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GBA Audio: Adjust PSG sampling rate with SOUNDBIAS

This commit is contained in:
Vicki Pfau 2022-06-01 02:13:16 -07:00
parent cbbaa42641
commit cbbcf7478e
3 changed files with 59 additions and 41 deletions
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1
CHANGES
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@ -28,6 +28,7 @@ Emulation fixes:
- GBA: Improve timing when not booting from BIOS
- GBA: Fix expected entry point for multiboot ELFs (fixes mgba.io/i/2450)
- GBA: Fix booting multiboot ROMs with no JOY entrypoint
- GBA Audio: Adjust PSG sampling rate with SOUNDBIAS
- GBA BIOS: Work around IRQ handling hiccup in Mario & Luigi (fixes mgba.io/i/1059)
- GBA BIOS: Initial HLE timing estimation of UnLz77 functions (fixes mgba.io/i/2141)
- GBA DMA: Fix DMA source direction bits being cleared (fixes mgba.io/i/2410)

2
src/gb/audio.c
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@ -674,7 +674,6 @@ void GBAudioUpdateFrame(struct GBAudio* audio) {
}
void GBAudioSamplePSG(struct GBAudio* audio, int16_t* left, int16_t* right) {
GBAudioRun(audio, mTimingCurrentTime(audio->timing), 0xF);
int dcOffset = audio->style == GB_AUDIO_GBA ? 0 : -0x8;
int sampleLeft = dcOffset;
int sampleRight = dcOffset;
@ -731,6 +730,7 @@ static void _sample(struct mTiming* timing, void* user, uint32_t cyclesLate) {
struct GBAudio* audio = user;
int16_t sampleLeft = 0;
int16_t sampleRight = 0;
GBAudioRun(audio, mTimingCurrentTime(audio->timing), 0xF);
GBAudioSamplePSG(audio, &sampleLeft, &sampleRight);
sampleLeft = (sampleLeft * audio->masterVolume * 6) >> 7;
sampleRight = (sampleRight * audio->masterVolume * 6) >> 7;

