diff --git a/Builds/LinuxMakefile/Makefile b/Builds/LinuxMakefile/Makefile
index e857246f..9a2bcf3a 100644
--- a/Builds/LinuxMakefile/Makefile
+++ b/Builds/LinuxMakefile/Makefile
@@ -35,7 +35,7 @@ ifeq ($(CONFIG),Debug)
     TARGET_ARCH := 
   endif
 
-  JUCE_CPPFLAGS := $(DEPFLAGS) "-DLINUX=1" "-DDEBUG=1" "-D_DEBUG=1" "-DPIP_JUCE_EXAMPLES_DIRECTORY=QzpcdG9vbHNcSlVDRVxleGFtcGxlcw==" "-DSAMPLER_SKIP_UI" "-DJUCE_MODAL_LOOPS_PERMITTED" "-DHAVE_LIBSAMPLERATE" "-DUSE_BUILTIN_FFT" "-DUSE_PTHREADS" "-DBUILD_DAWDREAMER_FAUST" "-DBUILD_DAWDREAMER_RUBBERBAND" "-DJUCER_LINUX_MAKE_6D53C8B4=1" "-DJUCE_APP_VERSION=0.6.5" "-DJUCE_APP_VERSION_HEX=0x605" $(shell pkg-config --cflags alsa freetype2) -pthread -I../../JuceLibraryCode/modules/juce_audio_processors/format_types/VST3_SDK -I../../thirdparty/JUCE/modules/juce_audio_processors/format_types/VST3_SDK -I../../JuceLibraryCode -I../../JuceLibraryCode/modules -I../../thirdparty/pybind11/include -I../../thirdparty/faust/architecture -I../../thirdparty/faust/compiler -I../../thirdparty/faust/compiler/utils -I../../thirdparty/libsamplerate/src -I../../thirdparty/libsamplerate/include -I../../thirdparty/rubberband -I../../thirdparty/rubberband/rubberband -I../../thirdparty/rubberband/src/kissfft -I../../thirdparty/rubberband/src -I../../thirdparty/portable_endian/include $(CPPFLAGS)
+  JUCE_CPPFLAGS := $(DEPFLAGS) "-DLINUX=1" "-DDEBUG=1" "-D_DEBUG=1" "-DPIP_JUCE_EXAMPLES_DIRECTORY=QzpcdG9vbHNcSlVDRVxleGFtcGxlcw==" "-DSAMPLER_SKIP_UI" "-DJUCE_MODAL_LOOPS_PERMITTED" "-DHAVE_LIBSAMPLERATE" "-DUSE_BUILTIN_FFT" "-DUSE_PTHREADS" "-DBUILD_DAWDREAMER_FAUST" "-DBUILD_DAWDREAMER_RUBBERBAND" "-DJUCER_LINUX_MAKE_6D53C8B4=1" "-DJUCE_APP_VERSION=0.6.6" "-DJUCE_APP_VERSION_HEX=0x606" $(shell pkg-config --cflags alsa freetype2) -pthread -I../../JuceLibraryCode/modules/juce_audio_processors/format_types/VST3_SDK -I../../thirdparty/JUCE/modules/juce_audio_processors/format_types/VST3_SDK -I../../JuceLibraryCode -I../../JuceLibraryCode/modules -I../../thirdparty/pybind11/include -I../../thirdparty/faust/architecture -I../../thirdparty/faust/compiler -I../../thirdparty/faust/compiler/utils -I../../thirdparty/libsamplerate/src -I../../thirdparty/libsamplerate/include -I../../thirdparty/rubberband -I../../thirdparty/rubberband/rubberband -I../../thirdparty/rubberband/src/kissfft -I../../thirdparty/rubberband/src -I../../thirdparty/portable_endian/include $(CPPFLAGS)
   JUCE_CPPFLAGS_DYNAMIC_LIBRARY :=  "-DJucePlugin_Build_VST=0" "-DJucePlugin_Build_VST3=0" "-DJucePlugin_Build_AU=0" "-DJucePlugin_Build_AUv3=0" "-DJucePlugin_Build_RTAS=0" "-DJucePlugin_Build_AAX=0" "-DJucePlugin_Build_Standalone=0" "-DJucePlugin_Build_Unity=0"
   JUCE_CFLAGS_DYNAMIC_LIBRARY := -fPIC -fvisibility=hidden
   JUCE_LDFLAGS_DYNAMIC_LIBRARY := -shared
@@ -58,7 +58,7 @@ ifeq ($(CONFIG),Release)
     TARGET_ARCH := 
   endif
 
-  JUCE_CPPFLAGS := $(DEPFLAGS) "-DLINUX=1" "-DNDEBUG=1" "-DPIP_JUCE_EXAMPLES_DIRECTORY=QzpcdG9vbHNcSlVDRVxleGFtcGxlcw==" "-DSAMPLER_SKIP_UI" "-DJUCE_MODAL_LOOPS_PERMITTED" "-DHAVE_LIBSAMPLERATE" "-DUSE_BUILTIN_FFT" "-DUSE_PTHREADS" "-DBUILD_DAWDREAMER_FAUST" "-DBUILD_DAWDREAMER_RUBBERBAND" "-DJUCER_LINUX_MAKE_6D53C8B4=1" "-DJUCE_APP_VERSION=0.6.5" "-DJUCE_APP_VERSION_HEX=0x605" $(shell pkg-config --cflags alsa freetype2) -pthread -I../../JuceLibraryCode/modules/juce_audio_processors/format_types/VST3_SDK -I../../thirdparty/JUCE/modules/juce_audio_processors/format_types/VST3_SDK -I../../JuceLibraryCode -I../../JuceLibraryCode/modules -I../../thirdparty/pybind11/include -I../../thirdparty/faust/architecture -I../../thirdparty/faust/compiler -I../../thirdparty/faust/compiler/utils -I../../thirdparty/libsamplerate/src -I../../thirdparty/libsamplerate/include -I../../thirdparty/rubberband -I../../thirdparty/rubberband/rubberband -I../../thirdparty/rubberband/src/kissfft -I../../thirdparty/rubberband/src -I../../thirdparty/portable_endian/include $(CPPFLAGS)
+  JUCE_CPPFLAGS := $(DEPFLAGS) "-DLINUX=1" "-DNDEBUG=1" "-DPIP_JUCE_EXAMPLES_DIRECTORY=QzpcdG9vbHNcSlVDRVxleGFtcGxlcw==" "-DSAMPLER_SKIP_UI" "-DJUCE_MODAL_LOOPS_PERMITTED" "-DHAVE_LIBSAMPLERATE" "-DUSE_BUILTIN_FFT" "-DUSE_PTHREADS" "-DBUILD_DAWDREAMER_FAUST" "-DBUILD_DAWDREAMER_RUBBERBAND" "-DJUCER_LINUX_MAKE_6D53C8B4=1" "-DJUCE_APP_VERSION=0.6.6" "-DJUCE_APP_VERSION_HEX=0x606" $(shell pkg-config --cflags alsa freetype2) -pthread -I../../JuceLibraryCode/modules/juce_audio_processors/format_types/VST3_SDK -I../../thirdparty/JUCE/modules/juce_audio_processors/format_types/VST3_SDK -I../../JuceLibraryCode -I../../JuceLibraryCode/modules -I../../thirdparty/pybind11/include -I../../thirdparty/faust/architecture -I../../thirdparty/faust/compiler -I../../thirdparty/faust/compiler/utils -I../../thirdparty/libsamplerate/src -I../../thirdparty/libsamplerate/include -I../../thirdparty/rubberband -I../../thirdparty/rubberband/rubberband -I../../thirdparty/rubberband/src/kissfft -I../../thirdparty/rubberband/src -I../../thirdparty/portable_endian/include $(CPPFLAGS)
   JUCE_CPPFLAGS_DYNAMIC_LIBRARY :=  "-DJucePlugin_Build_VST=0" "-DJucePlugin_Build_VST3=0" "-DJucePlugin_Build_AU=0" "-DJucePlugin_Build_AUv3=0" "-DJucePlugin_Build_RTAS=0" "-DJucePlugin_Build_AAX=0" "-DJucePlugin_Build_Standalone=0" "-DJucePlugin_Build_Unity=0"
   JUCE_CFLAGS_DYNAMIC_LIBRARY := -fPIC -fvisibility=hidden
   JUCE_LDFLAGS_DYNAMIC_LIBRARY := -shared
@@ -101,8 +101,10 @@ OBJECTS_DYNAMIC_LIBRARY := \
   $(JUCE_OBJDIR)/SamplerAudioProcessorEditor_39db550d.o \
   $(JUCE_OBJDIR)/FaustProcessor_2056a3e0.o \
   $(JUCE_OBJDIR)/FilterProcessor_8755f5b.o \
+  $(JUCE_OBJDIR)/PlaybackWarpProcessor_eedba470.o \
   $(JUCE_OBJDIR)/PluginProcessor_a059e380.o \
   $(JUCE_OBJDIR)/ProcessorBase_65bf9464.o \
+  $(JUCE_OBJDIR)/CustomParameters_5970e144.o \
   $(JUCE_OBJDIR)/custom_pybind_wrappers_ac32b5b4.o \
   $(JUCE_OBJDIR)/RenderEngine_d1c4d401.o \
   $(JUCE_OBJDIR)/RenderEngineWrapper_9b21dedc.o \
@@ -281,6 +283,11 @@ $(JUCE_OBJDIR)/FilterProcessor_8755f5b.o: ../../Source/FilterProcessor.cpp
 	@echo "Compiling FilterProcessor.cpp"
 	$(V_AT)$(CXX) $(JUCE_CXXFLAGS) $(JUCE_CPPFLAGS_DYNAMIC_LIBRARY) $(JUCE_CFLAGS_DYNAMIC_LIBRARY) -o "$@" -c "$<"
 
+$(JUCE_OBJDIR)/PlaybackWarpProcessor_eedba470.o: ../../Source/PlaybackWarpProcessor.cpp
+	-$(V_AT)mkdir -p $(JUCE_OBJDIR)
+	@echo "Compiling PlaybackWarpProcessor.cpp"
+	$(V_AT)$(CXX) $(JUCE_CXXFLAGS) $(JUCE_CPPFLAGS_DYNAMIC_LIBRARY) $(JUCE_CFLAGS_DYNAMIC_LIBRARY) -o "$@" -c "$<"
+
 $(JUCE_OBJDIR)/PluginProcessor_a059e380.o: ../../Source/PluginProcessor.cpp
 	-$(V_AT)mkdir -p $(JUCE_OBJDIR)
 	@echo "Compiling PluginProcessor.cpp"
@@ -291,6 +298,11 @@ $(JUCE_OBJDIR)/ProcessorBase_65bf9464.o: ../../Source/ProcessorBase.cpp
 	@echo "Compiling ProcessorBase.cpp"
 	$(V_AT)$(CXX) $(JUCE_CXXFLAGS) $(JUCE_CPPFLAGS_DYNAMIC_LIBRARY) $(JUCE_CFLAGS_DYNAMIC_LIBRARY) -o "$@" -c "$<"
 
+$(JUCE_OBJDIR)/CustomParameters_5970e144.o: ../../Source/CustomParameters.cpp
+	-$(V_AT)mkdir -p $(JUCE_OBJDIR)
+	@echo "Compiling CustomParameters.cpp"
+	$(V_AT)$(CXX) $(JUCE_CXXFLAGS) $(JUCE_CPPFLAGS_DYNAMIC_LIBRARY) $(JUCE_CFLAGS_DYNAMIC_LIBRARY) -o "$@" -c "$<"
+
 $(JUCE_OBJDIR)/custom_pybind_wrappers_ac32b5b4.o: ../../Source/custom_pybind_wrappers.cpp
 	-$(V_AT)mkdir -p $(JUCE_OBJDIR)
 	@echo "Compiling custom_pybind_wrappers.cpp"
diff --git a/Builds/MacOSX/DawDreamer.xcodeproj/project.pbxproj b/Builds/MacOSX/DawDreamer.xcodeproj/project.pbxproj
index b7dafce9..86a8fe3b 100644
--- a/Builds/MacOSX/DawDreamer.xcodeproj/project.pbxproj
+++ b/Builds/MacOSX/DawDreamer.xcodeproj/project.pbxproj
@@ -11,6 +11,8 @@
 		03E7579908465C2C4C83596E /* include_juce_dsp.mm */ = {isa = PBXBuildFile; fileRef = 2AA8CBD57F5CC5B7912A192F; };
 		0C01D9069A27A2ECB0B24253 /* StretcherChannelData.cpp */ = {isa = PBXBuildFile; fileRef = 5558FF156E977213C6DE9C14; };
 		158D137D07F738CDB6AD2028 /* Accelerate.framework */ = {isa = PBXBuildFile; fileRef = 553C0A3501582EE95328719B; };
+		1B14253412716097BBB38A9B /* CustomParameters.cpp */ = {isa = PBXBuildFile; fileRef = C5A12760CCD5CE6DD270F92F; };
+		210F38443F9D77354A4B28EB /* PlaybackWarpProcessor.cpp */ = {isa = PBXBuildFile; fileRef = AB39C0017B2EE44B9E510D89; };
 		21FB7BCF683F51D403A7475D /* StretcherProcess.cpp */ = {isa = PBXBuildFile; fileRef = 1B00487B34E2B8C9A26AFA49; };
 		24E83A217028D2AEF88FF90E /* include_juce_data_structures.mm */ = {isa = PBXBuildFile; fileRef = DE2959972547E114EB1E1FCD; };
 		2706724BA659554B6F4EE85E /* include_juce_audio_formats.mm */ = {isa = PBXBuildFile; fileRef = 03BD258F06159505E43073E1; };
@@ -172,7 +174,6 @@
 		8A778A5F6395BCBDE3BE5982 /* Scavenger.h */ /* Scavenger.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = Scavenger.h; path = ../../thirdparty/rubberband/src/base/Scavenger.h; sourceTree = SOURCE_ROOT; };
 		8B852E83502C66F8B990F051 /* SilentAudioCurve.h */ /* SilentAudioCurve.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = SilentAudioCurve.h; path = ../../thirdparty/rubberband/src/audiocurves/SilentAudioCurve.h; sourceTree = SOURCE_ROOT; };
 		8D80555F3F275D566655DEC2 /* getopt.c */ /* getopt.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = getopt.c; path = ../../thirdparty/rubberband/src/getopt/getopt.c; sourceTree = SOURCE_ROOT; };
-		8D88F06BD34407D802DFB597 /* AllProcessors.h */ /* AllProcessors.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = AllProcessors.h; path = ../../Source/AllProcessors.h; sourceTree = SOURCE_ROOT; };
 		8F78B14DAA164DE1331AD3FD /* Main.cpp */ /* Main.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = Main.cpp; path = ../../Source/Sampler/Source/Main.cpp; sourceTree = SOURCE_ROOT; };
 		8F7B057EB4717EDC7E8E8048 /* StretcherImpl.h */ /* StretcherImpl.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = StretcherImpl.h; path = ../../thirdparty/rubberband/src/StretcherImpl.h; sourceTree = SOURCE_ROOT; };
 		94091A9630E5C9649E958A16 /* kiss_fftr.h */ /* kiss_fftr.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = kiss_fftr.h; path = ../../thirdparty/rubberband/src/kissfft/kiss_fftr.h; sourceTree = SOURCE_ROOT; };
@@ -195,6 +196,7 @@
 		A64A3D2EF69F5E968839751A /* sysutils.h */ /* sysutils.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = sysutils.h; path = ../../thirdparty/rubberband/src/system/sysutils.h; sourceTree = SOURCE_ROOT; };
 		A997E92A77CDFA70A6536BBD /* MPESettingsDataModel.cpp */ /* MPESettingsDataModel.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = MPESettingsDataModel.cpp; path = ../../Source/Sampler/Source/DataModels/MPESettingsDataModel.cpp; sourceTree = SOURCE_ROOT; };
 		AA2C7D7E9590EB6623631A35 /* CommandFifo.h */ /* CommandFifo.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = CommandFifo.h; path = ../../Source/Sampler/Source/CommandFifo.h; sourceTree = SOURCE_ROOT; };
+		AB39C0017B2EE44B9E510D89 /* PlaybackWarpProcessor.cpp */ /* PlaybackWarpProcessor.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = PlaybackWarpProcessor.cpp; path = ../../Source/PlaybackWarpProcessor.cpp; sourceTree = SOURCE_ROOT; };
 		AD248A6B37034D215326CB7F /* juce_audio_processors */ /* juce_audio_processors */ = {isa = PBXFileReference; lastKnownFileType = folder; name = juce_audio_processors; path = ../../JuceLibraryCode/modules/juce_audio_processors; sourceTree = SOURCE_ROOT; };
 		B1FAEBBC0B8073BAE84C5665 /* PlaybackProcessor.h */ /* PlaybackProcessor.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = PlaybackProcessor.h; path = ../../Source/PlaybackProcessor.h; sourceTree = SOURCE_ROOT; };
 		B2CCBF7C3614EE8B865D41C0 /* include_juce_graphics.mm */ /* include_juce_graphics.mm */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.objcpp; name = include_juce_graphics.mm; path = ../../JuceLibraryCode/include_juce_graphics.mm; sourceTree = SOURCE_ROOT; };
@@ -208,6 +210,7 @@
 		C379050966C094B7C05FEED8 /* Resampler.h */ /* Resampler.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = Resampler.h; path = ../../thirdparty/rubberband/src/dsp/Resampler.h; sourceTree = SOURCE_ROOT; };
 		C503A9D9F3009D679DBFC556 /* FilterProcessor.h */ /* FilterProcessor.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = FilterProcessor.h; path = ../../Source/FilterProcessor.h; sourceTree = SOURCE_ROOT; };
 		C519012CDDE2CA92FF58903A /* include_juce_video.mm */ /* include_juce_video.mm */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.objcpp; name = include_juce_video.mm; path = ../../JuceLibraryCode/include_juce_video.mm; sourceTree = SOURCE_ROOT; };
+		C5A12760CCD5CE6DD270F92F /* CustomParameters.cpp */ /* CustomParameters.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = CustomParameters.cpp; path = ../../Source/CustomParameters.cpp; sourceTree = SOURCE_ROOT; };
 		C5E04D72E37F04C0663E5EA6 /* CoreMIDI.framework */ /* CoreMIDI.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = CoreMIDI.framework; path = System/Library/Frameworks/CoreMIDI.framework; sourceTree = SDKROOT; };
 		C6695EB398FC46133B7FAC70 /* PlaybackPositionOverlay.h */ /* PlaybackPositionOverlay.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = PlaybackPositionOverlay.h; path = ../../Source/Sampler/Source/Components/PlaybackPositionOverlay.h; sourceTree = SOURCE_ROOT; };
 		C713F7F77E549E8C08771E49 /* AbletonClipInfo.h */ /* AbletonClipInfo.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = AbletonClipInfo.h; path = ../../Source/AbletonClipInfo.h; sourceTree = SOURCE_ROOT; };
@@ -410,6 +413,7 @@
 				561257822B09113041A6909C,
 				E553BE5C53ED6D87F71CC515,
 				7C90363A6D8BA8EA0D91A985,
+				C5A12760CCD5CE6DD270F92F,
 				3319047264FBD43706A45A08,
 				A2CE988FDEA16DA46477FB4B,
 				094BECA334087BBDDA1CA022,
@@ -570,13 +574,13 @@
 				C20139B2E8CF4DC6B19AC0A6,
 				8038F1D9B3CEBD503DE7AD1F,
 				4915A3C580F0DAA69FAAED98,
-				8D88F06BD34407D802DFB597,
 				9ABCCF2DBAA01DE762D8BD60,
 				BFD142EBB8AFBADC8AB04A8A,
 				7332613820553661C4FA3CD8,
 				C503A9D9F3009D679DBFC556,
 				1799B9631459052DD116F71A,
 				F7EDC034912036BE18A95734,
+				AB39C0017B2EE44B9E510D89,
 				B1FAEBBC0B8073BAE84C5665,
 				EE65D34C5FCFBF1D5F6C12D8,
 				2F962B6F6EA479E59F083A16,
@@ -736,8 +740,10 @@
 				B942805B7846AE297800EF37,
 				E358C00D8D92D35AFBC9944C,
 				7C6656C8A61187E3C9113001,
+				210F38443F9D77354A4B28EB,
 				92F6A21BA484A258A4EF5F23,
 				BD1166D575CE2FE6886E0DFC,
+				1B14253412716097BBB38A9B,
 				9D34C1F40CAD6A7617F1B60B,
 				48CA111A7776D753A8BFF02E,
 				F69DC2F3034023CA38AFFC92,
@@ -788,8 +794,8 @@
 					"BUILD_DAWDREAMER_FAUST",
 					"BUILD_DAWDREAMER_RUBBERBAND",
 					"JUCER_XCODE_MAC_F6D2F4CF=1",
-					"JUCE_APP_VERSION=0.6.5",
-					"JUCE_APP_VERSION_HEX=0x605",
+					"JUCE_APP_VERSION=0.6.6",
+					"JUCE_APP_VERSION_HEX=0x606",
 					"JucePlugin_Build_VST=0",
 					"JucePlugin_Build_VST3=0",
 					"JucePlugin_Build_AU=0",
@@ -914,8 +920,8 @@
 					"BUILD_DAWDREAMER_FAUST",
 					"BUILD_DAWDREAMER_RUBBERBAND",
 					"JUCER_XCODE_MAC_F6D2F4CF=1",
-					"JUCE_APP_VERSION=0.6.5",
-					"JUCE_APP_VERSION_HEX=0x605",
+					"JUCE_APP_VERSION=0.6.6",
+					"JUCE_APP_VERSION_HEX=0x606",
 					"JucePlugin_Build_VST=0",
 					"JucePlugin_Build_VST3=0",
 					"JucePlugin_Build_AU=0",
diff --git a/Builds/VisualStudio2019/DawDreamer_DynamicLibrary.vcxproj b/Builds/VisualStudio2019/DawDreamer_DynamicLibrary.vcxproj
index 8e6b5716..10485d3c 100644
--- a/Builds/VisualStudio2019/DawDreamer_DynamicLibrary.vcxproj
+++ b/Builds/VisualStudio2019/DawDreamer_DynamicLibrary.vcxproj
@@ -66,7 +66,7 @@
       <Optimization>Disabled</Optimization>
       <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>
       <AdditionalIncludeDirectories>..\..\JuceLibraryCode\modules\juce_audio_processors\format_types\VST3_SDK;..\..\thirdparty\JUCE\modules\juce_audio_processors\format_types\VST3_SDK;..\..\JuceLibraryCode;..\..\JuceLibraryCode\modules;$(pythonLocation)\include;..\..\thirdparty\pybind11\include;..\..\thirdparty\faust\architecture;..\..\thirdparty\faust\compiler;..\..\thirdparty\faust\compiler\utils;..\..\thirdparty\libsamplerate\src;..\..\thirdparty\libsamplerate\include;..\..\thirdparty\rubberband;..\..\thirdparty\rubberband\rubberband;..\..\thirdparty\rubberband\src\kissfft;..\..\thirdparty\rubberband\src;..\..\thirdparty\portable_endian\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
