Merge branch 'master' of github.com:marcel303/framework

marcel303 · Jun 13, 2022 · 134036c · 134036c
2 parents 3be2bb7 + 5d71787
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diff --git a/README.md b/README.md
@@ -1,8 +1,9 @@
 | Platform | Build status
 |:--|:--|
-| macOS | [![Build status](https://ci.appveyor.com/api/projects/status/kqradcg3ruto0uhy?svg=true)](https://ci.appveyor.com/project/MarcelSmit/framework-macos) |
-| Ubuntu linux | [![Build status](https://ci.appveyor.com/api/projects/status/1spm8vp61ax61yox?svg=true)](https://ci.appveyor.com/project/MarcelSmit/framework-ubuntu) |
-| Windows | [![Build status](https://ci.appveyor.com/api/projects/status/c82webixs5kbw6xt?svg=true)](https://ci.appveyor.com/project/MarcelSmit/framework) |
+| macOS | [![Build status](https://ci.appveyor.com/api/projects/status/kqradcg3ruto0uhy?svg=true)](https://ci.appveyor.com/project/MarcelSmit/framework-macos) |
+| Ubuntu linux | [![Build status](https://ci.appveyor.com/api/projects/status/1spm8vp61ax61yox?svg=true)](https://ci.appveyor.com/project/MarcelSmit/framework-ubuntu) |
+| Windows - VS 2019 | [![Build status](https://ci.appveyor.com/api/projects/status/c82webixs5kbw6xt?svg=true)](https://ci.appveyor.com/project/MarcelSmit/framework) |
+| Windows - VS 2022 | [![Build status](https://ci.appveyor.com/api/projects/status/qsm90rcfvid4qoaf?svg=true)](https://ci.appveyor.com/project/MarcelSmit/framework-windows-vs2022) |
 
 # README #
 

diff --git a/audioGraph/audioGraphCore/audioTypes-compat.h b/audioGraph/audioGraphCore/audioTypes-compat.h
@@ -0,0 +1,73 @@
+#include "audioTypes.h"
+
+#if AUDIO_USE_SSE && defined(WIN32)
+
+// Clang and GCC support this nice syntax where vector types support the same basic operations as floats or integers. on Windows we need to re-implement a subset here to make the code compile
+
+inline __m128 operator+(__m128 a, __m128 b)
+{
+ return _mm_add_ps(a, b);
+}
+
+inline __m128 operator-(__m128 a, __m128 b)
+{
+ return _mm_sub_ps(a, b);
+}
+
+inline __m128 operator*(__m128 a, __m128 b)
+{
+ return _mm_mul_ps(a, b);
+}
+
+inline __m128d operator+(__m128d a, __m128d b)
+{
+ return _mm_add_pd(a, b);
+}
+
+inline __m128d operator-(__m128d a, __m128d b)
+{
+ return _mm_sub_pd(a, b);
+}
+
+inline __m128d operator*(__m128d a, __m128d b)
+{
+ return _mm_mul_pd(a, b);
+}
+
+#if __AVX__
+
+#include <immintrin.h>
+
+inline __m256 operator+(__m256 a, __m256 b)
+{
+ return _mm256_add_ps(a, b);
+}
+
+inline __m256 operator-(__m256 a, __m256 b)
+{
+ return _mm256_sub_ps(a, b);
+}
+
+inline __m256 operator*(__m256 a, __m256 b)
+{
+ return _mm256_mul_ps(a, b);
+}
+
+inline __m256d operator+(__m256d a, __m256d b)
+{
+ return _mm256_add_pd(a, b);
+}
+
+inline __m256d operator-(__m256d a, __m256d b)
+{
+ return _mm256_sub_pd(a, b);
+}
+
+inline __m256d operator*(__m256d a, __m256d b)
+{
+ return _mm256_mul_pd(a, b);
+}
+
+#endif
+
+#endif
diff --git a/audioGraph/audioGraphCore/audioTypes.h b/audioGraph/audioGraphCore/audioTypes.h
@@ -144,78 +144,6 @@
  #define SCOPED_FLUSH_DENORMALS do { } while (false)
 #endif
 