97
src/gba/audio.c
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@ -25,6 +25,7 @@ mLOG_DEFINE_CATEGORY(GBA_AUDIO, "GBA Audio", "gba.audio");
const unsigned GBA_AUDIO_SAMPLES = 2048;
const int GBA_AUDIO_VOLUME_MAX = 0x100;
static const int SAMPLE_INTERVAL = GBA_ARM7TDMI_FREQUENCY / 0x8000;
static const int CLOCKS_PER_FRAME = 0x800;
static int _applyBias(struct GBAAudio* audio, int sample);
@ -218,6 +219,7 @@ void GBAAudioWriteSOUNDCNT_X(struct GBAAudio* audio, uint16_t value) {
void GBAAudioWriteSOUNDBIAS(struct GBAAudio* audio, uint16_t value) {
audio->soundbias = value;
audio->sampleInterval = 0x200 >> GBARegisterSOUNDBIASGetResolution(value);
}
void GBAAudioWriteWaveRAM(struct GBAAudio* audio, int address, uint32_t value) {
@ -318,59 +320,74 @@ static int _applyBias(struct GBAAudio* audio, int sample) {
static void _sample(struct mTiming* timing, void* user, uint32_t cyclesLate) {
struct GBAAudio* audio = user;
int16_t sampleLeft = 0;
int16_t sampleRight = 0;
int psgShift = 4 - audio->volume;
GBAudioSamplePSG(&audio->psg, &sampleLeft, &sampleRight);
sampleLeft >>= psgShift;
sampleRight >>= psgShift;
int16_t samplesLeft[8];
int16_t samplesRight[8];
int32_t timestamp = mTimingCurrentTime(&audio->p->timing) - cyclesLate - SAMPLE_INTERVAL;
int sample;
for (sample = 0; sample * audio->sampleInterval < (int32_t) SAMPLE_INTERVAL; ++sample) {
int16_t sampleLeft = 0;
int16_t sampleRight = 0;
int psgShift = 4 - audio->volume;
GBAudioRun(&audio->psg, timestamp + (sample + 1) * audio->sampleInterval, 0xF);
GBAudioSamplePSG(&audio->psg, &sampleLeft, &sampleRight);
sampleLeft >>= psgShift;
sampleRight >>= psgShift;
if (audio->mixer) {
audio->mixer->step(audio->mixer);
}
if (!audio->externalMixing) {
if (!audio->forceDisableChA) {
if (audio->chALeft) {
sampleLeft += (audio->chA.sample << 2) >> !audio->volumeChA;
if (audio->mixer) {
audio->mixer->step(audio->mixer);
}
if (!audio->externalMixing) {
if (!audio->forceDisableChA) {
if (audio->chALeft) {
sampleLeft += (audio->chA.sample << 2) >> !audio->volumeChA;
}
if (audio->chARight) {
sampleRight += (audio->chA.sample << 2) >> !audio->volumeChA;
}
}
if (audio->chARight) {
sampleRight += (audio->chA.sample << 2) >> !audio->volumeChA;
if (!audio->forceDisableChB) {
if (audio->chBLeft) {
sampleLeft += (audio->chB.sample << 2) >> !audio->volumeChB;
}
if (audio->chBRight) {
sampleRight += (audio->chB.sample << 2) >> !audio->volumeChB;
}
}
}
if (!audio->forceDisableChB) {
if (audio->chBLeft) {
sampleLeft += (audio->chB.sample << 2) >> !audio->volumeChB;
}
if (audio->chBRight) {
sampleRight += (audio->chB.sample << 2) >> !audio->volumeChB;
}
}
sampleLeft = _applyBias(audio, sampleLeft);
sampleRight = _applyBias(audio, sampleRight);
samplesLeft[sample] = sampleLeft;
samplesRight[sample] = sampleRight;
}
sampleLeft = _applyBias(audio, sampleLeft);
sampleRight = _applyBias(audio, sampleRight);
mCoreSyncLockAudio(audio->p->sync);
unsigned produced;
if ((size_t) blip_samples_avail(audio->psg.left) < audio->samples) {
blip_add_delta(audio->psg.left, audio->clock, sampleLeft - audio->lastLeft);
blip_add_delta(audio->psg.right, audio->clock, sampleRight - audio->lastRight);
audio->lastLeft = sampleLeft;
audio->lastRight = sampleRight;
audio->clock += audio->sampleInterval;
if (audio->clock >= CLOCKS_PER_FRAME) {
blip_end_frame(audio->psg.left, CLOCKS_PER_FRAME);
blip_end_frame(audio->psg.right, CLOCKS_PER_FRAME);
audio->clock -= CLOCKS_PER_FRAME;
int i;
for (i = 0; i < sample; ++i) {
int16_t sampleLeft = samplesLeft[i];
int16_t sampleRight = samplesRight[i];
if ((size_t) blip_samples_avail(audio->psg.left) < audio->samples) {
blip_add_delta(audio->psg.left, audio->clock, sampleLeft - audio->lastLeft);
blip_add_delta(audio->psg.right, audio->clock, sampleRight - audio->lastRight);
audio->lastLeft = sampleLeft;
audio->lastRight = sampleRight;
audio->clock += audio->sampleInterval;
if (audio->clock >= CLOCKS_PER_FRAME) {
blip_end_frame(audio->psg.left, CLOCKS_PER_FRAME);
blip_end_frame(audio->psg.right, CLOCKS_PER_FRAME);
audio->clock -= CLOCKS_PER_FRAME;
}
}
}
produced = blip_samples_avail(audio->psg.left);
// TODO: Post all frames
if (audio->p->stream && audio->p->stream->postAudioFrame) {
audio->p->stream->postAudioFrame(audio->p->stream, sampleLeft, sampleRight);
audio->p->stream->postAudioFrame(audio->p->stream, samplesLeft[sample - 1], samplesRight[sample - 1]);
}
produced = blip_samples_avail(audio->psg.left);
bool wait = produced >= audio->samples;
if (!mCoreSyncProduceAudio(audio->p->sync, audio->psg.left, audio->samples)) {
// Interrupted
@ -381,7 +398,7 @@ static void _sample(struct mTiming* timing, void* user, uint32_t cyclesLate) {
audio->p->stream->postAudioBuffer(audio->p->stream, audio->psg.left, audio->psg.right);
}
mTimingSchedule(timing, &audio->sampleEvent, audio->sampleInterval - cyclesLate);
mTimingSchedule(timing, &audio->sampleEvent, SAMPLE_INTERVAL - cyclesLate);
}
void GBAAudioSerialize(const struct GBAAudio* audio, struct GBASerializedState* state) {