Merge pull request #5631 from LAGonauta/openal-real-latency

Fixed latency setting and cleaned-up OpenAL backend

5 files changed, 115 insertions, 129 deletions

diff --git a/Source/Core/AudioCommon/OpenALStream.cpp b/Source/Core/AudioCommon/OpenALStream.cpp
index 21b2ae0acc..191452b7f3 100644
--- a/Source/Core/AudioCommon/OpenALStream.cpp
+++ b/Source/Core/AudioCommon/OpenALStream.cpp
@@ -100,27 +100,27 @@ bool OpenALStream::Start()
     return false;
   }
 
-  const char* defaultDeviceName = palcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
-  INFO_LOG(AUDIO, "Found OpenAL device %s", defaultDeviceName);
+  const char* default_device_dame = palcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
+  INFO_LOG(AUDIO, "Found OpenAL device %s", default_device_dame);
 
-  ALCdevice* pDevice = palcOpenDevice(defaultDeviceName);
-  if (!pDevice)
+  ALCdevice* device = palcOpenDevice(default_device_dame);
+  if (!device)
   {
-    PanicAlertT("OpenAL: can't open device %s", defaultDeviceName);
+    PanicAlertT("OpenAL: can't open device %s", default_device_dame);
     return false;
   }
 
-  ALCcontext* pContext = palcCreateContext(pDevice, nullptr);
-  if (!pContext)
+  ALCcontext* context = palcCreateContext(device, nullptr);
+  if (!context)
   {
-    palcCloseDevice(pDevice);
-    PanicAlertT("OpenAL: can't create context for device %s", defaultDeviceName);
+    palcCloseDevice(device);
+    PanicAlertT("OpenAL: can't create context for device %s", default_device_dame);
     return false;
   }
 
-  palcMakeContextCurrent(pContext);
+  palcMakeContextCurrent(context);
   m_run_thread.Set();
-  thread = std::thread(&OpenALStream::SoundLoop, this);
+  m_thread = std::thread(&OpenALStream::SoundLoop, this);
   return true;
 }
 
@@ -128,37 +128,37 @@ void OpenALStream::Stop()
 {
   m_run_thread.Clear();
   // kick the thread if it's waiting
-  soundSyncEvent.Set();
+  m_sound_sync_event.Set();
 
-  thread.join();
+  m_thread.join();
 
-  palSourceStop(uiSource);
-  palSourcei(uiSource, AL_BUFFER, 0);
+  palSourceStop(m_source);
+  palSourcei(m_source, AL_BUFFER, 0);
 
   // Clean up buffers and sources
-  palDeleteSources(1, &uiSource);
-  uiSource = 0;
-  palDeleteBuffers(numBuffers, uiBuffers);
+  palDeleteSources(1, &m_source);
+  m_source = 0;
+  palDeleteBuffers(OAL_BUFFERS, m_buffers.data());
 
-  ALCcontext* pContext = palcGetCurrentContext();
-  ALCdevice* pDevice = palcGetContextsDevice(pContext);
+  ALCcontext* context = palcGetCurrentContext();
+  ALCdevice* device = palcGetContextsDevice(context);
 
   palcMakeContextCurrent(nullptr);
-  palcDestroyContext(pContext);
-  palcCloseDevice(pDevice);
+  palcDestroyContext(context);
+  palcCloseDevice(device);
 }
 
 void OpenALStream::SetVolume(int volume)
 {
-  fVolume = (float)volume / 100.0f;
+  m_volume = (float)volume / 100.0f;
 
-  if (uiSource)
-    palSourcef(uiSource, AL_GAIN, fVolume);
+  if (m_source)
+    palSourcef(m_source, AL_GAIN, m_volume);
 }
 
 void OpenALStream::Update()
 {
-  soundSyncEvent.Set();
+  m_sound_sync_event.Set();
 }
 
 void OpenALStream::Clear(bool mute)
@@ -167,11 +167,11 @@ void OpenALStream::Clear(bool mute)
 
   if (m_muted)
   {
-    palSourceStop(uiSource);
+    palSourceStop(m_source);
   }
   else
   {
-    palSourcePlay(uiSource);
+    palSourcePlay(m_source);
   }
 }
 
