Merge develop into master

Merge pull request #480 from Drewol/develop

.github/workflows/build.yml

@@ -86,8 +86,10 @@ jobs:
  uses: actions/checkout@v1
  - name: Checkout submodules
  run: git submodule update --init --recursive
+ - name: brew update
+ run: brew update
  - name: Install packages
- run: brew install cmake freetype libvorbis sdl2 libpng jpeg libarchive
+ run: brew install freetype libvorbis sdl2 libpng jpeg libarchive
  - name: cmake
  run: cmake . -DLibArchive_LIBRARY=/usr/local/opt/libarchive/lib/libarchive.dylib -DLibArchive_INCLUDE_DIR=/usr/local/opt/libarchive/include -DCMAKE_BUILD_TYPE=Release
  - name: make

.gitignore

@@ -20,6 +20,7 @@ bin/skins/*
 !bin/skins/Default
 bin/skins/Default/skin.cfg
 bin/replays
+bin/profiles
 
 # Intermediate Files
 x64

Audio/CMakeLists.txt

@@ -59,6 +59,9 @@ target_link_libraries(Audio ${SDL2_LIBRARY})
 target_link_libraries(Audio ${OGG_LIBRARIES})
 target_link_libraries(Audio ${Vorbis_LIBRARIES})
 
-if(WIN32)
- target_compile_options(Graphics PRIVATE /Zi)
-endif()
+target_link_libraries(Audio cc-common)
+
+# Enable multiprocess compiling
+if(MSVC)
+ target_compile_options(Audio PRIVATE /MP)
+endif(MSVC)

Audio/include/Audio/AudioBase.hpp

@@ -7,33 +7,33 @@ class DSP
 {
 protected:
  DSP() = default; // Abstract
- DSP(const DSP&) = delete;
+ DSP(const DSP &) = delete;
 
- inline void SetSampleRate(uint32 sampleRate) { m_sampleRate = sampleRate; }
  uint32 GetStartSample() const;
  uint32 GetCurrentSample() const;
 
  // Smpling rate of m_audio (not m_audioBase)
  // Only use this for initializing parameters
  uint32 m_sampleRate = 0;
 
- class AudioBase* m_audioBase = nullptr;
- class Audio_Impl* m_audio = nullptr;
+ class AudioBase *m_audioBase = nullptr;
 
 public:
  virtual ~DSP();
+ static bool Sorter(DSP *&a, DSP *&b);
 
- void SetAudio(class Audio_Impl* audio);
- void SetAudioBase(class AudioBase* audioBase);
+ void SetAudioBase(class AudioBase *audioBase);
  inline void RemoveAudioBase() { m_audioBase = nullptr; }
+ inline void SetSampleRate(uint32 sampleRate) { m_sampleRate = sampleRate; }
 
  // Process <numSamples> amount of samples in stereo float format
- virtual void Process(float* out, uint32 numSamples) = 0;
- virtual const char* GetName() const = 0;
+ virtual void Process(float *out, uint32 numSamples) = 0;
+ virtual const char *GetName() const = 0;
 
  float mix = 1.0f;
  uint32 priority = 0;
  uint32 startTime = 0;
+ uint32 endTime = 0;
  int32 chartOffset = 0;
  int32 lastTimingPoint = 0;
 };
@@ -46,27 +46,31 @@ class AudioBase
 public:
  virtual ~AudioBase();
  // Process <numSamples> amount of samples in stereo float format
- virtual void Process(float* out, uint32 numSamples) = 0;
- 
+ virtual void Process(float *out, uint32 numSamples) = 0;
+
  // Gets the playback position in millisecond
  virtual int32 GetPosition() const = 0;
 
  // Get the sample rate of this audio stream
  virtual uint32 GetSampleRate() const = 0;
 
+ // Get the exact playback position in samples
+ virtual uint64 GetSamplePos() const = 0;
+
  // Get the sample rate of the audio connected to this
  uint32 GetAudioSampleRate() const;
 
