixBuffer[s + sz1] += buf2[s];
}
}
}
if (firstChannel) {
memset(mixBuffer, 0, hwBlockSize * 2 * sizeof(s32));
}
if (g_Config.bEnableSound) {
lock_guard guard(section);
if (outAudioQueue.room() >= hwBlockSize * 2) {
s16 *buf1 = 0, *buf2 = 0;
size_t sz1, sz2;
outAudioQueue.pushPointers(hwBlockSize * 2, &buf1, &sz1, &buf2, &sz2);
for (size_t s = 0; s < sz1; s++)
buf1[s] = clamp_s16(mixBuffer[s]);
if (buf2) {
for (size_t s = 0; s < sz2; s++)
buf2[s] = clamp_s16(mixBuffer[s + sz1]);
}
} else {
// This happens quite a lot. There's still something slightly off
// about the amount of audio we produce.
}
}
}
int eventAudioUpdate = -1;
int eventHostAudioUpdate = -1;
int mixFrequency = 44100;
const int hwSampleRate = 44100;
const int hwBlockSize = 64;
const int hostAttemptBlockSize = 256;
const int audioIntervalUs = (int)(1000000ULL * hwBlockSize / hwSampleRate);
const int audioHostIntervalUs = (int)(1000000ULL * hostAttemptBlockSize / hwSampleRate);
// High and low watermarks, basically.
const int chanQueueMaxSizeFactor = 4;
const int chanQueueMinSizeFactor = 1;
FixedSizeQueue<s16, hostAttemptBlockSize * 16> outAudioQueue;
void hleAudioUpdate(u64 userdata, int cyclesLate)
{
__AudioUpdate();
CoreTiming::ScheduleEvent(usToCycles(audioIntervalUs) - cyclesLate, eventAudioUpdate, 0);
}
void hleHostAudioUpdate(u64 userdata, int cyclesLate)
{
host->UpdateSound();
CoreTiming::ScheduleEvent(usToCycles(audioHostIntervalUs) - cyclesLate, eventHostAudioUpdate, 0);
}