ppsspp/Core/HW/StereoResampler.cpp
Unknown W. Brackets fb831b0f48 Add a global volume config setting.
This applies to all audio, UI, game, etc.  Fixes #7749.
2016-01-17 22:47:29 -08:00

243 lines
8.3 KiB
C++

// Copyright (c) 2015- PPSSPP Project and 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 or later versions.
// 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 git repository and contact information can be found at
// https://github.com/hrydgard/ppsspp and http://www.ppsspp.org/.
// Adapted from Dolphin.
#include <string.h>
#include "base/logging.h"
#include "base/NativeApp.h"
#include "Common/ChunkFile.h"
#include "Common/MathUtil.h"
#include "Common/Atomics.h"
#include "Core/Config.h"
#include "Core/HW/StereoResampler.h"
#include "Core/HLE/__sceAudio.h"
#include "Core/System.h"
#include "Globals.h"
#ifdef _M_SSE
#include <emmintrin.h>
#endif
StereoResampler::StereoResampler()
: m_dma_mixer(this, 44100)
{
// Some Android devices are v-synced to non-60Hz framerates. We simply timestretch audio to fit.
// TODO: should only do this if auto frameskip is off?
float refresh = System_GetPropertyInt(SYSPROP_DISPLAY_REFRESH_RATE) / 1000.0f;
// If framerate is "close"...
if (refresh != 60.0f && refresh > 50.0f && refresh < 70.0f) {
m_dma_mixer.SetInputSampleRate((int)(44100 * (refresh / 60.0f)));
}
}
template<bool useShift>
inline void ClampBufferToS16(s16 *out, const s32 *in, size_t size, s8 volShift) {
#ifdef _M_SSE
// Size will always be 16-byte aligned as the hwBlockSize is.
while (size >= 8) {
__m128i in1 = _mm_loadu_si128((__m128i *)in);
__m128i in2 = _mm_loadu_si128((__m128i *)(in + 4));
__m128i packed = _mm_packs_epi32(in1, in2);
if (useShift) {
packed = _mm_srai_epi16(packed, volShift);
}
_mm_storeu_si128((__m128i *)out, packed);
out += 8;
in += 8;
size -= 8;
}
#endif
// This does the remainder if SSE was used, otherwise it does it all.
for (size_t i = 0; i < size; i++) {
out[i] = clamp_s16(useShift ? (in[i] >> volShift) : in[i]);
}
}
inline void ClampBufferToS16WithVolume(s16 *out, const s32 *in, size_t size) {
if (g_Config.iGlobalVolume >= VOLUME_MAX) {
ClampBufferToS16<false>(out, in, size, 0);
} else if (g_Config.iGlobalVolume <= VOLUME_OFF) {
memset(out, 0, size * sizeof(s16));
} else {
ClampBufferToS16<true>(out, in, size, VOLUME_MAX - (s8)g_Config.iGlobalVolume);
}
}
void StereoResampler::MixerFifo::Clear() {
memset(m_buffer, 0, sizeof(m_buffer));
}
// Executed from sound stream thread
unsigned int StereoResampler::MixerFifo::Mix(short* samples, unsigned int numSamples, bool consider_framelimit, int sample_rate) {
unsigned int currentSample = 0;
// 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);
// We force on the audio resampler if the output sample rate doesn't match the input.
if (!g_Config.bAudioResampler && sample_rate == (int)m_input_sample_rate) {
for (; currentSample < numSamples * 2 && ((indexW - indexR) & INDEX_MASK) > 2; currentSample += 2) {
s16 l1 = m_buffer[indexR & INDEX_MASK]; //current
s16 r1 = m_buffer[(indexR + 1) & INDEX_MASK]; //current
samples[currentSample] = l1;
samples[currentSample + 1] = r1;
indexR += 2;
}
aid_sample_rate_ = sample_rate;
} else {
// Drift prevention mechanism
float numLeft = (float)(((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;
aid_sample_rate_ = m_input_sample_rate + offset;
/* Hm?
u32 framelimit = SConfig::GetInstance().m_Framelimit;
if (consider_framelimit && framelimit > 1) {
aid_sample_rate = aid_sample_rate * (framelimit - 1) * 5 / 59.994;
}*/
const u32 ratio = (u32)(65536.0f * aid_sample_rate_ / (float)sample_rate);
// TODO: consider a higher-quality resampling algorithm.
