DSPHLE: Added an option for 48khz output and supporting upsampling code.
Games that have higher frequency sounds and music should sound a bit better using 48k. I don't have any games that use DTKMusic so that upsampling code untested. If you get strange sounds only at 48k try toggling dtk music to see if that isolates the problem and let me know. git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@6383 8ced0084-cf51-0410-be5f-012b33b47a6e
This commit is contained in:
mylek4
parent
c6d95c9087
commit
67f850dd37
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@ -26,14 +26,19 @@ void AudioCommonConfig::Load(IniFile &file) {
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file.Get("Config", "Volume", &m_Volume, 75);
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#ifdef _WIN32
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file.Get("Config", "Backend", &sBackend, BACKEND_DIRECTSOUND);
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file.Get("Config", "Frequency", &sFrequency, "32,000 Hz");
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#elif defined __APPLE__
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std::string temp;
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file.Get("Config", "Backend", &temp, BACKEND_COREAUDIO);
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strncpy(sBackend, temp.c_str(), 128);
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file.Get("Config", "Frequency", &temp, "32,000 Hz");
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strncpy(sFrequency, temp.c_str(), 128);
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#elif defined __linux__
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file.Get("Config", "Backend", &sBackend, BACKEND_ALSA);
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file.Get("Config", "Frequency", &sFrequency, "32,000 Hz");
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#else
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file.Get("Config", "Backend", &sBackend, BACKEND_OPENAL);
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file.Get("Config", "Frequency", &sFrequency, "32,000 Hz");
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#endif
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}
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@ -43,6 +48,7 @@ void AudioCommonConfig::Set(IniFile &file) {
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file.Set("Config", "EnableThrottle", m_EnableThrottle);
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file.Set("Config", "EnableJIT", m_EnableJIT);
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file.Set("Config", "Backend", sBackend);
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file.Set("Config", "Frequency", sFrequency);
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file.Set("Config", "Volume", m_Volume);
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}
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@ -39,8 +39,10 @@ struct AudioCommonConfig
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int m_Volume;
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#ifdef __APPLE__
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char sBackend[128];
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char sFrequency[128];
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#else
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std::string sBackend;
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std::string sFrequency;
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#endif
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// Load from given file
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@ -57,74 +57,39 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
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samples[i] = Common::swap16(m_buffer[(m_indexR + i) & INDEX_MASK]);
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m_indexR += numLeft * 2;
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}
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else
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else //linear interpolation
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{
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// AyuanX: Up-sampling is not implemented yet
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PanicAlert("Mixer: Up-sampling is not implemented yet!");
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/*
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static int PV1l=0,PV2l=0,PV3l=0,PV4l=0;
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static int PV1r=0,PV2r=0,PV3r=0,PV4r=0;
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static int acc=0;
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//render numleft sample pairs to samples[]
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//advance m_indexR with sample position
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//remember fractional offset
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while (num_stereo_samples) {
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acc += core_sample_rate;
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while (num_stereo_samples && (acc >= 48000)) {
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PV4l=PV3l;
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PV3l=PV2l;
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PV2l=PV1l;
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PV1l=*(samples++); //32bit processing
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PV4r=PV3r;
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PV3r=PV2r;
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PV2r=PV1r;
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PV1r=*(samples++); //32bit processing
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num_stereo_samples--;
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acc-=48000;
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}
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// defaults to nearest
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s32 DataL = PV1l;
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s32 DataR = PV1r;
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if (m_mode == 1) { //linear
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DataL = PV1l + ((PV2l - PV1l)*acc)/48000;