  // Gets pcm data from a decoded stream, nullptr if not available
- virtual float* GetPCM() = 0;
+ virtual float *GetPCM() = 0;
+ // Gets pcm sample count
  virtual uint64 GetPCMCount() const = 0;
+ virtual void PreRenderDSPs(Vector<DSP *> &DSPs) = 0;
 
- void ProcessDSPs(float* out, uint32 numSamples);
+ void ProcessDSPs(float *out, uint32 numSamples);
  // Adds a signal processor to the audio
- void AddDSP(DSP* dsp);
+ void AddDSP(DSP *dsp);
  // Removes a signal processor from the audio
- void RemoveDSP(DSP* dsp);
-
+ void RemoveDSP(DSP *dsp);
 
  void Deregister();
 
@@ -80,9 +84,10 @@ class AudioBase
  return m_volume;
  }
 
- Vector<DSP*> DSPs;
+ Vector<DSP *> DSPs;
  float PlaybackSpeed = 1.0;
- class Audio_Impl* audio = nullptr;
+ class Audio_Impl *audio = nullptr;
+
 private:
  float m_volume = 1.0f;
 };

Audio/include/Audio/AudioStream.hpp

@@ -10,7 +10,8 @@ class Audio;
 class AudioStream : public AudioBase
 {
 public:
- static Ref<AudioStream> Create(Audio* audio, const String& path, bool preload);
+ static Ref<AudioStream> Create(Audio *audio, const String &path, bool preload);
+ static Ref<AudioStream> Clone(Audio *audio, Ref<AudioStream> source);
  virtual ~AudioStream() = default;
  // Starts playback of the stream or continues a paused stream
  virtual void Play() = 0;

Audio/include/Audio/DSP.hpp

@@ -10,8 +10,8 @@ class PanDSP : public DSP
 public:
  // -1 to 1 LR pan value
  float panning = 0.0f;
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "PanDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "PanDSP"; }
 };
 
 // Biquad Filter
@@ -28,8 +28,8 @@ class BQFDSP : public DSP
  float a1 = 0.0f;
  float a2 = 0.0f;
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "BQFDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "BQFDSP"; }
 
  // Sets the filter parameters
  void SetPeaking(float q, float freq, float gain);
@@ -39,13 +39,14 @@ class BQFDSP : public DSP
  void SetPeaking(float q, float freq, float gain, float sampleRate);
  void SetLowPass(float q, float freq, float sampleRate);
  void SetHighPass(float q, float freq, float sampleRate);
+
 private:
  // Delayed samples
  static const uint32 order = 2;
  // FIR Delay buffers
- float zb[2][order];
+ float zb[2][order]{};
  // IIR Delay buffers
- float za[2][order];
+ float za[2][order]{};
 };
 
 // Combinded Low/High-pass and Peaking filter
@@ -55,9 +56,10 @@ class CombinedFilterDSP : public DSP
  CombinedFilterDSP(uint32 sampleRate);
  void SetLowPass(float q, float freq, float peakQ, float peakGain);
  void SetHighPass(float q, float freq, float peakQ, float peakGain);
- virtual const char* GetName() const { return "CombinedFilterDSP"; }
+ virtual const char *GetName() const { return "CombinedFilterDSP"; }
+
+ virtual void Process(float *out, uint32 numSamples);
 
- virtual void Process(float* out, uint32 numSamples);
 private:
  BQFDSP a;
  BQFDSP peak;
@@ -70,8 +72,9 @@ class LimiterDSP : public DSP
  LimiterDSP(uint32 sampleRate);
 
  float releaseTime = 0.1f;
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "LimiterDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "LimiterDSP"; }
+
 private:
  float m_currentMaxVolume = 1.0f;
  float m_currentReleaseTimer = releaseTime;
@@ -84,12 +87,13 @@ class BitCrusherDSP : public DSP
 
  // Duration of samples, <1 = disable
  void SetPeriod(float period = 0);
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "BitCrusherDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "BitCrusherDSP"; }
+
 private:
  uint32 m_period = 1;
  uint32 m_increment = 0;
- float m_sampleBuffer[2] = { 0.0f };
+ float m_sampleBuffer[2] = {0.0f};
  uint32 m_currentDuration = 0;
 };
 