-      <PreprocessorDefinitions>_CRT_SECURE_NO_WARNINGS;WIN32;_WINDOWS;DEBUG;_DEBUG;PIP_JUCE_EXAMPLES_DIRECTORY=QzpcdG9vbHNcSlVDRVxleGFtcGxlcw==;SAMPLER_SKIP_UI;JUCE_MODAL_LOOPS_PERMITTED;_WIN32;__SSE__;__SSE2__;BUILD_DAWDREAMER_FAUST;BUILD_DAWDREAMER_RUBBERBAND;NOMINMAX;HAVE_LIBSAMPLERATE;HAVE_KISSFFT;JUCER_VS2019_78A5026=1;JUCE_APP_VERSION=0.6.5;JUCE_APP_VERSION_HEX=0x605;JucePlugin_Build_VST=0;JucePlugin_Build_VST3=0;JucePlugin_Build_AU=0;JucePlugin_Build_AUv3=0;JucePlugin_Build_RTAS=0;JucePlugin_Build_AAX=0;JucePlugin_Build_Standalone=0;JucePlugin_Build_Unity=0;_LIB;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <PreprocessorDefinitions>_CRT_SECURE_NO_WARNINGS;WIN32;_WINDOWS;DEBUG;_DEBUG;PIP_JUCE_EXAMPLES_DIRECTORY=QzpcdG9vbHNcSlVDRVxleGFtcGxlcw==;SAMPLER_SKIP_UI;JUCE_MODAL_LOOPS_PERMITTED;_WIN32;__SSE__;__SSE2__;BUILD_DAWDREAMER_FAUST;BUILD_DAWDREAMER_RUBBERBAND;NOMINMAX;HAVE_LIBSAMPLERATE;HAVE_KISSFFT;JUCER_VS2019_78A5026=1;JUCE_APP_VERSION=0.6.6;JUCE_APP_VERSION_HEX=0x606;JucePlugin_Build_VST=0;JucePlugin_Build_VST3=0;JucePlugin_Build_AU=0;JucePlugin_Build_AUv3=0;JucePlugin_Build_RTAS=0;JucePlugin_Build_AAX=0;JucePlugin_Build_Standalone=0;JucePlugin_Build_Unity=0;_LIB;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <RuntimeLibrary>MultiThreadedDebugDLL</RuntimeLibrary>
       <RuntimeTypeInfo>true</RuntimeTypeInfo>
       <PrecompiledHeader>NotUsing</PrecompiledHeader>
@@ -116,7 +116,7 @@ copy &quot;..\..\thirdparty\libfaust\win-x64\Debug\bin\faust.dll&quot; &quot;$(p
     <ClCompile>
       <Optimization>Full</Optimization>
       <AdditionalIncludeDirectories>..\..\JuceLibraryCode\modules\juce_audio_processors\format_types\VST3_SDK;..\..\thirdparty\JUCE\modules\juce_audio_processors\format_types\VST3_SDK;..\..\JuceLibraryCode;..\..\JuceLibraryCode\modules;$(pythonLocation)\include;..\..\thirdparty\pybind11\include;..\..\thirdparty\faust\architecture;..\..\thirdparty\faust\compiler;..\..\thirdparty\faust\compiler\utils;..\..\thirdparty\libsamplerate\src;..\..\thirdparty\libsamplerate\include;..\..\thirdparty\rubberband;..\..\thirdparty\rubberband\rubberband;..\..\thirdparty\rubberband\src\kissfft;..\..\thirdparty\rubberband\src;..\..\thirdparty\portable_endian\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
-      <PreprocessorDefinitions>_CRT_SECURE_NO_WARNINGS;WIN32;_WINDOWS;NDEBUG;PIP_JUCE_EXAMPLES_DIRECTORY=QzpcdG9vbHNcSlVDRVxleGFtcGxlcw==;SAMPLER_SKIP_UI;JUCE_MODAL_LOOPS_PERMITTED;_WIN32;__SSE__;__SSE2__;BUILD_DAWDREAMER_FAUST;BUILD_DAWDREAMER_RUBBERBAND;NOMINMAX;HAVE_LIBSAMPLERATE;HAVE_KISSFFT;JUCER_VS2019_78A5026=1;JUCE_APP_VERSION=0.6.5;JUCE_APP_VERSION_HEX=0x605;JucePlugin_Build_VST=0;JucePlugin_Build_VST3=0;JucePlugin_Build_AU=0;JucePlugin_Build_AUv3=0;JucePlugin_Build_RTAS=0;JucePlugin_Build_AAX=0;JucePlugin_Build_Standalone=0;JucePlugin_Build_Unity=0;_LIB;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <PreprocessorDefinitions>_CRT_SECURE_NO_WARNINGS;WIN32;_WINDOWS;NDEBUG;PIP_JUCE_EXAMPLES_DIRECTORY=QzpcdG9vbHNcSlVDRVxleGFtcGxlcw==;SAMPLER_SKIP_UI;JUCE_MODAL_LOOPS_PERMITTED;_WIN32;__SSE__;__SSE2__;BUILD_DAWDREAMER_FAUST;BUILD_DAWDREAMER_RUBBERBAND;NOMINMAX;HAVE_LIBSAMPLERATE;HAVE_KISSFFT;JUCER_VS2019_78A5026=1;JUCE_APP_VERSION=0.6.6;JUCE_APP_VERSION_HEX=0x606;JucePlugin_Build_VST=0;JucePlugin_Build_VST3=0;JucePlugin_Build_AU=0;JucePlugin_Build_AUv3=0;JucePlugin_Build_RTAS=0;JucePlugin_Build_AAX=0;JucePlugin_Build_Standalone=0;JucePlugin_Build_Unity=0;_LIB;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <RuntimeLibrary>MultiThreadedDLL</RuntimeLibrary>
       <RuntimeTypeInfo>true</RuntimeTypeInfo>
       <PrecompiledHeader>NotUsing</PrecompiledHeader>
@@ -200,8 +200,10 @@ copy &quot;..\..\thirdparty\libfaust\win-x64\Release\bin\faust.dll&quot; &quot;$
     <ClCompile Include="..\..\Source\Sampler\Source\SamplerAudioProcessorEditor.cpp"/>
     <ClCompile Include="..\..\Source\FaustProcessor.cpp"/>
     <ClCompile Include="..\..\Source\FilterProcessor.cpp"/>
+    <ClCompile Include="..\..\Source\PlaybackWarpProcessor.cpp"/>
     <ClCompile Include="..\..\Source\PluginProcessor.cpp"/>
     <ClCompile Include="..\..\Source\ProcessorBase.cpp"/>
+    <ClCompile Include="..\..\Source\CustomParameters.cpp"/>
     <ClCompile Include="..\..\Source\custom_pybind_wrappers.cpp"/>
     <ClCompile Include="..\..\Source\RenderEngine.cpp"/>
     <ClCompile Include="..\..\Source\RenderEngineWrapper.cpp"/>
@@ -2349,7 +2351,6 @@ copy &quot;..\..\thirdparty\libfaust\win-x64\Release\bin\faust.dll&quot; &quot;$
     <ClInclude Include="..\..\Source\ReverbProcessor.h"/>
     <ClInclude Include="..\..\Source\AddProcessor.h"/>
     <ClInclude Include="..\..\Source\CompressorProcessor.h"/>
-    <ClInclude Include="..\..\Source\AllProcessors.h"/>
     <ClInclude Include="..\..\Source\FaustProcessor.h"/>
     <ClInclude Include="..\..\Source\FilterProcessor.h"/>
     <ClInclude Include="..\..\Source\OscillatorProcessor.h"/>
diff --git a/Builds/VisualStudio2019/DawDreamer_DynamicLibrary.vcxproj.filters b/Builds/VisualStudio2019/DawDreamer_DynamicLibrary.vcxproj.filters
index a986a562..52729c88 100644
--- a/Builds/VisualStudio2019/DawDreamer_DynamicLibrary.vcxproj.filters
+++ b/Builds/VisualStudio2019/DawDreamer_DynamicLibrary.vcxproj.filters
@@ -631,12 +631,18 @@
     <ClCompile Include="..\..\Source\FilterProcessor.cpp">
       <Filter>DawDreamer\Processors</Filter>
     </ClCompile>
+    <ClCompile Include="..\..\Source\PlaybackWarpProcessor.cpp">
+      <Filter>DawDreamer\Processors</Filter>
+    </ClCompile>
     <ClCompile Include="..\..\Source\PluginProcessor.cpp">
       <Filter>DawDreamer\Processors</Filter>
     </ClCompile>
     <ClCompile Include="..\..\Source\ProcessorBase.cpp">
       <Filter>DawDreamer\Processors</Filter>
     </ClCompile>
+    <ClCompile Include="..\..\Source\CustomParameters.cpp">
+      <Filter>DawDreamer</Filter>
+    </ClCompile>
     <ClCompile Include="..\..\Source\custom_pybind_wrappers.cpp">
       <Filter>DawDreamer</Filter>
     </ClCompile>
@@ -3078,9 +3084,6 @@
     <ClInclude Include="..\..\Source\CompressorProcessor.h">
       <Filter>DawDreamer\Processors</Filter>
     </ClInclude>
-    <ClInclude Include="..\..\Source\AllProcessors.h">
-      <Filter>DawDreamer\Processors</Filter>
-    </ClInclude>
     <ClInclude Include="..\..\Source\FaustProcessor.h">
       <Filter>DawDreamer\Processors</Filter>
     </ClInclude>
diff --git a/Builds/VisualStudio2019/resources.rc b/Builds/VisualStudio2019/resources.rc
index ea211781..eb91b857 100644
--- a/Builds/VisualStudio2019/resources.rc
+++ b/Builds/VisualStudio2019/resources.rc
@@ -9,16 +9,16 @@
 #include <windows.h>
 
 VS_VERSION_INFO VERSIONINFO
-FILEVERSION  0,6,5,0
+FILEVERSION  0,6,6,0
 BEGIN
   BLOCK "StringFileInfo"
   BEGIN
     BLOCK "040904E4"
     BEGIN
       VALUE "FileDescription",  "DawDreamer\0"
-      VALUE "FileVersion",  "0.6.5\0"
+      VALUE "FileVersion",  "0.6.6\0"
       VALUE "ProductName",  "DawDreamer\0"
-      VALUE "ProductVersion",  "0.6.5\0"
+      VALUE "ProductVersion",  "0.6.6\0"
     END
   END
 
diff --git a/DawDreamer.jucer b/DawDreamer.jucer
index 060da481..bfe13e43 100644
--- a/DawDreamer.jucer
+++ b/DawDreamer.jucer
@@ -1,6 +1,6 @@
 <?xml version="1.0" encoding="UTF-8"?>
 
-<JUCERPROJECT id="EHAJpP" name="DawDreamer" projectType="dll" version="0.6.5"
+<JUCERPROJECT id="EHAJpP" name="DawDreamer" projectType="dll" version="0.6.6"
               bundleIdentifier="design.dirt.DawDreamer" includeBinaryInAppConfig="1"
               displaySplashScreen="1" reportAppUsage="0" splashScreenColour="Dark"
               cppLanguageStandard="17" companyCopyright="" jucerFormatVersion="1"
@@ -198,7 +198,6 @@
       <FILE id="jkEvDu" name="AddProcessor.h" compile="0" resource="0" file="Source/AddProcessor.h"/>
       <FILE id="WFptFr" name="CompressorProcessor.h" compile="0" resource="0"
             file="Source/CompressorProcessor.h"/>
-      <FILE id="VdWCsf" name="AllProcessors.h" compile="0" resource="0" file="Source/AllProcessors.h"/>
       <FILE id="wA54Bg" name="FaustProcessor.h" compile="0" resource="0"
             file="Source/FaustProcessor.h"/>
       <FILE id="TwJEHz" name="FaustProcessor.cpp" compile="1" resource="0"
@@ -211,6 +210,8 @@
             file="Source/OscillatorProcessor.h"/>
       <FILE id="tlHHTL" name="PlaybackWarpProcessor.h" compile="0" resource="0"
             file="Source/PlaybackWarpProcessor.h"/>
+      <FILE id="qEKHvR" name="PlaybackWarpProcessor.cpp" compile="1" resource="0"
+            file="Source/PlaybackWarpProcessor.cpp"/>
       <FILE id="D0T0n5" name="PlaybackProcessor.h" compile="0" resource="0"
             file="Source/PlaybackProcessor.h"/>
       <FILE id="sHn6tW" name="PluginProcessor.h" compile="0" resource="0"
@@ -224,6 +225,8 @@
     <GROUP id="{6A50F3BF-C55C-AF7D-5BA6-E62FF469B8C4}" name="Source"/>
     <FILE id="ejbgC9" name="CustomParameters.h" compile="0" resource="0"
           file="Source/CustomParameters.h"/>
+    <FILE id="jUQDYH" name="CustomParameters.cpp" compile="1" resource="0"
+          file="Source/CustomParameters.cpp"/>
     <FILE id="rDTPMs" name="custom_pybind_wrappers.h" compile="0" resource="0"
           file="Source/custom_pybind_wrappers.h"/>
     <FILE id="p932XA" name="custom_pybind_wrappers.cpp" compile="1" resource="0"
diff --git a/JuceLibraryCode/JuceHeader.h b/JuceLibraryCode/JuceHeader.h
index eae8ef8b..4d62661e 100644
--- a/JuceLibraryCode/JuceHeader.h
+++ b/JuceLibraryCode/JuceHeader.h
@@ -46,7 +46,7 @@ namespace ProjectInfo
 {
     const char* const  projectName    = "DawDreamer";
     const char* const  companyName    = "";
-    const char* const  versionString  = "0.6.5";
-    const int          versionNumber  = 0x605;
+    const char* const  versionString  = "0.6.6";
+    const int          versionNumber  = 0x606;
 }
 #endif
diff --git a/README.md b/README.md
index 6d71bbef..7f6e7f24 100644
--- a/README.md
+++ b/README.md
@@ -75,7 +75,7 @@ wavfile.write('sine_demo.wav', SAMPLE_RATE, audio.transpose())
 
 ## Advanced Example
 
-Let's demonstate audio playback, graph-building, VST instruments/effects, and automation. You need to change many hard-coded paths for this to work.
+Let's demonstrate audio playback, graph-building, VST instruments/effects, and automation. You need to change many hard-coded paths for this to work.
 
 ```python
 import dawdreamer as daw
@@ -84,9 +84,9 @@ from scipy.io import wavfile
 import librosa
 
 SAMPLE_RATE = 44100
-BUFFER_SIZE = 128 # Parameters will undergo automation at this buffer/block size. It can be as small as 1 sample.
-SYNTH_PLUGIN = "C:/path/to/synth.dll"  # for instruments, DLLs work.
-REVERB_PLUGIN = "C:/path/to/reverb.dll"  # extensions: .dll, .vst3, .component
+BUFFER_SIZE = 128 # Parameters will undergo automation at this buffer/block size.
+SYNTH_PLUGIN = "C:/path/to/synth.dll"  # extensions: .dll, .vst3, .vst, .component
+REVERB_PLUGIN = "C:/path/to/reverb.dll"  # extensions: .dll, .vst3, .vst, .component
 VOCALS_PATH = "C:/path/to/vocals.wav"
 PIANO_PATH = "C:/path/to/piano.wav"
 
@@ -102,7 +102,13 @@ def make_sine(freq: float, duration: float, sr=SAMPLE_RATE):
 
 # Make an engine. We'll only need one.
 engine = daw.RenderEngine(SAMPLE_RATE, BUFFER_SIZE)
-engine.set_bpm(120.)  # default is 120.
+engine.set_bpm(120.)  # default is 120 beats per minute.
+
+# The BPM can also be set as a numpy array that is interpreted
+# with a fixed PPQN (Pulses Per Quarter Note).
+# If we choose ppqn=960 and the numpy array abruptly changes values every 960 samples,
+# the tempo will abruptly change "on the beat".
+engine.set_bpm(120.+60.*make_sine(1./2., duration*10., ppqn=960))
 
 DURATION = 10 # How many seconds we want to render.
 
@@ -126,16 +132,35 @@ synth.load_vst3_preset('C:/path/to/preset.vstpreset')
 
 # Get a list of dictionaries where each dictionary describes a controllable parameter.
 print(synth.get_plugin_parameters_description()) 
-print(synth.get_parameter_name(1)) # For Serum, returns "A Pan" (the panning of oscillator A)
+print(synth.get_parameter_name(1)) # For Serum, returns "A Pan" (oscillator A's panning)
 # Note that Plugin Processor parameters are between [0, 1], even "discrete" parameters.
-# The Plugin Processor can set automation according to a parameter index.
-synth.set_automation(1, 0.5+.5*make_sine(.5, DURATION)) # 0.5 Hz sine wave remapped to [0, 1]
-# We can simply use one value constantly:
+# We can simply set a constant value.
 synth.set_parameter(1, 0.1234)
+# The Plugin Processor can set automation with data at audio rate.
+synth.set_automation(1, 0.5+.5*make_sine(.5, DURATION)) # 0.5 Hz sine wave remapped to [0, 1]
+
+# It's also possible to set automation in alignment with the tempo.
+# Let's make a numpy array whose PPQN is 960.
+# Each 960 values in the array correspond to a quarter note of time progressing.
+# Let's make a parameter alternate between 0.25 and 0.75 four times per beat.
+automation = make_sine(4, DURATION, sr=960)
+automation = 0.25+.5*(automation > 0).astype(np.float32)
+synth.set_automation(1, automation, ppqn=960)
+
+# Load a MIDI file and convert the timing to absolute seconds. Changes to the Render Engine's BPM
+# won't affect the timing. The kwargs below are defaults.
+synth.load_midi("C:/path/to/song.mid", clear_previous=True, convert_to_sec=True, all_events=True)
+
+# Load a MIDI file and keep the timing in units of beats. Changes to the Render Engine's BPM
+# will affect the timing.
+synth.load_midi("C:/path/to/song.mid", convert_to_sec=False)
+
+# We can also add one note at a time, specifying a start time and duration, both in seconds
+synth.add_midi_note(60, 127, 0.5, .25) # (MIDI note, velocity, start, duration)
 
-synth.load_midi("C:/path/to/song.mid")
-# We can also add notes one at a time.
-synth.add_midi_note(67, 127, 0.5, .25) # (MIDI note, velocity, start sec, duration sec)
+# With `convert_to_sec=False`, we can use beats as the unit for the start time and duration.
+# Rest for a beat and then play a note for a half beat.
+synth.add_midi_note(67, 127, 1, .5, convert_to_sec=False)
 
 # For any processor type, we can get the number of inputs and outputs
 print("synth num inputs: ", synth.get_num_input_channels())
@@ -153,7 +178,7 @@ filter_processor = engine.make_filter_processor("filter", "high", 7000.0, .5, 1.
 filter_processor.freq = 7123.  # Some parameters can be get/set like this.
 freq_automation = make_sine(.5, DURATION)*5000. + 7000. # 0.5 Hz sine wave centered at 7000 w/ amp 5000.
 filter_processor.set_automation("freq", freq_automation) # argument is single channel numpy array.
-freq_automation = filter_processor.get_automation("freq") # You can get automation of most processor parameters.
+freq_automation = filter_processor.get_automation("freq") # Get automation of most processor parameters.
 filter_processor.record = True  # This will allow us to access the filter processor's audio after a render.
 
 # A graph is a meaningfully ordered list of tuples.
@@ -164,7 +189,7 @@ filter_processor.record = True  # This will allow us to access the filter proces
 # The audio from the last tuple's processor will be accessed automatically later by engine.get_audio()
 graph = [
   (synth, []),  # synth takes no inputs, so we give an empty list.
-  (engine.make_reverb_processor("reverb"), [synth.get_name()]), # Apply JUCE reverb to the synth named earlier
+  (engine.make_reverb_processor("reverb"), [synth.get_name()]), # Apply JUCE reverb to synth from earlier
   (engine.make_plugin_processor("more_reverb", REVERB_PLUGIN), ["reverb"]), # Apply VST reverb
   (engine.make_playback_processor("vocals", vocals), []), # Playback has no inputs.
   (filter_processor, ["vocals"]), # High-pass filter with automation set earlier.
@@ -216,7 +241,7 @@ faust_processor = engine.make_faust_processor("faust")
 faust_processor.set_dsp(DSP_PATH)  # You can do this anytime.
 
