Emulator-Archive/ppsspp
1
0
Fork 0
mirror of https://github.com/hrydgard/ppsspp.git synced 2024-11-23 05:19:56 +00:00
Code Issues Actions 2 Packages Projects Releases Wiki Activity

libretro: Rip out the audio speed control logic, libretro should handle that internally.

Browse Source
This commit is contained in:
Henrik Rydgård 2023-04-03 10:01:49 +02:00
parent ecfd4759dd
commit 1d0afd4867
1 changed files with 12 additions and 171 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

183
libretro/libretro.cpp
View File

@ -69,13 +69,6 @@
#define SAMPLERATE 44100
#define AUDIO_RING_BUFFER_SIZE (1 << 16)
#define AUDIO_RING_BUFFER_SIZE_MASK (AUDIO_RING_BUFFER_SIZE - 1)
// An alpha factor of 1/180 is *somewhat* equivalent
// to calculating the average for the last 180
// frames, or 3 seconds of runtime...
#define AUDIO_FRAMES_MOVING_AVG_ALPHA (1.0f / 180.0f)
// Calculated swap interval is 'stable' if the same
// value is recorded for a number of retro_run()
// calls equal to VSYNC_SWAP_INTERVAL_FRAMES
@ -104,10 +97,7 @@ namespace Libretro
static retro_audio_sample_batch_t audio_batch_cb;
static retro_input_poll_t input_poll_cb;
static retro_input_state_t input_state_cb;
} // namespace Libretro
namespace Libretro
{
static bool detectVsyncSwapInterval = false;
static bool detectVsyncSwapIntervalOptShown = true;
@ -239,149 +229,6 @@ namespace Libretro
}
} // namespace Libretro
namespace Libretro
{
static std::mutex audioSampleLock_;
static int16_t audioRingBuffer[AUDIO_RING_BUFFER_SIZE] = {0};
static uint32_t audioRingBufferBase = 0;
static uint32_t audioRingBufferIndex = 0;
static int16_t *audioOutBuffer = NULL;
static uint32_t audioOutBufferSize = 0;
static float audioOutFramesAvg = 0.0f;
// Set this to an arbitrarily large value,
// it will be fine tuned in AudioUploadSamples()
static uint32_t audioBatchFramesMax = AUDIO_RING_BUFFER_SIZE >> 1;
static void AudioBufferFlush()
{
const std::lock_guard<std::mutex> lock(audioSampleLock_);
audioRingBufferBase = 0;
audioRingBufferIndex = 0;
audioOutFramesAvg = (float)SAMPLERATE / (60.0f / 1.001f);
}
static void AudioBufferInit()
{
audioOutFramesAvg = (float)SAMPLERATE / (60.0f / 1.001f);
audioOutBufferSize = ((uint32_t)audioOutFramesAvg + 1) * 2;
audioOutBuffer = (int16_t *)malloc(audioOutBufferSize * sizeof(int16_t));
audioBatchFramesMax = AUDIO_RING_BUFFER_SIZE >> 1;
AudioBufferFlush();
}
static void AudioBufferDeinit()
{
if (audioOutBuffer)
free(audioOutBuffer);
audioOutBuffer = NULL;
audioOutBufferSize = 0;
audioOutFramesAvg = 0.0f;
audioBatchFramesMax = AUDIO_RING_BUFFER_SIZE >> 1;
AudioBufferFlush();
}
static uint32_t AudioBufferOccupancy()
{
const std::lock_guard<std::mutex> lock(audioSampleLock_);
uint32_t occupancy = (audioRingBufferIndex - audioRingBufferBase) &
AUDIO_RING_BUFFER_SIZE_MASK;
return occupancy >> 1;
}
static void AudioBufferWrite(int16_t *audio, uint32_t frames)
{
const std::lock_guard<std::mutex> lock(audioSampleLock_);
uint32_t frameIndex;
uint32_t bufferIndex = audioRingBufferIndex;
for (frameIndex = 0; frameIndex < frames; frameIndex++)
{
audioRingBuffer[audioRingBufferIndex] = *(audio++);
audioRingBuffer[audioRingBufferIndex + 1] = *(audio++);
audioRingBufferIndex = (audioRingBufferIndex + 2) % AUDIO_RING_BUFFER_SIZE;
}
}
static uint32_t AudioBufferRead(int16_t *audio, uint32_t frames)
{
const std::lock_guard<std::mutex> lock(audioSampleLock_);
uint32_t framesAvailable = ((audioRingBufferIndex - audioRingBufferBase) &
AUDIO_RING_BUFFER_SIZE_MASK) >> 1;
uint32_t frameIndex;
if (frames > framesAvailable)
frames = framesAvailable;
for(frameIndex = 0; frameIndex < frames; frameIndex++)
{
uint32_t bufferIndex = (audioRingBufferBase + (frameIndex << 1)) &
AUDIO_RING_BUFFER_SIZE_MASK;
*(audio++) = audioRingBuffer[bufferIndex];
*(audio++) = audioRingBuffer[bufferIndex + 1];
}
audioRingBufferBase += frames << 1;
audioRingBufferBase &= AUDIO_RING_BUFFER_SIZE_MASK;
return frames;
}
static void AudioUploadSamples()
{
// - If 'Detect Frame Rate Changes' is disabled, then
// the core specifies a fixed frame rate of (60.0f / 1.001f)
// - At the audio sample rate of 44100, this means the
// frontend expects exactly 735.735 sample frames per call of
// retro_run()
// - If g_Config.bRenderDuplicateFrames is enabled and
