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audiocommon: sync mixer by fifo instead of estimate values

This commit is contained in:
degasus 2014-02-06 06:55:45 +01:00
parent 2956ffa5be
commit d20dbbc92f
2 changed files with 34 additions and 84 deletions
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115
Source/Core/AudioCommon/Mixer.cpp
View File

@ -32,16 +32,7 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
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;
}
unsigned int currentSample = 0;
// Cache access in non-volatile variable
// This is the only function changing the read value, so it's safe to
@ -53,72 +44,55 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
u32 indexR = Common::AtomicLoad(m_indexR);
u32 indexW = Common::AtomicLoad(m_indexW);
if (m_AIplaying) {
numLeft = (numLeft > numSamples) ? numSamples : numLeft;
//render numleft sample pairs to samples[]
//advance indexR with sample position
//remember fractional offset
//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 );
static u32 frac = 0;
const u32 ratio = (u32)( 65536.0f * (float)AudioInterface::GetAIDSampleRate() / (float)m_sampleRate );
if(ratio > 0x10000)
ERROR_LOG(AUDIO, "ratio out of range");
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;
for (; currentSample < numSamples*2 && ((indexW-indexR) & INDEX_MASK) > 2; currentSample+=2) {
u32 indexR2 = indexR + 2; //next sample
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 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[currentSample+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;
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[currentSample] = sampleR;
frac += ratio;
indexR += 2 * (u16)(frac >> 16);
frac &= 0xffff;
}
} else {
numLeft = 0;
frac += ratio;
indexR += 2 * (u16)(frac >> 16);
frac &= 0xffff;
}
// 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 (; currentSample < numSamples*2; currentSample+=2)
{
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);
samples[currentSample] = s[0];
samples[currentSample+1] = s[1];
}
// Flush cached variable
Common::AtomicStore(m_indexR, indexR);
//when logging, also throttle HLE audio
if (m_logAudio) {
if (m_AIplaying) {
Premix(samples, numLeft);
// Add the DSPHLE sound, re-sampling is done inside
Premix(samples, numSamples);
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);
}
// Add the DTK Music
// Re-sampling is done inside
AudioInterface::Callback_GetStreaming(samples, numSamples, m_sampleRate);
if (m_logAudio)
g_wave_writer.AddStereoSamples(samples, numSamples);
return numSamples;
}
@ -170,24 +144,3 @@ void CMixer::PushSamples(const short *samples, unsigned int num_samples)
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;
}

3
Source/Core/AudioCommon/Mixer.h
View File

@ -24,7 +24,6 @@ public:
, m_logAudio(0)
, m_indexW(0)
, m_indexR(0)
, m_AIplaying(true)
{
// AyuanX: The internal (Core & DSP) sample rate is fixed at 32KHz
// So when AI/DAC sample rate differs than 32KHz, we have to do re-sampling
@ -40,7 +39,6 @@ public:
// Called from audio threads
virtual unsigned int Mix(short* samples, unsigned int numSamples);
virtual void Premix(short * /*samples*/, unsigned int /*numSamples*/) {}
unsigned int GetNumSamples();
// Called from main thread
virtual void PushSamples(const short* samples, unsigned int num_samples);
@ -98,7 +96,6 @@ protected:
volatile u32 m_indexW;
volatile u32 m_indexR;
bool m_AIplaying;
std::mutex m_csMixing;
volatile float m_speed; // Current rate of the emulation (1.0 = 100% speed)