Replaced r8brain with libsoxr

Signed-off-by: Christopher Snowhill <kode54@gmail.com>

.gitignore

@@ -49,3 +49,4 @@ Xcode-config/DEVELOPMENT_TEAM.xcconfig
 /ThirdParty/speex/libspeex.a
 /ThirdParty/vorbis/lib/libvorbisfile.3.dylib
 /ThirdParty/vorbis/lib/libvorbis.0.dylib
+/ThirdParty/soxr/lib/libsoxr.0.dylib

.gitmodules

@@ -19,6 +19,3 @@
 [submodule "Frameworks/libsidplayfp/sidplayfp"]
 	path = Frameworks/libsidplayfp/sidplayfp
 	url = https://github.com/kode54/libsidplayfp.git
-[submodule "Audio/ThirdParty/r8brain-free-src"]
-	path = Audio/ThirdParty/r8brain-free-src
-	url = https://github.com/kode54/r8brain-free-src

Audio/CogAudio.xcodeproj/project.pbxproj

@@ -41,34 +41,9 @@
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 		17F94DD60B8D0F7000A34E87 /* PluginController.mm in Sources */ = {isa = PBXBuildFile; fileRef = 17F94DD40B8D0F7000A34E87 /* PluginController.mm */; };
 		17F94DDD0B8D101100A34E87 /* Plugin.h in Headers */ = {isa = PBXBuildFile; fileRef = 17F94DDC0B8D101100A34E87 /* Plugin.h */; settings = {ATTRIBUTES = (Public, ); }; };
-		831A4FDC2865A7DC0049CFE4 /* CDSPProcessor.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4F9D2865A7DC0049CFE4 /* CDSPProcessor.h */; };
+		831A50142865A7FD0049CFE4 /* rsstate.hpp in Headers */ = {isa = PBXBuildFile; fileRef = 831A50132865A7FD0049CFE4 /* rsstate.hpp */; };
-		831A4FDD2865A7DC0049CFE4 /* CDSPRealFFT.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4F9E2865A7DC0049CFE4 /* CDSPRealFFT.h */; };
+		831A50162865A8800049CFE4 /* rsstate.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 831A50152865A8800049CFE4 /* rsstate.cpp */; };
-		831A4FDE2865A7DC0049CFE4 /* pffft_double.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FA02865A7DC0049CFE4 /* pffft_double.h */; };
+		831A50182865A8B30049CFE4 /* rsstate.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A50172865A8B30049CFE4 /* rsstate.h */; };
-		831A4FDF2865A7DC0049CFE4 /* pf_neon_double_from_avx.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FA22865A7DC0049CFE4 /* pf_neon_double_from_avx.h */; };
-		831A4FE02865A7DC0049CFE4 /* pf_double.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FA32865A7DC0049CFE4 /* pf_double.h */; };
-		831A4FE12865A7DC0049CFE4 /* pf_neon_double.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FA42865A7DC0049CFE4 /* pf_neon_double.h */; };
-		831A4FE22865A7DC0049CFE4 /* pf_sse2_double.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FA52865A7DC0049CFE4 /* pf_sse2_double.h */; };
-		831A4FE32865A7DC0049CFE4 /* pf_avx_double.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FA62865A7DC0049CFE4 /* pf_avx_double.h */; };
-		831A4FE42865A7DC0049CFE4 /* pf_scalar_double.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FA72865A7DC0049CFE4 /* pf_scalar_double.h */; };
-		831A4FE52865A7DC0049CFE4 /* pffft_priv_impl.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FA82865A7DC0049CFE4 /* pffft_priv_impl.h */; };
-		831A4FE62865A7DC0049CFE4 /* pffft_double.c in Sources */ = {isa = PBXBuildFile; fileRef = 831A4FA92865A7DC0049CFE4 /* pffft_double.c */; };
-		831A4FF32865A7DC0049CFE4 /* r8butil.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FB82865A7DC0049CFE4 /* r8butil.h */; };