@@ -229,86 +229,109 @@ void OpenALStream::SoundLoop()
   // we just check if one is being used.
   bool fixed32_capable = IsCreativeXFi();
 
-  u32 ulFrequency = m_mixer->GetSampleRate();
-  numBuffers = SConfig::GetInstance().iLatency + 2;  // OpenAL requires a minimum of two buffers
+  u32 frequency = m_mixer->GetSampleRate();
 
-  memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
-  uiSource = 0;
+  u32 frames_per_buffer;
+  // Can't have zero samples per buffer
+  if (SConfig::GetInstance().iLatency > 0)
+  {
+    frames_per_buffer = frequency / 1000 * SConfig::GetInstance().iLatency / OAL_BUFFERS;
+  }
+  else
+  {
+    frames_per_buffer = frequency / 1000 * 1 / OAL_BUFFERS;
+  }
+
+  if (frames_per_buffer > OAL_MAX_FRAMES)
+  {
+    frames_per_buffer = OAL_MAX_FRAMES;
+  }
+
+  // DPL2 needs a minimum number of samples to work (FWRDURATION)
+  if (use_surround && frames_per_buffer < 240)
+  {
+    frames_per_buffer = 240;
+  }
+
+  INFO_LOG(AUDIO, "Using %d buffers, each with %d audio frames for a total of %d.", OAL_BUFFERS,
+           frames_per_buffer, frames_per_buffer * OAL_BUFFERS);
+
+  // Should we make these larger just in case the mixer ever sends more samples
+  // than what we request?
+  m_realtime_buffer.resize(frames_per_buffer * STEREO_CHANNELS);
+  m_source = 0;
 
   // Clear error state before querying or else we get false positives.
   ALenum err = palGetError();
 
   // Generate some AL Buffers for streaming
-  palGenBuffers(numBuffers, (ALuint*)uiBuffers);
+  palGenBuffers(OAL_BUFFERS, (ALuint*)m_buffers.data());
   err = CheckALError("generating buffers");
 
   // Generate a Source to playback the Buffers
-  palGenSources(1, &uiSource);
+  palGenSources(1, &m_source);
   err = CheckALError("generating sources");
 
   // Set the default sound volume as saved in the config file.
-  palSourcef(uiSource, AL_GAIN, fVolume);
+  palSourcef(m_source, AL_GAIN, m_volume);
 
   // TODO: Error handling
   // ALenum err = alGetError();
 
-  unsigned int nextBuffer = 0;
-  unsigned int numBuffersQueued = 0;
-  ALint iState = 0;
+  unsigned int next_buffer = 0;
+  unsigned int num_buffers_queued = 0;
+  ALint state = 0;
 
   while (m_run_thread.IsSet())
   {
     // Block until we have a free buffer
-    int numBuffersProcessed;
-    palGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
-    if (numBuffers == numBuffersQueued && !numBuffersProcessed)
+    int num_buffers_processed;
+    palGetSourcei(m_source, AL_BUFFERS_PROCESSED, &num_buffers_processed);
+    if (num_buffers_queued == OAL_BUFFERS && !num_buffers_processed)
     {
-      soundSyncEvent.Wait();
+      std::this_thread::sleep_for(std::chrono::milliseconds(1));
       continue;
     }
 
     // Remove the Buffer from the Queue.
-    if (numBuffersProcessed)
+    if (num_buffers_processed)
     {
-      ALuint unqueuedBufferIds[OAL_MAX_BUFFERS];
-      palSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds);
+      std::array<ALuint, OAL_BUFFERS> unqueued_buffer_ids;
+      palSourceUnqueueBuffers(m_source, num_buffers_processed, unqueued_buffer_ids.data());
       err = CheckALError("unqueuing buffers");
 
-      numBuffersQueued -= numBuffersProcessed;
+      num_buffers_queued -= num_buffers_processed;
     }
 