-#if AUDIO_USE_SSE && defined(WIN32)
-
-// Clang and GCC support this nice syntax where vector types support the same basic operations as floats or integers. on Windows we need to re-implement a subset here to make the code compile
-
-inline __m128 operator+(__m128 a, __m128 b)
-{
- return _mm_add_ps(a, b);
-}
-
-inline __m128 operator-(__m128 a, __m128 b)
-{
- return _mm_sub_ps(a, b);
-}
-
-inline __m128 operator*(__m128 a, __m128 b)
-{
- return _mm_mul_ps(a, b);
-}
-
-inline __m128d operator+(__m128d a, __m128d b)
-{
- return _mm_add_pd(a, b);
-}
-
-inline __m128d operator-(__m128d a, __m128d b)
-{
- return _mm_sub_pd(a, b);
-}
-
-inline __m128d operator*(__m128d a, __m128d b)
-{
- return _mm_mul_pd(a, b);
-}
-
-#if __AVX__
-
-#include <immintrin.h>
-
-inline __m256 operator+(__m256 a, __m256 b)
-{
- return _mm256_add_ps(a, b);
-}
-
-inline __m256 operator-(__m256 a, __m256 b)
-{
- return _mm256_sub_ps(a, b);
-}
-
-inline __m256 operator*(__m256 a, __m256 b)
-{
- return _mm256_mul_ps(a, b);
-}
-
-inline __m256d operator+(__m256d a, __m256d b)
-{
- return _mm256_add_pd(a, b);
-}
-
-inline __m256d operator-(__m256d a, __m256d b)
-{
- return _mm256_sub_pd(a, b);
-}
-
-inline __m256d operator*(__m256d a, __m256d b)
-{
- return _mm256_mul_pd(a, b);
-}
-
-#endif
-
-#endif
-
 struct AudioControlValue
 {
  enum Type
@@ -280,4 +208,3 @@ struct AudioRNG
 #else
  #define ALIGNED_AUDIO_NEW_AND_DELETE()
 #endif
-
diff --git a/audioGraph/audioGraphCore/soundmix.cpp b/audioGraph/audioGraphCore/soundmix.cpp
@@ -25,6 +25,7 @@
  OTHER DEALINGS IN THE SOFTWARE.
 */
 
+#include "audioTypes-compat.h"
 #include "soundfile/SoundIO.h"
 #include "soundmix.h"
 

diff --git a/audioGraph/objects/wavefield.cpp b/audioGraph/objects/wavefield.cpp
@@ -25,6 +25,7 @@
  OTHER DEALINGS IN THE SOFTWARE.
 */
 
+#include "audioTypes-compat.h"
 #include "Noise.h"
 #include "wavefield.h"
 #include <algorithm>

diff --git a/framework/framework.cpp b/framework/framework.cpp
@@ -2159,6 +2159,7 @@ void Framework::present()
  {
  if (window->getWindowData()->needsPresent)
  {
+ SDL_GL_MakeCurrent(window->getWindow(), globals.glContext);
  SDL_GL_SwapWindow(window->getWindow());
  window->getWindowData()->needsPresent = false;
  }

diff --git a/users/marcel/flacdecode1/main.cpp b/users/marcel/flacdecode1/main.cpp
@@ -31,22 +31,41 @@
 
 #include <stdio.h>
 
+#ifdef _MSC_VER
+ #include <intrin.h>
+#endif
+
 struct Integer
 {
  static int numberOfLeadingZeros(const uint32_t value)
  {
  return
  value == 0
  ? 32
+ #ifdef _MSC_VER
+ : __lzcnt(value);
+ #else
  : __builtin_clz(value);
+ #endif
  }
 
  static int numberOfLeadingZeros64(const uint64_t value)
  {
+ #ifdef _MSC_VER
+ if (value == 0)
+ return 64;
+
+ const int n = numberOfLeadingZeros(uint32_t(value >> 32));
+ if (n != 0)
+ return n;
+
+ return 32 + numberOfLeadingZeros(uint32_t(value));
+ #else
  return
  value == 0
  ? 64
  : __builtin_clzll(value);
+ #endif
  }
 };
 

