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// 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;
}
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