-    unsigned int numSamples = OAL_MAX_SAMPLES;
+    unsigned int min_frames = frames_per_buffer;
 
     if (use_surround)
     {
-      // DPL2 accepts 240 samples minimum (FWRDURATION)
-      unsigned int minSamples = 240;
-
-      float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
-      numSamples = m_mixer->MixSurround(dpl2, numSamples);
+      std::array<float, OAL_MAX_FRAMES * SURROUND_CHANNELS> dpl2;
+      u32 rendered_frames = m_mixer->MixSurround(dpl2.data(), min_frames);
 
-      if (numSamples < minSamples)
+      if (rendered_frames < min_frames)
         continue;
 
       // zero-out the subwoofer channel - DPL2Decode generates a pretty
       // good 5.0 but not a good 5.1 output.  Sadly there is not a 5.0
       // AL_FORMAT_50CHN32 to make this super-explicit.
       // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
-      for (u32 i = 0; i < numSamples; ++i)
+      for (u32 i = 0; i < rendered_frames; ++i)
       {
         dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
       }
 
       if (float32_capable)
       {
-        palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
-                      numSamples * FRAME_SURROUND_FLOAT, ulFrequency);
+        palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, dpl2.data(),
+                      rendered_frames * FRAME_SURROUND_FLOAT, frequency);
       }
       else if (fixed32_capable)
       {
-        int surround_int32[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
+        std::array<int, OAL_MAX_FRAMES * SURROUND_CHANNELS> surround_int32;
 
-        for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i)
+        for (u32 i = 0; i < rendered_frames * SURROUND_CHANNELS; ++i)
         {
           // For some reason the ffdshow's DPL2 decoder outputs samples bigger than 1.
           // Most are close to 2.5 and some go up to 8. Hard clamping here, we need to
@@ -319,17 +342,17 @@ void OpenALStream::SoundLoop()
           else if (dpl2[i] < INT_MIN)
             surround_int32[i] = INT_MIN;
           else
-            surround_int32[i] = (int)dpl2[i];
+            surround_int32[i] = static_cast<int>(dpl2[i]);
         }
 
-        palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, surround_int32,
-                      numSamples * FRAME_SURROUND_INT32, ulFrequency);
+        palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, surround_int32.data(),
+                      rendered_frames * FRAME_SURROUND_INT32, frequency);
       }
       else
       {
-        short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
+        std::array<short, OAL_MAX_FRAMES * SURROUND_CHANNELS> surround_short;
 
-        for (u32 i = 0; i < numSamples * SURROUND_CHANNELS; ++i)
+        for (u32 i = 0; i < rendered_frames * SURROUND_CHANNELS; ++i)
         {
           dpl2[i] = dpl2[i] * (1 << 15);
           if (dpl2[i] > SHRT_MAX)
@@ -337,11 +360,11 @@ void OpenALStream::SoundLoop()
           else if (dpl2[i] < SHRT_MIN)
             surround_short[i] = SHRT_MIN;
           else
-            surround_short[i] = (int)dpl2[i];
+            surround_short[i] = static_cast<int>(dpl2[i]);
         }
 
-        palBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
-                      numSamples * FRAME_SURROUND_SHORT, ulFrequency);
+        palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN16, surround_short.data(),
+                      rendered_frames * FRAME_SURROUND_SHORT, frequency);
       }
 
       err = CheckALError("buffering data");
@@ -355,59 +378,26 @@ void OpenALStream::SoundLoop()
     }
     else
     {
-      numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
-
-      // Convert the samples from short to float
-      for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
-        sampleBuffer[i] = static_cast<float>(realtimeBuffer[i]) / (1 << 15);
+      u32 rendered_frames = m_mixer->Mix(m_realtime_buffer.data(), min_frames);
 
-      if (!numSamples)
+      if (!rendered_frames)
         continue;
 