diff --git a/users/marcel/ovr-cubes/data/mod/Chrysilis.s3m b/users/marcel/ovr-cubes/data/mod/Chrysilis.s3m
diff --git a/users/marcel/ovr-cubes/data/veemsounds.ogg b/users/marcel/ovr-cubes/data/veemsounds.ogg
diff --git a/users/marcel/ovr-cubes/main6.cpp b/users/marcel/ovr-cubes/main6.cpp
@@ -332,8 +332,8 @@ struct CelestialSphereObject
 
  for (int i = 0; i < numStars; ++i)
  {
- const float a1 = random<float>(-M_PI, +M_PI);
- const float a2 = random<float>(-M_PI, +M_PI);
+ const float a1 = random<float>(-float(M_PI), +float(M_PI));
+ const float a2 = random<float>(-float(M_PI), +float(M_PI));
 
  const Mat4x4 rotation = Mat4x4(true).RotateX(a1).RotateY(a2);
  const Vec4 position = rotation * Vec4(0, 0, 1, 1);
@@ -1191,7 +1191,8 @@ void Scene::create()
  //jgplayer.init("K_vision.s3m");
  //jgplayer.init("25 - Surfacing.s3m");
  //jgplayer.init("05 - Shared Dig.s3m");
- jgplayer.init("UNATCO.it");
+ //jgplayer.init("UNATCO.it");
+ jgplayer.init("mod/Chrysilis.s3m");
 
  parameterMgr.addChild(&jgplayer.parameterMgr, ++jgplayerIdx);
  }
@@ -1799,9 +1800,6 @@ struct SpatialAudioSystemAudioStream : AudioStream
 
  virtual int Provide(int numSamples, AudioSample * __restrict samples) override final
  {
- ALIGN16 float outputSamplesL[numSamples];
- ALIGN16 float outputSamplesR[numSamples];
-
  // note : we update the audio in steps of 32 samples, to improve the timing for notes when the sample size is
  // large. hopefully this fixes strange timing issues during Oculus Vr movie recordings. also this will
  // reduce (the maximum) binauralization latency by a little and make it more stable/less jittery
@@ -1814,37 +1812,40 @@ struct SpatialAudioSystemAudioStream : AudioStream
  ? numSamplesRemaining
  : kUpdateSize;
 
+ ALIGN16 float outputSamplesL[kUpdateSize];
+ ALIGN16 float outputSamplesR[kUpdateSize];
+
  spatialAudioSystem->generateLR(
- outputSamplesL + i,
- outputSamplesR + i,
+ outputSamplesL,
+ outputSamplesR,
  numSamplesThisUpdate);
 
- i += numSamplesThisUpdate;
- }
-
- #if AUDIO_BUFFER_VALIDATION
- for (int i = 0; i < numSamples; ++i)
- Assert(isfinite(outputSamplesL[i]) && isfinite(outputSamplesR[i]));
- #endif
+ #if AUDIO_BUFFER_VALIDATION
+ for (int j = 0; j < numSamplesThisUpdate; ++j)
+ Assert(isfinite(outputSamplesL[j]) && isfinite(outputSamplesR[j]));
+ #endif
 
- limiter.chunk_analyze(outputSamplesL, numSamples, 16);
- limiter.chunk_analyze(outputSamplesR, numSamples, 16);
- limiter.chunk_applyInPlace(outputSamplesL, numSamples, 16, 1.f);
- limiter.chunk_applyInPlace(outputSamplesR, numSamples, 16, 1.f);
- limiter.chunk_end(powf(.995f, numSamples/256.f));
+  limiter.chunk_analyze(outputSamplesL, numSamplesThisUpdate, 16);
+  limiter.chunk_analyze(outputSamplesR, numSamplesThisUpdate, 16);
+  limiter.chunk_applyInPlace(outputSamplesL, numSamplesThisUpdate, 16, 1.f);
+  limiter.chunk_applyInPlace(outputSamplesR, numSamplesThisUpdate, 16, 1.f);
+  limiter.chunk_end(powf(.995f, numSamplesThisUpdate/256.f));
 
