pierre 37e31f2df6 AudioCommon: Improve pad silence when ppc does not keep up with realtime
Uses the last sample from the ppc buffer to fill the samples the ppc
didn't deliver data for, avoids clicking on underruns.


git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@7338 8ced0084-cf51-0410-be5f-012b33b47a6e
2011-03-12 22:02:46 +00:00

215 lines
5.7 KiB
C++

// 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;
// Do re-sampling if needed
if (m_sampleRate == 32000)
{
#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 * 32000.0f / (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 (m_sampleRate == 32000)
Common::AtomicAdd(m_numSamples, num_samples);
else // Assume 48000 otherwise
Common::AtomicAdd(m_numSamples, num_samples * 3 / 2);
return;
}
unsigned int CMixer::GetNumSamples()
{
return Common::AtomicLoad(m_numSamples);
}