mirror of
https://github.com/hrydgard/ppsspp.git
synced 2024-12-13 16:45:46 +00:00
452 lines
14 KiB
C++
452 lines
14 KiB
C++
// Copyright (c) 2012- PPSSPP Project.
|
|
|
|
// This program is free software: you can redistribute it and/or modify
|
|
// it under the terms of the GNU General Public License as published by
|
|
// the Free Software Foundation, version 2.0 or later versions.
|
|
|
|
// This program is distributed in the hope that it will be useful,
|
|
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
// GNU General Public License 2.0 for more details.
|
|
|
|
// A copy of the GPL 2.0 should have been included with the program.
|
|
// If not, see http://www.gnu.org/licenses/
|
|
|
|
// Official git repository and contact information can be found at
|
|
// https://github.com/hrydgard/ppsspp and http://www.ppsspp.org/.
|
|
|
|
#include "__sceAudio.h"
|
|
#include "sceAudio.h"
|
|
#include "sceKernel.h"
|
|
#include "sceKernelThread.h"
|
|
#include "base/mutex.h"
|
|
#include "CommonTypes.h"
|
|
#include "../CoreTiming.h"
|
|
#include "../MemMap.h"
|
|
#include "../Host.h"
|
|
#include "../Config.h"
|
|
#include "ChunkFile.h"
|
|
#include "FixedSizeQueue.h"
|
|
#include "Common/Atomics.h"
|
|
#include "../../native/base/mutex.h"
|
|
|
|
// Should be used to lock anything related to the outAudioQueue.
|
|
//atomic locks are used on the lock. TODO: make this lock-free
|
|
atomic_flag atomicLock_;
|
|
recursive_mutex mutex_;
|
|
|
|
int eventAudioUpdate = -1;
|
|
int eventHostAudioUpdate = -1;
|
|
int mixFrequency = 44100;
|
|
|
|
const int hwSampleRate = 44100;
|
|
|
|
int hwBlockSize = 64;
|
|
int hostAttemptBlockSize = 512;
|
|
|
|
static int audioIntervalUs;
|
|
static int audioHostIntervalUs;
|
|
|
|
static s32 *mixBuffer;
|
|
|
|
// High and low watermarks, basically. For perfect emulation, the correct values are 0 and 1, respectively.
|
|
// TODO: Tweak. Hm, there aren't actually even used currently...
|
|
static int chanQueueMaxSizeFactor;
|
|
static int chanQueueMinSizeFactor;
|
|
|
|
// TODO: Need to replace this with something lockless. Mutexes in the audio pipeline
|
|
// is bad mojo.
|
|
FixedSizeQueue<s16, 512 * 16> outAudioQueue;
|
|
|
|
bool __gainAudioQueueLock();
|
|
void __releaseAcquiredLock();
|
|
void __blockForAudioQueueLock();
|
|
|
|
static inline s16 adjustvolume(s16 sample, int vol) {
|
|
#ifdef ARM
|
|
register int r;
|
|
asm volatile("smulwb %0, %1, %2\n\t" \
|
|
"ssat %0, #16, %0" \
|
|
: "=r"(r) : "r"(vol), "r"(sample));
|
|
return r;
|
|
#else
|
|
return clamp_s16((sample * vol) >> 16);
|
|
#endif
|
|
}
|
|
|
|
void hleAudioUpdate(u64 userdata, int cyclesLate) {
|
|
__AudioUpdate();
|
|
|
|
CoreTiming::ScheduleEvent(usToCycles(audioIntervalUs) - cyclesLate, eventAudioUpdate, 0);
|
|
}
|
|
|
|
void hleHostAudioUpdate(u64 userdata, int cyclesLate) {
|
|
// Not all hosts need this call to poke their audio system once in a while, but those that don't
|
|
// can just ignore it.