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DataR = PV1r + ((PV2r - PV1r)*acc)/48000;
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}
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else if (m_mode == 2) {//cubic
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s32 a0l = PV1l - PV2l - PV4l + PV3l;
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s32 a0r = PV1r - PV2r - PV4r + PV3r;
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s32 a1l = PV4l - PV3l - a0l;
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s32 a1r = PV4r - PV3r - a0r;
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s32 a2l = PV1l - PV4l;
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s32 a2r = PV1r - PV4r;
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s32 a3l = PV2l;
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s32 a3r = PV2r;
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s32 t0l = ((a0l )*acc)/48000;
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s32 t0r = ((a0r )*acc)/48000;
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s32 t1l = ((t0l+a1l)*acc)/48000;
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s32 t1r = ((t0r+a1r)*acc)/48000;
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s32 t2l = ((t1l+a2l)*acc)/48000;
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s32 t2r = ((t1r+a2r)*acc)/48000;
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s32 t3l = ((t2l+a3l));
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s32 t3r = ((t2r+a3r));
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DataL = t3l;
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DataR = t3r;
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}
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int l = DataL, r = DataR;
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if (l < -32767) l = -32767;
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if (r < -32767) r = -32767;
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if (l > 32767) l = 32767;
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if (r > 32767) r = 32767;
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sample_queue.push(l);
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sample_queue.push(r);
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m_queueSize += 2;
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}
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*/
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static u32 frac = 0;
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const u32 ratio = (u32)( 65536.0f * 32000.0f / (float)m_sampleRate );
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for (u32 i = 0; i < numLeft * 2; i+=2) {
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u32 m_indexR2 = m_indexR + 2; //next sample
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if ((m_indexR2 & INDEX_MASK) == (m_indexW & INDEX_MASK)) //..if it exists
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m_indexR2 = m_indexR;
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s16 l1 = Common::swap16(m_buffer[m_indexR & INDEX_MASK]); //current
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s16 l2 = Common::swap16(m_buffer[m_indexR2 & INDEX_MASK]); //next
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int sampleL = (l1 << 16) + (l2 - l1) * (u16)frac >> 16;
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samples[i] = sampleL;
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s16 r1 = Common::swap16(m_buffer[(m_indexR + 1) & INDEX_MASK]); //current
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s16 r2 = Common::swap16(m_buffer[(m_indexR2 + 1) & INDEX_MASK]); //next
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int sampleR = (r1 << 16) + (r2 - r1) * (u16)frac >> 16;
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samples[i+1] = sampleR;
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frac += ratio;
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m_indexR += 2 * (u16)(frac >> 16);
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frac &= 0xffff;
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}
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}
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} else {
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numLeft = 0;
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}
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@ -154,7 +119,7 @@ void CMixer::PushSamples(short *samples, unsigned int num_samples)
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if (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 (Common::AtomicLoad(m_numSamples) >= MAX_SAMPLES - RESERVED_SAMPLES)
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while (Common::AtomicLoad(m_numSamples) + RESERVED_SAMPLES >= MAX_SAMPLES)
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{
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if (g_dspInitialize.pEmulatorState)
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{
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@ -171,7 +136,7 @@ void CMixer::PushSamples(short *samples, unsigned int num_samples)
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}
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// Check if we have enough free space
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if (num_samples > MAX_SAMPLES - Common::AtomicLoad(m_numSamples))
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if (num_samples + Common::AtomicLoad(m_numSamples) > MAX_SAMPLES)
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return;
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// AyuanX: Actual re-sampling work has been moved to sound thread
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@ -190,12 +155,12 @@ void CMixer::PushSamples(short *samples, unsigned int num_samples)
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m_indexW += num_samples * 2;
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if (m_sampleRate != 32000)
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{
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PanicAlert("Mixer: Up-sampling is not implemented yet!");
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}
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Common::AtomicAdd(m_numSamples, num_samples);
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if (m_sampleRate == 32000)