 # Using compile() isn't necessary, but it's an early warning check.
-faust_processor.compile()
+faust_processor.compile() # throws a catchable Python Runtime Error for bad Faust code
 
 print(faust_processor.get_parameters_description())
 
diff --git a/Source/AllProcessors.h b/Source/AllProcessors.h
deleted file mode 100644
index c11d03ac..00000000
--- a/Source/AllProcessors.h
+++ /dev/null
@@ -1,15 +0,0 @@
-#pragma once
-
-#include "ProcessorBase.h"
-#include "AddProcessor.h"
-#include "CompressorProcessor.h"
-#include "DelayProcessor.h"
-#include "FaustProcessor.h"
-#include "FilterProcessor.h"
-#include "OscillatorProcessor.h"
-#include "PlaybackProcessor.h"
-#include "PlaybackWarpProcessor.h"
-#include "PluginProcessor.h"
-#include "ReverbProcessor.h"
-#include "PannerProcessor.h"
-#include "SamplerProcessor.h"
diff --git a/Source/CompressorProcessor.h b/Source/CompressorProcessor.h
index 079b6114..76bd9410 100644
--- a/Source/CompressorProcessor.h
+++ b/Source/CompressorProcessor.h
@@ -42,10 +42,10 @@ class CompressorProcessor : public ProcessorBase
         AudioPlayHead::CurrentPositionInfo posInfo;
         getPlayHead()->getCurrentPosition(posInfo);
 
-        *myThreshold = getAutomationVal("threshold", posInfo.timeInSamples);
-        *myRatio = getAutomationVal("ratio", posInfo.timeInSamples);
-        *myAttack = getAutomationVal("attack", posInfo.timeInSamples);
-        *myRelease = getAutomationVal("release", posInfo.timeInSamples);
+        *myThreshold = getAutomationVal("threshold", posInfo);
+        *myRatio = getAutomationVal("ratio", posInfo);
+        *myAttack = getAutomationVal("attack", posInfo);
+        *myRelease = getAutomationVal("release", posInfo);
         updateParameters();
     }
 
@@ -56,16 +56,16 @@ class CompressorProcessor : public ProcessorBase
     const juce::String getName() { return "CompressorProcessor"; };
 
     void setThreshold(float threshold) { setAutomationVal("threshold", threshold); }
-    float getThreshold() { return getAutomationVal("threshold", 0); }
+    float getThreshold() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("threshold", posInfo); }
 
     void setRatio(float ratio) { setAutomationVal("ratio", ratio); }
-    float getRatio() { return getAutomationVal("ratio", 0); }
+    float getRatio() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("ratio", posInfo); }
 
     void setAttack(float attack) { setAutomationVal("attack", attack); }
-    float getAttack() { return getAutomationVal("attack", 0); }
+    float getAttack() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("attack", posInfo); }
 
     void setRelease(float release) { setAutomationVal("release", release); }
-    float getRelease() { return getAutomationVal("release", 0); }
+    float getRelease() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("release", posInfo); }
 
 
 private:
diff --git a/Source/CustomParameters.cpp b/Source/CustomParameters.cpp
new file mode 100644
index 00000000..1bd918ef
--- /dev/null
+++ b/Source/CustomParameters.cpp
@@ -0,0 +1,59 @@
+#include "CustomParameters.h"
+#include "ProcessorBase.h"
+
+#include <algorithm>
+
+bool
+AutomateParameter::setAutomation(py::array_t<float> input, std::uint32_t newPPQN) {
+
+    if (newPPQN < 0) {
+        throw std::runtime_error("The PPQN must be greater than or equal to zero. Received: " + std::to_string(newPPQN));
+    }
+
+    m_ppqn = newPPQN;
+
+    try
+    {
+        myAutomation.clear();
+
+        auto numSamples = input.shape(0);
+
+        myAutomation = std::vector<float>(numSamples, 0.f);
+
+        memcpy(myAutomation.data(), (float*)input.data(), numSamples * sizeof(float));
+    }
+    catch (const std::exception& e)
+    {
+        throw std::runtime_error(std::string("Error: setAutomation: ") + e.what());
+        return false;
+    }
+
+    return true;
+}
+
+void
+AutomateParameter::setAutomation(const float val) {
+    myAutomation.clear();
+    myAutomation.push_back(val);
+}
+
+std::vector<float>
+AutomateParameter::getAutomation() {
+    return myAutomation;
+}
+
+float
+AutomateParameter::sample(juce::AudioPlayHead::CurrentPositionInfo& posInfo) {
+
+    size_t i;
+    if (m_ppqn > 0) {
+        i = std::min(myAutomation.size() - 1, size_t(posInfo.ppqPosition * m_ppqn));
+    }
+    else {
+        i = std::min(myAutomation.size() - 1, size_t(posInfo.timeInSamples));
+    }
+
+    i = std::max((size_t)0, i);
+    return myAutomation.at(i);
+
+}
\ No newline at end of file
diff --git a/Source/CustomParameters.h b/Source/CustomParameters.h
index bb3fe540..2adaa667 100644
--- a/Source/CustomParameters.h
+++ b/Source/CustomParameters.h
@@ -1,6 +1,7 @@
 #pragma once
 
 #include "../JuceLibraryCode/JuceHeader.h"
+#include "custom_pybind_wrappers.h"
 
 using juce::ADSR;
 using juce::AbstractFifo;
@@ -112,6 +113,7 @@ using juce::roundFloatToInt;
 using juce::roundToInt;
 using juce::uint8;
 
+
 class AutomateParameter
 {
 
@@ -119,51 +121,20 @@ class AutomateParameter
 
     AutomateParameter() {}
 
-    bool setAutomation(py::array_t<float> input) {
-
-        try
-        {
-            float* input_ptr = (float*)input.data();
-            myAutomation.clear();
-
-            myAutomation = std::vector<float>(input.shape(0), 0.f);
+    bool setAutomation(py::array_t<float> input, std::uint32_t newPPQN);
 
-            for (int x = 0; x < input.shape(0); x++) {
-                myAutomation[x] = *(input_ptr++);
-            }
+    void setAutomation(const float val);
 
-        }
-        catch (const std::exception& e)
-        {
-            throw std::runtime_error(std::string("Error: setAutomation: ") + e.what());
-            return false;
-        }
+    std::vector<float> getAutomation();
 
-        return true;
-    }
+    float sample(juce::AudioPlayHead::CurrentPositionInfo& posInfo);
 
-    void setAutomation(const float val) {
-        myAutomation.clear();
-        myAutomation.push_back(val);
-    }
-
-    std::vector<float> getAutomation() {
-        return myAutomation;
-    }
-
-    float sample(size_t index) {
-        auto i = std::min(myAutomation.size() - 1, index);
-        i = std::max((size_t)0, i);
-        return myAutomation.at(i);
-    }
+    ~AutomateParameter() {}
 
 protected:
 
     std::vector<float> myAutomation;
-
-    ~AutomateParameter()
-    {
-    }
+    std::uint32_t m_ppqn = 0;
 
 };
 
diff --git a/Source/DelayProcessor.h b/Source/DelayProcessor.h
index 91324b04..ef82d805 100644
--- a/Source/DelayProcessor.h
+++ b/Source/DelayProcessor.h
@@ -50,8 +50,8 @@ class DelayProcessor : public ProcessorBase
         AudioPlayHead::CurrentPositionInfo posInfo;
         getPlayHead()->getCurrentPosition(posInfo);
 
-        *myWetLevel = getAutomationVal("wet_level", posInfo.timeInSamples);
-        *myDelaySize = getAutomationVal("delay", posInfo.timeInSamples);
+        *myWetLevel = getAutomationVal("wet_level", posInfo);
+        *myDelaySize = getAutomationVal("delay", posInfo);
         updateParameters();
     }
 
@@ -62,10 +62,10 @@ class DelayProcessor : public ProcessorBase
     const juce::String getName() { return "DelayProcessor"; };
 
     void setDelay(float newDelaySize) { setAutomationVal("delay", newDelaySize); }
-    float getDelay() { return getAutomationVal("delay", 0); }
+    float getDelay() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("delay", posInfo); }
 
     void setWet(float newWet) { setAutomationVal("wet_level", newWet); }
-    float getWet() { return getAutomationVal("wet_level", 0); }
+    float getWet() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("wet_level", posInfo); }
 
 
 private:
diff --git a/Source/FaustProcessor.cpp b/Source/FaustProcessor.cpp
index 13a996a0..969116cb 100644
--- a/Source/FaustProcessor.cpp
+++ b/Source/FaustProcessor.cpp
@@ -41,7 +41,17 @@ FaustProcessor::FaustProcessor(std::string newUniqueName, double sampleRate, int
 
 FaustProcessor::~FaustProcessor() {
 	clear();
-	delete myMidiIterator;
+	delete myMidiIteratorQN;
+	delete myMidiIteratorSec;
+}
+
+bool
+FaustProcessor::setAutomation(std::string parameterName, py::array input, std::uint32_t ppqn) {
+
+	if (!m_isCompiled) {
+		this->compile();
+	}
+	return ProcessorBase::setAutomation(parameterName, input, ppqn);
 }
 
 bool
@@ -52,6 +62,9 @@ FaustProcessor::canApplyBusesLayout(const juce::AudioProcessor::BusesLayout& lay
 void
 FaustProcessor::prepareToPlay(double sampleRate, int samplesPerBlock)
 {
+	if (!m_isCompiled) {
+		this->compile();
+	}
 }
 
 void
@@ -63,45 +76,70 @@ FaustProcessor::processBlock(juce::AudioSampleBuffer& buffer, juce::MidiBuffer&
 	getPlayHead()->getCurrentPosition(posInfo);
 
 	if (!m_isCompiled) {
-		ProcessorBase::processBlock(buffer, midiBuffer);
-		return;
+		throw std::runtime_error("Faust Processor called processBlock but it wasn't compiled.");
 	}
 
 	if (m_nvoices < 1) {
 		if (m_dsp != NULL) {
 			m_dsp->compute(buffer.getNumSamples(), (float**)buffer.getArrayOfReadPointers(), buffer.getArrayOfWritePointers());
 		}
+		else {
+			throw std::runtime_error("Faust Processor: m_dsp is null");
+		}
 	}
 	else if (m_dsp_poly != NULL) {
-		long long int start = posInfo.timeInSamples;
+		auto start = posInfo.timeInSamples;
+
+		auto pulseStart = std::floor(posInfo.ppqPosition * PPQN);
+		auto pulseStep = (posInfo.bpm * PPQN) / (mySampleRate * 60.);
 
-		// render one sample at a time
+		// render one sample at a time because we want accurate timing of keyOn/keyOff.
+
+		auto oneSampReadPtrs = (float**)oneSampleInBuffer.getArrayOfReadPointers();
+		auto oneSampWritePtrs = (float**)oneSampleOutBuffer.getArrayOfWritePointers();
+		const int midiChannel = 0;
 
 		for (size_t i = 0; i < buffer.getNumSamples(); i++)
 		{
-			myIsMessageBetween = myMidiMessagePosition >= start && myMidiMessagePosition < start + 1;
-			do {
-				if (myIsMessageBetween) {
+			{
+				myIsMessageBetweenSec = myMidiMessagePositionSec >= start && myMidiMessagePositionSec < start + 1;
+				while (myIsMessageBetweenSec && myMidiEventsDoRemainSec) {
 
-					int midiChannel = 0;
-					if (myMidiMessage.isNoteOn()) {
-						m_dsp_poly->keyOn(midiChannel, myMidiMessage.getNoteNumber(), myMidiMessage.getVelocity());
+					if (myMidiMessageSec.isNoteOn()) {
+						m_dsp_poly->keyOn(midiChannel, myMidiMessageSec.getNoteNumber(), myMidiMessageSec.getVelocity());
 					}
-					else if (myMidiMessage.isNoteOff()) {
-						m_dsp_poly->keyOff(midiChannel, myMidiMessage.getNoteNumber(), myMidiMessage.getVelocity());
+					else if (myMidiMessageSec.isNoteOff()) {
+						m_dsp_poly->keyOff(midiChannel, myMidiMessageSec.getNoteNumber(), myMidiMessageSec.getVelocity());
 					}
 
-					myMidiEventsDoRemain = myMidiIterator->getNextEvent(myMidiMessage, myMidiMessagePosition);
-					myIsMessageBetween = myMidiMessagePosition >= start && myMidiMessagePosition < start + 1;
+					myMidiEventsDoRemainSec = myMidiIteratorSec->getNextEvent(myMidiMessageSec, myMidiMessagePositionSec);
+					myIsMessageBetweenSec = myMidiMessagePositionSec >= start && myMidiMessagePositionSec < start + 1;
 				}
-			} while (myIsMessageBetween && myMidiEventsDoRemain);
+			}
+
+			{
+				myIsMessageBetweenQN = myMidiMessagePositionQN >= pulseStart && myMidiMessagePositionQN < pulseStart + 1;
+				while (myIsMessageBetweenQN && myMidiEventsDoRemainQN) {
+
+					if (myMidiMessageQN.isNoteOn()) {
+						m_dsp_poly->keyOn(midiChannel, myMidiMessageQN.getNoteNumber(), myMidiMessageQN.getVelocity());
+					}
+					else if (myMidiMessageQN.isNoteOff()) {
+						m_dsp_poly->keyOff(midiChannel, myMidiMessageQN.getNoteNumber(), myMidiMessageQN.getVelocity());
+					}
+
+					myMidiEventsDoRemainQN = myMidiIteratorQN->getNextEvent(myMidiMessageQN, myMidiMessagePositionQN);
+
+					myIsMessageBetweenQN = myMidiMessagePositionQN >= pulseStart && myMidiMessagePositionQN < pulseStart + 1;
+				}
+			}
 
 			for (size_t chan = 0; chan < m_numInputChannels; chan++)
 			{
 				oneSampleInBuffer.setSample(chan, 0, buffer.getSample(chan, i));
 			}
 
-			m_dsp_poly->compute(1, (float**)oneSampleInBuffer.getArrayOfReadPointers(), (float**)oneSampleOutBuffer.getArrayOfWritePointers());
+			m_dsp_poly->compute(1, oneSampReadPtrs, oneSampWritePtrs);
 
 			for (size_t chan = 0; chan < m_numOutputChannels; chan++)
 			{
@@ -109,8 +147,12 @@ FaustProcessor::processBlock(juce::AudioSampleBuffer& buffer, juce::MidiBuffer&
 			}
 
 			start += 1;
+			pulseStart += pulseStep;
 		}
 	}
+	else {
+		throw std::runtime_error("Faust Processor: m_dsp_poly is null");
+	}
 
 	ProcessorBase::processBlock(buffer, midiBuffer);
 }
@@ -147,7 +189,7 @@ FaustProcessor::automateParameters() {
 		int faustIndex = m_map_juceIndex_to_faustIndex[i];
 
 		if (theParameter) {
-			m_ui->setParamValue(faustIndex, theParameter->sample(posInfo.timeInSamples));
+			m_ui->setParamValue(faustIndex, theParameter->sample(posInfo));
 		}
 		else {
 			auto theName = this->getParameterName(i);
@@ -172,6 +214,7 @@ FaustProcessor::automateParameters() {
 void
 FaustProcessor::reset()
 {
+
 	if (m_dsp) {
 		m_dsp->instanceClear();
 	}
@@ -180,9 +223,13 @@ FaustProcessor::reset()
 		m_dsp_poly->instanceClear();
 	}
 
-	delete myMidiIterator;
-	myMidiIterator = new MidiBuffer::Iterator(myMidiBuffer); // todo: deprecated.
-	myMidiEventsDoRemain = myMidiIterator->getNextEvent(myMidiMessage, myMidiMessagePosition);
+	delete myMidiIteratorQN;
+	myMidiIteratorQN = new MidiBuffer::Iterator(myMidiBufferQN); // todo: deprecated.
+	myMidiEventsDoRemainQN = myMidiIteratorQN->getNextEvent(myMidiMessageQN, myMidiMessagePositionQN);
+
+	delete myMidiIteratorSec;
+	myMidiIteratorSec = new MidiBuffer::Iterator(myMidiBufferSec); // todo: deprecated.
+	myMidiEventsDoRemainSec = myMidiIteratorSec->getNextEvent(myMidiMessageSec, myMidiMessagePositionSec);
 
 	if (!m_isCompiled) {
 		this->compile();
@@ -250,8 +297,6 @@ FaustProcessor::setDSPString(const std::string& code)
 	return true;
 }
 
-#define FAUSTPROCESSOR_FAIL_COMPILE clear(); return false;
-
 bool
 FaustProcessor::compile()
 {
@@ -266,16 +311,16 @@ FaustProcessor::compile()
 
 	auto pathToFaustLibraries = getPathToFaustLibraries();
 
-	int argc = 0;
-	const char** argv = new const char* [256];
-	if (pathToFaustLibraries.compare(std::string("")) != 0) {
-		argv[argc++] = "-I";
-		argv[argc++] = pathToFaustLibraries.c_str();
-	}
-	else {
+	if (pathToFaustLibraries.compare(std::string("")) == 0) {
 		throw std::runtime_error("FaustProcessor::compile(): Error for path for faust libraries: " + pathToFaustLibraries);
 	}
 
+	int argc = 0;
+	const char** argv = new const char* [256];
+
+	argv[argc++] = "-I";
+	argv[argc++] = pathToFaustLibraries.c_str();
+
 	if (m_faustLibrariesPath.compare(std::string("")) != 0) {
 		argv[argc++] = "-I";
 		argv[argc++] = m_faustLibrariesPath.c_str();
@@ -308,11 +353,17 @@ FaustProcessor::compile()
 			argc, argv, target.c_str(), m_errorString, optimize);
 	}
 
+	for (int i = 0; i < argc; i++) {
+		argv[i] = NULL;
+	}
+	delete[] argv;
+	argv = nullptr;
+
 	// check for error
 	if (m_errorString != "") {
 		// output error
+		clear();
 		throw std::runtime_error("FaustProcessor::compile(): " + m_errorString + ". Check the faustlibraries path: " + pathToFaustLibraries);
-		FAUSTPROCESSOR_FAIL_COMPILE
 	}
 
 	//// print where faustlib is looking for stdfaust.lib and the other lib files.
@@ -327,18 +378,12 @@ FaustProcessor::compile()
 	//    std::cout << name << "\n" << std::endl;
 	//}
 
-	for (int i = 0; i < argc; i++) {
-		argv[i] = NULL;
-	}
-	delete[] argv;
-	argv = nullptr;
-
 	if (is_polyphonic) {
 		// (false, true) works
 		m_dsp_poly = m_poly_factory->createPolyDSPInstance(m_nvoices, true, m_groupVoices);
 		if (!m_dsp_poly) {
+			clear();
 			throw std::runtime_error("FaustProcessor::compile(): Cannot create Poly DSP instance.");
-			FAUSTPROCESSOR_FAIL_COMPILE
 		}
 		m_dsp_poly->setReleaseLength(m_releaseLengthSec);
 	}
@@ -346,8 +391,8 @@ FaustProcessor::compile()
 		// create DSP instance
 		m_dsp = m_factory->createDSPInstance();
 		if (!m_dsp) {
+			clear();
 			throw std::runtime_error("FaustProcessor::compile(): Cannot create DSP instance.");
-			FAUSTPROCESSOR_FAIL_COMPILE
 		}
 	}
 
@@ -395,6 +440,11 @@ bool
 FaustProcessor::setDSPFile(const std::string& path)
 {
 	m_isCompiled = false;
+
+	if (!std::filesystem::exists(path.c_str())) {
+		throw std::runtime_error("File not found: " + path);
+	}
+
 	if (std::strcmp(path.c_str(), "") == 0) {
 		throw std::runtime_error("Path to DSP file is empty.");
 	}
@@ -459,7 +509,9 @@ FaustProcessor::getParamWithIndex(const int index)
 
 	auto& parAddress = it->second;
 
-	return this->getAutomationVal(parAddress, 0);
+	AudioPlayHead::CurrentPositionInfo posInfo;
+
+	return this->getAutomationVal(parAddress, posInfo);
 }
 
 float
@@ -471,7 +523,9 @@ FaustProcessor::getParamWithPath(const std::string& n)
 	}
 	if (!m_ui) return 0; // todo: better handling
 
-	return this->getAutomationVal(n, 0);
+	AudioPlayHead::CurrentPositionInfo posInfo;
+
+	return this->getAutomationVal(n, posInfo);
 }
 
 std::string
@@ -589,7 +643,8 @@ FaustProcessor::getPluginParametersDescription()
 			myDictionary["min"] = m_ui->getParamMin(faustIndex);
 			myDictionary["max"] = m_ui->getParamMax(faustIndex);
 			myDictionary["step"] = m_ui->getParamStep(faustIndex);
-			myDictionary["value"] = this->getAutomationVal(theName, 0);
+			AudioPlayHead::CurrentPositionInfo posInfo;
+			myDictionary["value"] = this->getAutomationVal(theName, posInfo);
 
 			myList.append(myDictionary);
 		}
@@ -605,25 +660,54 @@ FaustProcessor::getPluginParametersDescription()
 int
 FaustProcessor::getNumMidiEvents()
 {
-	return myMidiBuffer.getNumEvents();
+	return myMidiBufferSec.getNumEvents() + myMidiBufferQN.getNumEvents();
 };
 
 bool
-FaustProcessor::loadMidi(const std::string& path)
+FaustProcessor::loadMidi(const std::string& path, bool clearPrevious, bool convertToSeconds, bool allEvents)
 {
+	if (!std::filesystem::exists(path.c_str())) {
+		throw std::runtime_error("File not found: " + path);
+	}
+
 	File file = File(path);
 	FileInputStream fileStream(file);
 	MidiFile midiFile;
 	midiFile.readFrom(fileStream);
-	midiFile.convertTimestampTicksToSeconds();
-	myMidiBuffer.clear();
-
-	for (int t = 0; t < midiFile.getNumTracks(); t++) {
-		const MidiMessageSequence* track = midiFile.getTrack(t);
-		for (int i = 0; i < track->getNumEvents(); i++) {
-			MidiMessage& m = track->getEventPointer(i)->message;
-			int sampleOffset = (int)(mySampleRate * m.getTimeStamp());
-			myMidiBuffer.addEvent(m, sampleOffset);
+
+	if (clearPrevious) {
+		myMidiBufferSec.clear();
+		myMidiBufferQN.clear();
+	}
+
+	if (convertToSeconds) {
+		midiFile.convertTimestampTicksToSeconds();
+		
+		for (int t = 0; t < midiFile.getNumTracks(); t++) {
+			const MidiMessageSequence* track = midiFile.getTrack(t);
+			for (int i = 0; i < track->getNumEvents(); i++) {
+				MidiMessage& m = track->getEventPointer(i)->message;
+				int sampleOffset = (int)(mySampleRate * m.getTimeStamp());
+				if (allEvents || m.isNoteOff() || m.isNoteOn()) {
+					myMidiBufferSec.addEvent(m, sampleOffset);
+				}
+			}
+		}
+	}
+	else {
+		auto timeFormat = midiFile.getTimeFormat(); // the ppqn (Ableton makes midi files with 96 ppqn)
+
+		for (int t = 0; t < midiFile.getNumTracks(); t++) {
+			const MidiMessageSequence* track = midiFile.getTrack(t);
+			for (int i = 0; i < track->getNumEvents(); i++) {
+				MidiMessage& m = track->getEventPointer(i)->message;
+
+				if (allEvents || m.isNoteOff() || m.isNoteOn()) {
+					// convert timestamp from its original time format to our high resolution PPQN
+					auto timeStamp = m.getTimeStamp() * PPQN / timeFormat;
+					myMidiBufferQN.addEvent(m, timeStamp);
+				}
+			}
 		}
 	}
 
@@ -632,14 +716,16 @@ FaustProcessor::loadMidi(const std::string& path)
 
 void
 FaustProcessor::clearMidi() {
-	myMidiBuffer.clear();
+	myMidiBufferSec.clear();
+	myMidiBufferQN.clear();
 }
 
 bool
 FaustProcessor::addMidiNote(uint8  midiNote,
 	uint8  midiVelocity,
 	const double noteStart,
-	const double noteLength) {
+	const double noteLength,
+	bool convert_to_sec) {
 
 	if (midiNote > 255) midiNote = 255;
 	if (midiNote < 0) midiNote = 0;
@@ -658,11 +744,20 @@ FaustProcessor::addMidiNote(uint8  midiNote,
 		midiNote,
 		midiVelocity);
 
-	auto startTime = noteStart * mySampleRate;
-	onMessage.setTimeStamp(startTime);
-	offMessage.setTimeStamp(startTime + noteLength * mySampleRate);
-	myMidiBuffer.addEvent(onMessage, (int)onMessage.getTimeStamp());
-	myMidiBuffer.addEvent(offMessage, (int)offMessage.getTimeStamp());
+	if (convert_to_sec) {
+		auto startTime = noteStart * mySampleRate;
+		onMessage.setTimeStamp(startTime);
+		offMessage.setTimeStamp(startTime + noteLength * mySampleRate);
+		myMidiBufferSec.addEvent(onMessage, (int)onMessage.getTimeStamp());
+		myMidiBufferSec.addEvent(offMessage, (int)offMessage.getTimeStamp());
+	}
+	else {
+		auto startTime = noteStart * PPQN;
+		onMessage.setTimeStamp(startTime);
+		offMessage.setTimeStamp(startTime + noteLength * PPQN);
+		myMidiBufferQN.addEvent(onMessage, (int)onMessage.getTimeStamp());
+		myMidiBufferQN.addEvent(offMessage, (int)offMessage.getTimeStamp());
+	}
 
 	return true;
 }
@@ -754,7 +849,6 @@ FaustProcessor::getPathToFaustLibraries() {
 	catch (...) {
 		throw std::runtime_error("Error getting path to faustlibraries.");
 	}
-	return "";
 }
 
 using  myaudiotype = py::array_t<float, py::array::c_style | py::array::forcecast>;
diff --git a/Source/FaustProcessor.h b/Source/FaustProcessor.h
index 553c78f8..a6801eb7 100644
--- a/Source/FaustProcessor.h
+++ b/Source/FaustProcessor.h
@@ -117,6 +117,8 @@ class FaustProcessor : public ProcessorBase
 
     const juce::String getName() const { return "FaustProcessor"; }
 
+    bool setAutomation(std::string parameterName, py::array input, std::uint32_t ppqn);
+
     // faust stuff
     void clear();
     bool compile();
@@ -139,7 +141,7 @@ class FaustProcessor : public ProcessorBase
     void setAutoImport(const std::string& s) { m_autoImport = s; }
     std::string getAutoImport() { return m_autoImport; }
 
-    bool loadMidi(const std::string& path);
+    bool loadMidi(const std::string& path, bool clearPrevious, bool convertToSeconds, bool allEvents);
 
     void clearMidi();
 
@@ -148,7 +150,8 @@ class FaustProcessor : public ProcessorBase
     bool addMidiNote(const uint8  midiNote,
         const uint8  midiVelocity,
         const double noteStart,
-        const double noteLength);
+        const double noteLength,
+        bool convert_to_sec);
 
     void setSoundfiles(py::dict);
 
@@ -199,12 +202,23 @@ class FaustProcessor : public ProcessorBase
     bool m_groupVoices = true;
     bool m_isCompiled = false;
 
-    MidiBuffer myMidiBuffer;
-    MidiMessage myMidiMessage;
-    int myMidiMessagePosition = -1;
-    MidiBuffer::Iterator* myMidiIterator = nullptr;
-    bool myIsMessageBetween = false;
-    bool myMidiEventsDoRemain = false;
+    MidiBuffer myMidiBufferQN;
+    MidiBuffer myMidiBufferSec;
+
+    MidiMessage myMidiMessageQN;
+    MidiMessage myMidiMessageSec;
+
+    int myMidiMessagePositionQN = -1;
+    int myMidiMessagePositionSec = -1;
+
+    MidiBuffer::Iterator* myMidiIteratorQN = nullptr;
+    MidiBuffer::Iterator* myMidiIteratorSec = nullptr;
+
+    bool myIsMessageBetweenQN = false;
+    bool myIsMessageBetweenSec = false;
+
+    bool myMidiEventsDoRemainQN = false;
+    bool myMidiEventsDoRemainSec = false;
 
     juce::AudioSampleBuffer oneSampleInBuffer;
     juce::AudioSampleBuffer oneSampleOutBuffer;
diff --git a/Source/FilterProcessor.cpp b/Source/FilterProcessor.cpp
index bb4eaca1..3ce0679c 100644
--- a/Source/FilterProcessor.cpp
+++ b/Source/FilterProcessor.cpp
@@ -48,9 +48,9 @@ void FilterProcessor::automateParameters() {
     AudioPlayHead::CurrentPositionInfo posInfo;
     getPlayHead()->getCurrentPosition(posInfo);
 
-    *myFreq = getAutomationVal("freq", posInfo.timeInSamples);
-    *myQ = getAutomationVal("q", posInfo.timeInSamples);