|
|
host->UpdateSound();
|
|
CoreTiming::ScheduleEvent(usToCycles(audioHostIntervalUs) - cyclesLate, eventHostAudioUpdate, 0);
|
|
}
|
|
|
|
void __AudioInit() {
|
|
mixFrequency = 44100;
|
|
|
|
if (g_Config.bLowLatencyAudio) {
|
|
chanQueueMaxSizeFactor = 1;
|
|
chanQueueMinSizeFactor = 1;
|
|
hwBlockSize = 16;
|
|
hostAttemptBlockSize = 256;
|
|
} else {
|
|
chanQueueMaxSizeFactor = 2;
|
|
chanQueueMinSizeFactor = 1;
|
|
hwBlockSize = 64;
|
|
hostAttemptBlockSize = 512;
|
|
}
|
|
|
|
audioIntervalUs = (int)(1000000ULL * hwBlockSize / hwSampleRate);
|
|
audioHostIntervalUs = (int)(1000000ULL * hostAttemptBlockSize / hwSampleRate);
|
|
|
|
eventAudioUpdate = CoreTiming::RegisterEvent("AudioUpdate", &hleAudioUpdate);
|
|
eventHostAudioUpdate = CoreTiming::RegisterEvent("AudioUpdateHost", &hleHostAudioUpdate);
|
|
|
|
CoreTiming::ScheduleEvent(usToCycles(audioIntervalUs), eventAudioUpdate, 0);
|
|
CoreTiming::ScheduleEvent(usToCycles(audioHostIntervalUs), eventHostAudioUpdate, 0);
|
|
for (u32 i = 0; i < PSP_AUDIO_CHANNEL_MAX + 1; i++)
|
|
chans[i].clear();
|
|
|
|
mixBuffer = new s32[hwBlockSize * 2];
|
|
memset(mixBuffer, 0, hwBlockSize * 2 * sizeof(s32));
|
|
|
|
__blockForAudioQueueLock();
|
|
outAudioQueue.clear();
|
|
__releaseAcquiredLock();
|
|
}
|
|
|
|
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);
|
|
|
|
{
|
|
//block until a lock is achieved. Not a good idea at all, but
|
|
//can't think of a better one...
|
|
__blockForAudioQueueLock();
|
|
|
|
outAudioQueue.DoState(p);
|
|
|
|
//release the atomic lock
|
|
__releaseAcquiredLock();
|
|
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
void __AudioShutdown() {
|
|
delete [] mixBuffer;
|
|
|
|
mixBuffer = 0;
|
|
for (u32 i = 0; i < PSP_AUDIO_CHANNEL_MAX + 1; i++)
|
|
chans[i].clear();
|
|
}
|
|
|
|
u32 __AudioEnqueue(AudioChannel &chan, int chanNum, bool blocking) {
|
|
u32 ret = chan.sampleCount;
|
|
|
|
if (chan.sampleAddress == 0) {
|
|
// For some reason, multichannel audio lies and returns the sample count here.
|
|
if (chanNum == PSP_AUDIO_CHANNEL_SRC || chanNum == PSP_AUDIO_CHANNEL_OUTPUT2) {
|
|
ret = 0;
|
|
}
|
|
}
|
|
|
|
// If there's anything on the queue at all, it should be busy, but we try to be a bit lax.
|
|
//if (chan.sampleQueue.size() > chan.sampleCount * 2 * chanQueueMaxSizeFactor || chan.sampleAddress == 0) {
|
|
if (chan.sampleQueue.size() > 0) {
|
|
if (blocking) {
|
|
// TODO: Regular multichannel audio seems to block for 64 samples less? Or enqueue the first 64 sync?
|
|
int blockSamples = (int)chan.sampleQueue.size() / 2 / chanQueueMinSizeFactor;
|
|
|
|
if (__KernelIsDispatchEnabled()) {
|
|
AudioChannelWaitInfo waitInfo = {__KernelGetCurThread(), blockSamples};
|
|
chan.waitingThreads.push_back(waitInfo);
|
|
// Also remember the value to return in the waitValue.
|
|
__KernelWaitCurThread(WAITTYPE_AUDIOCHANNEL, (SceUID)chanNum + 1, ret, 0, false, "blocking audio");
|
|
} else {
|
|
// TODO: Maybe we shouldn't take this audio after all?
|
|
ret = SCE_KERNEL_ERROR_CAN_NOT_WAIT;
|
|
}
|
|
|
|
// Fall through to the sample queueing, don't want to lose the samples even though
|
|
// we're getting full. The PSP would enqueue after blocking.
