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d43ecd0bd1
The primary motivation here is to make sure we submit samples from the CPU thread. This makes sure the timing of related interrupts accurate, and generally keeps the different kinds of audio synchronized. This will also allow improvements to audio dumping functionality. The new code is also more concise because it gets rid of some duplicated audio mixing code.
179 lines
5.7 KiB
C++
179 lines
5.7 KiB
C++
// Copyright 2013 Dolphin Emulator Project
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// Licensed under GPLv2
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// Refer to the license.txt file included.
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#include "AudioCommon/AudioCommon.h"
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#include "AudioCommon/Mixer.h"
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#include "Common/Atomic.h"
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#include "Common/CPUDetect.h"
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#include "Common/MathUtil.h"
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#include "Core/ConfigManager.h"
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#include "Core/Core.h"
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#include "Core/HW/AudioInterface.h"
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#include "Core/HW/VideoInterface.h"
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// UGLINESS
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#include "Core/PowerPC/PowerPC.h"
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#if _M_SSE >= 0x301 && !(defined __GNUC__ && !defined __SSSE3__)
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#include <tmmintrin.h>
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#endif
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// Executed from sound stream thread
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unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples, bool consider_framelimit)
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{
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unsigned int currentSample = 0;
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// Cache access in non-volatile variable
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// This is the only function changing the read value, so it's safe to
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// cache it locally although it's written here.
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// The writing pointer will be modified outside, but it will only increase,
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// so we will just ignore new written data while interpolating.
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// Without this cache, the compiler wouldn't be allowed to optimize the
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// interpolation loop.
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u32 indexR = Common::AtomicLoad(m_indexR);
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u32 indexW = Common::AtomicLoad(m_indexW);
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float numLeft = ((indexW - indexR) & INDEX_MASK) / 2;
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m_numLeftI = (numLeft + m_numLeftI*(CONTROL_AVG-1)) / CONTROL_AVG;
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float offset = (m_numLeftI - LOW_WATERMARK) * CONTROL_FACTOR;
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if (offset > MAX_FREQ_SHIFT) offset = MAX_FREQ_SHIFT;
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if (offset < -MAX_FREQ_SHIFT) offset = -MAX_FREQ_SHIFT;
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//render numleft sample pairs to samples[]
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//advance indexR with sample position
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//remember fractional offset
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u32 framelimit = SConfig::GetInstance().m_Framelimit;
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float aid_sample_rate = m_input_sample_rate + offset;
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if (consider_framelimit && framelimit > 2)
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{
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aid_sample_rate = aid_sample_rate * (framelimit - 1) * 5 / VideoInterface::TargetRefreshRate;
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}
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static u32 frac = 0;
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const u32 ratio = (u32)( 65536.0f * aid_sample_rate / (float)m_mixer->m_sampleRate );
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if (ratio > 0x10000)
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ERROR_LOG(AUDIO, "ratio out of range");
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for (; currentSample < numSamples*2 && ((indexW-indexR) & INDEX_MASK) > 2; currentSample+=2) {
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u32 indexR2 = indexR + 2; //next sample
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s16 l1 = Common::swap16(m_buffer[indexR & INDEX_MASK]); //current
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s16 l2 = Common::swap16(m_buffer[indexR2 & INDEX_MASK]); //next
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int sampleL = ((l1 << 16) + (l2 - l1) * (u16)frac) >> 16;
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sampleL += samples[currentSample + 1];
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MathUtil::Clamp(&sampleL, -32767, 32767);
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samples[currentSample+1] = sampleL;
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s16 r1 = Common::swap16(m_buffer[(indexR + 1) & INDEX_MASK]); //current
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s16 r2 = Common::swap16(m_buffer[(indexR2 + 1) & INDEX_MASK]); //next
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int sampleR = ((r1 << 16) + (r2 - r1) * (u16)frac) >> 16;
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sampleR += samples[currentSample];
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MathUtil::Clamp(&sampleR, -32767, 32767);
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samples[currentSample] = sampleR;
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frac += ratio;
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indexR += 2 * (u16)(frac >> 16);
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frac &= 0xffff;
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}
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// Padding
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short s[2];
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s[0] = Common::swap16(m_buffer[(indexR - 1) & INDEX_MASK]);
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s[1] = Common::swap16(m_buffer[(indexR - 2) & INDEX_MASK]);
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for (; currentSample < numSamples*2; currentSample+=2)
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{
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int sampleR = s[0] + samples[currentSample];
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MathUtil::Clamp(&sampleR, -32767, 32767);
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samples[currentSample] = sampleR;
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int sampleL = s[1] + samples[currentSample + 1];
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MathUtil::Clamp(&sampleL, -32767, 32767);
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samples[currentSample + 1] = sampleL;
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}
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// Flush cached variable
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Common::AtomicStore(m_indexR, indexR);
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return numSamples;
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}
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unsigned int CMixer::Mix(short* samples, unsigned int num_samples, bool consider_framelimit)
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{
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if (!samples)
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return 0;
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std::lock_guard<std::mutex> lk(m_csMixing);
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memset(samples, 0, num_samples * 2 * sizeof(short));
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if (PowerPC::GetState() != PowerPC::CPU_RUNNING)
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{
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// Silence
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return num_samples;
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}
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m_dma_mixer.Mix(samples, num_samples, consider_framelimit);
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m_streaming_mixer.Mix(samples, num_samples, consider_framelimit);
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if (m_logAudio)
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g_wave_writer.AddStereoSamples(samples, num_samples);
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return num_samples;
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}
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void CMixer::MixerFifo::PushSamples(const short *samples, unsigned int num_samples)
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{
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// Cache access in non-volatile variable
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// indexR isn't allowed to cache in the audio throttling loop as it
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// needs to get updates to not deadlock.
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u32 indexW = Common::AtomicLoad(m_indexW);
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if (m_mixer->m_throttle)
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{
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// The auto throttle function. This loop will put a ceiling on the CPU MHz.
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while (num_samples * 2 + ((indexW - Common::AtomicLoad(m_indexR)) & INDEX_MASK) >= MAX_SAMPLES * 2)
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{
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if (*PowerPC::GetStatePtr() != PowerPC::CPU_RUNNING || soundStream->IsMuted())
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break;
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// Shortcut key for Throttle Skipping
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if (Core::GetIsFramelimiterTempDisabled())
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break;
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SLEEP(1);
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soundStream->Update();
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}
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}
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// Check if we have enough free space
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// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
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if (num_samples * 2 + ((indexW - Common::AtomicLoad(m_indexR)) & INDEX_MASK) >= MAX_SAMPLES * 2)
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return;
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// AyuanX: Actual re-sampling work has been moved to sound thread
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// to alleviate the workload on main thread
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// and we simply store raw data here to make fast mem copy
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int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
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if (over_bytes > 0)
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{
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memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
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memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
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}
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else
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{
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memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
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}
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Common::AtomicAdd(m_indexW, num_samples * 2);
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return;
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}
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void CMixer::PushSamples(const short *samples, unsigned int num_samples)
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{
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m_dma_mixer.PushSamples(samples, num_samples);
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}
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void CMixer::PushStreamingSamples(const short *samples, unsigned int num_samples)
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{
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m_streaming_mixer.PushSamples(samples, num_samples);
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}
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