222 lines
6.6 KiB
C++
222 lines
6.6 KiB
C++
// Copyright 2013 Dolphin Emulator Project
|
|
// Licensed under GPLv2
|
|
// Refer to the license.txt file included.
|
|
|
|
#include "Atomic.h"
|
|
#include "Mixer.h"
|
|
#include "AudioCommon.h"
|
|
#include "CPUDetect.h"
|
|
#include "../../Core/Src/Host.h"
|
|
|
|
#include "../../Core/Src/HW/AudioInterface.h"
|
|
|
|
// UGLINESS
|
|
#include "../../Core/Src/PowerPC/PowerPC.h"
|
|
|
|
#if _M_SSE >= 0x301 && !(defined __GNUC__ && !defined __SSSE3__)
|
|
#include <tmmintrin.h>
|
|
#endif
|
|
|
|
// Executed from sound stream thread
|
|
unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
|
|
{
|
|
if (!samples)
|
|
return 0;
|
|
|
|
std::lock_guard<std::mutex> lk(m_csMixing);
|
|
|
|
if (PowerPC::GetState() != PowerPC::CPU_RUNNING)
|
|
{
|
|
// Silence
|
|
memset(samples, 0, numSamples * 4);
|
|
return numSamples;
|
|
}
|
|
|
|
unsigned int numLeft = GetNumSamples();
|
|
if (m_AIplaying) {
|
|
if (numLeft < numSamples)//cannot do much about this
|
|
m_AIplaying = false;
|
|
if (numLeft < MAX_SAMPLES/4)//low watermark
|
|
m_AIplaying = false;
|
|
} else {
|
|
if (numLeft > MAX_SAMPLES/2)//high watermark
|
|
m_AIplaying = true;
|
|
}
|
|
|
|
// Cache access in non-volatile variable
|
|
// This is the only function changing the read value, so it's safe to
|
|
// cache it locally although it's written here.
|
|
// The writing pointer will be modified outside, but it will only increase,
|
|
// so we will just ignore new written data while interpolating.
|
|
// Without this cache, the compiler wouldn't be allowed to optimize the
|
|
// interpolation loop.
|
|
u32 indexR = Common::AtomicLoad(m_indexR);
|
|
u32 indexW = Common::AtomicLoad(m_indexW);
|
|
|
|
if (m_AIplaying) {
|
|
numLeft = (numLeft > numSamples) ? numSamples : numLeft;
|
|
|
|
if (AudioInterface::GetAIDSampleRate() == m_sampleRate) // (1:1)
|
|
{
|
|
#if _M_SSE >= 0x301
|
|
if (cpu_info.bSSSE3 && !((numLeft * 2) % 8))
|
|
{
|
|
static const __m128i sr_mask =
|
|
_mm_set_epi32(0x0C0D0E0FL, 0x08090A0BL,
|
|
0x04050607L, 0x00010203L);
|
|
|
|
for (unsigned int i = 0; i < numLeft * 2; i += 8)
|
|
{
|
|
_mm_storeu_si128((__m128i *)&samples[i], _mm_shuffle_epi8(_mm_loadu_si128((__m128i *)&m_buffer[(indexR + i) & INDEX_MASK]), sr_mask));
|
|
}
|
|
}
|
|
else
|
|
#endif
|
|
{
|
|
for (unsigned int i = 0; i < numLeft * 2; i+=2)
|
|
{
|
|
samples[i] = Common::swap16(m_buffer[(indexR + i + 1) & INDEX_MASK]);
|
|
samples[i+1] = Common::swap16(m_buffer[(indexR + i) & INDEX_MASK]);
|
|
}
|
|
}
|
|
indexR += numLeft * 2;
|
|
}
|
|
else //linear interpolation
|
|
{
|
|
//render numleft sample pairs to samples[]
|
|
//advance indexR with sample position
|
|
//remember fractional offset
|
|
|
|
static u32 frac = 0;
|
|
const u32 ratio = (u32)( 65536.0f * (float)AudioInterface::GetAIDSampleRate() / (float)m_sampleRate );
|
|
|
|
for (u32 i = 0; i < numLeft * 2; i+=2) {
|
|
u32 indexR2 = indexR + 2; //next sample
|
|
if ((indexR2 & INDEX_MASK) == (indexW & INDEX_MASK)) //..if it exists
|
|
indexR2 = indexR;
|
|
|
|
s16 l1 = Common::swap16(m_buffer[indexR & INDEX_MASK]); //current
|
|
s16 l2 = Common::swap16(m_buffer[indexR2 & INDEX_MASK]); //next
|
|
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)frac) >> 16;
|
|
samples[i+1] = sampleL;
|
|
|
|
s16 r1 = Common::swap16(m_buffer[(indexR + 1) & INDEX_MASK]); //current
|
|
s16 r2 = Common::swap16(m_buffer[(indexR2 + 1) & INDEX_MASK]); //next
|
|
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)frac) >> 16;
|
|
samples[i] = sampleR;
|
|
|
|
frac += ratio;
|
|