// TODO: Add a fast path for 1:1.
for (; currentSample < numSamples * 2 && ((indexW - indexR) & INDEX_MASK) > 2; currentSample += 2) {
u32 indexR2 = indexR + 2; //next sample
s16 l1 = m_buffer[indexR & INDEX_MASK]; //current
s16 r1 = m_buffer[(indexR + 1) & INDEX_MASK]; //current
s16 l2 = m_buffer[indexR2 & INDEX_MASK]; //next
s16 r2 = m_buffer[(indexR2 + 1) & INDEX_MASK]; //next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)m_frac) >> 16;
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)m_frac) >> 16;
samples[currentSample] = sampleL;
samples[currentSample + 1] = sampleR;
m_frac += ratio;
indexR += 2 * (u16)(m_frac >> 16);
m_frac &= 0xffff;
}
}
int realSamples = currentSample;
if (currentSample < numSamples * 2)
underrunCount_++;
// Padding with the last value to reduce clicking
short s[2];
s[0] = clamp_s16(m_buffer[(indexR - 1) & INDEX_MASK]);
s[1] = clamp_s16(m_buffer[(indexR - 2) & INDEX_MASK]);
for (; currentSample < numSamples * 2; currentSample += 2) {
samples[currentSample] = s[0];
samples[currentSample + 1] = s[1];
}
// Flush cached variable
Common::AtomicStore(m_indexR, indexR);
//if (realSamples != numSamples * 2) {
// ILOG("Underrun! %i / %i", realSamples / 2, numSamples);
//}
lastBufSize_ = (m_indexW - m_indexR) & INDEX_MASK;
return realSamples / 2;
}
unsigned int StereoResampler::Mix(short* samples, unsigned int num_samples, bool consider_framelimit, int sample_rate) {
if (!samples)
return 0;
return m_dma_mixer.Mix(samples, num_samples, consider_framelimit, sample_rate);
}
void StereoResampler::MixerFifo::PushSamples(const s32 *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);
u32 cap = MAX_SAMPLES * 2;
// If unthottling, no need to fill up the entire buffer, just screws up timing after releasing unthrottle.
if (PSP_CoreParameter().unthrottle)
cap = LOW_WATERMARK * 2;
// 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) >= cap) {
if (!PSP_CoreParameter().unthrottle)
overrunCount_++;
// TODO: "Timestretch" by doing a windowed overlap with existing buffer content?
return;
}
int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
if (over_bytes > 0) {
ClampBufferToS16WithVolume(&m_buffer[indexW & INDEX_MASK], samples, (num_samples * 4 - over_bytes) / 2);
ClampBufferToS16WithVolume(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes / 2);
} else {
ClampBufferToS16WithVolume(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 2);
}
Common::AtomicAdd(m_indexW, num_samples * 2);
lastPushSize_ = num_samples;
}
void StereoResampler::MixerFifo::GetAudioDebugStats(AudioDebugStats *stats) {
stats->buffered = lastBufSize_;
stats->underrunCount += underrunCount_;
underrunCount_ = 0;
stats->overrunCount += overrunCount_;
overrunCount_ = 0;
stats->watermark = LOW_WATERMARK;
stats->bufsize = MAX_SAMPLES * 2;
stats->instantSampleRate = aid_sample_rate_;
stats->lastPushSize = lastPushSize_;
}
void StereoResampler::PushSamples(const int *samples, unsigned int num_samples) {
m_dma_mixer.PushSamples(samples, num_samples);
}
void StereoResampler::MixerFifo::SetInputSampleRate(unsigned int rate) {
m_input_sample_rate = rate;
}
void StereoResampler::DoState(PointerWrap &p) {
auto s = p.Section("resampler", 1);
if (!s)
return;
}
void StereoResampler::GetAudioDebugStats(AudioDebugStats *stats) {
m_dma_mixer.GetAudioDebugStats(stats);
}