-    *myGain = getAutomationVal("gain", posInfo.timeInSamples);
+    *myFreq = getAutomationVal("freq", posInfo);
+    *myQ = getAutomationVal("q", posInfo);
+    *myGain = getAutomationVal("gain", posInfo);
     
     switch (myMode)
     {
@@ -160,14 +160,14 @@ FilterProcessor::getMode() {
 void
 FilterProcessor::setFrequency(float freq) { setAutomationVal("freq", freq);}
 float
-FilterProcessor::getFrequency() { return getAutomationVal("freq", 0); }
+FilterProcessor::getFrequency() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("freq", posInfo); }
 
 void
 FilterProcessor::setQ(float q) { setAutomationVal("q", q);}
 float
-FilterProcessor::getQ() { return getAutomationVal("q", 0); }
+FilterProcessor::getQ() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("q", posInfo); }
 
 void
 FilterProcessor::setGain(float gain) { setAutomationVal("gain", gain);}
 float
-FilterProcessor::getGain() { return getAutomationVal("gain", 0);}
+FilterProcessor::getGain() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("gain", posInfo);}
diff --git a/Source/PannerProcessor.h b/Source/PannerProcessor.h
index 1228a0f6..3ba2c902 100644
--- a/Source/PannerProcessor.h
+++ b/Source/PannerProcessor.h
@@ -37,7 +37,7 @@ class PannerProcessor : public ProcessorBase
         AudioPlayHead::CurrentPositionInfo posInfo;
         getPlayHead()->getCurrentPosition(posInfo);
 
-        *myPan = getAutomationVal("pan", posInfo.timeInSamples);
+        *myPan = getAutomationVal("pan", posInfo);
         updateParameters();
     }
 
@@ -48,7 +48,7 @@ class PannerProcessor : public ProcessorBase
     const juce::String getName() { return "PannerProcessor"; };
 
     void setPan(float newPanVal) { setAutomationVal("pan", newPanVal); }
-    float getPan() { return getAutomationVal("pan", 0); }
+    float getPan() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("pan", posInfo); }
 
     void setRule(std::string newRule) {
         myRule = stringToRule(newRule);
diff --git a/Source/PlaybackWarpProcessor.cpp b/Source/PlaybackWarpProcessor.cpp
new file mode 100644
index 00000000..67d0b41b
--- /dev/null
+++ b/Source/PlaybackWarpProcessor.cpp
@@ -0,0 +1,404 @@
+#include "PlaybackWarpProcessor.h"
+#ifdef BUILD_DAWDREAMER_RUBBERBAND
+
+PlaybackWarpProcessor::PlaybackWarpProcessor(std::string newUniqueName, std::vector<std::vector<float>> inputData, double sr) : ProcessorBase{ createParameterLayout, newUniqueName }
+{
+    m_numChannels = (int)inputData.size();
+    setMainBusInputsAndOutputs(0, m_numChannels);
+    const int numSamples = (int)inputData.at(0).size();
+
+    myPlaybackData.setSize(m_numChannels, numSamples);
+    for (int chan = 0; chan < m_numChannels; chan++) {
+        myPlaybackData.copyFrom(chan, 0, inputData.at(chan).data(), numSamples);
+    }
+
+    init(sr);
+}
+
+PlaybackWarpProcessor::PlaybackWarpProcessor(std::string newUniqueName, py::array_t<float, py::array::c_style | py::array::forcecast> input, double sr) : ProcessorBase{ createParameterLayout, newUniqueName }
+{
+    setData(input);
+    init(sr);
+}
+
+void
+PlaybackWarpProcessor::init(double sr) {
+    m_sample_rate = sr;
+    setAutomationVal("transpose", 0.);
+    myTranspose = myParameters.getRawParameterValue("transpose");
+    setupRubberband(sr, m_numChannels);
+    setClipPositionsDefault();
+}
+
+
+void
+PlaybackWarpProcessor::setClipPositionsDefault() {
+
+    std::vector<std::tuple<float, float, float>> positions;
+
+    positions.push_back(std::tuple<float, float, float>(0.f, 65536.f, 0.f));
+
+    setClipPositions(positions);
+}
+
+void
+PlaybackWarpProcessor::prepareToPlay(double, int) {
+
+}
+
+void
+PlaybackWarpProcessor::automateParameters() {
+
+    AudioPlayHead::CurrentPositionInfo posInfo;
+    getPlayHead()->getCurrentPosition(posInfo);
+
+    *myTranspose = getAutomationVal("transpose", posInfo);
+    double scale = std::pow(2., *myTranspose / 12.);
+    m_rbstretcher->setPitchScale(scale);
+}
+
+
+py::array_t<float>
+PlaybackWarpProcessor::getWarpMarkers() {
+
+    py::array_t<float, py::array::c_style> arr({ (int)m_clipInfo.warp_markers.size(), 2 });
+
+    auto ra = arr.mutable_unchecked();
+
+    int i = 0;
+    for (auto& warp_marker : m_clipInfo.warp_markers) {
+        ra(i, 0) = warp_marker.first; // time in seconds in the audio
+        ra(i, 1) = warp_marker.second; // time in beats in the audio, relative to 1.1.1
+        i++;
+    }
+
+    return arr;
+}
+
+void
+PlaybackWarpProcessor::resetWarpMarkers(double bpm) {
+    m_clipInfo.warp_markers.clear();
+
+    m_clipInfo.warp_markers.push_back(std::make_pair(0, 0));
+    double numSamples = 128;
+    double beats = bpm / (60. * numSamples / m_sample_rate);
+    m_clipInfo.warp_markers.push_back(std::make_pair(numSamples, beats));
+}
+
+void
+PlaybackWarpProcessor::setWarpMarkers(py::array_t<float, py::array::c_style | py::array::forcecast> input) {
+
+    if (input.ndim() != 2) {
+        throw std::runtime_error("The warp markers must be two-dimensional and shaped (num_markers, 2).");
+        return;
+    }
+
+    const int numPairs = (int)input.shape(0);
+
+    if (numPairs < 2) {
+        throw std::runtime_error("The number of warp markers must be greater than one.");
+        return;
+    }
+
+    if (input.shape(1) != 2) {
+        throw std::runtime_error("The dimensions of the passed warp markers are incorrect.");
+        return;
+    }
+
+    std::vector<std::pair<double, double>> warp_markers;
+
+    double beat, new_beat;
+    double pos, new_pos;
+    beat = new_beat = pos = new_pos = -999999.;
+
+    float* input_ptr = (float*)input.data();
+
+    for (int pair_i = 0; pair_i < numPairs; pair_i++) {
+
+        new_pos = *input_ptr++;
+        new_beat = *input_ptr++;
+
+        if (new_beat <= beat || new_pos <= pos) {
+            throw std::runtime_error("The warp markers must be monotonically increasing. new_beat: " + std::to_string(new_beat) + " beat: " + std::to_string(beat) + " new_pos: " + std::to_string(new_pos) + " pos: " + std::to_string(pos));
+        }
+
+        pos = new_pos;
+        beat = new_beat;
+
+        warp_markers.push_back(std::make_pair(pos, beat));
+    }
+
+    m_clipInfo.warp_markers = warp_markers;
+}
+
+
+bool
+PlaybackWarpProcessor::setClipPositions(std::vector<std::tuple<float, float, float>> positions) {
+
+    // a position is a (clip start, clip end, clip offset)
+    // clip start: The position in beats relative to the engine's timeline where the clip starts
+    // clip end: The position in beats relative to the engine's timeline where the clip ends
+    // clip offset: A clip's first sample is determined by the "start marker" in the ASD file.
+    //              This is an offset to that start marker.
+
+    m_clips.clear();
+
+    for (auto& position : positions) {
+
+        Clip clip = Clip((double)std::get<0>(position), (double)std::get<1>(position), (double)std::get<2>(position));
+        m_clips.push_back(clip);
+    }
+
+    return true;
+}
+
+void
+PlaybackWarpProcessor::processBlock(juce::AudioSampleBuffer& buffer, juce::MidiBuffer& midiBuffer)
+{
+    AudioPlayHead::CurrentPositionInfo posInfo;
+    getPlayHead()->getCurrentPosition(posInfo);
+
+    automateParameters();
+
+    if (m_clips.size() == 0) {
+        ProcessorBase::processBlock(buffer, midiBuffer);
+        return;
+    }
+
+    if (m_clipIndex >= m_clips.size()) {
+        // we've already passed the last clip.
+        ProcessorBase::processBlock(buffer, midiBuffer);
+        return;
+    }
+
+    double movingPPQ = posInfo.ppqPosition;
+
+    double nextPPQ = posInfo.ppqPosition + (double(buffer.getNumSamples()) / m_sample_rate) * posInfo.bpm / 60.;
+
+    std::uint32_t numAvailable = 0;
+    const std::uint32_t numSamplesNeeded = buffer.getNumSamples();
+
+    std::uint32_t numWritten = 0;
+    std::uint64_t numToRetrieve = 0;
+
+    while (numWritten < numSamplesNeeded) {
+        // In this loop, figure out just one sample at a time.
+        // There are a lot of things to juggle including:
+        // The rubberband stretcher: does it have samples available? what samples should we tell it to process?
+        // The global clip position: are we inside a region that should be producing any kind of audio at all or silence?
+        // The local clip position: Do we have samples for the requested sample index, or do we need to loop to another position, or fake zeros?
+        // The clip info: is warping enabled, is looping enabled.
+
+        numAvailable = m_rbstretcher->available();
+
+        numToRetrieve = std::min(numAvailable, numSamplesNeeded - numWritten);
+        numToRetrieve = std::min(numToRetrieve, (std::uint64_t)(std::ceil((m_currentClip.end_pos - movingPPQ) / (posInfo.bpm) * 60. * m_sample_rate)));
+
+        if (numToRetrieve > 0) {
+            m_nonInterleavedBuffer.setSize(m_numChannels, numToRetrieve);
+            numToRetrieve = m_rbstretcher->retrieve(m_nonInterleavedBuffer.getArrayOfWritePointers(), numToRetrieve);
+
+            for (int chan = 0; chan < m_numChannels; chan++) {
+                auto chanPtr = m_nonInterleavedBuffer.getReadPointer(chan);
+                buffer.copyFrom(chan, numWritten, chanPtr, numToRetrieve);
+            }
+
+            numWritten += numToRetrieve;
+            movingPPQ += double(numToRetrieve) * posInfo.bpm / (m_sample_rate * 60.);
+            continue;
+        }
+
+        while (movingPPQ >= m_currentClip.end_pos) {
+            m_clipIndex += 1;
+            if (m_clipIndex < m_clips.size()) {
+                m_currentClip = m_clips.at(m_clipIndex);
+                setupRubberband(m_sample_rate, m_numChannels);
+                if (m_clipInfo.warp_on) {
+                    sampleReadIndex = m_clipInfo.beat_to_sample(m_clipInfo.start_marker + m_currentClip.start_marker_offset, m_sample_rate);
+                }
+                else {
+                    sampleReadIndex = 0;
+                }
+            }
+            else {
+                ProcessorBase::processBlock(buffer, midiBuffer);
+                return;
+            }
+        }
+
+        if (nextPPQ < m_currentClip.start_pos || movingPPQ < m_currentClip.start_pos) {
+            // write some zeros into the output
+            for (int chan = 0; chan < m_numChannels; chan++) {
+                buffer.setSample(chan, numWritten, 0.f);
+            }
+            numWritten += 1;
+            movingPPQ += posInfo.bpm / (m_sample_rate * 60.);
+            continue;
+        }
+
+        double ppqPosition = movingPPQ - m_currentClip.start_pos + m_currentClip.start_marker_offset;
+
+        bool past_end_marker_and_loop_off = ppqPosition > m_clipInfo.end_marker && !m_clipInfo.loop_on;
+        if (past_end_marker_and_loop_off || movingPPQ > m_currentClip.end_pos) {
+            m_clipIndex += 1;
+            if (m_clipIndex < m_clips.size()) {
+                // Use the next clip position.
+                m_currentClip = m_clips.at(m_clipIndex);
+                setupRubberband(m_sample_rate, m_numChannels);
+                if (m_clipInfo.warp_on) {
+                    sampleReadIndex = m_clipInfo.beat_to_sample(m_clipInfo.start_marker + m_currentClip.start_marker_offset, m_sample_rate);
+                }
+                else {
+                    sampleReadIndex = 0;
+                }
+                continue;
+            }
+            else {
+                for (int chan = 0; chan < m_numChannels; chan++) {
+                    buffer.setSample(chan, numWritten, 0.f);
+                }
+                numWritten += 1;
+                movingPPQ += posInfo.bpm / (m_sample_rate * 60.);
+                continue;
+            }
+        }
+
+        if (m_clipInfo.warp_on) {
+            // todo: if the playback data sample rate is different than the engine's sr
+            // then that would affect the call to setTimeRatio.
+
+            double instant_bpm;
+            double _;
+            m_clipInfo.beat_to_seconds(ppqPosition, _, instant_bpm);
+            m_rbstretcher->setTimeRatio(instant_bpm / posInfo.bpm);
+        }
+        else {
+            m_rbstretcher->setTimeRatio(m_time_ratio_if_warp_off);
+        }
+
+        if (m_clipInfo.loop_on) {
+            int loop_end_sample = m_clipInfo.beat_to_sample(m_clipInfo.loop_end, m_sample_rate);
+            if (sampleReadIndex > loop_end_sample) {
+                int loop_start_sample = m_clipInfo.beat_to_sample(m_clipInfo.loop_start, m_sample_rate);
+                sampleReadIndex = loop_start_sample;
+            }
+        }
+        else {
+            int end_marker_sample = myPlaybackData.getNumSamples() - 1;
+            if (sampleReadIndex > end_marker_sample) {
+                continue;
+            }
+        }
+
+        m_nonInterleavedBuffer.setSize(m_numChannels, 1);
+
+        // can we read from the playback data or are we out of bounds and we need to pass zeros to rubberband?
+        const int last_sample = myPlaybackData.getNumSamples() - 1;
+        if (sampleReadIndex > -1 && sampleReadIndex <= last_sample) {
+            for (int chan = 0; chan < m_numChannels; chan++) {
+                m_nonInterleavedBuffer.copyFrom(chan, 0, myPlaybackData, chan, sampleReadIndex, 1);
+            }
+        }
+        else {
+            // pass zeros because the requested clip loop parameters are asking for out of bounds samples.
+            m_nonInterleavedBuffer.clear();
+        }
+
+        m_rbstretcher->process(m_nonInterleavedBuffer.getArrayOfReadPointers(), m_nonInterleavedBuffer.getNumSamples(), false);
+
+        sampleReadIndex += 1;
+    }
+
+    ProcessorBase::processBlock(buffer, midiBuffer);
+}
+
+void
+PlaybackWarpProcessor::reset() {
+
+    setupRubberband(m_sample_rate, m_numChannels);
+
+    m_clipIndex = 0;
+    sampleReadIndex = 0;
+
+    if (m_clipIndex < m_clips.size()) {
+        m_currentClip = m_clips.at(0);
+        if (m_clipInfo.warp_on) {
+            sampleReadIndex = m_clipInfo.beat_to_sample(m_clipInfo.start_marker + m_currentClip.start_marker_offset, m_sample_rate);
+        }
+        else {
+            sampleReadIndex = 0;
+        }
+    }
+}
+
+void
+PlaybackWarpProcessor::setData(py::array_t<float, py::array::c_style | py::array::forcecast> input) {
+    float* input_ptr = (float*)input.data();
+
+    m_numChannels = (int)input.shape(0);
+    setMainBusInputsAndOutputs(0, m_numChannels);
+    const int numSamples = (int)input.shape(1);
+
+    myPlaybackData.setSize(m_numChannels, numSamples);
+    for (int chan = 0; chan < m_numChannels; chan++) {
+        myPlaybackData.copyFrom(chan, 0, input_ptr, numSamples);
+        input_ptr += numSamples;
+    }
+}
+
+bool
+PlaybackWarpProcessor::loadAbletonClipInfo(const char* filepath) {
+    return m_clipInfo.readWarpFile(filepath);;
+}
+
+void
+PlaybackWarpProcessor::setupRubberband(float sr, int numChannels) {
+    // Note that we call this instead of calling m_rbstretcher->reset() because
+    // that method doesn't seem to work correctly.
+    // It's better to just create a whole new stretcher object.
+    using namespace RubberBand;
+
+    RubberBandStretcher::Options options = 0;
+    options |= RubberBandStretcher::OptionProcessRealTime;
+    options |= RubberBandStretcher::OptionStretchPrecise;
+    //options |= RubberBandStretcher::OptionPhaseIndependent;
+    //options |= RubberBandStretcher::OptionWindowLong;
+    //options |= RubberBandStretcher::OptionWindowShort;
+    //options |= RubberBandStretcher::OptionSmoothingOn;
+    //options |= RubberBandStretcher::OptionFormantPreserved;
+    options |= RubberBandStretcher::OptionPitchHighQuality;
+    options |= RubberBandStretcher::OptionChannelsTogether;  // enabling this is NOT the default
+
+    // Pick one of these:
+    //options |= RubberBandStretcher::OptionThreadingAuto;
+    options |= RubberBandStretcher::OptionThreadingNever;
+    //options |= RubberBandStretcher::OptionThreadingAlways;
+
+    // Pick one of these:
+    options |= RubberBandStretcher::OptionTransientsSmooth;
+    //options |= RubberBandStretcher::OptionTransientsMixed;
+    //options |= RubberBandStretcher::OptionTransientsCrisp;
+
+    // Pick one of these:
+    options |= RubberBandStretcher::OptionDetectorCompound;
+    //options |= RubberBandStretcher::OptionDetectorPercussive;
+    //options |= RubberBandStretcher::OptionDetectorSoft;
+
+    m_rbstretcher = std::make_unique<RubberBand::RubberBandStretcher>(
+        sr,
+        numChannels,
+        options,
+        1.,
+        1.);
+}
+
+juce::AudioProcessorValueTreeState::ParameterLayout
+PlaybackWarpProcessor::createParameterLayout()
+{
+    juce::AudioProcessorValueTreeState::ParameterLayout params;
+
+    params.add(std::make_unique<AutomateParameterFloat>("transpose", "transpose", NormalisableRange<float>(-96.f, 96.f), 0.f));
+    return params;
+}
+
+#endif
\ No newline at end of file
diff --git a/Source/PlaybackWarpProcessor.h b/Source/PlaybackWarpProcessor.h
index 152c79c0..f55adc7a 100644
--- a/Source/PlaybackWarpProcessor.h
+++ b/Source/PlaybackWarpProcessor.h
@@ -11,69 +11,27 @@
 class PlaybackWarpProcessor : public ProcessorBase
 {
 public:
-    PlaybackWarpProcessor(std::string newUniqueName, std::vector<std::vector<float>> inputData, double sr) : ProcessorBase{ createParameterLayout, newUniqueName }
-    {
-        m_numChannels = (int) inputData.size();
-        setMainBusInputsAndOutputs(0, m_numChannels);
-        const int numSamples = (int)inputData.at(0).size();
-        
-        myPlaybackData.setSize(m_numChannels, numSamples);
-        for (int chan = 0; chan < m_numChannels; chan++) {
-            myPlaybackData.copyFrom(chan, 0, inputData.at(chan).data(), numSamples);
-        }
-
-        init(sr);
-    }
-
-    PlaybackWarpProcessor(std::string newUniqueName, py::array_t<float, py::array::c_style | py::array::forcecast> input, double sr) : ProcessorBase{ createParameterLayout, newUniqueName }
-    {
-        setData(input);
-        init(sr);
-    }
+    PlaybackWarpProcessor(std::string newUniqueName, std::vector<std::vector<float>> inputData, double sr);
 
-private:
-    void init(double sr) {
-        m_sample_rate = sr;
-        setAutomationVal("transpose", 0.);
-        myTranspose = myParameters.getRawParameterValue("transpose");
-        setupRubberband(sr, m_numChannels);
-        setClipPositionsDefault();
-    }
-
-    void setClipPositionsDefault() {
-
-        std::vector<std::tuple<float, float, float>> positions;
-
-        positions.push_back(std::tuple<float, float, float>(0.f, 65536.f, 0.f));
+    PlaybackWarpProcessor(std::string newUniqueName, py::array_t<float, py::array::c_style | py::array::forcecast> input, double sr);
 
-        setClipPositions(positions);
-    }
+    void prepareToPlay(double, int);
 
-public:
-    void
-    prepareToPlay(double, int) {
+    void automateParameters();
 
-    }
+    void processBlock(juce::AudioSampleBuffer& buffer, juce::MidiBuffer& midiBuffer);
 
-    void setTimeRatio(double ratio) {
-        m_time_ratio_if_warp_off = ratio;
-    }
-    double getTimeRatio() {
-        return m_time_ratio_if_warp_off;
-    }
+    void reset();
 
-    void automateParameters() {
+    const juce::String getName() const { return "PlaybackWarpProcessor"; }
 
-        AudioPlayHead::CurrentPositionInfo posInfo;
-        getPlayHead()->getCurrentPosition(posInfo);
+    void setData(py::array_t<float, py::array::c_style | py::array::forcecast> input);
 
-        *myTranspose = getAutomationVal("transpose", posInfo.timeInSamples);
-        double scale = std::pow(2., *myTranspose / 12.);
-        m_rbstretcher->setPitchScale(scale);
-    }
+    void setTimeRatio(double ratio) { m_time_ratio_if_warp_off = ratio; }
+    double getTimeRatio() { return m_time_ratio_if_warp_off; }
 
     void setTranspose(float newVal) { setAutomationVal("transpose", newVal); }
-    float getTranspose() { return getAutomationVal("transpose", 0); }
+    float getTranspose() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("transpose", posInfo); }
 
     bool getWarpOn() { return m_clipInfo.warp_on; }
     void setWarpOn(bool warpOn) { m_clipInfo.warp_on = warpOn; }
@@ -89,308 +47,30 @@ class PlaybackWarpProcessor : public ProcessorBase
     double getEndMarker() { return m_clipInfo.end_marker; }
     void setEndMarker(double endMarker) { m_clipInfo.end_marker = endMarker;}
 
-    py::array_t<float> getWarpMarkers() {
-
-        py::array_t<float, py::array::c_style> arr({ (int)m_clipInfo.warp_markers.size(), 2 });
-
-        auto ra = arr.mutable_unchecked();
-
-        int i = 0;
-        for (auto& warp_marker : m_clipInfo.warp_markers) {
-            ra(i, 0) = warp_marker.first; // time in seconds in the audio
-            ra(i, 1) = warp_marker.second; // time in beats in the audio, relative to 1.1.1
-            i++;
-        }
-
-        return arr;
-    }
-
-    void resetWarpMarkers(double bpm) {
-        m_clipInfo.warp_markers.clear();
-
-        m_clipInfo.warp_markers.push_back(std::make_pair(0, 0));
-        double numSamples = 128;
-        double beats = bpm / (60. * numSamples / m_sample_rate);
-        m_clipInfo.warp_markers.push_back(std::make_pair(numSamples, beats));
-    }
-
-    void setWarpMarkers(py::array_t<float, py::array::c_style | py::array::forcecast> input) {
-
-        if (input.ndim() != 2) {
-            throw std::runtime_error("The warp markers must be two-dimensional and shaped (num_markers, 2).");
-            return;
-        }
-
-        const int numPairs = (int)input.shape(0);
-
-        if (numPairs < 2) {
-            throw std::runtime_error("The number of warp markers must be greater than one.");
-            return;
-        }
+    py::array_t<float> getWarpMarkers();
 
-        if (input.shape(1) != 2) {
-            throw std::runtime_error("The dimensions of the passed warp markers are incorrect.");
-            return;
-        }
+    void resetWarpMarkers(double bpm);
 
-        std::vector<std::pair<double, double>> warp_markers;
+    void setWarpMarkers(py::array_t<float, py::array::c_style | py::array::forcecast> input);
 
-        double beat, new_beat;
-        double pos, new_pos;
-        beat = new_beat = pos = new_pos = -999999.;
+    bool setClipPositions(std::vector<std::tuple<float, float, float>> positions);
 
-        float* input_ptr = (float*)input.data();
-
-        for (int pair_i = 0; pair_i < numPairs; pair_i++) {
-
-            new_pos = *input_ptr++;
-            new_beat = *input_ptr++;
-
-            if (new_beat <= beat || new_pos <= pos) {
-                throw std::runtime_error("The warp markers must be monotonically increasing. new_beat: " + std::to_string(new_beat) + " beat: " + std::to_string(beat) + " new_pos: " + std::to_string(new_pos) + " pos: " + std::to_string(pos));
-            }
-
-            pos = new_pos;
-            beat = new_beat;
-
-            warp_markers.push_back(std::make_pair(pos, beat));
-        }
-
-        m_clipInfo.warp_markers = warp_markers;
-    }
+    bool loadAbletonClipInfo(const char* filepath);
 
 private:
-    class Clip {
-    public:
-        Clip(double startPos, double endPos, double startMarkerOffset) : start_pos{ startPos }, end_pos{ endPos }, start_marker_offset{ startMarkerOffset } {};
-        Clip() : start_pos { 0. }, end_pos{ 4. }, start_marker_offset{ 0. } {};
-        double start_pos = 0.;
-        double end_pos = 4.;
-        double start_marker_offset = 0.;
-    };
-public:
 
-    bool setClipPositions(std::vector<std::tuple<float, float, float>> positions) {
-
-        // a position is a (clip start, clip end, clip offset)
-        // clip start: The position in beats relative to the engine's timeline where the clip starts
-        // clip end: The position in beats relative to the engine's timeline where the clip ends
-        // clip offset: A clip's first sample is determined by the "start marker" in the ASD file.
