(view as text)
diff --git a/Source/Core/AudioCommon/Mixer.cpp b/Source/Core/AudioCommon/Mixer.cpp
index 788a3a2..31698a9 100644
--- a/Source/Core/AudioCommon/Mixer.cpp
+++ b/Source/Core/AudioCommon/Mixer.cpp
@@ -7,6 +7,8 @@
#include "AudioCommon.h"
#include "CPUDetect.h"
#include "../Core/Host.h"
+#include "ConfigManager.h"
+#include "HW/VideoInterface.h"
#include "../Core/HW/AudioInterface.h"
@@ -32,16 +34,7 @@ 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
@@ -53,100 +46,68 @@ 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;
+ float numLeft = ((indexW - indexR) & INDEX_MASK) / 2;
+ m_numLeftI = (numLeft + m_numLeftI*(CONTROL_AVG-1)) / CONTROL_AVG;
+ float offset = (m_numLeftI - LOW_WATERMARK) * CONTROL_FACTOR;
+ if(offset > MAX_FREQ_SHIFT) offset = MAX_FREQ_SHIFT;
+ if(offset < -MAX_FREQ_SHIFT) offset = -MAX_FREQ_SHIFT;
- 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;
- }
- }
+ //render numleft sample pairs to samples[]
+ //advance indexR with sample position
+ //remember fractional offset
- } else {
- numLeft = 0;
+ u32 framelimit = SConfig::GetInstance().m_Framelimit;
+ float aid_sample_rate = AudioInterface::GetAIDSampleRate() + offset;
+ if (framelimit > 2)
+ {
+ aid_sample_rate = aid_sample_rate * (framelimit - 1) * 5 / VideoInterface::TargetRefreshRate;
+ }
+
+ static u32 frac = 0;
+ const u32 ratio = (u32)( 65536.0f * aid_sample_rate / (float)m_sampleRate );
+
+ if(ratio > 0x10000)
+ ERROR_LOG(AUDIO, "ratio out of range");
+
+ 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[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[currentSample] = sampleR;
+
+ 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);
-
- 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 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;
}
@@ -198,24 +159,3 @@ 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;
-}
-
diff --git a/Source/Core/AudioCommon/Mixer.h b/Source/Core/AudioCommon/Mixer.h
index 6b2c4b9..4f31d66 100644
--- a/Source/Core/AudioCommon/Mixer.h
+++ b/Source/Core/AudioCommon/Mixer.h
@@ -8,9 +8,13 @@
#include "StdMutex.h"
// 16 bit Stereo
-#define MAX_SAMPLES (1024 * 8)
+#define MAX_SAMPLES (1024 * 2) // 64ms
#define INDEX_MASK (MAX_SAMPLES * 2 - 1)
-#define RESERVED_SAMPLES (256)
+
+#define LOW_WATERMARK 1280 // 40 ms
+#define MAX_FREQ_SHIFT 200 // per 32000 Hz
+#define CONTROL_FACTOR 0.2 // in freq_shift per fifo size offset
+#define CONTROL_AVG 32
class CMixer {
@@ -24,7 +28,7 @@ public:
, m_logAudio(0)
, m_indexW(0)
, m_indexR(0)
- , m_AIplaying(true)
+ , m_numLeftI(0.0f)
{
// 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 +44,6 @@ 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);
@@ -98,8 +101,8 @@ protected:
volatile u32 m_indexW;
volatile u32 m_indexR;
- bool m_AIplaying;
std::mutex m_csMixing;
+ float m_numLeftI;
volatile float m_speed; // Current rate of the emulation (1.0 = 100% speed)
private: