AudioCommon: move DPL2 decoding into Mixer

Source/Core/AudioCommon/CubebStream.cpp

@@ -21,29 +21,9 @@ long CubebStream::DataCallback(cubeb_stream* stream, void* user_data, const void
   auto* self = static_cast<CubebStream*>(user_data);
 
   if (self->m_stereo)
-  {
     self->m_mixer->Mix(static_cast<short*>(output_buffer), num_frames);
-  }
   else
-  {
+    self->m_mixer->MixSurround(static_cast<float*>(output_buffer), num_frames);
-    size_t required_capacity = num_frames * 2;
-    if (required_capacity > self->m_short_buffer.capacity() ||
-        required_capacity > self->m_floatstereo_buffer.capacity())
-    {
-      INFO_LOG(AUDIO, "Expanding conversion buffers size: %li frames", num_frames);
-      self->m_short_buffer.reserve(required_capacity);
-      self->m_floatstereo_buffer.reserve(required_capacity);
-    }
-
-    self->m_mixer->Mix(self->m_short_buffer.data(), num_frames);
-
-    // s16 to float
-    for (size_t i = 0; i < static_cast<size_t>(num_frames) * 2; ++i)
-      self->m_floatstereo_buffer[i] = self->m_short_buffer[i] / static_cast<float>(1 << 15);
-
-    // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
-    DPL2Decode(self->m_floatstereo_buffer.data(), num_frames, static_cast<float*>(output_buffer));
-  }
 
   return num_frames;
 }

Source/Core/AudioCommon/Mixer.cpp

@@ -7,6 +7,7 @@
 #include <cmath>
 #include <cstring>
 
+#include "AudioCommon/DPL2Decoder.h"
 #include "Common/CommonTypes.h"
 #include "Common/Logging/Log.h"
 #include "Common/MathUtil.h"
@@ -25,6 +26,8 @@ CMixer::CMixer(unsigned int BackendSampleRate) : m_sampleRate(BackendSampleRate)
   m_sound_touch.setSetting(SETTING_SEQUENCE_MS, 62);
   m_sound_touch.setSetting(SETTING_SEEKWINDOW_MS, 28);
   m_sound_touch.setSetting(SETTING_OVERLAP_MS, 8);
+
+  DPL2Reset();
 }
 
 CMixer::~CMixer()
@@ -159,6 +162,25 @@ unsigned int CMixer::Mix(short* samples, unsigned int num_samples)
   return num_samples;
 }
 
+unsigned int CMixer::MixSurround(float* samples, unsigned int num_samples)
+{
+  if (!num_samples)
+    return 0;
+
+  memset(samples, 0, num_samples * 6 * sizeof(float));
+
+  unsigned int available_samples = Mix(m_stretch_buffer.data(), num_samples);
+  for (size_t i = 0; i < static_cast<size_t>(available_samples) * 2; ++i)
+  {
+    m_float_conversion_buffer[i] =
+        m_stretch_buffer[i] / static_cast<float>(std::numeric_limits<short>::max());
+  }
+
+  DPL2Decode(m_float_conversion_buffer.data(), available_samples, samples);
+
+  return available_samples;
+}
+
 void CMixer::StretchAudio(const short* in, unsigned int num_in, short* out, unsigned int num_out)
 {
   const double time_delta = static_cast<double>(num_out) / m_sampleRate;  // seconds

Source/Core/AudioCommon/Mixer.h

@@ -21,6 +21,7 @@ public:
 
   // Called from audio threads
   unsigned int Mix(short* samples, unsigned int numSamples);
+  unsigned int MixSurround(float* samples, unsigned int num_samples);
 
   // Called from main thread
   void PushSamples(const short* samples, unsigned int num_samples);
@@ -87,6 +88,7 @@ private:
   double m_stretch_ratio = 1.0;
   std::array<short, 2> m_last_stretched_sample = {};
   std::array<short, MAX_SAMPLES * 2> m_stretch_buffer;
+  std::array<float, MAX_SAMPLES * 2> m_float_conversion_buffer;
 
   WaveFileWriter m_wave_writer_dtk;
   WaveFileWriter m_wave_writer_dsp;

Source/Core/AudioCommon/OpenALStream.cpp

@@ -6,7 +6,6 @@
 #include <cstring>
 #include <thread>
 
-#include "AudioCommon/DPL2Decoder.h"
 #include "AudioCommon/OpenALStream.h"
 #include "AudioCommon/aldlist.h"
 #include "Common/Logging/Log.h"
@@ -66,9 +65,6 @@ bool OpenALStream::Start()
     PanicAlertT("OpenAL: can't find sound devices");
   }
 
-  // Initialize DPL2 parameters
-  DPL2Reset();
-
   return bReturn;
 }
 
@@ -228,23 +224,18 @@ void OpenALStream::SoundLoop()
       numBuffersQueued -= numBuffersProcessed;
     }
 
-    // DPL2 accepts 240 samples minimum (FWRDURATION)
-    unsigned int minSamples = surround_capable ? 240 : 0;
-
     unsigned int numSamples = OAL_MAX_SAMPLES;
-    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);
-
-    if (numSamples <= minSamples)
-      continue;
 
     if (surround_capable)
     {
+      // DPL2 accepts 240 samples minimum (FWRDURATION)
+      unsigned int minSamples = 240;
+
       float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
-      DPL2Decode(sampleBuffer, numSamples, dpl2);
+      numSamples = m_mixer->MixSurround(dpl2, numSamples);
+
+      if (numSamples < minSamples)
+        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
@@ -311,6 +302,15 @@ 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);
+
+      if (!numSamples)
+        continue;
+
       if (float32_capable)
       {
         alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,

Source/Core/AudioCommon/PulseAudioStream.cpp

@@ -4,7 +4,6 @@
 
 #include <cstring>
 
-#include "AudioCommon/DPL2Decoder.h"
 #include "AudioCommon/PulseAudioStream.h"
 #include "Common/CommonTypes.h"
 #include "Common/Logging/Log.h"
@@ -30,9 +29,6 @@ bool PulseAudio::Start()
   m_run_thread.Set();
   m_thread = std::thread(&PulseAudio::SoundLoop, this);
 
-  // Initialize DPL2 parameters
-  DPL2Reset();
-
   return true;
 }
 
@@ -194,23 +190,12 @@ void PulseAudio::WriteCallback(pa_stream* s, size_t length)
   }
   else
   {
-    // get a floating point mix
-    s16 s16buffer_stereo[frames * 2];
-    m_mixer->Mix(s16buffer_stereo, frames);  // implicitly mixes to 16-bit stereo
-
-    float floatbuffer_stereo[frames * 2];
-    // s16 to float
-    for (int i = 0; i < frames * 2; ++i)
-    {
-      floatbuffer_stereo[i] = s16buffer_stereo[i] / float(1 << 15);
-    }
-
     if (m_channels == 5)  // Extract dpl2/5.0 Surround
     {
       float floatbuffer_6chan[frames * 6];
-      // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
+      m_mixer->MixSurround(floatbuffer_6chan, frames);
-      DPL2Decode(floatbuffer_stereo, frames, floatbuffer_6chan);
 
+      // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
       // Discard the subwoofer channel - DPL2Decode generates a pretty
       // good 5.0 but not a good 5.1 output.
       const int dpl2_to_5chan[] = {0, 1, 2, 4, 5};