-        //              This is an offset to that start marker.
-
-        m_clips.clear();
-
-        for (auto& position : positions) {
-
-            Clip clip = Clip((double)std::get<0>(position), (double)std::get<1>(position), (double)std::get<2>(position));
-            m_clips.push_back(clip);
-        }
-
-        return true;
-    }
-   
-    void
-    processBlock(juce::AudioSampleBuffer& buffer, juce::MidiBuffer& midiBuffer)
-    {
-        AudioPlayHead::CurrentPositionInfo posInfo;
-        getPlayHead()->getCurrentPosition(posInfo);
-
-        automateParameters();
-
-        if (m_clips.size() == 0) {
-            ProcessorBase::processBlock(buffer, midiBuffer);
-            return;
-        }
-
-        if (m_clipIndex >= m_clips.size()) {
-            // we've already passed the last clip.
-            ProcessorBase::processBlock(buffer, midiBuffer);
-            return;
-        }
-
-        double movingPPQ = posInfo.ppqPosition;
-
-        double nextPPQ = posInfo.ppqPosition + (double(buffer.getNumSamples())/ m_sample_rate) * posInfo.bpm / 60.;
-        
-        int numAvailable = 0;
-        const int numSamplesNeeded = buffer.getNumSamples();
-
-        int numWritten = 0;
-
-        while (numWritten < numSamplesNeeded) {
-            // In this loop, figure out just one sample at a time.
-            // There are a lot of things to juggle including:
-            // The rubberband stretcher: does it have samples available? what samples should we tell it to process?
-            // The global clip position: are we inside a region that should be producing any kind of audio at all or silence?
-            // The local clip position: Do we have samples for the requested sample index, or do we need to loop to another position, or fake zeros?
-            // The clip info: is warping enabled, is looping enabled.
-
-            numAvailable = m_rbstretcher->available();
-
-            int numToRetrieve = std::min(numAvailable, numSamplesNeeded - numWritten);
-            numToRetrieve = std::min(numToRetrieve,int(std::ceil( (m_currentClip.end_pos-movingPPQ)/(posInfo.bpm)*60.*m_sample_rate)));
-
-            if (numToRetrieve > 0) {
-                m_nonInterleavedBuffer.setSize(m_numChannels, numToRetrieve);
-                numToRetrieve = m_rbstretcher->retrieve(m_nonInterleavedBuffer.getArrayOfWritePointers(), numToRetrieve);
-
-                for (int chan = 0; chan < m_numChannels; chan++) {
-                    auto chanPtr = m_nonInterleavedBuffer.getReadPointer(chan);
-                    buffer.copyFrom(chan, numWritten, chanPtr, numToRetrieve);
-                }
-
-                numWritten += numToRetrieve;
-                movingPPQ += (double)(numToRetrieve)*posInfo.bpm / (m_sample_rate * 60.);
-                continue;
-            }
-
-            while (movingPPQ >= m_currentClip.end_pos) {
-                m_clipIndex += 1;
-                if (m_clipIndex < m_clips.size()) {
-                    m_currentClip = m_clips.at(m_clipIndex);
-                    setupRubberband(m_sample_rate, m_numChannels);
-                    if (m_clipInfo.warp_on) {
-                        sampleReadIndex = m_clipInfo.beat_to_sample(m_clipInfo.start_marker + m_currentClip.start_marker_offset, m_sample_rate);
-                    }
-                    else {
-                        sampleReadIndex = 0;
-                    }
-                }
-                else {
-                    ProcessorBase::processBlock(buffer, midiBuffer);
-                    return;
-                }
-            }
-
-            if (nextPPQ < m_currentClip.start_pos || movingPPQ < m_currentClip.start_pos) {
-                // write some zeros into the output
-                for (int chan = 0; chan < m_numChannels; chan++) {
-                    buffer.setSample(chan, numWritten, 0.f);
-                }
-                numWritten += 1;
-                movingPPQ += posInfo.bpm / (m_sample_rate * 60.);
-
-                continue;
-            }
-
-            double ppqPosition = movingPPQ - m_currentClip.start_pos + m_currentClip.start_marker_offset;
-
-            bool past_end_marker_and_loop_off = ppqPosition > m_clipInfo.end_marker && !m_clipInfo.loop_on;
-            if (past_end_marker_and_loop_off || movingPPQ > m_currentClip.end_pos) {
-                m_clipIndex += 1;
-                if (m_clipIndex < m_clips.size()) {
-                    // Use the next clip position.
-                    m_currentClip = m_clips.at(m_clipIndex);
-                    setupRubberband(m_sample_rate, m_numChannels);
-                    if (m_clipInfo.warp_on) {
-                        sampleReadIndex = m_clipInfo.beat_to_sample(m_clipInfo.start_marker + m_currentClip.start_marker_offset, m_sample_rate);
-                    }
-                    else {
-                        sampleReadIndex = 0;
-                    }
-                    
-                    continue;
-                }
-                else {
-                    for (int chan = 0; chan < m_numChannels; chan++) {
-                        buffer.setSample(chan, numWritten, 0.f);
-                    }
-                    numWritten += 1;
-                    movingPPQ += posInfo.bpm / (m_sample_rate * 60.);
-
-                    continue;
-                }
-            }
-
-            if (m_clipInfo.warp_on) {
-                // todo: if the playback data sample rate is different than the engine's sr
-                // then that would affect the call to setTimeRatio.
-
-                double instant_bpm;
-                double _;
-                m_clipInfo.beat_to_seconds(ppqPosition, _, instant_bpm);
-                m_rbstretcher->setTimeRatio(instant_bpm / posInfo.bpm);
-            }
-            else {
-                m_rbstretcher->setTimeRatio(m_time_ratio_if_warp_off);
-            }
-
-            if (m_clipInfo.loop_on) {
-                int loop_end_sample = m_clipInfo.beat_to_sample(m_clipInfo.loop_end, m_sample_rate);
-                if (sampleReadIndex > loop_end_sample) {
-                    int loop_start_sample = m_clipInfo.beat_to_sample(m_clipInfo.loop_start, m_sample_rate);
-                    sampleReadIndex = loop_start_sample;
-                }
-            }
-            else {
-                int end_marker_sample = myPlaybackData.getNumSamples() - 1;
-                if (sampleReadIndex > end_marker_sample) {
-                    continue;
-                }
-            }
-
-            m_nonInterleavedBuffer.setSize(m_numChannels, 1);
-
-            // can we read from the playback data or are we out of bounds and we need to pass zeros to rubberband?
-            const int last_sample = myPlaybackData.getNumSamples() - 1;
-            if (sampleReadIndex > -1 && sampleReadIndex <= last_sample) {
-                for (int chan = 0; chan < m_numChannels; chan++) {
-                    m_nonInterleavedBuffer.copyFrom(chan, 0, myPlaybackData, chan, sampleReadIndex, 1);
-                }
-            }
-            else {
-                // pass zeros because the requested clip loop parameters are asking for out of bounds samples.
-                m_nonInterleavedBuffer.clear();
-            }
-
-            m_rbstretcher->process(m_nonInterleavedBuffer.getArrayOfReadPointers(), m_nonInterleavedBuffer.getNumSamples(), false);
-            
-            sampleReadIndex += 1;
-        }
-
-        ProcessorBase::processBlock(buffer, midiBuffer);
-    }
-    
-    void
-    reset() {
-        
-        setupRubberband(m_sample_rate, m_numChannels);
-
-        m_clipIndex = 0;
-        sampleReadIndex = 0;
-
-        if (m_clipIndex < m_clips.size()) {
-            m_currentClip = m_clips.at(0);
-            if (m_clipInfo.warp_on) {
-                sampleReadIndex = m_clipInfo.beat_to_sample(m_clipInfo.start_marker + m_currentClip.start_marker_offset, m_sample_rate);
-            }
-            else {
-                sampleReadIndex = 0;
-            }
-        }
-    }
+    void init(double sr);
 
-    const juce::String getName() const { return "PlaybackWarpProcessor"; }
-
-    void setData(py::array_t<float, py::array::c_style | py::array::forcecast> input) {
-        float* input_ptr = (float*)input.data();
-        
-        m_numChannels = (int) input.shape(0);
-        setMainBusInputsAndOutputs(0, m_numChannels);
-        const int numSamples = (int) input.shape(1);
-
-        myPlaybackData.setSize(m_numChannels, numSamples);
-        for (int chan = 0; chan < m_numChannels; chan++) {
-            myPlaybackData.copyFrom(chan, 0, input_ptr, numSamples);
-            input_ptr += numSamples;
-        }
-    }
-
-    bool loadAbletonClipInfo(const char* filepath) {
-        return m_clipInfo.readWarpFile(filepath);;
-    }
+    void setClipPositionsDefault();
 
-private:
+    class Clip {
+        public:
+            Clip(double startPos, double endPos, double startMarkerOffset) : start_pos{ startPos }, end_pos{ endPos }, start_marker_offset{ startMarkerOffset } {};
+            Clip() : start_pos { 0. }, end_pos{ 4. }, start_marker_offset{ 0. } {};
+            double start_pos = 0.;
+            double end_pos = 4.;
+            double start_marker_offset = 0.;
+        };
 
     juce::AudioSampleBuffer myPlaybackData;
 
@@ -411,53 +91,9 @@ class PlaybackWarpProcessor : public ProcessorBase
     int m_clipIndex = 0;
     Clip m_currentClip;
 
-    // Note that we call this instead of calling m_rbstretcher->reset() because
-    // that method doesn't seem to work correctly.
-    // It's better to just create a whole new stretcher object.
-    void setupRubberband(float sr, int numChannels) {
-        using namespace RubberBand;
-
-        RubberBandStretcher::Options options = 0;
-        options |= RubberBandStretcher::OptionProcessRealTime;
-        options |= RubberBandStretcher::OptionStretchPrecise;
-        //options |= RubberBandStretcher::OptionPhaseIndependent;
-        //options |= RubberBandStretcher::OptionWindowLong;
-        //options |= RubberBandStretcher::OptionWindowShort;
-        //options |= RubberBandStretcher::OptionSmoothingOn;
-        //options |= RubberBandStretcher::OptionFormantPreserved;
-        options |= RubberBandStretcher::OptionPitchHighQuality;
-        options |= RubberBandStretcher::OptionChannelsTogether;  // enabling this is NOT the default
-
-        // Pick one of these:
-        options |= RubberBandStretcher::OptionThreadingAuto;
-        //options |= RubberBandStretcher::OptionThreadingNever;
-        //options |= RubberBandStretcher::OptionThreadingAlways;
-
-        // Pick one of these:
-        options |= RubberBandStretcher::OptionTransientsSmooth;
-        //options |= RubberBandStretcher::OptionTransientsMixed;
-        //options |= RubberBandStretcher::OptionTransientsCrisp;
-
-        // Pick one of these:
-        options |= RubberBandStretcher::OptionDetectorCompound;
-        //options |= RubberBandStretcher::OptionDetectorPercussive;
-        //options |= RubberBandStretcher::OptionDetectorSoft;
-
-        m_rbstretcher = std::make_unique<RubberBand::RubberBandStretcher>(
-            sr,
-            numChannels,
-            options,
-            1.,
-            1.);
-    }
-
-    static juce::AudioProcessorValueTreeState::ParameterLayout createParameterLayout()
-    {
-        juce::AudioProcessorValueTreeState::ParameterLayout params;
-
-        params.add(std::make_unique<AutomateParameterFloat>("transpose", "transpose", NormalisableRange<float>(-96.f, 96.f), 0.f));
-        return params;
-    }
+    void setupRubberband(float sr, int numChannels);
+
+    static juce::AudioProcessorValueTreeState::ParameterLayout createParameterLayout();
 };
 
 #endif
diff --git a/Source/PluginProcessor.cpp b/Source/PluginProcessor.cpp
index ff4f7851..7aa0fd29 100644
--- a/Source/PluginProcessor.cpp
+++ b/Source/PluginProcessor.cpp
@@ -130,7 +130,8 @@ PluginProcessor::~PluginProcessor() {
         std::lock_guard<std::mutex> lock(PLUGIN_INSTANCE_MUTEX);
         myPlugin.reset();
     }
-    delete myMidiIterator;
+    delete myMidiIteratorQN;
+    delete myMidiIteratorSec;
 }
 
 void PluginProcessor::setPlayHead(AudioPlayHead* newPlayHead)
@@ -174,28 +175,35 @@ PluginProcessor::processBlock(juce::AudioSampleBuffer& buffer, juce::MidiBuffer&
     AudioPlayHead::CurrentPositionInfo posInfo;
     getPlayHead()->getCurrentPosition(posInfo);
     myRenderMidiBuffer.clear();
-    
+
     if (!myPlugin.get() || !isLoaded) {
-        buffer.clear();
-        
-        if (posInfo.ppqPosition == 0) {
-            throw std::runtime_error("Error: no plugin was processed for processor named " + this->getUniqueName());
-        }
-        return;
+        throw std::runtime_error("Error: no plugin was processed for processor named " + this->getUniqueName());
     }
-    
+
     automateParameters();
 
-    long long int start = posInfo.timeInSamples;
-    long long int end = start + buffer.getNumSamples();
-    myIsMessageBetween = myMidiMessagePosition >= start && myMidiMessagePosition < end;
-    do {
-        if (myIsMessageBetween) {
-            myRenderMidiBuffer.addEvent(myMidiMessage, int(myMidiMessagePosition - start));
-            myMidiEventsDoRemain = myMidiIterator->getNextEvent(myMidiMessage, myMidiMessagePosition);
-            myIsMessageBetween = myMidiMessagePosition >= start && myMidiMessagePosition < end;
+    {
+        auto start = posInfo.timeInSamples;
+        auto end = start + buffer.getNumSamples();
+        myIsMessageBetweenSec = myMidiMessagePositionSec >= start && myMidiMessagePositionSec < end;
+        while (myIsMessageBetweenSec && myMidiEventsDoRemainSec) {
+            myRenderMidiBuffer.addEvent(myMidiMessageSec, int(myMidiMessagePositionSec - start));
+            myMidiEventsDoRemainSec = myMidiIteratorSec->getNextEvent(myMidiMessageSec, myMidiMessagePositionSec);
+            myIsMessageBetweenSec = myMidiMessagePositionSec >= start && myMidiMessagePositionSec < end;
+        }
+    }
+
+    {
+        auto pulseStart = std::floor(posInfo.ppqPosition * PPQN);
+        auto pulseEnd = pulseStart + buffer.getNumSamples() * (posInfo.bpm * PPQN) / (mySampleRate * 60.);
+
+        myIsMessageBetweenQN = myMidiMessagePositionQN >= pulseStart && myMidiMessagePositionQN < pulseEnd;
+        while (myIsMessageBetweenQN && myMidiEventsDoRemainQN) {
+            myRenderMidiBuffer.addEvent(myMidiMessageQN, int(myMidiMessagePositionQN - pulseStart));
+            myMidiEventsDoRemainQN = myMidiIteratorQN->getNextEvent(myMidiMessageQN, myMidiMessagePositionQN);
+            myIsMessageBetweenQN = myMidiMessagePositionQN >= pulseStart && myMidiMessagePositionQN < pulseEnd;
         }
-    } while (myIsMessageBetween && myMidiEventsDoRemain);
+    }
 
     myPlugin->processBlock(buffer, myRenderMidiBuffer);
     
@@ -217,7 +225,7 @@ PluginProcessor::automateParameters() {
             auto theParameter = ((AutomateParameterFloat*)myParameters.getParameter(paramID));
             if (theParameter) {
                 // todo: change to setParameterNotifyingHost?
-                myPlugin->setParameter(i, theParameter->sample(posInfo.timeInSamples));
+                myPlugin->setParameter(i, theParameter->sample(posInfo));
             }
             else {
                 throw std::runtime_error("Error automateParameters: " + myPlugin->getParameterName(i).toStdString());
@@ -233,10 +241,16 @@ PluginProcessor::reset()
         myPlugin->reset();
     }
 
-    delete myMidiIterator;
-    myMidiIterator = new MidiBuffer::Iterator(myMidiBuffer); // todo: deprecated.
+    delete myMidiIteratorSec;
+    myMidiIteratorSec = new MidiBuffer::Iterator(myMidiBufferSec); // todo: deprecated.
+
+    myMidiEventsDoRemainSec = myMidiIteratorSec->getNextEvent(myMidiMessageSec, myMidiMessagePositionSec);
+
+    delete myMidiIteratorQN;
+    myMidiIteratorQN = new MidiBuffer::Iterator(myMidiBufferQN); // todo: deprecated.