@@ -105,14 +109,15 @@ class GateDSP : public DSP
  // Low volume
  float low = 0.1f;
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "GateDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "GateDSP"; }
+
 private:
  float m_gating = 0.5f;
  uint32 m_length = 0;
- uint32 m_fadeIn = 0; // Fade In mark
+ uint32 m_fadeIn = 0;  // Fade In mark
  uint32 m_fadeOut = 0; // Fade Out mark
- uint32 m_halfway; // Halfway mark
+ uint32 m_halfway{};  // Halfway mark
  uint32 m_currentSample = 0;
 };
 
@@ -123,8 +128,9 @@ class TapeStopDSP : public DSP
 
  void SetLength(double length);
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "TapeStopDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "TapeStopDSP"; }
+
 private:
  uint32 m_length = 0;
  Vector<float> m_sampleBuffer;
@@ -142,8 +148,9 @@ class RetriggerDSP : public DSP
  void SetGating(float gating);
  void SetMaxLength(uint32 length);
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "RetriggerDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "RetriggerDSP"; }
+
 private:
  float m_gating = 0.75f;
  uint32 m_length = 0;
@@ -166,16 +173,17 @@ class WobbleDSP : public BQFDSP
  float fmax = 20000.0f;
  float q = 1.414f;
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "WobbleDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "WobbleDSP"; }
+
 private:
- uint32 m_length;
+ uint32 m_length{};
  uint32 m_currentSample = 0;
 };
 
 // Referenced http://www.musicdsp.org/files/phaser.cpp
 class PhaserDSP : public DSP
-{ 
+{
 public:
  PhaserDSP(uint32 sampleRate);
 
@@ -184,13 +192,13 @@ class PhaserDSP : public DSP
  // Frequency range
  float dmin = 1000.0f;
  float dmax = 4000.0f;
- float fb = 0.2f; //feedback
+ float fb = 0.2f; //feedback
  float lmix = 0.33f; //local mix
 
  void SetLength(double length);
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "PhaserDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "PhaserDSP"; }
 
 private:
  uint32 m_length = 0;
@@ -204,7 +212,7 @@ class PhaserDSP : public DSP
  };
 
  APF filters[2][6];
- float za[2] = { 0.0f };
+ float za[2] = {0.0f};
 };
 
 class FlangerDSP : public DSP
@@ -215,8 +223,9 @@ class FlangerDSP : public DSP
  void SetLength(double length);
  void SetDelayRange(uint32 min, uint32 max);
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "FlangerDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "FlangerDSP"; }
+
 private:
  uint32 m_length = 0;
 
@@ -239,16 +248,16 @@ class EchoDSP : public DSP
 
  float feedback = 0.6f;
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "EchoDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "EchoDSP"; }
+
 private:
  uint32 m_bufferLength = 0;
  size_t m_bufferOffset = 0;
  uint32 m_numLoops = 0;
  Vector<float> m_sampleBuffer;
 };
 
-
 class SidechainDSP : public DSP
 {
 public:
@@ -260,10 +269,11 @@ class SidechainDSP : public DSP
  // Volume multiplier for the sidechaing
  float amount = 0.25f;
 
- Interpolation::CubicBezier curve;
+ Interpolation::CubicBezier curve{};
+
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "SidechainDSP"; }
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "SidechainDSP"; }
 private:
  uint32 m_length = 0;
  size_t m_time = 0;
@@ -279,8 +289,9 @@ class PitchShiftDSP : public DSP
 
  ~PitchShiftDSP();
 
- virtual void Process(float* out, uint32 numSamples);
- virtual const char* GetName() const { return "PitchShiftDSP"; }
+ virtual void Process(float *out, uint32 numSamples);
+ virtual const char *GetName() const { return "PitchShiftDSP"; }
+
 private:
- class PitchShiftDSP_Impl* m_impl;
+ class PitchShiftDSP_Impl *m_impl;
 };