|
|
} else {
|
|
// Non-blocking doesn't even enqueue, but it's not commonly used.
|
|
return SCE_ERROR_AUDIO_CHANNEL_BUSY;
|
|
}
|
|
}
|
|
|
|
if (chan.sampleAddress == 0) {
|
|
return ret;
|
|
}
|
|
|
|
int leftVol = chan.leftVolume;
|
|
int rightVol = chan.rightVolume;
|
|
|
|
if (leftVol == (1 << 15) && rightVol == (1 << 15) && chan.format == PSP_AUDIO_FORMAT_STEREO && IS_LITTLE_ENDIAN) {
|
|
// TODO: Add mono->stereo conversion to this path.
|
|
|
|
// Good news: the volume doesn't affect the values at all.
|
|
// We can just do a direct memory copy.
|
|
const u32 totalSamples = chan.sampleCount * (chan.format == PSP_AUDIO_FORMAT_STEREO ? 2 : 1);
|
|
s16 *buf1 = 0, *buf2 = 0;
|
|
size_t sz1, sz2;
|
|
chan.sampleQueue.pushPointers(totalSamples, &buf1, &sz1, &buf2, &sz2);
|
|
|
|
if (Memory::IsValidAddress(chan.sampleAddress + (totalSamples - 1) * sizeof(s16_le))) {
|
|
Memory::Memcpy(buf1, chan.sampleAddress, (u32)sz1 * sizeof(s16));
|
|
if (buf2)
|
|
Memory::Memcpy(buf2, chan.sampleAddress + (u32)sz1 * sizeof(s16), (u32)sz2 * sizeof(s16));
|
|
}
|
|
} else {
|
|
// Remember that maximum volume allowed is 0xFFFFF so left shift is no issue.
|
|
// This way we can optimally shift by 16.
|
|
leftVol <<=1;
|
|
rightVol <<=1;
|
|
|
|
if (chan.format == PSP_AUDIO_FORMAT_STEREO) {
|
|
const u32 totalSamples = chan.sampleCount * 2;
|
|
|
|
s16_le *sampleData = (s16_le *) Memory::GetPointer(chan.sampleAddress);
|
|
|
|
// Walking a pointer for speed. But let's make sure we wouldn't trip on an invalid ptr.
|
|
if (Memory::IsValidAddress(chan.sampleAddress + (totalSamples - 1) * sizeof(s16_le))) {
|
|
s16 *buf1 = 0, *buf2 = 0;
|
|
size_t sz1, sz2;
|
|
chan.sampleQueue.pushPointers(totalSamples, &buf1, &sz1, &buf2, &sz2);
|
|
|
|
// TODO: SSE/NEON (VQDMULH) implementations
|
|
for (u32 i = 0; i < sz1; i += 2) {
|
|
buf1[i] = adjustvolume(sampleData[i], leftVol);
|
|
buf1[i + 1] = adjustvolume(sampleData[i + 1], rightVol);
|
|
}
|
|
if (buf2) {
|
|
sampleData += sz1;
|
|
for (u32 i = 0; i < sz2; i += 2) {
|
|
buf2[i] = adjustvolume(sampleData[i], leftVol);
|
|
buf2[i + 1] = adjustvolume(sampleData[i + 1], rightVol);
|
|
}
|
|
}
|
|
}
|
|
} else if (chan.format == PSP_AUDIO_FORMAT_MONO) {
|
|
for (u32 i = 0; i < chan.sampleCount; i++) {
|
|
// Expand to stereo
|
|
s16 sample = (s16)Memory::Read_U16(chan.sampleAddress + 2 * i);
|
|
chan.sampleQueue.push(adjustvolume(sample, leftVol));
|
|
chan.sampleQueue.push(adjustvolume(sample, rightVol));
|
|
}
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
inline void __AudioWakeThreads(AudioChannel &chan, int result, int step) {
|
|
u32 error;
|
|
for (size_t w = 0; w < chan.waitingThreads.size(); ++w) {
|
|
AudioChannelWaitInfo &waitInfo = chan.waitingThreads[w];
|
|
waitInfo.numSamples -= step;
|
|
|
|
// If it's done (there will still be samples on queue) and actually still waiting, wake it up.