+
+    myMidiEventsDoRemainQN = myMidiIteratorQN->getNextEvent(myMidiMessageQN, myMidiMessagePositionQN);
 
-    myMidiEventsDoRemain = myMidiIterator->getNextEvent(myMidiMessage, myMidiMessagePosition);
     myRenderMidiBuffer.clear();
 }
 
@@ -434,6 +448,8 @@ PluginProcessor::getPatch() {
     params.clear();
     params.reserve(myPlugin->getNumParameters());
 
+
+    AudioPlayHead::CurrentPositionInfo posInfo;
     for (int i = 0; i < myPlugin->AudioProcessor::getNumParameters(); i++) {
 
         auto theName = myPlugin->getParameterName(i);
@@ -444,7 +460,7 @@ PluginProcessor::getPatch() {
 
         auto parameter = ((AutomateParameterFloat*)myParameters.getParameter(theName));
         if (parameter) {
-            float val = parameter->sample(0);
+            float val = parameter->sample(posInfo);
             if (parameter) {
                 params.push_back(std::make_pair(i, val));
             }
@@ -476,26 +492,52 @@ PluginProcessor::getPluginParameterSize()
 
 int
 PluginProcessor::getNumMidiEvents() {
-    return myMidiBuffer.getNumEvents();
+    return myMidiBufferSec.getNumEvents() + myMidiBufferQN.getNumEvents();
 };
 
 bool
-PluginProcessor::loadMidi(const std::string& path, bool allEvents)
+PluginProcessor::loadMidi(const std::string& path, bool clearPrevious, bool convertToSeconds, bool allEvents)
 {
+    if (!std::filesystem::exists(path.c_str())) {
+        throw std::runtime_error("File not found: " + path);
+    }
+
     File file = File(path);
     FileInputStream fileStream(file);
     MidiFile midiFile;
     midiFile.readFrom(fileStream);
-    midiFile.convertTimestampTicksToSeconds();
-    myMidiBuffer.clear();
-
-    for (int t = 0; t < midiFile.getNumTracks(); t++) {
-        const MidiMessageSequence* track = midiFile.getTrack(t);
-        for (int i = 0; i < track->getNumEvents(); i++) {
-            MidiMessage& m = track->getEventPointer(i)->message;
-            int sampleOffset = (int)(mySampleRate * m.getTimeStamp());
-            if (allEvents || m.isNoteOff() || m.isNoteOn()) {
-                myMidiBuffer.addEvent(m, sampleOffset);
+
+    if (clearPrevious) {
+        myMidiBufferSec.clear();
+        myMidiBufferQN.clear();
+    }
+
+    if (convertToSeconds) {
+        midiFile.convertTimestampTicksToSeconds();
+
+        for (int t = 0; t < midiFile.getNumTracks(); t++) {
+            const MidiMessageSequence* track = midiFile.getTrack(t);
+            for (int i = 0; i < track->getNumEvents(); i++) {
+                MidiMessage& m = track->getEventPointer(i)->message;
+                int sampleOffset = (int)(mySampleRate * m.getTimeStamp());
+                if (allEvents || m.isNoteOff() || m.isNoteOn()) {
+                    myMidiBufferSec.addEvent(m, sampleOffset);
+                }
+            }
+        }
+    }
+    else {
+        auto timeFormat = midiFile.getTimeFormat(); // the ppqn (Ableton makes midi files with 96 ppqn)
+
+        for (int t = 0; t < midiFile.getNumTracks(); t++) {
+            const MidiMessageSequence* track = midiFile.getTrack(t);
+            for (int i = 0; i < track->getNumEvents(); i++) {
+                MidiMessage& m = track->getEventPointer(i)->message;
+                if (allEvents || m.isNoteOff() || m.isNoteOn()) {
+                    // convert timestamp from its original time format to our high resolution PPQN
+                    auto timeStamp = m.getTimeStamp() * PPQN / timeFormat;
+                    myMidiBufferQN.addEvent(m, timeStamp);
+                }
             }
         }
     }
@@ -505,14 +547,16 @@ PluginProcessor::loadMidi(const std::string& path, bool allEvents)
 
 void
 PluginProcessor::clearMidi() {
-    myMidiBuffer.clear();
+    myMidiBufferSec.clear();
+    myMidiBufferQN.clear();
 }
 
 bool
 PluginProcessor::addMidiNote(uint8  midiNote,
     uint8  midiVelocity,
     const double noteStart,
-    const double noteLength) {
+    const double noteLength,
+    bool convert_to_sec) {
 
     if (midiNote > 255) midiNote = 255;
     if (midiNote < 0) midiNote = 0;
@@ -531,11 +575,20 @@ PluginProcessor::addMidiNote(uint8  midiNote,
         midiNote,
         midiVelocity);
 
-    auto startTime = noteStart * mySampleRate;
-    onMessage.setTimeStamp(startTime);
-    offMessage.setTimeStamp(startTime + noteLength * mySampleRate);
-    myMidiBuffer.addEvent(onMessage, (int)onMessage.getTimeStamp());
-    myMidiBuffer.addEvent(offMessage, (int)offMessage.getTimeStamp());
+    if (convert_to_sec) {
+        auto startTime = noteStart * mySampleRate;
+        onMessage.setTimeStamp(startTime);
+        offMessage.setTimeStamp(startTime + noteLength * mySampleRate);
+        myMidiBufferSec.addEvent(onMessage, (int)onMessage.getTimeStamp());
+        myMidiBufferSec.addEvent(offMessage, (int)offMessage.getTimeStamp());
+    }
+    else {
+        auto startTime = noteStart * PPQN;
+        onMessage.setTimeStamp(startTime);
+        offMessage.setTimeStamp(startTime + noteLength * PPQN);
+        myMidiBufferQN.addEvent(onMessage, (int)onMessage.getTimeStamp());
+        myMidiBufferQN.addEvent(offMessage, (int)offMessage.getTimeStamp());
+    }
 
     return true;
 }
@@ -573,7 +626,9 @@ PluginProcessorWrapper::wrapperGetParameter(int parameterIndex)
         return 0.;
     }
 
-    return ProcessorBase::getAutomationVal(std::to_string(parameterIndex), 0);
+    AudioPlayHead::CurrentPositionInfo posInfo;
+
+    return ProcessorBase::getAutomationVal(std::to_string(parameterIndex), posInfo);
 }
 
 std::string
@@ -595,8 +650,8 @@ PluginProcessorWrapper::wrapperSetParameter(int parameter, float value)
 }
 
 bool
-PluginProcessorWrapper::wrapperSetAutomation(int parameterIndex, py::array input) {
-    return PluginProcessorWrapper::setAutomation(std::to_string(parameterIndex), input);
+PluginProcessorWrapper::wrapperSetAutomation(int parameterIndex, py::array input, std::uint32_t ppqn) {
+    return PluginProcessorWrapper::setAutomation(std::to_string(parameterIndex), input, ppqn);
 }
 
 int
diff --git a/Source/PluginProcessor.h b/Source/PluginProcessor.h
index 8a9150b8..cf2c06f0 100644
--- a/Source/PluginProcessor.h
+++ b/Source/PluginProcessor.h
@@ -64,7 +64,7 @@ class PluginProcessor : public ProcessorBase
 
     const juce::String getName() const { return "PluginProcessor"; }
 
-    bool loadMidi(const std::string& path, bool allEvents);
+    bool loadMidi(const std::string& path, bool clearPrevious, bool convertToSeconds, bool allEvents);
 
     void clearMidi();
 
@@ -73,7 +73,8 @@ class PluginProcessor : public ProcessorBase
     bool addMidiNote(const uint8  midiNote,
         const uint8  midiVelocity,
         const double noteStart,
-        const double noteLength);
+        const double noteLength,
+        bool convert_to_sec);
 
     void setPlayHead(AudioPlayHead* newPlayHead);
 
@@ -91,13 +92,25 @@ class PluginProcessor : public ProcessorBase
     std::string myPluginPath;
     double mySampleRate;
 
-    MidiBuffer myMidiBuffer;
+    MidiBuffer myMidiBufferQN;
+    MidiBuffer myMidiBufferSec;
+
     MidiBuffer myRenderMidiBuffer;
-    MidiMessage myMidiMessage;
-    int myMidiMessagePosition = -1;
-    MidiBuffer::Iterator* myMidiIterator = nullptr;
-    bool myIsMessageBetween = false;
-    bool myMidiEventsDoRemain = false;
+
+    MidiMessage myMidiMessageQN;
+    MidiMessage myMidiMessageSec;
+
+    int myMidiMessagePositionQN = -1;
+    int myMidiMessagePositionSec = -1;
+
+    MidiBuffer::Iterator* myMidiIteratorQN = nullptr;
+    MidiBuffer::Iterator* myMidiIteratorSec = nullptr;
+
+    bool myIsMessageBetweenQN = false;
+    bool myIsMessageBetweenSec = false;
+
+    bool myMidiEventsDoRemainQN = false;
+    bool myMidiEventsDoRemainSec = false;
 
     void automateParameters();
 
@@ -124,7 +137,7 @@ class PluginProcessorWrapper : public PluginProcessor
 
     bool wrapperSetParameter(int parameter, float value);
 
-    bool wrapperSetAutomation(int parameterIndex, py::array input);
+    bool wrapperSetAutomation(int parameterIndex, py::array input, std::uint32_t ppqn);
 
     int wrapperGetPluginParameterSize();
 
diff --git a/Source/ProcessorBase.cpp b/Source/ProcessorBase.cpp
index 505408b9..01c331b8 100644
--- a/Source/ProcessorBase.cpp
+++ b/Source/ProcessorBase.cpp
@@ -23,14 +23,14 @@ ProcessorBase::setStateInformation(const void* data, int sizeInBytes)
             myParameters.replaceState(juce::ValueTree::fromXml(*xmlState));
 }
 
-bool ProcessorBase::setAutomation(std::string parameterName, py::array input) {
+bool ProcessorBase::setAutomation(std::string parameterName, py::array input, std::uint32_t ppqn) {
 
     try
     {
         auto parameter = (AutomateParameterFloat*)myParameters.getParameter(parameterName);  // todo: why do we have to cast to AutomateParameterFloat instead of AutomateParameter
 
         if (parameter) {
-            return parameter->setAutomation(input);
+            return parameter->setAutomation(input, ppqn);
         }
         else {
             throw std::runtime_error("Failed to find parameter: " + parameterName);
@@ -75,11 +75,11 @@ std::vector<float> ProcessorBase::getAutomation(std::string parameterName) {
     }
 }
 
-float ProcessorBase::getAutomationVal(std::string parameterName, int index) {
+float ProcessorBase::getAutomationVal(std::string parameterName, AudioPlayHead::CurrentPositionInfo& posInfo) {
     auto parameter = (AutomateParameterFloat*)myParameters.getParameter(parameterName);  // todo: why do we have to cast to AutomateParameterFloat instead of AutomateParameter
 
     if (parameter) {
-        return parameter->sample(index);
+        return parameter->sample(posInfo);
     }
     else {
         throw std::runtime_error("Failed to get automation value for parameter: " + parameterName);
diff --git a/Source/ProcessorBase.h b/Source/ProcessorBase.h
index 4c04a372..901302f9 100644
--- a/Source/ProcessorBase.h
+++ b/Source/ProcessorBase.h
@@ -29,7 +29,7 @@ class ProcessorBase : public juce::AudioProcessor
         if (!m_recordEnable) {
             return;
         }
-        AudioPlayHead::CurrentPositionInfo posInfo;
+        juce::AudioPlayHead::CurrentPositionInfo posInfo;
         getPlayHead()->getCurrentPosition(posInfo);
 
         const int numberChannels = myRecordBuffer.getNumChannels();
@@ -71,11 +71,11 @@ class ProcessorBase : public juce::AudioProcessor
     void getStateInformation(juce::MemoryBlock&);
     void setStateInformation(const void*, int);
 
-    bool setAutomation(std::string parameterName, py::array input);
+    bool setAutomation(std::string parameterName, py::array input, std::uint32_t ppqn);
 
-    bool setAutomationVal(std::string parameterName, float val);
+    virtual bool setAutomationVal(std::string parameterName, float val);
 
-    float getAutomationVal(std::string parameterName, int index);
+    float getAutomationVal(std::string parameterName, juce::AudioPlayHead::CurrentPositionInfo& posInfo);
 
     std::vector<float> getAutomation(std::string parameterName);
     py::array_t<float> getAutomationNumpy(std::string parameterName);
@@ -148,7 +148,6 @@ class ProcessorBase : public juce::AudioProcessor
         }
         else {
             throw std::invalid_argument(this->getUniqueName() + " CANNOT ApplyBusesLayout inputs: " + std::to_string(inputs) + " outputs: " + std::to_string(outputs));
-            return false;
         }
     }
 
@@ -157,6 +156,18 @@ class ProcessorBase : public juce::AudioProcessor
         return this->canApplyBusesLayout(busesLayout);
     }
 
+    // todo: this is not a good thing to hard-code.
+    // Ableton saves MIDI at a PPQN of 96, which is somewhat low.
+    // To be easily compatible with higher resolution PPQN MIDI files that
+    // are loaded with `load_midi`, we use an internal high rate PPQN of 960.
+    // It's easy to "upsample" to this higher resolution, as we do in `load_midi` and `add_midi_note`.
+    // Another bad design of the code is that FaustProcessor, PluginProcessor, and SamplerProcessor
+    // have a lot of common code related to MIDIBuffers.
+    // There's one set of variables for keeping track of everything in absolute time (when
+    // `convert_to_sec` is True), and another for relative-to-tempo timing (when note start
+    // times and durations are measured in "quarter notes" (QN)).
+    const static std::uint32_t PPQN = 3840;
+
 private:
     //==============================================================================
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR(ProcessorBase)
diff --git a/Source/RenderEngine.cpp b/Source/RenderEngine.cpp
index d1e7c34a..d62f6e13 100644
--- a/Source/RenderEngine.cpp
+++ b/Source/RenderEngine.cpp
@@ -1,5 +1,4 @@
 #include "RenderEngine.h"
-#include "AllProcessors.h"
 
 #include <unordered_map>
 
@@ -10,6 +9,19 @@ RenderEngine::RenderEngine(double sr, int bs) :
 {
     myMainProcessorGraph->setNonRealtime(true);
     myMainProcessorGraph->setPlayHead(this);
+
+    bpmAutomation.setSize(1, 1);
+    bpmAutomation.setSample(0, 0, 120.); // default 120 bpm
+}
+
+bool
+RenderEngine::removeProcessor(const std::string& name) {
+    if (m_UniqueNameToNodeID.find(name) != m_UniqueNameToNodeID.end()) {
+        myMainProcessorGraph->removeNode(m_UniqueNameToNodeID[name]);
+        m_UniqueNameToNodeID.erase(name);
+        return true;
+    }
+    return false;
 }
 
 bool
@@ -40,9 +52,6 @@ RenderEngine::loadGraph(DAG inDagNodes) {
         );
     }
 
-    myMainProcessorGraph->enableAllBuses();
-    // NB: don't enableAllBuses on all the nodes in the graph because
-    // it will actually mess them up (FaustProcessor)
     return success;
 }
 
@@ -88,6 +97,7 @@ RenderEngine::connectGraph() {
         }
                 
         processor->setPlayHead(this);
+        processor->prepareToPlay(mySampleRate, myBufferSize);
         processor->automateParameters();
         int numOutputAudioChans = processor->getMainBusNumOutputChannels();
         int expectedInputChannels = processor->getMainBusNumInputChannels();
@@ -128,23 +138,29 @@ RenderEngine::connectGraph() {
 
     myMainProcessorGraph->setPlayConfigDetails(0, 0, mySampleRate, myBufferSize);
     myMainProcessorGraph->prepareToPlay(mySampleRate, myBufferSize);
-    for (auto entry : m_stringDag) {
-        auto node = myMainProcessorGraph->getNodeForId(m_UniqueNameToNodeID[entry.first]);
-        node->getProcessor()->prepareToPlay(mySampleRate, myBufferSize);
-    }
     
     return true;
 }
 
+float RenderEngine::getBPM(double ppqPosition) {
+
+    int index = int(myBPMPPQN * ppqPosition);
+    index = std::min(bpmAutomation.getNumSamples() - 1, index);
+
+    auto bpm = bpmAutomation.getSample(0, index);
+
+    return bpm;
+}
+
 bool
 RenderEngine::render(const double renderLength) {
 
-    int numRenderedSamples = renderLength * mySampleRate;
+    std::uint64_t numRenderedSamples = renderLength * mySampleRate;
     if (numRenderedSamples <= 0) {
         throw std::runtime_error("Render length must be greater than zero.");
     }
     
-    int numberOfBuffers = myBufferSize == 1 ? numRenderedSamples : int(std::ceil((numRenderedSamples -1.) / myBufferSize));
+    std::uint64_t numberOfBuffers = myBufferSize == 1 ? numRenderedSamples : std::uint64_t(std::ceil((numRenderedSamples -1.) / myBufferSize));
 
     bool graphIsConnected = true;
     int audioBufferNumChans = 0;
@@ -168,26 +184,21 @@ RenderEngine::render(const double renderLength) {
         else {
             throw std::runtime_error("Unable to cast to Processor Base during render.");
         }
-
-        auto faustProcessor = dynamic_cast<FaustProcessor*> (processor);
-        if (faustProcessor && (!faustProcessor->isCompiled())) {
-            if (!faustProcessor->compile()) {
-                return false;
-            }
-        }
     }
 
     myMainProcessorGraph->reset();
     myMainProcessorGraph->setPlayHead(this);
 
     myCurrentPositionInfo.resetToDefault();
-    myCurrentPositionInfo.bpm = myBPM;
+    myCurrentPositionInfo.ppqPosition = 0.;
     myCurrentPositionInfo.isPlaying = true;
     myCurrentPositionInfo.isRecording = true;
+    myCurrentPositionInfo.timeInSeconds = 0;
     myCurrentPositionInfo.timeInSamples = 0;
     myCurrentPositionInfo.timeSigNumerator = 4;
     myCurrentPositionInfo.timeSigDenominator = 4;
     myCurrentPositionInfo.isLooping = false;
+    myCurrentPositionInfo.bpm = getBPM(myCurrentPositionInfo.ppqPosition);
     
     if (!graphIsConnected) {
         bool result = connectGraph();
@@ -197,6 +208,8 @@ RenderEngine::render(const double renderLength) {
     }
     
     AudioSampleBuffer audioBuffer(audioBufferNumChans, myBufferSize);
+
+    bool lastProcessorRecordEnable = false;
     
     for (auto entry : m_stringDag) {
         auto node = myMainProcessorGraph->getNodeForId(m_UniqueNameToNodeID[entry.first]);
@@ -205,6 +218,7 @@ RenderEngine::render(const double renderLength) {
         if (processor) {
             if (entry == m_stringDag.at(m_stringDag.size()-1)) {
                 // Always force the last processor to record.
+                lastProcessorRecordEnable = processor->getRecordEnable();
                 processor->setRecordEnable(true);
             }
             processor->setRecorderLength(processor->getRecordEnable() ? numRenderedSamples : 0);
@@ -215,17 +229,33 @@ RenderEngine::render(const double renderLength) {
     }
 
     MidiBuffer renderMidiBuffer;
+
+    auto stepInSeconds = double(myBufferSize) / mySampleRate;
     
-    for (long long int i = 0; i < numberOfBuffers; ++i)
+    for (std::uint64_t i = 0; i < numberOfBuffers; ++i)
     {
+        myCurrentPositionInfo.bpm = getBPM(myCurrentPositionInfo.ppqPosition);
+
         myMainProcessorGraph->processBlock(audioBuffer, renderMidiBuffer);
 
         myCurrentPositionInfo.timeInSamples += myBufferSize;
-        myCurrentPositionInfo.ppqPosition = (myCurrentPositionInfo.timeInSamples / (mySampleRate * 60.)) * myBPM;
+        myCurrentPositionInfo.timeInSeconds += stepInSeconds;
+        myCurrentPositionInfo.ppqPosition += (stepInSeconds / 60.) * myCurrentPositionInfo.bpm;
     }
 
     myCurrentPositionInfo.isPlaying = false;
     myCurrentPositionInfo.isRecording = false;
+
+    // restore the record-enable of the last processor.
+    if (m_stringDag.size()) {
+        
+        auto node = myMainProcessorGraph->getNodeForId(m_UniqueNameToNodeID[m_stringDag.at(m_stringDag.size() - 1).first]);
+        auto processor = dynamic_cast<ProcessorBase*> (node->getProcessor());
+
+        if (processor) {
+            processor->setRecordEnable(lastProcessorRecordEnable);
+        }
+    }
     
     return true;
 }
@@ -235,7 +265,25 @@ void RenderEngine::setBPM(double bpm) {
         throw std::runtime_error("BPM must be positive.");
         return;
     }
-    myBPM = bpm;
+    bpmAutomation.setSize(1, 1);
+    bpmAutomation.setSample(0, 0, bpm);
+}
+
+bool RenderEngine::setBPMwithPPQN(py::array_t<float> input, std::uint32_t ppqn) {
+
+    if (ppqn <= 0) {
+        throw std::runtime_error("The BPM's PPQN cannot be less than or equal to zero.");
+    }
+
+    myBPMPPQN = ppqn;
+
+    auto numSamples = input.shape(0);
+
+    bpmAutomation.setSize(1, numSamples);
+
+    bpmAutomation.copyFrom(0, 0, (float*)input.data(), numSamples);
+
+    return true;
 }
 
 py::array_t<float>
diff --git a/Source/RenderEngine.h b/Source/RenderEngine.h
index 7b183574..4b98df5f 100644
--- a/Source/RenderEngine.h
+++ b/Source/RenderEngine.h
@@ -3,14 +3,15 @@
 #include "../JuceLibraryCode/JuceHeader.h"
 #include "custom_pybind_wrappers.h"
 #include "CustomParameters.h"
+
+#include "ProcessorBase.h"
+
 #include <random>
 #include <array>
 #include <iomanip>
 #include <sstream>
 #include <string>
 
-class ProcessorBase;
-
 class DAGNode {
 public:
     ProcessorBase* processorBase;
@@ -28,11 +29,15 @@ class RenderEngine : AudioPlayHead
     RenderEngine(double sr, int bs);
 
     bool loadGraph(DAG dagNodes);
+
+    bool removeProcessor(const std::string& name);
     
     bool render (const double renderLength);
 
     void setBPM(double bpm);
 
+    bool setBPMwithPPQN(py::array_t<float> input, std::uint32_t ppqn);
+
     py::array_t<float> getAudioFrames();
 
     py::array_t<float> getAudioFramesForName(std::string& name);
@@ -47,7 +52,6 @@ class RenderEngine : AudioPlayHead
 
     double mySampleRate;
     int myBufferSize;
-    double myBPM = 120.;
     std::unordered_map<std::string, juce::AudioProcessorGraph::NodeID> m_UniqueNameToNodeID;
     
     bool connectGraph();
@@ -59,4 +63,8 @@ class RenderEngine : AudioPlayHead
 private:
 
     CurrentPositionInfo myCurrentPositionInfo;
+    AudioSampleBuffer bpmAutomation;
+    std::uint32_t myBPMPPQN = 960;
+
+    float getBPM(double ppqPosition);
 };
diff --git a/Source/RenderEngineWrapper.cpp b/Source/RenderEngineWrapper.cpp
index f85b6d00..c43af770 100644
--- a/Source/RenderEngineWrapper.cpp
+++ b/Source/RenderEngineWrapper.cpp
@@ -10,10 +10,9 @@ RenderEngineWrapper::RenderEngineWrapper(double sr, int bs) :
 void
 RenderEngineWrapper::prepareProcessor(ProcessorBase* processor, const std::string& name)
 {
-    if (m_UniqueNameToNodeID.find(name) != m_UniqueNameToNodeID.end()) {
-        myMainProcessorGraph->removeNode(m_UniqueNameToNodeID[name]);
-        m_UniqueNameToNodeID.erase(name);
-    }
+    if (this->removeProcessor(name)) {
+        // todo: maybe warn the user that a processor was removed.
+    };
 
     auto node = myMainProcessorGraph->addNode((std::unique_ptr<ProcessorBase>)(processor));
     m_UniqueNameToNodeID[name] = node->nodeID;
diff --git a/Source/RenderEngineWrapper.h b/Source/RenderEngineWrapper.h
index ffc9e310..94e184b8 100644
--- a/Source/RenderEngineWrapper.h
+++ b/Source/RenderEngineWrapper.h
@@ -1,9 +1,22 @@
 #pragma once
 
 #include "RenderEngine.h"
-#include "AllProcessors.h"
 #include "custom_pybind_wrappers.h"
 
+#include "ProcessorBase.h"
+#include "AddProcessor.h"
+#include "CompressorProcessor.h"
+#include "DelayProcessor.h"
+#include "FaustProcessor.h"
+#include "FilterProcessor.h"
+#include "OscillatorProcessor.h"
+#include "PlaybackProcessor.h"
+#include "PlaybackWarpProcessor.h"
+#include "PluginProcessor.h"
+#include "ReverbProcessor.h"
+#include "PannerProcessor.h"
+#include "SamplerProcessor.h"
+
 class RenderEngineWrapper : public RenderEngine
 {
 public:
diff --git a/Source/ReverbProcessor.h b/Source/ReverbProcessor.h
index 902a0a5c..e52c9d00 100644
--- a/Source/ReverbProcessor.h
+++ b/Source/ReverbProcessor.h
@@ -43,11 +43,11 @@ class ReverbProcessor : public ProcessorBase
         AudioPlayHead::CurrentPositionInfo posInfo;
         getPlayHead()->getCurrentPosition(posInfo);
 
-        *myRoomSize = getAutomationVal("room_size", posInfo.timeInSamples);
-        *myDamping = getAutomationVal("damping", posInfo.timeInSamples);
-        *myDryLevel = getAutomationVal("dry_level", posInfo.timeInSamples);
-        *myWetLevel = getAutomationVal("wet_level", posInfo.timeInSamples);
-        *myWidth = getAutomationVal("width", posInfo.timeInSamples);
+        *myRoomSize = getAutomationVal("room_size", posInfo);
+        *myDamping = getAutomationVal("damping", posInfo);
+        *myDryLevel = getAutomationVal("dry_level", posInfo);
+        *myWetLevel = getAutomationVal("wet_level", posInfo);
+        *myWidth = getAutomationVal("width", posInfo);
 
         updateParameters();
     }
@@ -59,19 +59,19 @@ class ReverbProcessor : public ProcessorBase
     const juce::String getName() { return "ReverbProcessor"; };
 
     void setRoomSize(float roomSize) { setAutomationVal("room_size", roomSize); }
-    float getRoomSize() { return getAutomationVal("room_size", 0); }
+    float getRoomSize() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("room_size", posInfo); }
 
     void setDamping(float damping) { setAutomationVal("damping", damping); }
-    float getDamping() { return getAutomationVal("damping", 0); }
+    float getDamping() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("damping", posInfo); }
 
     void setWetLevel(float wetLevel) { setAutomationVal("wet_level", wetLevel); }
-    float getWetLevel() { return getAutomationVal("wet_level", 0); }
+    float getWetLevel() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("wet_level", posInfo); }
 
     void setDryLevel(float dryLevel) { setAutomationVal("dry_level", dryLevel); }
-    float getDryLevel() { return getAutomationVal("dry_level", 0);  }
+    float getDryLevel() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("dry_level", posInfo);  }
 
     void setWidth(float width) { setAutomationVal("width", width); }
-    float getWidth() { return getAutomationVal("width", 0);  }
+    float getWidth() { AudioPlayHead::CurrentPositionInfo posInfo; return getAutomationVal("width", posInfo);  }
 
 
 private:
diff --git a/Source/SamplerProcessor.h b/Source/SamplerProcessor.h
index 28556fcf..1dd9f8bb 100644
--- a/Source/SamplerProcessor.h
+++ b/Source/SamplerProcessor.h
@@ -2,6 +2,7 @@
 
 #include "ProcessorBase.h"
 #include "../Source/Sampler/Source/SamplerAudioProcessor.h"
+#include <filesystem>
 
 class SamplerProcessor : public ProcessorBase
 {
@@ -22,7 +23,10 @@ class SamplerProcessor : public ProcessorBase
         setMainBusInputsAndOutputs(0, input.shape(0));
     }
 
-    ~SamplerProcessor() {}
+    ~SamplerProcessor() {
+        delete myMidiIteratorSec;
+        delete myMidiIteratorQN;
+    }
 
     bool acceptsMidi() const { return true; }
     bool producesMidi() const { return true; }
@@ -37,7 +41,9 @@ class SamplerProcessor : public ProcessorBase
 
         auto parameterName = sampler.getParameterName(parameterIndex);
 
-        return ProcessorBase::getAutomationVal(parameterName.toStdString(), 0);
+        juce::AudioPlayHead::CurrentPositionInfo posInfo;
+
+        return ProcessorBase::getAutomationVal(parameterName.toStdString(), posInfo);
     }
 
     void
@@ -69,12 +75,16 @@ class SamplerProcessor : public ProcessorBase
     void reset() {
         sampler.reset();
 
-        if (myMidiIterator) {
-            delete myMidiIterator;
-        }
+        delete myMidiIteratorSec;
+        myMidiIteratorSec = new MidiBuffer::Iterator(myMidiBufferSec); // todo: deprecated.