// frameskip is disabled, the mean of the buffer occupancy
// willapproximate to this value in most cases
uint32_t framesAvailable = AudioBufferOccupancy();
if (framesAvailable > 0)
{
// Update 'running average' of buffer occupancy.
// Note that this is not a true running
// average, but just a leaky-integrator/
// exponential moving average, used because
// it is simple and fast (i.e. requires no
// window of samples).
audioOutFramesAvg = (AUDIO_FRAMES_MOVING_AVG_ALPHA * (float)framesAvailable) +
((1.0f - AUDIO_FRAMES_MOVING_AVG_ALPHA) * audioOutFramesAvg);
uint32_t frames = (uint32_t)audioOutFramesAvg;
if (audioOutBufferSize < (frames << 1))
{
audioOutBufferSize = (frames << 1);
audioOutBuffer = (int16_t *)realloc(audioOutBuffer,
audioOutBufferSize * sizeof(int16_t));
}
frames = AudioBufferRead(audioOutBuffer, frames);
int16_t *audioOutBufferPtr = audioOutBuffer;
while (frames > 0)
{
uint32_t framesToWrite = (frames > audioBatchFramesMax) ?
audioBatchFramesMax : frames;
uint32_t framesWritten = audio_batch_cb(audioOutBufferPtr,
framesToWrite);
if ((framesWritten < framesToWrite) &&
(framesWritten > 0))
audioBatchFramesMax = framesWritten;
frames -= framesToWrite;
audioOutBufferPtr += framesToWrite << 1;
}
}
}
} // namespace Libretro
using namespace Libretro;
class PrintfLogger : public LogListener
@ -1197,7 +1044,6 @@ void retro_set_input_state(retro_input_state_t cb) { input_state_cb = cb; }
void retro_init(void)
{
VsyncSwapIntervalReset();
AudioBufferInit();
g_threadManager.Init(cpu_info.num_cores, cpu_info.logical_cpu_count);
@ -1279,7 +1125,6 @@ void retro_deinit(void)
libretro_supports_option_categories = false;
VsyncSwapIntervalReset();
AudioBufferDeinit();
}
void retro_set_controller_port_device(unsigned port, unsigned device)
@ -1385,7 +1230,7 @@ namespace Libretro
// Need to keep eating frames to allow the EmuThread to exit correctly.
while (ctx->ThreadFrame())
AudioBufferFlush();
;
emuThread.join();
emuThread = std::thread();
@ -1400,7 +1245,6 @@ namespace Libretro
emuThreadState = EmuThreadState::PAUSE_REQUESTED;
ctx->ThreadFrame(); // Eat 1 frame
AudioBufferFlush();
while (emuThreadState != EmuThreadState::PAUSED)
sleep_ms(1);
@ -1590,7 +1434,6 @@ void retro_run(void)
emuThreadState == EmuThreadState::PAUSE_REQUESTED)
{
VsyncSwapIntervalDetect();
AudioUploadSamples();
ctx->SwapBuffers();
return;
}
@ -1601,7 +1444,6 @@ void retro_run(void)
if (!ctx->ThreadFrame())
{
VsyncSwapIntervalDetect();
AudioUploadSamples();
return;
}
}
@ -1609,7 +1451,6 @@ void retro_run(void)
EmuFrame();
VsyncSwapIntervalDetect();
AudioUploadSamples();
ctx->SwapBuffers();
}
@ -1646,15 +1487,14 @@ bool retro_serialize(void *data, size_t size)
return false;
}
bool retVal;
SaveState::SaveStart state;
// TODO: Libretro API extension to use the savestate queue
if (useEmuThread)
EmuThreadPause(); // Does nothing if already paused
size_t measuredSize;
SaveState::SaveStart state;
auto err = CChunkFileReader::MeasureAndSavePtr(state, (u8 **)&data, &measuredSize);
retVal = err == CChunkFileReader::ERROR_NONE;
bool retVal = err == CChunkFileReader::ERROR_NONE;
if (useEmuThread)
{
@ -1662,21 +1502,18 @@ bool retro_serialize(void *data, size_t size)
sleep_ms(4);
}
AudioBufferFlush();
return retVal;
}
bool retro_unserialize(const void *data, size_t size)
{
bool retVal;
SaveState::SaveStart state;
// TODO: Libretro API extension to use the savestate queue
if (useEmuThread)
EmuThreadPause(); // Does nothing if already paused
std::string errorString;
retVal = CChunkFileReader::LoadPtr((u8 *)data, state, &errorString)
SaveState::SaveStart state;
bool retVal = CChunkFileReader::LoadPtr((u8 *)data, state, &errorString)
== CChunkFileReader::ERROR_NONE;
if (useEmuThread)
@ -1685,8 +1522,6 @@ bool retro_unserialize(const void *data, size_t size)
sleep_ms(4);
}
AudioBufferFlush();
return retVal;
}
@ -1877,7 +1712,13 @@ void System_AudioPushSamples(const int32_t *audio, int numSamples) {
buffer[i * 2] = Clamp16(audio[i * 2]);
buffer[i * 2 + 1] = Clamp16(audio[i * 2 + 1]);
}
AudioBufferWrite(buffer, blockSize);
int framesToWrite = blockSize;
uint32_t framesWritten = audio_batch_cb(buffer, framesToWrite);
if (framesWritten != framesToWrite) {
// Let's just stop, and eat any left-over samples.
break;
}
numSamples -= blockSize;
}
}