-      if (float32_capable)
-      {
-        palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
-                      numSamples * FRAME_STEREO_FLOAT, ulFrequency);
-
-        err = CheckALError("buffering float32 data");
-        if (err == AL_INVALID_ENUM)
-        {
-          float32_capable = false;
-        }
-      }
-      else if (fixed32_capable)
-      {
-        // Clamping is not necessary here, samples are always between (-1,1)
-        int stereo_int32[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
-        for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
-          stereo_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31));
-
-        palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO32, stereo_int32,
-                      numSamples * FRAME_STEREO_INT32, ulFrequency);
-      }
-      else
-      {
-        // Convert the samples from float to short
-        short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
-        for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
-          stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
-
-        palBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
-                      numSamples * FRAME_STEREO_SHORT, ulFrequency);
-      }
+      palBufferData(m_buffers[next_buffer], AL_FORMAT_STEREO16, m_realtime_buffer.data(),
+                    rendered_frames * FRAME_STEREO_SHORT, frequency);
     }
 
-    palSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
+    palSourceQueueBuffers(m_source, 1, &m_buffers[next_buffer]);
     err = CheckALError("queuing buffers");
 
-    numBuffersQueued++;
-    nextBuffer = (nextBuffer + 1) % numBuffers;
+    num_buffers_queued++;
+    next_buffer = (next_buffer + 1) % OAL_BUFFERS;
 
-    palGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
-    if (iState != AL_PLAYING)
+    palGetSourcei(m_source, AL_SOURCE_STATE, &state);
+    if (state != AL_PLAYING)
     {
       // Buffer underrun occurred, resume playback
-      palSourcePlay(uiSource);
+      palSourcePlay(m_source);
       err = CheckALError("occurred resuming playback");
     }
   }
diff --git a/Source/Core/AudioCommon/OpenALStream.h b/Source/Core/AudioCommon/OpenALStream.h
index 78c0a616a1..3b9364c571 100644
--- a/Source/Core/AudioCommon/OpenALStream.h
+++ b/Source/Core/AudioCommon/OpenALStream.h
@@ -17,17 +17,15 @@
 #include <OpenAL/include/alc.h>
 #include <OpenAL/include/alext.h>
 
-#define SFX_MAX_SOURCE 1
-#define OAL_MAX_BUFFERS 32
-#define OAL_MAX_SAMPLES 256
+// OpenAL requires a minimum of two buffers, three or more recommended
+#define OAL_BUFFERS 3
+#define OAL_MAX_FRAMES 4096
 #define STEREO_CHANNELS 2
 #define SURROUND_CHANNELS 6  // number of channels in surround mode
 #define SIZE_SHORT 2
 #define SIZE_INT32 4
 #define SIZE_FLOAT 4  // size of a float in bytes
 #define FRAME_STEREO_SHORT STEREO_CHANNELS* SIZE_SHORT
-#define FRAME_STEREO_FLOAT STEREO_CHANNELS* SIZE_FLOAT
-#define FRAME_STEREO_INT32 STEREO_CHANNELS* SIZE_INT32
 #define FRAME_SURROUND_FLOAT SURROUND_CHANNELS* SIZE_FLOAT
 #define FRAME_SURROUND_SHORT SURROUND_CHANNELS* SIZE_SHORT
 #define FRAME_SURROUND_INT32 SURROUND_CHANNELS* SIZE_INT32
@@ -56,7 +54,7 @@ class OpenALStream final : public SoundStream
 {
 #ifdef _WIN32
 public:
-  OpenALStream() : uiSource(0) {}
+  OpenALStream() : m_source(0) {}
   bool Start() override;
   void SoundLoop() override;
   void SetVolume(int volume) override;
@@ -67,17 +65,15 @@ public:
   static bool isValid();
 
 private:
-  std::thread thread;
+  std::thread m_thread;
   Common::Flag m_run_thread;
 
-  Common::Event soundSyncEvent;
+  Common::Event m_sound_sync_event;
 
-  short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
-  float sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
-  ALuint uiBuffers[OAL_MAX_BUFFERS];
-  ALuint uiSource;
-  ALfloat fVolume;
+  std::vector<short> m_realtime_buffer;
+  std::array<ALuint, OAL_BUFFERS> m_buffers;
+  ALuint m_source;
+  ALfloat m_volume;
 