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Common::AtomicAdd(m_numSamples, num_samples);
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else if (m_sampleRate == 48000)
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Common::AtomicAdd(m_numSamples, num_samples * 1.5);
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else
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PanicAlert("Mixer: Unsupported sample rate.");
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return;
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}
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@ -26,7 +26,7 @@
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class CMixer {
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public:
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CMixer(unsigned int AISampleRate = 48000, unsigned int DACSampleRate = 48000)
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CMixer(unsigned int AISampleRate = 48000, unsigned int DACSampleRate = 48000, unsigned int BackendSampleRate = 32000)
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: m_aiSampleRate(AISampleRate)
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, m_dacSampleRate(DACSampleRate)
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, m_bits(16)
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@ -39,9 +39,8 @@ public:
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{
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// AyuanX: The internal (Core & DSP) sample rate is fixed at 32KHz
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// So when AI/DAC sample rate differs than 32KHz, we have to do re-sampling
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// I prefer speed so let's do down-sampling instead of up-sampling
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// If you like better sound than speed, feel free to implement the up-sampling code
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m_sampleRate = 32000;
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m_sampleRate = BackendSampleRate;
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INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized (AISampleRate:%i, DACSampleRate:%i)", AISampleRate, DACSampleRate);
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}
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@ -281,17 +281,11 @@ unsigned int Callback_GetStreaming(short* _pDestBuffer, unsigned int _numSamples
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const int lvolume = g_AudioRegister.m_Volume.leftVolume;
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const int rvolume = g_AudioRegister.m_Volume.rightVolume;
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if (g_AISampleRate == 48000 && _sampleRate == 32000)
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{
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_dbg_assert_msg_(AUDIO_INTERFACE, !(_numSamples & 1), "Number of Samples: %i must be even!", _numSamples);
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_numSamples = _numSamples * 3 / 2;
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}
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else if (g_AISampleRate == 32000 && _sampleRate == 48000)
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{
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// AyuanX: Up-sampling is not implemented yet
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PanicAlert("AUDIO_INTERFACE: Up-sampling is not implemented yet!");
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}
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int pcm_l = 0, pcm_r = 0;
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for (unsigned int i = 0; i < _numSamples; i++)
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@ -299,45 +293,82 @@ unsigned int Callback_GetStreaming(short* _pDestBuffer, unsigned int _numSamples
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if (pos == 0)
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ReadStreamBlock(pcm);
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if (g_AISampleRate == 48000 && _sampleRate == 32000)
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if (g_AISampleRate == 48000 && _sampleRate == 32000) //downsample 48>32
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{
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if (i % 3)
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{
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pcm_l = (((pcm_l + (int)pcm[pos*2]) / 2 * lvolume) >> 8) + (int)(*_pDestBuffer);
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if (pcm_l > 32767)
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pcm_l = 32767;
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else if (pcm_l < -32767)
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pcm_l = -32767;
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if (pcm_l > 32767) pcm_l = 32767;
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else if (pcm_l < -32767) pcm_l = -32767;
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*_pDestBuffer++ = pcm_l;
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pcm_r = (((pcm_r + (int)pcm[pos*2+1]) / 2 * rvolume) >> 8) + (int)(*_pDestBuffer);
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if (pcm_r > 32767)
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pcm_r = 32767;
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else if (pcm_r < -32767)
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pcm_r = -32767;
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if (pcm_r > 32767) pcm_r = 32767;
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else if (pcm_r < -32767) pcm_r = -32767;
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*_pDestBuffer++ = pcm_r;
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}
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pcm_l = pcm[pos*2];
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pcm_r = pcm[pos*2+1];
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pos++;
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}
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else
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else if (g_AISampleRate == 32000 && _sampleRate == 48000) //upsample 32>48
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{
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//starts with one sample of 0