-		831A4FF52865A7DC0049CFE4 /* r8bbase.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FBA2865A7DC0049CFE4 /* r8bbase.h */; };
-		831A4FFE2865A7DC0049CFE4 /* CDSPSincFilterGen.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FC42865A7DC0049CFE4 /* CDSPSincFilterGen.h */; };
-		831A50072865A7DC0049CFE4 /* LICENSE in Resources */ = {isa = PBXBuildFile; fileRef = 831A4FD02865A7DC0049CFE4 /* LICENSE */; };
-		831A50082865A7DC0049CFE4 /* CDSPResampler.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FD12865A7DC0049CFE4 /* CDSPResampler.h */; };
-		831A50092865A7DC0049CFE4 /* CDSPHBUpsampler.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FD22865A7DC0049CFE4 /* CDSPHBUpsampler.h */; };
-		831A500B2865A7DC0049CFE4 /* CDSPBlockConvolver.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FD42865A7DC0049CFE4 /* CDSPBlockConvolver.h */; };
-		831A500C2865A7DC0049CFE4 /* fft4g.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FD52865A7DC0049CFE4 /* fft4g.h */; };
-		831A500D2865A7DC0049CFE4 /* CDSPHBDownsampler.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FD62865A7DC0049CFE4 /* CDSPHBDownsampler.h */; };
-		831A500E2865A7DC0049CFE4 /* r8bconf.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FD72865A7DC0049CFE4 /* r8bconf.h */; };
-		831A500F2865A7DC0049CFE4 /* CDSPFracInterpolator.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FD82865A7DC0049CFE4 /* CDSPFracInterpolator.h */; };
-		831A50102865A7DC0049CFE4 /* CDSPFIRFilter.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FD92865A7DC0049CFE4 /* CDSPFIRFilter.h */; };
-		831A50112865A7DC0049CFE4 /* r8bbase.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 831A4FDA2865A7DC0049CFE4 /* r8bbase.cpp */; };
-		831A50122865A7DC0049CFE4 /* pffft.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A4FDB2865A7DC0049CFE4 /* pffft.h */; };
-		831A50142865A7FD0049CFE4 /* r8bstate.hpp in Headers */ = {isa = PBXBuildFile; fileRef = 831A50132865A7FD0049CFE4 /* r8bstate.hpp */; };
-		831A50162865A8800049CFE4 /* r8bstate.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 831A50152865A8800049CFE4 /* r8bstate.cpp */; };
-		831A50182865A8B30049CFE4 /* r8bstate.h in Headers */ = {isa = PBXBuildFile; fileRef = 831A50172865A8B30049CFE4 /* r8bstate.h */; };
 		8328995327CB511000D7F028 /* RedundantPlaylistDataStore.m in Sources */ = {isa = PBXBuildFile; fileRef = 8328995127CB510F00D7F028 /* RedundantPlaylistDataStore.m */; };
 		8328995427CB511000D7F028 /* RedundantPlaylistDataStore.h in Headers */ = {isa = PBXBuildFile; fileRef = 8328995227CB511000D7F028 /* RedundantPlaylistDataStore.h */; };
 		8328995727CB51B700D7F028 /* SHA256Digest.h in Headers */ = {isa = PBXBuildFile; fileRef = 8328995527CB51B700D7F028 /* SHA256Digest.h */; };
@@ -93,6 +68,7 @@
 		835C88B2279811A500E28EAE /* hdcd_decode2.c in Sources */ = {isa = PBXBuildFile; fileRef = 835C88B0279811A500E28EAE /* hdcd_decode2.c */; };
 		835FAC5E27BCA14D00BA8562 /* BadSampleCleaner.h in Headers */ = {isa = PBXBuildFile; fileRef = 835FAC5C27BCA14D00BA8562 /* BadSampleCleaner.h */; };
 		835FAC5F27BCA14D00BA8562 /* BadSampleCleaner.m in Sources */ = {isa = PBXBuildFile; fileRef = 835FAC5D27BCA14D00BA8562 /* BadSampleCleaner.m */; };