+
+        myMidiEventsDoRemainSec = myMidiIteratorSec->getNextEvent(myMidiMessageSec, myMidiMessagePositionSec);
+
+        delete myMidiIteratorQN;
+        myMidiIteratorQN = new MidiBuffer::Iterator(myMidiBufferQN); // todo: deprecated.
+
+        myMidiEventsDoRemainQN = myMidiIteratorQN->getNextEvent(myMidiMessageQN, myMidiMessagePositionQN);
 
-        myMidiIterator = new MidiBuffer::Iterator(myMidiBuffer); // todo: deprecated.
-        myMidiEventsDoRemain = myMidiIterator->getNextEvent(myMidiMessage, myMidiMessagePosition);
         myRenderMidiBuffer.clear();
     }
 
@@ -88,23 +98,34 @@ class SamplerProcessor : public ProcessorBase
 
         buffer.clear(); // todo: why did this become necessary?
         midiBuffer.clear();
+        myRenderMidiBuffer.clear();
 
-        long long int start = posInfo.timeInSamples;
-        long long int end = start + buffer.getNumSamples();
+        {
+            auto start = posInfo.timeInSamples;
+            auto end = start + buffer.getNumSamples();
+
+            myIsMessageBetweenSec = myMidiMessagePositionSec >= start && myMidiMessagePositionSec < end;
+            while (myIsMessageBetweenSec && myMidiEventsDoRemainSec) {
+                myRenderMidiBuffer.addEvent(myMidiMessageSec, int(myMidiMessagePositionSec - start));
+                myMidiEventsDoRemainSec = myMidiIteratorSec->getNextEvent(myMidiMessageSec, myMidiMessagePositionSec);
+                myIsMessageBetweenSec = myMidiMessagePositionSec >= start && myMidiMessagePositionSec < end;
+            }
+        }
 
-        myIsMessageBetween = myMidiMessagePosition >= start && myMidiMessagePosition < end;
-        do {
-            if (myIsMessageBetween) {
-                myRenderMidiBuffer.addEvent(myMidiMessage, int(myMidiMessagePosition - start));
-                myMidiEventsDoRemain = myMidiIterator->getNextEvent(myMidiMessage, myMidiMessagePosition);
-                myIsMessageBetween = myMidiMessagePosition >= start && myMidiMessagePosition < end;
+        {
+            auto pulseStart = std::floor(posInfo.ppqPosition * PPQN);
+            auto pulseEnd = pulseStart + buffer.getNumSamples() * (posInfo.bpm * PPQN) / (mySampleRate * 60.);
+
+            myIsMessageBetweenQN = myMidiMessagePositionQN >= pulseStart && myMidiMessagePositionQN < pulseEnd;
+            while (myIsMessageBetweenQN && myMidiEventsDoRemainQN) {
+                myRenderMidiBuffer.addEvent(myMidiMessageQN, int(myMidiMessagePositionQN - pulseStart));
+                myMidiEventsDoRemainQN = myMidiIteratorQN->getNextEvent(myMidiMessageQN, myMidiMessagePositionQN);
+                myIsMessageBetweenQN = myMidiMessagePositionQN >= pulseStart && myMidiMessagePositionQN < pulseEnd;
             }
-        } while (myIsMessageBetween && myMidiEventsDoRemain);
+        }
 
         sampler.processBlock(buffer, myRenderMidiBuffer);
 
-        myRenderMidiBuffer.clear();
-
         ProcessorBase::processBlock(buffer, midiBuffer);
     }
 
@@ -128,25 +149,54 @@ class SamplerProcessor : public ProcessorBase
     int
     getNumMidiEvents()
     {
-        return myMidiBuffer.getNumEvents();
+        return myMidiBufferSec.getNumEvents() + myMidiBufferQN.getNumEvents();
     };
 
     bool
-    loadMidi(const std::string& path)
+    loadMidi(const std::string& path, bool clearPrevious, bool convertToSeconds, bool allEvents)
     {
+
+        if (!std::filesystem::exists(path.c_str())) {
+            throw std::runtime_error("File not found: " + path);
+        }
+
         File file = File(path);
         FileInputStream fileStream(file);
         MidiFile midiFile;
         midiFile.readFrom(fileStream);
-        midiFile.convertTimestampTicksToSeconds();
-        myMidiBuffer.clear();
-
-        for (int t = 0; t < midiFile.getNumTracks(); t++) {
-            const MidiMessageSequence* track = midiFile.getTrack(t);
-            for (int i = 0; i < track->getNumEvents(); i++) {
-                MidiMessage& m = track->getEventPointer(i)->message;
-                int sampleOffset = (int)(mySampleRate * m.getTimeStamp());
-                myMidiBuffer.addEvent(m, sampleOffset);
+
+        if (clearPrevious) {
+            myMidiBufferSec.clear();
+            myMidiBufferQN.clear();
+        }
+
+        if (convertToSeconds) {
+            midiFile.convertTimestampTicksToSeconds();
+
+            for (int t = 0; t < midiFile.getNumTracks(); t++) {
+                const MidiMessageSequence* track = midiFile.getTrack(t);
+                for (int i = 0; i < track->getNumEvents(); i++) {
+                    MidiMessage& m = track->getEventPointer(i)->message;
+                    int sampleOffset = (int)(mySampleRate * m.getTimeStamp());
+                    if (allEvents || m.isNoteOff() || m.isNoteOn()) {
+                        myMidiBufferSec.addEvent(m, sampleOffset);
+                    }
+                }
+            }
+        }
+        else {
+            auto timeFormat = midiFile.getTimeFormat(); // the ppqn (Ableton makes midi files with 96 ppqn)
+            for (int t = 0; t < midiFile.getNumTracks(); t++) {
+                const MidiMessageSequence* track = midiFile.getTrack(t);
+                for (int i = 0; i < track->getNumEvents(); i++) {
+                    MidiMessage& m = track->getEventPointer(i)->message;
+
+                    if (allEvents || m.isNoteOff() || m.isNoteOn()) {
+                        // convert timestamp from its original time format to our high resolution PPQN
+                        auto timeStamp = m.getTimeStamp() * PPQN / timeFormat;
+                        myMidiBufferQN.addEvent(m, timeStamp);
+                    }
+                }
             }
         }
 
@@ -155,14 +205,16 @@ class SamplerProcessor : public ProcessorBase
 
     void
     clearMidi() {
-        myMidiBuffer.clear();
+        myMidiBufferSec.clear();
+        myMidiBufferQN.clear();
     }
 
     bool
     addMidiNote(uint8  midiNote,
             uint8  midiVelocity,
             const double noteStart,
-            const double noteLength) {
+            const double noteLength,
+            bool convert_to_sec) {
 
         if (midiNote > 255) midiNote = 255;
         if (midiNote < 0) midiNote = 0;
@@ -170,7 +222,6 @@ class SamplerProcessor : public ProcessorBase
         if (midiVelocity < 0) midiVelocity = 0;
         if (noteLength <= 0) {
             throw std::runtime_error("The note length must be greater than zero.");
-            return false;
         }
 
         // Get the note on midiBuffer.
@@ -182,11 +233,20 @@ class SamplerProcessor : public ProcessorBase
             midiNote,
             midiVelocity);
 
-        auto startTime = noteStart * mySampleRate;
-        onMessage.setTimeStamp(startTime);
-        offMessage.setTimeStamp(startTime + noteLength * mySampleRate);
-        myMidiBuffer.addEvent(onMessage, (int)onMessage.getTimeStamp());
-        myMidiBuffer.addEvent(offMessage, (int)offMessage.getTimeStamp());
+        if (convert_to_sec) {
+            auto startTime = noteStart * mySampleRate;
+            onMessage.setTimeStamp(startTime);
+            offMessage.setTimeStamp(startTime + noteLength * mySampleRate);
+            myMidiBufferSec.addEvent(onMessage, (int)onMessage.getTimeStamp());
+            myMidiBufferSec.addEvent(offMessage, (int)offMessage.getTimeStamp());
+        }
+        else {
+            auto startTime = noteStart * PPQN;
+            onMessage.setTimeStamp(startTime);
+            offMessage.setTimeStamp(startTime + noteLength * PPQN);
+            myMidiBufferQN.addEvent(onMessage, (int)onMessage.getTimeStamp());
+            myMidiBufferQN.addEvent(offMessage, (int)offMessage.getTimeStamp());
+        }
 
         return true;
     }
@@ -270,7 +330,7 @@ class SamplerProcessor : public ProcessorBase
 
             auto theParameter = ((AutomateParameterFloat*)myParameters.getParameter(theName));
             if (theParameter) {
-                sampler.setParameterRawNotifyingHost(i, theParameter->sample(posInfo.timeInSamples));
+                sampler.setParameterRawNotifyingHost(i, theParameter->sample(posInfo));
             }
             else {
                 std::cerr << "Error automateParameters: " << theName << std::endl;
@@ -286,11 +346,23 @@ class SamplerProcessor : public ProcessorBase
 
     SamplerAudioProcessor sampler;
 
-    MidiBuffer myMidiBuffer;
+    MidiBuffer myMidiBufferQN;
+    MidiBuffer myMidiBufferSec;
+
     MidiBuffer myRenderMidiBuffer;
-    MidiMessage myMidiMessage;
-    int myMidiMessagePosition = -1;
-    MidiBuffer::Iterator* myMidiIterator = nullptr;
-    bool myIsMessageBetween = false;
-    bool myMidiEventsDoRemain = false;
+
+    MidiMessage myMidiMessageQN;
+    MidiMessage myMidiMessageSec;
+
+    int myMidiMessagePositionQN = -1;
+    int myMidiMessagePositionSec = -1;
+
+    MidiBuffer::Iterator* myMidiIteratorQN = nullptr;
+    MidiBuffer::Iterator* myMidiIteratorSec = nullptr;
+
+    bool myIsMessageBetweenQN = false;
+    bool myIsMessageBetweenSec = false;
+
+    bool myMidiEventsDoRemainQN = false;
+    bool myMidiEventsDoRemainSec = false;
 };
diff --git a/Source/source.cpp b/Source/source.cpp
index c95f944c..ca1e96da 100644
--- a/Source/source.cpp
+++ b/Source/source.cpp
@@ -30,15 +30,17 @@ PYBIND11_MODULE(dawdreamer, m)
     )pbdoc";
 
     py::class_<ProcessorBase>(m, "ProcessorBase")
-        .def("set_automation", &ProcessorBase::setAutomation, arg("parameter_name"), arg("data"), R"pbdoc(
+        .def("set_automation", &ProcessorBase::setAutomation, arg("parameter_name"), arg("data"), kw_only(), arg("ppqn")=0, R"pbdoc(
     Set a parameter's automation with a numpy array.
 
     Parameters
     ----------
     parameter_name : str
         The name of the parameter.
-    data : str
+    data : np.array
         An array of data for the parameter automation.
+    ppqn : integer
+        If specified, it is the pulses-per-quarter-note rate of the automation data. If not specified or zero, the data will be interpreted at audio rate.
 
     Returns
     -------
@@ -156,6 +158,9 @@ but the filter mode cannot under automation.";
             "A list of gain levels to apply to the corresponding inputs.")
         .doc() = "An Add Processor adds one or more stereo inputs with corresponding gain parameters.";
 
+    auto add_midi_description = "Add a single MIDI note whose note and velocity are integers between 0 and 127. By default, `convert_to_sec` is True, so the start_time and duration are measured in seconds. If `convert_to_sec` is False, they are measured in beats.";
+    auto load_midi_description = "Load MIDI from a file. If `all_events` is True, then all events (not just Note On/Off) will be loaded. By default, `convert_to_sec` is True, so notes will be converted to absolute times and will not be affected by the Render Engine's BPM. By default `clear_previous` is True.";
+
     py::class_<PluginProcessorWrapper, ProcessorBase>(m, "PluginProcessor")
         .def("can_set_bus", &PluginProcessorWrapper::canApplyBusInputsAndOutputs, arg("inputs"), arg("outputs"), "Return bool for whether this combination of input and output channels can be set.")
         .def("set_bus", &PluginProcessorWrapper::setMainBusInputsAndOutputs, arg("inputs"), arg("outputs"), "Set the number of input and output channels. An error will be thrown for an unaccepted option.")
@@ -171,18 +176,17 @@ but the filter mode cannot under automation.";
         .def("get_parameter_text", &PluginProcessorWrapper::getParameterAsText, arg("index"), "Get a parameter's value as text.")
         .def("set_parameter", &PluginProcessorWrapper::wrapperSetParameter, arg("index"), arg("value"),
             "Set a parameter's value to a constant.")
-        .def("set_automation", &PluginProcessorWrapper::wrapperSetAutomation, arg("parameter_index"), arg("data"),
+        .def("set_automation", &PluginProcessorWrapper::wrapperSetAutomation, arg("parameter_index"), arg("data"), kw_only(), arg("ppqn")=0,
             "Set the automation based on its index.")
         .def("get_plugin_parameter_size", &PluginProcessorWrapper::wrapperGetPluginParameterSize, "Get the number of parameters.")
         .def("get_plugin_parameters_description", &PluginProcessorWrapper::getPluginParametersDescription,
             "Get a list of dictionaries describing the plugin's parameters.")
         .def_property_readonly("n_midi_events", &PluginProcessorWrapper::getNumMidiEvents, "The number of MIDI events stored in the buffer. \
 Note that note-ons and note-offs are counted separately.")
-        .def("load_midi", &PluginProcessorWrapper::loadMidi, arg("filepath"), kw_only(), arg("all_events")=true, "Load MIDI from a file. If `all_events` is True, then all events (not just Note On/Off) will be loaded.")
+        .def("load_midi", &PluginProcessorWrapper::loadMidi, arg("filepath"), kw_only(), arg("clear_previous")=true, arg("convert_to_sec")=true, arg("all_events")=true, load_midi_description)
         .def("clear_midi", &PluginProcessorWrapper::clearMidi, "Remove all MIDI notes.")
         .def("add_midi_note", &PluginProcessorWrapper::addMidiNote,
-            arg("note"), arg("velocity"), arg("start_time"), arg("duration"),
-            "Add a single MIDI note whose note and velocity are integers between 0 and 127.")
+            arg("note"), arg("velocity"), arg("start_time"), arg("duration"), kw_only(), arg("convert_to_sec")=true, add_midi_description)
         .doc() = "A Plugin Processor can load VST \".dll\" and \".vst3\" files on Windows. It can load \".vst\", \".vst3\", and \".component\" files on macOS. The files can be for either instruments \
 or effects. Some plugins such as ones that do sidechain compression can accept two inputs when loading a graph.";
 
@@ -198,11 +202,10 @@ or effects. Some plugins such as ones that do sidechain compression can accept t
             "Get a list of dictionaries describing the plugin's parameters.")
         .def_property_readonly("n_midi_events", &SamplerProcessor::getNumMidiEvents, "The number of MIDI events stored in the buffer. \
 Note that note-ons and note-offs are counted separately.")
-        .def("load_midi", &SamplerProcessor::loadMidi, arg("filepath"), "Load MIDI from a file.")
+        .def("load_midi", &SamplerProcessor::loadMidi, arg("filepath"), kw_only(), arg("clear_previous")=true, arg("convert_to_sec")=true, arg("all_events")=true, load_midi_description)
         .def("clear_midi", &SamplerProcessor::clearMidi, "Remove all MIDI notes.")
         .def("add_midi_note", &SamplerProcessor::addMidiNote,
-            arg("note"), arg("velocity"), arg("start_time"), arg("duration"),
-            "Add a single MIDI note whose note and velocity are integers between 0 and 127.")
+            arg("note"), arg("velocity"), arg("start_time"), arg("duration"), kw_only(), arg("convert_to_sec")=true, add_midi_description)
         .doc() = "The Sampler Processor works like a basic Sampler instrument. It takes a typically short audio sample and can play it back \
 at different pitches and speeds. It has parameters for an ADSR envelope controlling the amplitude and another for controlling a low-pass filter cutoff. \
 Unlike a VST, the parameters don't need to be between 0 and 1. For example, you can set an envelope attack parameter to 50 to represent 50 milliseconds.";
@@ -220,6 +223,7 @@ Unlike a VST, the parameters don't need to be between 0 and 1. For example, you
         .def("get_parameter", &FaustProcessor::getParamWithPath, arg("parameter_path"))
         .def("set_parameter", &FaustProcessor::setParamWithIndex, arg("parameter_index"), arg("value"))
         .def("set_parameter", &FaustProcessor::setAutomationVal, arg("parameter_path"), arg("value"))
+        .def("set_automation", &FaustProcessor::setAutomation, arg("parameter_name"), arg("data"), kw_only(), arg("ppqn") = 0)
         .def_property_readonly("compiled", &FaustProcessor::isCompiled, "Did the most recent DSP code compile?")
         .def_property_readonly("code", &FaustProcessor::code, "Get the most recently compiled Faust DSP code.")
         .def_property("num_voices", &FaustProcessor::getNumVoices, &FaustProcessor::setNumVoices, "The number of voices for polyphony. Set to zero to disable polyphony. One or more enables polyphony.")
@@ -228,10 +232,10 @@ Unlike a VST, the parameters don't need to be between 0 and 1. For example, you
         .def_property("faust_libraries_path", &FaustProcessor::getFaustLibrariesPath, &FaustProcessor::setFaustLibrariesPath, "Absolute path to directory containing your custom \".lib\" files containing Faust code.")
         .def_property_readonly("n_midi_events", &FaustProcessor::getNumMidiEvents, "The number of MIDI events stored in the buffer. \
 Note that note-ons and note-offs are counted separately.")
-        .def("load_midi", &FaustProcessor::loadMidi, arg("filepath"), "Load MIDI from a file.")
+        .def("load_midi", &FaustProcessor::loadMidi, arg("filepath"), kw_only(), arg("clear_previous")=true, arg("convert_to_sec")=true, arg("all_events")=true, load_midi_description)
         .def("clear_midi", &FaustProcessor::clearMidi, "Remove all MIDI notes.")
-        .def("add_midi_note", &FaustProcessor::addMidiNote, arg("note"), arg("velocity"), arg("start_time"), arg("duration"),
-    "Add a single MIDI note whose note and velocity are integers between 0 and 127.")
+        .def("add_midi_note", &FaustProcessor::addMidiNote,
+            arg("note"), arg("velocity"), arg("start_time"), arg("duration"), kw_only(), arg("convert_to_sec") = true, add_midi_description)
         .def("set_soundfiles", &FaustProcessor::setSoundfiles, arg("soundfile_dict"), "Set the audio data that the FaustProcessor can use with the `soundfile` primitive.")
         .doc() = "A Faust Processor can compile and execute FAUST code. See https://faust.grame.fr for more information.";
 #endif
@@ -243,9 +247,11 @@ Note that note-ons and note-offs are counted separately.")
     py::class_<RenderEngineWrapper>(m, "RenderEngine", "A Render Engine loads and runs a graph of audio processors.")
         .def(py::init<double, int>(), arg("sample_rate"), arg("block_size"))
         .def("render", &RenderEngineWrapper::render, arg("seconds"), "Render the most recently loaded graph.")
-        .def("set_bpm", &RenderEngineWrapper::setBPM, arg("bpm"), "Set the beats-per-minute of the engine.")
+        .def("set_bpm", &RenderEngineWrapper::setBPM, arg("bpm"), "Set the beats-per-minute of the engine as a constant rate.")
+        .def("set_bpm", &RenderEngineWrapper::setBPMwithPPQN, arg("bpm"), arg("ppqn"), "Set the beats-per-minute of the engine using a 1D numpy array and a constant PPQN. If the values in the array suddenly change every PPQN samples, the tempo change will occur \"on-the-beat.\"")
         .def("get_audio", &RenderEngine::getAudioFrames, "Get the most recently rendered audio as a numpy array.")
         .def("get_audio", &RenderEngine::getAudioFramesForName, arg("name"), "Get the most recently rendered audio for a specific processor.")
+        .def("remove_processor", &RenderEngine::removeProcessor, arg("name"), "Remove a processor based on its unique name. Existing Python references to the processor will become invalid.")
         .def("load_graph", &RenderEngineWrapper::loadGraphWrapper, arg("dag"), "Load a directed acyclic graph of processors.")
         .def("make_oscillator_processor", &RenderEngineWrapper::makeOscillatorProcessor, arg("name"), arg("frequency"),
             "Make an Oscillator Processor", returnPolicy)
diff --git a/tests/dawdreamer_utils.py b/tests/dawdreamer_utils.py
index 100f27de..fbe4bd4e 100644
--- a/tests/dawdreamer_utils.py
+++ b/tests/dawdreamer_utils.py
@@ -2,6 +2,7 @@
 from os import getenv
 from os.path import abspath, isfile, isdir, basename, splitext
 from pathlib import Path
+from itertools import product
 import pathlib
 import platform
 import random
diff --git a/tests/test_faust_poly.py b/tests/test_faust_poly.py
index 33fec92c..1d3a2e15 100644
--- a/tests/test_faust_poly.py
+++ b/tests/test_faust_poly.py
@@ -1,124 +1,146 @@
 from dawdreamer_utils import *
 
 def _test_faust_poly(file_path, group_voices=True, num_voices=8, buffer_size=1, cutoff=None,
-	automation=False, decay=None):
+    automation=False, decay=None):
 
-	engine = daw.RenderEngine(SAMPLE_RATE, buffer_size)
+    engine = daw.RenderEngine(SAMPLE_RATE, buffer_size)
 
-	dsp_path = abspath(FAUST_DSP / "polyphonic.dsp")
-	faust_processor = engine.make_faust_processor("faust")
-	faust_processor.set_dsp(dsp_path)
-	
-	# Group voices will affect the number of parameters.
-	# True will result in fewer parameters because all voices will
-	# share the same parameters.
-	faust_processor.group_voices = group_voices
-	faust_processor.num_voices = num_voices
-	
-	faust_processor.compile()
+    dsp_path = abspath(FAUST_DSP / "polyphonic.dsp")
+    faust_processor = engine.make_faust_processor("faust")
+    faust_processor.set_dsp(dsp_path)
+    
+    # Group voices will affect the number of parameters.
+    # True will result in fewer parameters because all voices will
+    # share the same parameters.