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const u32 ratio = (u32)( 65536.0f * 32000.0f / (float)_sampleRate );
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static u32 frac = 0;
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static s16 l1 = 0;
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static s16 l2 = 0;
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static s16 r1 = 0;
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static s16 r2 = 0;
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if ( frac >= 0x10000 || frac == 0)
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{
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frac &= 0xffff;
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l1 = l2; //current
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l2 = pcm[pos * 2]; //next
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r1 = r2; //current
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r2 = pcm[pos * 2 + 1]; //next
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}
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pcm_l = (l1 << 16) + (l2 - l1) * (u16)frac >> 16;
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pcm_r = (l1 << 16) + (l2 - l1) * (u16)frac >> 16;
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pcm_l = (pcm_l * lvolume >> 8) + (int)(*_pDestBuffer);
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if (pcm_l > 32767) pcm_l = 32767;
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else if (pcm_l < -32767) pcm_l = -32767;
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*_pDestBuffer++ = pcm_l;
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pcm_r = (pcm_r * lvolume >> 8) + (int)(*_pDestBuffer);
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if (pcm_r > 32767) pcm_r = 32767;
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else if (pcm_r < -32767) pcm_r = -32767;
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*_pDestBuffer++ = pcm_r;
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frac += ratio;
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pos += frac >> 16;
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}
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else //1:1 no resampling
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{
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pcm_l = (((int)pcm[pos*2] * lvolume) >> 8) + (int)(*_pDestBuffer);
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if (pcm_l > 32767)
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pcm_l = 32767;
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else if (pcm_l < -32767)
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pcm_l = -32767;
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if (pcm_l > 32767) pcm_l = 32767;
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else if (pcm_l < -32767) pcm_l = -32767;
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*_pDestBuffer++ = pcm_l;
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pcm_r = (((int)pcm[pos*2+1] * rvolume) >> 8) + (int)(*_pDestBuffer);
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if (pcm_r > 32767)
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pcm_r = 32767;
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else if (pcm_r < -32767)
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pcm_r = -32767;
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if (pcm_r > 32767) pcm_r = 32767;
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else if (pcm_r < -32767) pcm_r = -32767;
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*_pDestBuffer++ = pcm_r;
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pos++;
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}
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if (++pos == 28)
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if (pos == 28)
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pos = 0;
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}
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}
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@ -23,6 +23,7 @@ BEGIN_EVENT_TABLE(DSPConfigDialogHLE, wxDialog)
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EVT_CHECKBOX(ID_ENABLE_HLE_AUDIO, DSPConfigDialogHLE::SettingsChanged)
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EVT_CHECKBOX(ID_ENABLE_DTK_MUSIC, DSPConfigDialogHLE::SettingsChanged)
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EVT_CHECKBOX(ID_ENABLE_THROTTLE, DSPConfigDialogHLE::SettingsChanged)
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EVT_CHOICE(ID_FREQUENCY, DSPConfigDialogHLE::SettingsChanged)
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EVT_CHOICE(ID_BACKEND, DSPConfigDialogHLE::BackendChanged)
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EVT_COMMAND_SCROLL(ID_VOLUME, DSPConfigDialogHLE::VolumeChanged)
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END_EVENT_TABLE()
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@ -44,6 +45,12 @@ DSPConfigDialogHLE::DSPConfigDialogHLE(wxWindow *parent, wxWindowID id,
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wxDefaultPosition, wxDefaultSize, 0, wxDefaultValidator);
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m_buttonEnableThrottle = new wxCheckBox(this, ID_ENABLE_THROTTLE, wxT("Enable Audio Throttle"),
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wxDefaultPosition, wxDefaultSize, 0, wxDefaultValidator);
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wxStaticText *FrequencyText = new wxStaticText(this, wxID_ANY, wxT("Sample Rate"),
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wxDefaultPosition, wxDefaultSize, 0);
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m_FrequencySelection = new wxChoice(this, ID_FREQUENCY, wxDefaultPosition, wxSize(110, 20),
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wxArrayRates, 0, wxDefaultValidator, wxEmptyString);
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wxStaticText *BackendText = new wxStaticText(this, wxID_ANY, wxT("Audio Backend"),
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wxDefaultPosition, wxDefaultSize, 0);
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m_BackendSelection = new wxChoice(this, ID_BACKEND, wxDefaultPosition, wxSize(110, 20),
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@ -67,22 +74,41 @@ DSPConfigDialogHLE::DSPConfigDialogHLE(wxWindow *parent, wxWindowID id,