+		836DF618298F6F5F00CD0580 /* libsoxr.0.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 836DF615298F6E8900CD0580 /* libsoxr.0.dylib */; };
 		83725A9027AA16C90003F694 /* Accelerate.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 83725A7B27AA0D8A0003F694 /* Accelerate.framework */; };
 		83725A9127AA16D50003F694 /* AVFoundation.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 83725A7C27AA0D8E0003F694 /* AVFoundation.framework */; };
 		8377C64C27B8C51500E8BC0F /* fft_accelerate.c in Sources */ = {isa = PBXBuildFile; fileRef = 8377C64B27B8C51500E8BC0F /* fft_accelerate.c */; };
@@ -123,15 +99,6 @@
 /* End PBXBuildFile section */
 
 /* Begin PBXCopyFilesBuildPhase section */
-		17D21D2B0B8BE6A200D1EBDE /* CopyFiles */ = {
-			isa = PBXCopyFilesBuildPhase;
-			buildActionMask = 2147483647;
-			dstPath = "";
-			dstSubfolderSpec = 10;
-			files = (
-			);
-			runOnlyForDeploymentPostprocessing = 0;
-		};
 		83725A8D27AA0DDB0003F694 /* CopyFiles */ = {
 			isa = PBXCopyFilesBuildPhase;
 			buildActionMask = 2147483647;
@@ -182,35 +149,9 @@
 		17F94DD40B8D0F7000A34E87 /* PluginController.mm */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.cpp.objcpp; path = PluginController.mm; sourceTree = "<group>"; };
 		17F94DDC0B8D101100A34E87 /* Plugin.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = Plugin.h; sourceTree = "<group>"; };
 		32DBCF5E0370ADEE00C91783 /* CogAudio_Prefix.pch */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CogAudio_Prefix.pch; sourceTree = "<group>"; };
-		831A4F9D2865A7DC0049CFE4 /* CDSPProcessor.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPProcessor.h; sourceTree = "<group>"; };
+		831A50132865A7FD0049CFE4 /* rsstate.hpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.h; path = rsstate.hpp; sourceTree = "<group>"; };
-		831A4F9E2865A7DC0049CFE4 /* CDSPRealFFT.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPRealFFT.h; sourceTree = "<group>"; };
+		831A50152865A8800049CFE4 /* rsstate.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = rsstate.cpp; sourceTree = "<group>"; };
-		831A4FA02865A7DC0049CFE4 /* pffft_double.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pffft_double.h; sourceTree = "<group>"; };
+		831A50172865A8B30049CFE4 /* rsstate.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = rsstate.h; sourceTree = "<group>"; };
-		831A4FA22865A7DC0049CFE4 /* pf_neon_double_from_avx.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pf_neon_double_from_avx.h; sourceTree = "<group>"; };
-		831A4FA32865A7DC0049CFE4 /* pf_double.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pf_double.h; sourceTree = "<group>"; };
-		831A4FA42865A7DC0049CFE4 /* pf_neon_double.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pf_neon_double.h; sourceTree = "<group>"; };
-		831A4FA52865A7DC0049CFE4 /* pf_sse2_double.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pf_sse2_double.h; sourceTree = "<group>"; };
-		831A4FA62865A7DC0049CFE4 /* pf_avx_double.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pf_avx_double.h; sourceTree = "<group>"; };
-		831A4FA72865A7DC0049CFE4 /* pf_scalar_double.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pf_scalar_double.h; sourceTree = "<group>"; };
-		831A4FA82865A7DC0049CFE4 /* pffft_priv_impl.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pffft_priv_impl.h; sourceTree = "<group>"; };
