dolphin/Source/Core/AudioCommon/Src/Mixer.cpp

216 lines
5.9 KiB
C++
Raw Normal View History

// Copyright (C) 2003 Dolphin 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.
// 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 SVN repository and contact information can be found at
// http://code.google.com/p/dolphin-emu/
#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;
if (PowerPC::GetState() != 0)
{
// Silence
memset(samples, 0, numSamples * 4);
return numSamples;
}
unsigned int numLeft = Common::AtomicLoad(m_numSamples);
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;
}
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[(m_indexR + i) & INDEX_MASK]), sr_mask));
}
}
else
#endif
{
for (unsigned int i = 0; i < numLeft * 2; i+=2)
{
samples[i] = Common::swap16(m_buffer[(m_indexR + i + 1) & INDEX_MASK]);
samples[i+1] = Common::swap16(m_buffer[(m_indexR + i) & INDEX_MASK]);
}
}
m_indexR += numLeft * 2;
}
else //linear interpolation
{
//render numleft sample pairs to samples[]
//advance m_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 m_indexR2 = m_indexR + 2; //next sample
if ((m_indexR2 & INDEX_MASK) == (m_indexW & INDEX_MASK)) //..if it exists
m_indexR2 = m_indexR;
s16 l1 = Common::swap16(m_buffer[m_indexR & INDEX_MASK]); //current
s16 l2 = Common::swap16(m_buffer[m_indexR2 & INDEX_MASK]); //next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)frac) >> 16;
samples[i+1] = sampleL;
s16 r1 = Common::swap16(m_buffer[(m_indexR + 1) & INDEX_MASK]); //current
s16 r2 = Common::swap16(m_buffer[(m_indexR2 + 1) & INDEX_MASK]); //next
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)frac) >> 16;
samples[i] = sampleR;
frac += ratio;
m_indexR += 2 * (u16)(frac >> 16);
frac &= 0xffff;
}
}
} else {
numLeft = 0;
}
// Padding
if (numSamples > numLeft)
{
unsigned short s[2];
s[0] = Common::swap16(m_buffer[(m_indexR - 1) & INDEX_MASK]);
s[1] = Common::swap16(m_buffer[(m_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);
}
//when logging, also throttle HLE audio
if (m_logAudio) {
if (m_AIplaying) {
Premix(samples, numLeft);
if (m_EnableDTKMusic)
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
if (m_EnableDTKMusic)
{
// Re-sampling is done inside
AudioInterface::Callback_GetStreaming(samples, numSamples, m_sampleRate);
}
}
Common::AtomicAdd(m_numSamples, -(s32)numLeft);
return numSamples;
}
void CMixer::PushSamples(const short *samples, unsigned int num_samples)
{
if (m_throttle)
{
// The auto throttle function. This loop will put a ceiling on the CPU MHz.
while (num_samples + Common::AtomicLoad(m_numSamples) > MAX_SAMPLES)
{
if (*PowerPC::GetStatePtr() != 0)
break;
// Shortcut key for Throttle Skipping
if (Host_GetKeyState('\t'))
break;
SLEEP(1);
soundStream->Update();
}
}
// Check if we have enough free space
if (num_samples + Common::AtomicLoad(m_numSamples) > MAX_SAMPLES)
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 - (m_indexW & INDEX_MASK)) * sizeof(short);
if (over_bytes > 0)
{
memcpy(&m_buffer[m_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[m_indexW & INDEX_MASK], samples, num_samples * 4);
}
m_indexW += num_samples * 2;
if (AudioInterface::GetAIDSampleRate() == m_sampleRate)
Common::AtomicAdd(m_numSamples, num_samples);
else if ((AudioInterface::GetAIDSampleRate() == 32000) && (m_sampleRate == 48000))
Common::AtomicAdd(m_numSamples, num_samples * 3 / 2);
else
Common::AtomicAdd(m_numSamples, num_samples * 2 / 3);
return;
}
unsigned int CMixer::GetNumSamples()
{
return Common::AtomicLoad(m_numSamples);
}