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wxT("Disabling this could cause abnormal game speed, such as too fast.\n")
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wxT("But sometimes enabling this could cause constant noise.\n")
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wxT("\nKeyboard Shortcut <TAB>: Hold down to instantly disable Throttle."));
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m_FrequencySelection->
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SetToolTip(wxT("Changing this will have no effect while the emulator is running!"));
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m_BackendSelection->
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SetToolTip(wxT("Changing this will have no effect while the emulator is running!"));
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m_volumeSlider->SetToolTip(wxT("This setting only affects DSound, OpenAL, and PulseAudio."));
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m_volumeSlider->SetToolTip(wxT("This setting only affects DSound, OpenAL, XAudio2, and PulseAudio."));
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// Create sizer and add items to dialog
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wxBoxSizer *sMain = new wxBoxSizer(wxVERTICAL);
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wxBoxSizer *sSettings = new wxBoxSizer(wxHORIZONTAL);
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wxBoxSizer *sBackend = new wxBoxSizer(wxHORIZONTAL);
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wxBoxSizer *sBackend = new wxBoxSizer(wxVERTICAL);
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wxBoxSizer *sFrequency = new wxBoxSizer(wxVERTICAL);
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wxBoxSizer *sButtons = new wxBoxSizer(wxHORIZONTAL);
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sbSettings->Add(m_buttonEnableHLEAudio, 0, wxALL, 5);
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sbSettings->Add(m_buttonEnableDTKMusic, 0, wxALL, 5);
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sbSettings->Add(m_buttonEnableThrottle, 0, wxALL, 5);
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sBackend->Add(BackendText, 0, wxALIGN_CENTER|wxALL, 5);
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sFrequency->Add(FrequencyText, 0, wxALIGN_LEFT|wxALL, 1);
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sFrequency->Add(m_FrequencySelection, 0, wxALL, 1);
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m_FrequencySelection->Append(wxString::FromAscii("48,000 Hz"));
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m_FrequencySelection->Append(wxString::FromAscii("32,000 Hz"));
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#ifdef __APPLE__
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int num = m_FrequencySelection->FindString(wxString::FromAscii(ac_Config.sFrequency));
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#else
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int num = m_FrequencySelection->FindString(wxString::FromAscii(ac_Config.sFrequency.c_str()));
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#endif
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m_FrequencySelection->SetSelection(num);
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sbSettings->Add(sFrequency, 0, wxALL, 7);
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sBackend->Add(BackendText, 0, wxALIGN_LEFT|wxALL, 1);
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sBackend->Add(m_BackendSelection, 0, wxALL, 1);
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sbSettings->Add(sBackend, 0, wxALL, 2);
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sbSettings->Add(sBackend, 0, wxALL, 7);
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sbSettingsV->Add(m_volumeSlider, 1, wxLEFT|wxRIGHT|wxALIGN_CENTER, 6);
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sbSettingsV->Add(m_volumeText, 0, wxALL|wxALIGN_LEFT, 4);
|
||||
|
|
@ -138,8 +164,10 @@ void DSPConfigDialogHLE::SettingsChanged(wxCommandEvent& event)
|
|||
|
||||
#ifdef __APPLE__
|
||||
strncpy(ac_Config.sBackend, m_BackendSelection->GetStringSelection().mb_str(), 128);
|
||||
strncpy(ac_Config.sFrequency, m_FrequencySelection->GetStringSelection().mb_str(), 128);
|
||||
#else
|
||||
ac_Config.sBackend = m_BackendSelection->GetStringSelection().mb_str();
|
||||
ac_Config.sFrequency = m_FrequencySelection->GetStringSelection().mb_str();
|
||||
#endif
|
||||
ac_Config.Update();
|
||||
g_Config.Save();
|
||||
|
|
|
|||
|
|
@ -46,13 +46,16 @@ private:
|
|||
wxCheckBox* m_buttonEnableDTKMusic;
|
||||
wxCheckBox* m_buttonEnableThrottle;
|
||||
wxArrayString wxArrayBackends;
|
||||
wxArrayString wxArrayRates;
|
||||
wxChoice* m_BackendSelection;
|
||||
wxChoice* m_FrequencySelection;
|
||||
|
||||
enum
|
||||
{
|
||||
ID_ENABLE_HLE_AUDIO,
|
||||
ID_ENABLE_DTK_MUSIC,
|
||||
ID_ENABLE_THROTTLE,
|
||||
ID_FREQUENCY,
|
||||
ID_BACKEND,
|
||||
ID_VOLUME
|
||||
};
|
||||
|
|
|
|||
|
|
@ -6,8 +6,8 @@
|
|||
class HLEMixer : public CMixer
|
||||
{
|
||||
public:
|
||||
HLEMixer(unsigned int AISampleRate = 48000, unsigned int DACSampleRate = 48000)
|
||||
: CMixer(AISampleRate, DACSampleRate) {};
|
||||
HLEMixer(unsigned int AISampleRate = 48000, unsigned int DACSampleRate = 48000, unsigned int BackendSampleRate = 32000)
|
||||
: CMixer(AISampleRate, DACSampleRate, BackendSampleRate) {};
|
||||
|
||||
virtual void Premix(short *samples, unsigned int numSamples);
|
||||
};
|
||||
|
|
|
|||
|
|
@ -250,9 +250,15 @@ unsigned short DSP_WriteControlRegister(unsigned short _Value)
|
|||
{
|
||||
if (!Temp.DSPHalt && Temp.DSPInit)
|
||||
{
|
||||
unsigned int AISampleRate, DACSampleRate;
|
||||
unsigned int AISampleRate, DACSampleRate, BackendSampleRate;
|
||||
g_dspInitialize.pGetSampleRate(AISampleRate, DACSampleRate);
|
||||
soundStream = AudioCommon::InitSoundStream(new HLEMixer(AISampleRate, DACSampleRate));
|
||||
std::string frequency = ac_Config.sFrequency;
|
||||
if (frequency == "48,000 Hz")
|
||||
BackendSampleRate = 48000;
|
||||
else
|
||||
BackendSampleRate = 32000;
|
||||
|
||||
soundStream = AudioCommon::InitSoundStream(new HLEMixer(AISampleRate, DACSampleRate, BackendSampleRate));
|
||||
if(!soundStream) PanicAlert("Error starting up sound stream");
|
||||
// Mixer is initialized
|
||||
g_InitMixer = true;
|
||||
|
|
|
|||
Loading…
Reference in New Issue