-		831A4FA92865A7DC0049CFE4 /* pffft_double.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = pffft_double.c; sourceTree = "<group>"; };
-		831A4FB72865A7DC0049CFE4 /* CDSPHBUpsampler.inc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.pascal; path = CDSPHBUpsampler.inc; sourceTree = "<group>"; };
-		831A4FB82865A7DC0049CFE4 /* r8butil.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = r8butil.h; sourceTree = "<group>"; };
-		831A4FBA2865A7DC0049CFE4 /* r8bbase.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = r8bbase.h; sourceTree = "<group>"; };
-		831A4FC42865A7DC0049CFE4 /* CDSPSincFilterGen.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPSincFilterGen.h; sourceTree = "<group>"; };
-		831A4FD02865A7DC0049CFE4 /* LICENSE */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = LICENSE; sourceTree = "<group>"; };
-		831A4FD12865A7DC0049CFE4 /* CDSPResampler.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPResampler.h; sourceTree = "<group>"; };
-		831A4FD22865A7DC0049CFE4 /* CDSPHBUpsampler.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPHBUpsampler.h; sourceTree = "<group>"; };
-		831A4FD42865A7DC0049CFE4 /* CDSPBlockConvolver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPBlockConvolver.h; sourceTree = "<group>"; };
-		831A4FD52865A7DC0049CFE4 /* fft4g.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fft4g.h; sourceTree = "<group>"; };
-		831A4FD62865A7DC0049CFE4 /* CDSPHBDownsampler.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPHBDownsampler.h; sourceTree = "<group>"; };
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-		831A4FD82865A7DC0049CFE4 /* CDSPFracInterpolator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPFracInterpolator.h; sourceTree = "<group>"; };
-		831A4FD92865A7DC0049CFE4 /* CDSPFIRFilter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDSPFIRFilter.h; sourceTree = "<group>"; };
-		831A4FDA2865A7DC0049CFE4 /* r8bbase.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = r8bbase.cpp; sourceTree = "<group>"; };
-		831A4FDB2865A7DC0049CFE4 /* pffft.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = pffft.h; sourceTree = "<group>"; };
-		831A50132865A7FD0049CFE4 /* r8bstate.hpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.h; path = r8bstate.hpp; sourceTree = "<group>"; };
-		831A50152865A8800049CFE4 /* r8bstate.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = r8bstate.cpp; sourceTree = "<group>"; };
-		831A50172865A8B30049CFE4 /* r8bstate.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = r8bstate.h; sourceTree = "<group>"; };
 		8328995127CB510F00D7F028 /* RedundantPlaylistDataStore.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; name = RedundantPlaylistDataStore.m; path = ../../Utils/RedundantPlaylistDataStore.m; sourceTree = "<group>"; };
 		8328995227CB511000D7F028 /* RedundantPlaylistDataStore.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RedundantPlaylistDataStore.h; path = ../../Utils/RedundantPlaylistDataStore.h; sourceTree = "<group>"; };
 		8328995527CB51B700D7F028 /* SHA256Digest.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = SHA256Digest.h; path = ../../Utils/SHA256Digest.h; sourceTree = "<group>"; };