indexR += 2 * (u16)(frac >> 16);
|
|
frac &= 0xffff;
|
|
}
|
|
}
|
|
|
|
} else {
|
|
numLeft = 0;
|
|
}
|
|
|
|
// Padding
|
|
if (numSamples > numLeft)
|
|
{
|
|
unsigned short s[2];
|
|
s[0] = Common::swap16(m_buffer[(indexR - 1) & INDEX_MASK]);
|
|
s[1] = Common::swap16(m_buffer[(indexR - 2) & INDEX_MASK]);
|
|
for (unsigned int i = numLeft*2; i < numSamples*2; i+=2)
|
|
*(u32*)(samples+i) = *(u32*)(s);
|
|
// memset(&samples[numLeft * 2], 0, (numSamples - numLeft) * 4);
|
|
}
|
|
|
|
// Flush cached variable
|
|
Common::AtomicStore(m_indexR, indexR);
|
|
|
|
//when logging, also throttle HLE audio
|
|
if (m_logAudio) {
|
|
if (m_AIplaying) {
|
|
Premix(samples, numLeft);
|
|
|
|
AudioInterface::Callback_GetStreaming(samples, numLeft, m_sampleRate);
|
|
|
|
g_wave_writer.AddStereoSamples(samples, numLeft);
|
|
}
|
|
}
|
|
else { //or mix as usual
|
|
// Add the DSPHLE sound, re-sampling is done inside
|
|
Premix(samples, numSamples);
|
|
|
|
// Add the DTK Music
|
|
// Re-sampling is done inside
|
|
AudioInterface::Callback_GetStreaming(samples, numSamples, m_sampleRate);
|
|
}
|
|
|
|
return numSamples;
|
|
}
|
|
|
|
|
|
void CMixer::PushSamples(const short *samples, unsigned int num_samples)
|
|
{
|
|
// Cache access in non-volatile variable
|
|
// indexR isn't allowed to cache in the audio throttling loop as it
|
|
// needs to get updates to not deadlock.
|
|
u32 indexW = Common::AtomicLoad(m_indexW);
|
|
|
|
if (m_throttle)
|
|
{
|
|
// The auto throttle function. This loop will put a ceiling on the CPU MHz.
|
|
while (num_samples * 2 + ((indexW - Common::AtomicLoad(m_indexR)) & INDEX_MASK) >= MAX_SAMPLES * 2)
|
|
{
|
|
if (*PowerPC::GetStatePtr() != PowerPC::CPU_RUNNING || soundStream->IsMuted())
|
|
break;
|
|
// Shortcut key for Throttle Skipping
|
|
if (Host_GetKeyState('\t'))
|
|
break;
|
|
SLEEP(1);
|
|
soundStream->Update();
|
|
}
|
|
}
|
|
|
|
// Check if we have enough free space
|
|
// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
|
|
if (num_samples * 2 + ((indexW - Common::AtomicLoad(m_indexR)) & INDEX_MASK) >= MAX_SAMPLES * 2)
|
|
return;
|
|
|
|
// AyuanX: Actual re-sampling work has been moved to sound thread
|
|
// to alleviate the workload on main thread
|
|
// and we simply store raw data here to make fast mem copy
|
|
int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
|
|
if (over_bytes > 0)
|
|
{
|
|
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
|
|
memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
|
|
}
|
|
else
|
|
{
|
|
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
|
|
}
|
|
|
|
Common::AtomicAdd(m_indexW, num_samples * 2);
|
|
|
|
return;
|
|
}
|
|
|
|
unsigned int CMixer::GetNumSamples()
|
|
{
|
|
// Guess how many samples would be available after interpolation.
|
|
// As interpolation needs at least on sample from the future to
|
|
// linear interpolate between them, one sample less is available.
|
|
// We also can't say the current interpolation state (specially
|
|
// the frac), so to be sure, subtract one again to be sure not
|
|
// to underflow the fifo.
|
|
|
|
u32 numSamples = ((Common::AtomicLoad(m_indexW) - Common::AtomicLoad(m_indexR)) & INDEX_MASK) / 2;
|
|
|
|
if (AudioInterface::GetAIDSampleRate() == m_sampleRate)
|
|
; //numSamples = numSamples; // 1:1
|
|
else if (m_sampleRate == 48000 && AudioInterface::GetAIDSampleRate() == 32000)
|
|
numSamples = numSamples * 3 / 2 - 2; // most common case
|
|
else
|
|
numSamples = numSamples * m_sampleRate / AudioInterface::GetAIDSampleRate() - 2;
|
|
|
|
return numSamples;
|
|
}
|
|
|