- const float scale_s16 = (1 << 15) - 1;
+  const float scale_s16 = (1 << 15) - 1;
 
- for (int i = 0; i < numSamples; ++i)
- {
- const int32_t valueL = int32_t(outputSamplesL[i] * scale_s16);
- const int32_t valueR = int32_t(outputSamplesR[i] * scale_s16);
-
- Assert(valueL >= -((1 << 15) - 1) && valueL <= +((1 << 15) - 1));
- Assert(valueR >= -((1 << 15) - 1) && valueR <= +((1 << 15) - 1));
-
- samples[i].channel[0] = valueL;
- samples[i].channel[1] = valueR;
+ for (int j = 0; j < numSamplesThisUpdate; ++j)
+ {
+ const int32_t valueL = int32_t(outputSamplesL[j] * scale_s16);
+ const int32_t valueR = int32_t(outputSamplesR[j] * scale_s16);
+
+ Assert(valueL >= -((1 << 15) - 1) && valueL <= +((1 << 15) - 1));
+ Assert(valueR >= -((1 << 15) - 1) && valueR <= +((1 << 15) - 1));
+
+ samples[i + j].channel[0] = valueL;
+ samples[i + j].channel[1] = valueR;
+ }
+
+ i += numSamplesThisUpdate;
  }
 
  return numSamples;

diff --git a/users/marcel/pitchshift1/pshifter.cpp b/users/marcel/pitchshift1/pshifter.cpp
@@ -24,6 +24,7 @@
 
 #include "fourier.h"
 
+#include "Debugging.h"
 #include "Log.h"
 
 #ifdef HAVE_SSE_INTRINSICS
@@ -95,8 +96,12 @@ static int double2int(double value)
 
 template <typename T> static void complex_fft(T & buffer, double direction)
 {
- ALIGN16 double real[buffer.size()];
- ALIGN16 double imag[buffer.size()];
+ const int kMaxBufferSize = 4096;
+
+ Assert(buffer.size() <= kMaxBufferSize);
+
+ ALIGN16 double real[kMaxBufferSize];
+ ALIGN16 double imag[kMaxBufferSize];
 
  const int numBits = Fourier::integerLog2(buffer.size());
 
@@ -394,6 +399,8 @@ int main(int argc, char * argv[])
  provideInfo.outputSamples[c][i] = dst.values[i];
  }
  }
+
+ return provideInfo.numOutputChannels;
  }
  };
 

diff --git a/users/marcel/spatialAudioSystem-binaural/spatialAudioSystem-binaural.cpp b/users/marcel/spatialAudioSystem-binaural/spatialAudioSystem-binaural.cpp
@@ -170,14 +170,18 @@ void SpatialAudioSystem_Binaural::updatePanning()
 
 void SpatialAudioSystem_Binaural::generateLR(float * __restrict outputSamplesL, float * __restrict outputSamplesR, const int numSamples)
 {
+ const int kMaxSamples = binaural::Binauralizer::SampleBuffer::kBufferSize;
+
+ Assert(numSamples <= kMaxSamples);
+
  memset(outputSamplesL, 0, numSamples * sizeof(float));
  memset(outputSamplesR, 0, numSamples * sizeof(float));
 
  sources_mutex.lock();
  {
  for (Source * source = sources; source != nullptr; source = source->next)
  {
- ALIGN16 float inputSamples[numSamples];
+ ALIGN16 float inputSamples[kMaxSamples];
 
  // generate source audio
  source->audioSource->generate(inputSamples, numSamples);
@@ -200,8 +204,8 @@ void SpatialAudioSystem_Binaural::generateLR(float * __restrict outputSamplesL,
  source->elevation.load(),
  source->azimuth.load());
 
- ALIGN16 float samplesL[numSamples];
- ALIGN16 float samplesR[numSamples];
+ ALIGN16 float samplesL[kMaxSamples];
+ ALIGN16 float samplesR[kMaxSamples];
  source->binauralizer.generateLR(
  samplesL,
  samplesR,