@@ -235,6 +176,7 @@
 		835C88B0279811A500E28EAE /* hdcd_decode2.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = hdcd_decode2.c; sourceTree = "<group>"; };
 		835FAC5C27BCA14D00BA8562 /* BadSampleCleaner.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = BadSampleCleaner.h; path = Utils/BadSampleCleaner.h; sourceTree = SOURCE_ROOT; };
 		835FAC5D27BCA14D00BA8562 /* BadSampleCleaner.m */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.objc; name = BadSampleCleaner.m; path = Utils/BadSampleCleaner.m; sourceTree = SOURCE_ROOT; };
+		836DF615298F6E8900CD0580 /* libsoxr.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libsoxr.0.dylib; path = ../ThirdParty/soxr/lib/libsoxr.0.dylib; sourceTree = "<group>"; };
 		83725A7B27AA0D8A0003F694 /* Accelerate.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = Accelerate.framework; path = System/Library/Frameworks/Accelerate.framework; sourceTree = SDKROOT; };
 		83725A7C27AA0D8E0003F694 /* AVFoundation.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = AVFoundation.framework; path = System/Library/Frameworks/AVFoundation.framework; sourceTree = SDKROOT; };
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@@ -277,6 +219,7 @@
 				8350416D28646149006B32CC /* CoreMedia.framework in Frameworks */,
 				83725A9027AA16C90003F694 /* Accelerate.framework in Frameworks */,
 				17D21DAD0B8BE76800D1EBDE /* AudioToolbox.framework in Frameworks */,
+				836DF618298F6F5F00CD0580 /* libsoxr.0.dylib in Frameworks */,
 				17D21DAE0B8BE76800D1EBDE /* AudioUnit.framework in Frameworks */,
 				17D21DAF0B8BE76800D1EBDE /* CoreAudio.framework in Frameworks */,
 				17D21DB00B8BE76800D1EBDE /* CoreAudioKit.framework in Frameworks */,
@@ -371,6 +314,7 @@
 		1058C7B2FEA5585E11CA2CBB /* Other Frameworks */ = {
 			isa = PBXGroup;
 			children = (
+				836DF615298F6E8900CD0580 /* libsoxr.0.dylib */,
 				83725A7B27AA0D8A0003F694 /* Accelerate.framework */,
 				17D21DAA0B8BE76800D1EBDE /* AudioUnit.framework */,
 				17D21DA90B8BE76800D1EBDE /* AudioToolbox.framework */,
@@ -425,10 +369,9 @@
 			children = (
 				834A41A4287A90AB00EB9D9B /* fsurround */,
 				839E56E02879450300DFB5F4 /* hrtf */,
-				831A50152865A8800049CFE4 /* r8bstate.cpp */,
+				831A50152865A8800049CFE4 /* rsstate.cpp */,
-				831A50172865A8B30049CFE4 /* r8bstate.h */,
+				831A50172865A8B30049CFE4 /* rsstate.h */,
-				831A50132865A7FD0049CFE4 /* r8bstate.hpp */,
+				831A50132865A7FD0049CFE4 /* rsstate.hpp */,
-				831A4F9C2865A7DC0049CFE4 /* r8brain-free-src */,
 				8377C64A27B8C51500E8BC0F /* deadbeef */,
 				835C88AE279811A500E28EAE /* hdcd */,
 				17D21DC40B8BE79700D1EBDE /* CoreAudioUtils */,
@@ -474,55 +417,6 @@
 			name = "Other Sources";
 			sourceTree = "<group>";
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-			children = (
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-				831A4FBA2865A7DC0049CFE4 /* r8bbase.h */,
-				831A4FC42865A7DC0049CFE4 /* CDSPSincFilterGen.h */,
-				831A4FD02865A7DC0049CFE4 /* LICENSE */,
-				831A4FD12865A7DC0049CFE4 /* CDSPResampler.h */,
-				831A4FD22865A7DC0049CFE4 /* CDSPHBUpsampler.h */,
-				831A4FD42865A7DC0049CFE4 /* CDSPBlockConvolver.h */,
-				831A4FD52865A7DC0049CFE4 /* fft4g.h */,