+    faust_processor.group_voices = group_voices
+    faust_processor.num_voices = num_voices
+    
+    faust_processor.compile()
 
-	# for par in faust_processor.get_parameters_description():
-	# 	print(par)
+    # for par in faust_processor.get_parameters_description():
+    #   print(par)
 
-	 # (MIDI note, velocity, start sec, duration sec)
-	faust_processor.add_midi_note(60, 60, 0.0, .25)
-	faust_processor.add_midi_note(64, 80, 0.5, .5)
-	faust_processor.add_midi_note(67, 127, 0.75, .5)
+     # (MIDI note, velocity, start sec, duration sec)
+    faust_processor.add_midi_note(60, 60, 0.0, .25)
+    faust_processor.add_midi_note(64, 80, 0.5, .5)
+    faust_processor.add_midi_note(67, 127, 0.75, .5)
 
-	assert(faust_processor.n_midi_events == 3*2)  # multiply by 2 because of the off-notes.
+    assert(faust_processor.n_midi_events == 3*2)  # multiply by 2 because of the off-notes.
 
-	if cutoff is not None:
-		faust_processor.set_parameter("/Sequencer/DSP2/MyInstrument/cutoff", cutoff)
-	elif automation:
-		faust_processor.set_automation("/Sequencer/DSP2/MyInstrument/cutoff", 5000+4900*make_sine(30, 10.))
+    if cutoff is not None:
+        faust_processor.set_parameter("/Sequencer/DSP2/MyInstrument/cutoff", cutoff)
+    elif automation:
+        faust_processor.set_automation("/Sequencer/DSP2/MyInstrument/cutoff", 5000+4900*make_sine(30., 10.))
 
-	if decay is not None:
-		if group_voices:
-			faust_processor.set_parameter("/Sequencer/DSP1/Polyphonic/Voices/MyInstrument/decay", decay)
-		else:
-			for i in range(1, num_voices+1):
-				faust_processor.set_parameter(f"/Sequencer/DSP1/Polyphonic/V{i}/MyInstrument/decay", decay)
+    if decay is not None:
+        if group_voices:
+            faust_processor.set_parameter("/Sequencer/DSP1/Polyphonic/Voices/MyInstrument/decay", decay)
+        else:
+            for i in range(1, num_voices+1):
+                faust_processor.set_parameter(f"/Sequencer/DSP1/Polyphonic/V{i}/MyInstrument/decay", decay)
 
-	# for par in faust_processor.get_parameters_description():
-	# 	print(par)
+    # for par in faust_processor.get_parameters_description():
+    #   print(par)
 
-	graph = [
-	    (faust_processor, [])
-	]
+    graph = [
+        (faust_processor, [])
+    ]
 
-	engine.load_graph(graph)
+    engine.load_graph(graph)
 
-	render(engine, file_path=file_path, duration=3.)
+    render(engine, file_path=file_path, duration=3.)
 
-	return engine.get_audio()
+    return engine.get_audio()
 
 def test_faust_poly():
-	audio1 = _test_faust_poly(OUTPUT / 'test_faust_poly_grouped.wav', group_voices=True)
-	audio2 = _test_faust_poly(OUTPUT / 'test_faust_poly_ungrouped.wav', group_voices=False)
+    audio1 = _test_faust_poly(OUTPUT / 'test_faust_poly_grouped.wav', group_voices=True)
+    audio2 = _test_faust_poly(OUTPUT / 'test_faust_poly_ungrouped.wav', group_voices=False)
 
-	assert np.allclose(audio1, audio2)
+    assert np.allclose(audio1, audio2)
 
-	audio1 = _test_faust_poly(OUTPUT / 'test_faust_poly_2k_cutoff_grouped.wav', group_voices=True, cutoff=2000)
-	audio2 = _test_faust_poly(OUTPUT / 'test_faust_poly_2k_cutoff_ungrouped.wav', group_voices=False, cutoff=2000)
+    audio1 = _test_faust_poly(OUTPUT / 'test_faust_poly_2k_cutoff_grouped.wav', group_voices=True, cutoff=2000)
+    audio2 = _test_faust_poly(OUTPUT / 'test_faust_poly_2k_cutoff_ungrouped.wav', group_voices=False, cutoff=2000)
 
-	assert np.allclose(audio1, audio2)
+    assert np.allclose(audio1, audio2)
 
-	audio1 = _test_faust_poly(OUTPUT / 'test_faust_poly_automation_cutoff_grouped.wav', group_voices=True, automation=True)
-	audio2 = _test_faust_poly(OUTPUT / 'test_faust_poly_automation_cutoff_ungrouped.wav', group_voices=False, automation=True)
+    audio1 = _test_faust_poly(OUTPUT / 'test_faust_poly_automation_cutoff_grouped.wav', group_voices=True, automation=True)
+    audio2 = _test_faust_poly(OUTPUT / 'test_faust_poly_automation_cutoff_ungrouped.wav', group_voices=False, automation=True)
 
-	assert np.allclose(audio1, audio2)
+    assert np.allclose(audio1, audio2)
 
-	audio1 = _test_faust_poly(OUTPUT / 'test_faust_poly_automation_decay_grouped.wav', group_voices=True, decay=.5)
-	audio2 = _test_faust_poly(OUTPUT / 'test_faust_poly_automation_decay_ungrouped.wav', group_voices=False, decay=.5)
+    audio1 = _test_faust_poly(OUTPUT / 'test_faust_poly_decay_grouped.wav', group_voices=True, decay=.5)
+    audio2 = _test_faust_poly(OUTPUT / 'test_faust_poly_decay_ungrouped.wav', group_voices=False, decay=.5)
 
-	assert np.allclose(audio1, audio2)
+    assert np.allclose(audio1, audio2)
 
 
-@pytest.mark.parametrize("midi_path", [abspath(ASSETS / 'MIDI-Unprocessed_SMF_02_R1_2004_01-05_ORIG_MID--AUDIO_02_R1_2004_05_Track05_wav.midi')])
-def test_faust_sine(midi_path: str, buffer_size=1):
+@pytest.mark.parametrize("midi_path,bpm_automation,convert_to_sec,buffer_size",
+    product(
+        [abspath(ASSETS / 'MIDI-Unprocessed_SMF_02_R1_2004_01-05_ORIG_MID--AUDIO_02_R1_2004_05_Track05_wav.midi')],
+        [False, True],
+        [False, True],
+        [1, 128]
+    )
+)
+def test_faust_sine(midi_path: str, bpm_automation: bool, convert_to_sec: bool, buffer_size: int):
 
     engine = daw.RenderEngine(SAMPLE_RATE, buffer_size)
 
+    duration = 10.
+
+    ppqn = 960
+
+    if bpm_automation:
+        bpm_data = 120.+60.*make_sine(1./3., duration*10., sr=ppqn)
+        engine.set_bpm(bpm_data, ppqn=ppqn)
+
     faust_processor = engine.make_faust_processor("faust")
     faust_processor.num_voices = 16
     faust_processor.group_voices = True
     faust_processor.release_length = .5  # note that this is the maximum of the "release" hslider below
 
     faust_processor.set_dsp_string(
-    	f"""
-		declare name "MyInstrument";
+        f"""
+        declare name "MyInstrument";
 
-		declare options "[nvoices:8]"; // FaustProcessor has a property which will override this.
-		import("stdfaust.lib");
+        declare options "[nvoices:8]"; // FaustProcessor has a property which will override this.
+        import("stdfaust.lib");
 
-		freq = hslider("freq",200,50,1000,0.01); // note pitch
-		gain = hslider("gain",0.1,0,1,0.01);     // note velocity
-		gate = button("gate");                   // note on/off
+        freq = hslider("freq",200,50,1000,0.01); // note pitch
+        gain = hslider("gain",0.1,0,1,0.01);     // note velocity
+        gate = button("gate");                   // note on/off
 
-		attack = hslider("attack", .002, 0.001, 10., 0.);
-		decay = hslider("decay", .05, 0.001, 10., 0.);
-		sustain = hslider("sustain", 1.0, 0.0, 1., 0.);
-		release = hslider("release", .05, 0.001, {faust_processor.release_length}, 0.);
+        attack = hslider("attack", .002, 0.001, 10., 0.);
+        decay = hslider("decay", .05, 0.001, 10., 0.);
+        sustain = hslider("sustain", 1.0, 0.0, 1., 0.);
+        release = hslider("release", .05, 0.001, {faust_processor.release_length}, 0.);
 
-		envVol = 0.35*gain*en.adsr(attack, decay, sustain, release, gate);
+        envVol = 0.35*gain*en.adsr(attack, decay, sustain, release, gate);
 
-		process = os.osc(freq)*envVol <: _, _;
-		effect = _, _;
-    	"""
-    	)
+        process = os.osc(freq)*envVol <: _, _;
+        effect = _, _;
+        """
+        )
     faust_processor.compile()
     # desc = faust_processor.get_parameters_description()
     # for par in desc:
     #   print(par)
 
-    faust_processor.load_midi(midi_path)
+    faust_processor.load_midi(midi_path, convert_to_sec=convert_to_sec, clear_previous=True, all_events=False)
 
     graph = [
         (faust_processor, [])
     ]
 
     engine.load_graph(graph)
-    render(engine, file_path=OUTPUT / ('test_faust_sine_' + splitext(basename(midi_path))[0] + '.wav'), duration=10.)
+    file_path = OUTPUT / ''.join([
+        'test_faust_sine_',
+        'bpm_' if bpm_automation else '',
+        'conv_' if convert_to_sec else '',
+        f'bs_{buffer_size}_',
+        splitext(basename(midi_path))[0],
+        '.wav'])
+    render(engine, file_path=file_path, duration=duration)
 
     audio = engine.get_audio()
     assert(np.mean(np.abs(audio)) > .0001)
diff --git a/tests/test_faust_poly_sampler.py b/tests/test_faust_poly_sampler.py
index 08888e35..0faa284f 100644
--- a/tests/test_faust_poly_sampler.py
+++ b/tests/test_faust_poly_sampler.py
@@ -1,17 +1,20 @@
 from dawdreamer_utils import *
 
-BUFFER_SIZE = 1024
+BUFFER_SIZE = 128
 
-@pytest.mark.parametrize("sample_seq,output_path", [
-	(load_audio_file(ASSETS / "60988__folktelemetry__crash-fast-14.wav"), "test_faust_poly_sampler_cymbal.wav")
+@pytest.mark.parametrize("audio_file_path,output_path,convert_to_sec", [
+	(ASSETS / "60988__folktelemetry__crash-fast-14.wav", "test_faust_poly_sampler_cymbal.wav", False),
+	(ASSETS / "60988__folktelemetry__crash-fast-14.wav", "test_faust_poly_sampler_cymbal_convert_to_sec.wav", True)
 	])
-def test_faust_poly_sampler(sample_seq, output_path, lagrange_order=4):
+def test_faust_poly_sampler(audio_file_path: str, output_path: str, convert_to_sec: bool, lagrange_order=4):
 
 	engine = daw.RenderEngine(SAMPLE_RATE, BUFFER_SIZE)
 
 	faust_processor = engine.make_faust_processor("faust")
 	faust_processor.num_voices = 8
 
+	sample_seq = load_audio_file(ASSETS / "60988__folktelemetry__crash-fast-14.wav")
+
 	# set_soundfiles
 	if sample_seq.ndim == 1:
 		sample_seq = sample_seq.reshape(1, -1)
@@ -37,20 +40,16 @@ def test_faust_poly_sampler(sample_seq, output_path, lagrange_order=4):
 	# 	print(par)
 
 	 # (MIDI note, velocity, start sec, duration sec)
-	faust_processor.add_midi_note(60, 60, 0.0, .25)
-	faust_processor.add_midi_note(64, 80, 0.5, .5)
-	faust_processor.add_midi_note(67, 127, 0.75, .5)
-
-	assert(faust_processor.n_midi_events == 3*2)  # multiply by 2 because of the off-notes.
+	faust_processor.add_midi_note(60, 60, 0.0, .25, convert_to_sec=convert_to_sec)
+	faust_processor.add_midi_note(64, 80, 0.5, .5, convert_to_sec=convert_to_sec)
+	faust_processor.add_midi_note(67, 127, 0.75, .5, convert_to_sec=convert_to_sec)
 
 	graph = [
 	    (faust_processor, [])
 	]
 
 	engine.load_graph(graph)
-
 	render(engine, file_path=OUTPUT / output_path, duration=3.)
-
 	# check that it's non-silent
 	audio = engine.get_audio()
-	assert(np.mean(np.abs(audio)) > .01)
\ No newline at end of file
+	assert(np.mean(np.abs(audio)) > .001)
\ No newline at end of file
diff --git a/tests/test_faust_processor.py b/tests/test_faust_processor.py
index 481f79b8..8ce7cb8f 100644
--- a/tests/test_faust_processor.py
+++ b/tests/test_faust_processor.py
@@ -55,6 +55,33 @@ def test_faust_passthrough(duration, buffer_size):
 
     assert np.allclose(data, audio, atol=1e-07)
 
+def test_faust_automation_tempo():
+
+    duration = 10.
+
+    engine = daw.RenderEngine(SAMPLE_RATE, 128)
+
+    faust_processor = engine.make_faust_processor("faust")
+    faust_processor.set_dsp_string(
+        """
+        declare name "MyInstrument"; 
+        process = hslider("freq", 440., 0., 15000., 0.) : os.osc : _*.4 <: _, _;
+        """)
+
+    ppqn = 960
+    automation = make_sine(4, duration, sr=ppqn)
+    automation = 220.+220.*(automation > 0).astype(np.float32)
+    faust_processor.set_automation("/MyInstrument/freq", automation, ppqn=ppqn)
+    # print(faust_processor.get_parameters_description())
+
+    graph = [
+        (faust_processor, [])
+    ]
+
+    engine.load_graph(graph)
+
+    render(engine, file_path=OUTPUT / 'test_faust_automation_tempo.wav', duration=duration)
+
 @pytest.mark.parametrize("duration,buffer_size", product(
     [(44099.5/44100), (44100.5/44100), 1., (4./44100)],
     [1, 2, 4, 8, 16, 128, 2048]))
diff --git a/tests/test_playbackwarp_processor.py b/tests/test_playbackwarp_processor.py
index f26cdd7f..325463e9 100644
--- a/tests/test_playbackwarp_processor.py
+++ b/tests/test_playbackwarp_processor.py
@@ -1,14 +1,19 @@
 from dawdreamer_utils import *
 
-BUFFER_SIZE = 1
 
-def test_playbackwarp_processor1():
+@pytest.mark.parametrize("buffer_size", [1, 2048])
+def test_playbackwarp_processor1(buffer_size: int):
 
-	DURATION = 10.
+	DURATION = 15.
+
+	engine = daw.RenderEngine(SAMPLE_RATE, buffer_size)
 
-	engine = daw.RenderEngine(SAMPLE_RATE, BUFFER_SIZE)
+	ppqn = 960
 
-	engine.set_bpm(140.)
+	full_measure = ppqn * 4  # four beats is a measure
+	bpm = np.concatenate([140*np.ones(full_measure), 70.*np.ones(full_measure)])
+	bpm = np.tile(bpm, (100))
+	engine.set_bpm(bpm, ppqn=ppqn)
 
 	drums = engine.make_playbackwarp_processor("drums",
 		load_audio_file(ASSETS / "Music Delta - Disco" / "drums.wav", duration=DURATION))
@@ -54,28 +59,26 @@ def test_playbackwarp_processor1():
 	]
 
 	engine.load_graph(graph)
-
-	render(engine, file_path=OUTPUT / 'test_playbackwarp_processor1a.wav')
+	render(engine, file_path=OUTPUT / 'test_playbackwarp_processor1a.wav', duration=DURATION)
 
 	assert(np.mean(np.abs(engine.get_audio())) > .01)
 
 	other.transpose = 2.
-
-	render(engine, file_path=OUTPUT / 'test_playbackwarp_processor1b.wav')
+	render(engine, file_path=OUTPUT / 'test_playbackwarp_processor1b.wav', duration=DURATION)
 
 	assert(np.mean(np.abs(engine.get_audio())) > .01)
 
 	other.set_automation('transpose', make_sine(1., DURATION))
-
-	render(engine, file_path=OUTPUT / 'test_playbackwarp_processor1c.wav')
+	render(engine, file_path=OUTPUT / 'test_playbackwarp_processor1c.wav', duration=DURATION)
 
 	assert(np.mean(np.abs(engine.get_audio())) > .01)
 
-def test_playbackwarp_processor2():
+@pytest.mark.parametrize("buffer_size", [1])
+def test_playbackwarp_processor2(buffer_size: int):
 
 	DURATION = 10.
 
-	engine = daw.RenderEngine(SAMPLE_RATE, BUFFER_SIZE)
+	engine = daw.RenderEngine(SAMPLE_RATE, buffer_size)
 
 	# Pick 120 because it's easy to analyze for timing in Audacity.
 	# 1 second is two quarter notes.
@@ -120,8 +123,8 @@ def test_playbackwarp_processor2():
 
 	assert(np.mean(np.abs(engine.get_audio())) > .01)
 
-
-def test_playbackwarp_processor3():
+@pytest.mark.parametrize("buffer_size", [1])
+def test_playbackwarp_processor3(buffer_size: int):
 
 	"""
 	Test using the playback warp processor without a clip file and therefore without warping.
@@ -132,7 +135,7 @@ def test_playbackwarp_processor3():
 
 	DURATION = 3.
 
-	engine = daw.RenderEngine(SAMPLE_RATE, BUFFER_SIZE)
+	engine = daw.RenderEngine(SAMPLE_RATE, buffer_size)
 
 	drum_audio = load_audio_file(ASSETS / "575854__yellowtree__d-b-funk-loop.wav")
 	drum_audio = drum_audio[:, int(SAMPLE_RATE*.267):]  # manually trim to beginning
@@ -169,8 +172,8 @@ def test_playbackwarp_processor3():
 
 	assert(np.mean(np.abs(audio)) > .01)
 
-
-def test_playbackwarp_processor4():
+@pytest.mark.parametrize("buffer_size", [1])
+def test_playbackwarp_processor4(buffer_size: int):
 
 	"""
 	Test using the playback warp processor without a clip file and therefore without warping.
@@ -180,7 +183,7 @@ def test_playbackwarp_processor4():
 
 	DURATION = 10.
 
-	engine = daw.RenderEngine(SAMPLE_RATE, BUFFER_SIZE)
+	engine = daw.RenderEngine(SAMPLE_RATE, buffer_size)
 
 	drum_audio = load_audio_file(ASSETS / "Music Delta - Disco" / "drums.wav")
 	drum_audio = drum_audio[:, int(SAMPLE_RATE*.267):]  # manually trim to beginning
diff --git a/tests/test_plugins.py b/tests/test_plugins.py
index c2d0f8c4..c029af37 100644
--- a/tests/test_plugins.py
+++ b/tests/test_plugins.py
@@ -264,9 +264,10 @@ def test_plugin_upright_piano():
 
 @pytest.mark.parametrize("plugin_path",
     [
+    "C:/VSTPlugIns/TAL-NoiseMaker-64.vst3",
     "C:/VSTPlugIns/LABS (64 Bit).dll",
-    # "C:/VSTPlugIns/TAL-NoiseMaker-64.vst3",
-    # "C:/VSTPlugIns/sparta/sparta_ambiBIN.dll",
+    "C:/VSTPlugIns/Kontakt.dll",
+    # "C:/VSTPlugIns/Kontakt.vst3",  # not working
     ]
     )
 def test_plugin_editor(plugin_path: str):
@@ -299,20 +300,25 @@ def load_help(processor, filepath: str):
 
     DURATION = 5.
 
-    plugin_basename = splitext(basename(plugin_path))[0]
+    plugin_basename = basename(plugin_path)
 
     engine = daw.RenderEngine(SAMPLE_RATE, 128)
 
     synth = engine.make_plugin_processor("synth", plugin_path)
 
     plat_system = platform.system()
-    state_file_path = abspath(OUTPUT / (f'state_test_plugin_{plat_system}_{plugin_basename}'))
+    state_file_path = abspath(OUTPUT / (f'state_test_plugin_{plat_system}_{plugin_basename}.bin'))
+
+    # from time import sleep
+    # sleep(.5)
 
     load_help(synth, state_file_path)
 
+    # sleep(.5)
+
     # print(synth.get_plugin_parameters_description())
 
-    print('inputs: ', synth.get_num_input_channels(), ' outputs: ', synth.get_num_output_channels())
+    print('synth: ', plugin_basename, ' inputs: ', synth.get_num_input_channels(), ' outputs: ', synth.get_num_output_channels())
 
     # assert(synth.get_num_input_channels() == 0)
     # assert(synth.get_num_output_channels() == 2)
@@ -321,10 +327,23 @@ def load_help(processor, filepath: str):
     synth.add_midi_note(60, 60, 0.0, .25)
     synth.add_midi_note(64, 80, 0.5, .5)
     synth.add_midi_note(67, 127, 0.75, .5)
+    synth.add_midi_note(48, 80, 1.5, .5)
+    synth.add_midi_note(36, 80, 2.0, .5)
 
-    assert(synth.n_midi_events == 3*2)  # multiply by 2 because of the off-notes.
+    assert(synth.n_midi_events == 5*2)  # multiply by 2 because of the off-notes.
 
-    engine.load_graph([(synth, [])])
+    graph = [(synth, [])]
+
+    num_outputs = synth.get_num_output_channels()
+
+    if num_outputs > 2:
+        # Use faust to "fan-in" the number of channels from something larger than 2 to 2.
+        faust_processor = engine.make_faust_processor("faust")
+        num_outputs = synth.get_num_output_channels()
+        faust_processor.set_dsp_string(f"process = si.bus({num_outputs}) :> si.bus(2);")
+        graph.append((faust_processor, ["synth"]))
+
+    engine.load_graph(graph)
 
     render(engine, file_path=OUTPUT / (f'test_plugin_{plugin_basename}.wav'), duration=DURATION)
 
@@ -347,9 +366,9 @@ def test_plugin_iem(plugin_path1="C:/VSTPlugIns/IEMPluginSuite/VST2/IEM/MultiEnc
     ambisonics_encoder.record = True
     ambisonics_decoder.record = True
 
-    plugin_basename = splitext(basename(plugin_path1))[0]
+    plugin_basename = basename(plugin_path1)
 
-    state_file_path = abspath(OUTPUT / (f'state_test_plugin_{plugin_basename}'))
+    state_file_path = abspath(OUTPUT / (f'state_test_plugin_{plugin_basename}.bin'))
 
     if isfile(state_file_path):
         ambisonics_encoder.load_state(state_file_path)
@@ -375,8 +394,8 @@ def test_plugin_iem(plugin_path1="C:/VSTPlugIns/IEMPluginSuite/VST2/IEM/MultiEnc
     # print(ambisonics_encoder.get_plugin_parameters_description())
     # print('inputs: ', ambisonics_encoder.get_num_input_channels(), ' outputs: ', ambisonics_encoder.get_num_output_channels())
 
-    plugin_basename = splitext(basename(plugin_path2))[0]
-    state_file_path = abspath(OUTPUT / (f'state_test_plugin_{plugin_basename}'))
+    plugin_basename = basename(plugin_path2)
+    state_file_path = abspath(OUTPUT / (f'state_test_plugin_{plugin_basename}.bin'))
 
     if isfile(state_file_path):
         ambisonics_decoder.load_state(state_file_path)
diff --git a/thirdparty/rubberband b/thirdparty/rubberband
index a78dcaf4..4cff1741 160000
--- a/thirdparty/rubberband
+++ b/thirdparty/rubberband
@@ -1 +1 @@
-Subproject commit a78dcaf4c8efe6bb95432b5dfc2567c9018fd5a1
+Subproject commit 4cff1741664918791474da9a9d41f48cc3119229