-  u8 numBuffers;
 #endif  // _WIN32
 };
diff --git a/Source/Core/Core/ConfigManager.cpp b/Source/Core/Core/ConfigManager.cpp
index 19af3de572..c2e6940042 100644
--- a/Source/Core/Core/ConfigManager.cpp
+++ b/Source/Core/Core/ConfigManager.cpp
@@ -254,7 +254,7 @@ void SConfig::SaveCoreSettings(IniFile& ini)
   core->Set("SelectedLanguage", SelectedLanguage);
   core->Set("OverrideGCLang", bOverrideGCLanguage);
   core->Set("DPL2Decoder", bDPL2Decoder);
-  core->Set("Latency", iLatency);
+  core->Set("AudioLatency", iLatency);
   core->Set("AudioStretch", m_audio_stretch);
   core->Set("AudioStretchMaxLatency", m_audio_stretch_max_latency);
   core->Set("MemcardAPath", m_strMemoryCardA);
@@ -568,7 +568,7 @@ void SConfig::LoadCoreSettings(IniFile& ini)
   core->Get("SelectedLanguage", &SelectedLanguage, 0);
   core->Get("OverrideGCLang", &bOverrideGCLanguage, false);
   core->Get("DPL2Decoder", &bDPL2Decoder, false);
-  core->Get("Latency", &iLatency, 5);
+  core->Get("AudioLatency", &iLatency, 20);
   core->Get("AudioStretch", &m_audio_stretch, false);
   core->Get("AudioStretchMaxLatency", &m_audio_stretch_max_latency, 80);
   core->Get("MemcardAPath", &m_strMemoryCardA);
@@ -831,7 +831,7 @@ void SConfig::LoadDefaults()
   bOverrideGCLanguage = false;
   bWii = false;
   bDPL2Decoder = false;
-  iLatency = 14;
+  iLatency = 20;
   m_audio_stretch = false;
   m_audio_stretch_max_latency = 80;
 
diff --git a/Source/Core/Core/ConfigManager.h b/Source/Core/Core/ConfigManager.h
index 9a97b68df0..e40b6bf320 100644
--- a/Source/Core/Core/ConfigManager.h
+++ b/Source/Core/Core/ConfigManager.h
@@ -109,7 +109,7 @@ struct SConfig : NonCopyable
   bool bCopyWiiSaveNetplay = true;
 
   bool bDPL2Decoder = false;
-  int iLatency = 14;
+  int iLatency = 20;
   bool m_audio_stretch = false;
   int m_audio_stretch_max_latency = 80;
 
diff --git a/Source/Core/DolphinWX/Config/AudioConfigPane.cpp b/Source/Core/DolphinWX/Config/AudioConfigPane.cpp
index 5e940ffb22..c9f3c6b0e1 100644
--- a/Source/Core/DolphinWX/Config/AudioConfigPane.cpp
+++ b/Source/Core/DolphinWX/Config/AudioConfigPane.cpp
@@ -45,8 +45,8 @@ void AudioConfigPane::InitializeGUI()
   m_audio_backend_choice =
       new wxChoice(this, wxID_ANY, wxDefaultPosition, wxDefaultSize, m_audio_backend_strings);
   m_audio_latency_spinctrl =
-      new wxSpinCtrl(this, wxID_ANY, "", wxDefaultPosition, wxDefaultSize, wxSP_ARROW_KEYS, 0, 30);
-  m_audio_latency_label = new wxStaticText(this, wxID_ANY, _("Latency:"));
+      new wxSpinCtrl(this, wxID_ANY, "", wxDefaultPosition, wxDefaultSize, wxSP_ARROW_KEYS, 0, 200);
+  m_audio_latency_label = new wxStaticText(this, wxID_ANY, _("Latency (ms):"));
 
   m_stretch_checkbox = new wxCheckBox(this, wxID_ANY, _("Enable Audio Stretching"));
   m_stretch_label = new wxStaticText(this, wxID_ANY, _("Buffer Size:"));