-				831A4FD62865A7DC0049CFE4 /* CDSPHBDownsampler.h */,
-				831A4FD72865A7DC0049CFE4 /* r8bconf.h */,
-				831A4FD82865A7DC0049CFE4 /* CDSPFracInterpolator.h */,
-				831A4FD92865A7DC0049CFE4 /* CDSPFIRFilter.h */,
-				831A4FDA2865A7DC0049CFE4 /* r8bbase.cpp */,
-				831A4FDB2865A7DC0049CFE4 /* pffft.h */,
-			);
-			path = "r8brain-free-src";
-			sourceTree = "<group>";
-		};
-		831A4F9F2865A7DC0049CFE4 /* pffft_double */ = {
-			isa = PBXGroup;
-			children = (
-				831A4FA02865A7DC0049CFE4 /* pffft_double.h */,
-				831A4FA12865A7DC0049CFE4 /* simd */,
-				831A4FA82865A7DC0049CFE4 /* pffft_priv_impl.h */,
-				831A4FA92865A7DC0049CFE4 /* pffft_double.c */,
-			);
-			path = pffft_double;
-			sourceTree = "<group>";
-		};
-		831A4FA12865A7DC0049CFE4 /* simd */ = {
-			isa = PBXGroup;
-			children = (
-				831A4FA22865A7DC0049CFE4 /* pf_neon_double_from_avx.h */,
-				831A4FA32865A7DC0049CFE4 /* pf_double.h */,
-				831A4FA42865A7DC0049CFE4 /* pf_neon_double.h */,
-				831A4FA52865A7DC0049CFE4 /* pf_sse2_double.h */,
-				831A4FA62865A7DC0049CFE4 /* pf_avx_double.h */,
-				831A4FA72865A7DC0049CFE4 /* pf_scalar_double.h */,
-			);
-			path = simd;
-			sourceTree = "<group>";
-		};
 		834A41A4287A90AB00EB9D9B /* fsurround */ = {
 			isa = PBXGroup;
 			children = (
@@ -590,61 +484,39 @@
 			files = (
 				839E56E82879450300DFB5F4 /* IHrtfData.h in Headers */,
 				17D21CA10B8BE4BA00D1EBDE /* BufferChain.h in Headers */,
-				831A4FE02865A7DC0049CFE4 /* pf_double.h in Headers */,
+				831A50142865A7FD0049CFE4 /* rsstate.hpp in Headers */,
-				831A50142865A7FD0049CFE4 /* r8bstate.hpp in Headers */,
 				834A41AC287A90AB00EB9D9B /* channelmaps.h in Headers */,
 				17D21CA50B8BE4BA00D1EBDE /* InputNode.h in Headers */,
 				834A41A9287A90AB00EB9D9B /* freesurround_decoder.h in Headers */,
 				17D21CA70B8BE4BA00D1EBDE /* Node.h in Headers */,
 				8399CF2C27B5D1D5008751F1 /* NSDictionary+Merge.h in Headers */,
-				831A4FF32865A7DC0049CFE4 /* r8butil.h in Headers */,
 				17D21CA90B8BE4BA00D1EBDE /* OutputNode.h in Headers */,
 				8328995427CB511000D7F028 /* RedundantPlaylistDataStore.h in Headers */,
-				831A50102865A7DC0049CFE4 /* CDSPFIRFilter.h in Headers */,
-				831A4FFE2865A7DC0049CFE4 /* CDSPSincFilterGen.h in Headers */,
-				831A4FF52865A7DC0049CFE4 /* r8bbase.h in Headers */,
-				831A50082865A7DC0049CFE4 /* CDSPResampler.h in Headers */,
 				17D21CC50B8BE4BA00D1EBDE /* OutputAVFoundation.h in Headers */,
 				83504165286447DA006B32CC /* Downmix.h in Headers */,
-				831A4FDE2865A7DC0049CFE4 /* pffft_double.h in Headers */,
 				839E56E52879450300DFB5F4 /* HrtfData.h in Headers */,
 				83B74281289E027F005AAC28 /* CogAudio-Bridging-Header.h in Headers */,
-				831A4FE12865A7DC0049CFE4 /* pf_neon_double.h in Headers */,
 				17D21CC70B8BE4BA00D1EBDE /* Status.h in Headers */,
 				17D21CF30B8BE5EF00D1EBDE /* CogSemaphore.h in Headers */,
 				839E56E62879450300DFB5F4 /* Endianness.h in Headers */,
 				17D21DC70B8BE79700D1EBDE /* CoreAudioUtils.h in Headers */,
 				839E56ED2879515D00DFB5F4 /* HeadphoneFilter.h in Headers */,
 				17D21EBD0B8BF44000D1EBDE /* AudioPlayer.h in Headers */,
-				831A50182865A8B30049CFE4 /* r8bstate.h in Headers */,