|
|
u32 waitID = __KernelGetWaitID(waitInfo.threadID, WAITTYPE_AUDIOCHANNEL, error);
|
|
if (waitInfo.numSamples <= 0 && waitID != 0) {
|
|
// DEBUG_LOG(SCEAUDIO, "Woke thread %i for some buffer filling", waitingThread);
|
|
u32 ret = result == 0 ? __KernelGetWaitValue(waitInfo.threadID, error) : SCE_ERROR_AUDIO_CHANNEL_NOT_RESERVED;
|
|
__KernelResumeThreadFromWait(waitInfo.threadID, ret);
|
|
|
|
chan.waitingThreads.erase(chan.waitingThreads.begin() + w--);
|
|
}
|
|
// This means the thread stopped waiting, so stop trying to wake it.
|
|
else if (waitID == 0)
|
|
chan.waitingThreads.erase(chan.waitingThreads.begin() + w--);
|
|
}
|
|
}
|
|
|
|
void __AudioWakeThreads(AudioChannel &chan, int result) {
|
|
__AudioWakeThreads(chan, result, 0x7FFFFFFF);
|
|
}
|
|
|
|
void __AudioSetOutputFrequency(int freq) {
|
|
WARN_LOG(SCEAUDIO, "Switching audio frequency to %i", freq);
|
|
mixFrequency = freq;
|
|
}
|
|
|
|
// 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 > (int)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) {
|
|
|
|
__blockForAudioQueueLock();
|
|
/*
|
|
if (!__gainAudioQueueLock()){
|
|
return;
|
|
}
|
|
*/
|
|
|
|
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.
|
|
}
|
|
//release the atomic lock
|
|
__releaseAcquiredLock();
|
|
}
|
|
}
|
|
|
|
// 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;
|
|
{
|
|
|
|
//TODO: do rigorous testing to see whether just blind locking will improve speed.
|
|
if (!__gainAudioQueueLock()){
|
|
memset(outstereo, 0, numFrames * 2 * sizeof(short));
|
|
return 0;
|
|
}
|
|
|
|
outAudioQueue.popPointers(numFrames * 2, &buf1, &sz1, &buf2, &sz2);
|
|
|
|
memcpy(outstereo, buf1, sz1 * sizeof(s16));
|
|
if (buf2) {
|
|
memcpy(outstereo + sz1, buf2, sz2 * sizeof(s16));
|
|
}
|
|
|
|
//release the atomic lock
|
|
__releaseAcquiredLock();
|
|
}
|
|
|
|
int remains = (int)(numFrames * 2 - sz1 - sz2);
|
|
if (remains > 0)
|
|
memset(outstereo + numFrames * 2 - remains, 0, remains*sizeof(s16));
|
|
|
|
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;
|
|
}
|
|
|
|
|
|
|
|
/*returns whether the lock was successfully gained or not.
|
|
i.e - whether the lock belongs to you
|
|
*/
|
|
inline bool __gainAudioQueueLock(){
|
|
if (g_Config.bAtomicAudioLocks){
|
|
/*if the previous state was 0, that means the lock was "unlocked". So,
|
|
we return !0, which is true thanks to C's int to bool conversion
|
|
|
|
One the other hand, if it was locked, then the lock would return 1.
|
|
so, !1 = 0 = false.
|
|
*/
|
|
return atomicLock_.test_and_set() == 0;
|
|
} else {
|
|
mutex_.lock();
|
|
return true;
|
|
}
|
|
};
|
|
|
|
inline void __releaseAcquiredLock(){
|
|
if (g_Config.bAtomicAudioLocks){
|
|
atomicLock_.clear();
|
|
} else {
|
|
mutex_.unlock();
|
|
}
|
|
}
|
|
|
|
inline void __blockForAudioQueueLock(){
|
|
if (g_Config.bAtomicAudioLocks){
|
|
while ((atomicLock_.test_and_set() == 0)){ }
|
|
} else {
|
|
mutex_.lock();
|
|
}
|
|
}
|