+				831A50182865A8B30049CFE4 /* rsstate.h in Headers */,
-				831A4FE52865A7DC0049CFE4 /* pffft_priv_impl.h in Headers */,
-				831A4FE42865A7DC0049CFE4 /* pf_scalar_double.h in Headers */,
-				831A500D2865A7DC0049CFE4 /* CDSPHBDownsampler.h in Headers */,
-				831A500C2865A7DC0049CFE4 /* fft4g.h in Headers */,
 				834FD4F027AF93680063BC83 /* ChunkList.h in Headers */,
 				17F94DD50B8D0F7000A34E87 /* PluginController.h in Headers */,
-				831A50122865A7DC0049CFE4 /* pffft.h in Headers */,
-				831A4FDC2865A7DC0049CFE4 /* CDSPProcessor.h in Headers */,
-				831A4FDD2865A7DC0049CFE4 /* CDSPRealFFT.h in Headers */,
 				17F94DDD0B8D101100A34E87 /* Plugin.h in Headers */,
 				8328995727CB51B700D7F028 /* SHA256Digest.h in Headers */,
 				834FD4EB27AF8F380063BC83 /* AudioChunk.h in Headers */,
 				834A41AF287ABD6F00EB9D9B /* FSurroundFilter.h in Headers */,
 				17A2D3C50B8D1D37000778C4 /* AudioDecoder.h in Headers */,
-				831A50092865A7DC0049CFE4 /* CDSPHBUpsampler.h in Headers */,
-				831A500E2865A7DC0049CFE4 /* r8bconf.h in Headers */,
 				8347C7412796C58800FA8A7D /* NSFileHandle+CreateFile.h in Headers */,
 				17C940230B900909008627D6 /* AudioMetadataReader.h in Headers */,
-				831A500F2865A7DC0049CFE4 /* CDSPFracInterpolator.h in Headers */,
-				831A4FE22865A7DC0049CFE4 /* pf_sse2_double.h in Headers */,
 				839E56F7287974A100DFB5F4 /* SandboxBroker.h in Headers */,
 				839065F32853338700636FBB /* dsd2float.h in Headers */,
 				17B619300B909BC300BC003F /* AudioPropertiesReader.h in Headers */,
-				831A4FDF2865A7DC0049CFE4 /* pf_neon_double_from_avx.h in Headers */,
-				831A4FE32865A7DC0049CFE4 /* pf_avx_double.h in Headers */,
 				839366671815923C006DD712 /* CogPluginMulti.h in Headers */,
 				17ADB13C0B97926D00257CA2 /* AudioSource.h in Headers */,
-				831A500B2865A7DC0049CFE4 /* CDSPBlockConvolver.h in Headers */,
 				835C88B1279811A500E28EAE /* hdcd_decode2.h in Headers */,
 				8EC1225F0B993BD500C5B3AD /* ConverterNode.h in Headers */,
 				8384912718080FF100E7332D /* Logging.h in Headers */,
@@ -664,7 +536,6 @@
 			isa = PBXNativeTarget;
 			buildConfigurationList = 1DEB91AD08733DA50010E9CD /* Build configuration list for PBXNativeTarget "CogAudio Framework" */;
 			buildPhases = (
-				17D21D2B0B8BE6A200D1EBDE /* CopyFiles */,
 				8DC2EF500486A6940098B216 /* Headers */,
 				8DC2EF540486A6940098B216 /* Sources */,
 				8DC2EF560486A6940098B216 /* Frameworks */,
@@ -718,7 +589,6 @@
 			isa = PBXResourcesBuildPhase;
 			buildActionMask = 2147483647;
 			files = (
-				831A50072865A7DC0049CFE4 /* LICENSE in Resources */,
 			);
 			runOnlyForDeploymentPostprocessing = 0;
 		};
@@ -732,11 +602,10 @@
 				17D21CA20B8BE4BA00D1EBDE /* BufferChain.m in Sources */,
 				17D21CA60B8BE4BA00D1EBDE /* InputNode.m in Sources */,
 				839E56EE2879515D00DFB5F4 /* HeadphoneFilter.mm in Sources */,
-				831A50112865A7DC0049CFE4 /* r8bbase.cpp in Sources */,
 				83504166286447DA006B32CC /* Downmix.m in Sources */,
 				8399CF2D27B5D1D5008751F1 /* NSDictionary+Merge.m in Sources */,
 				834A41AB287A90AB00EB9D9B /* channelmaps.cpp in Sources */,
-				831A50162865A8800049CFE4 /* r8bstate.cpp in Sources */,
+				831A50162865A8800049CFE4 /* rsstate.cpp in Sources */,
 				17D21CA80B8BE4BA00D1EBDE /* Node.m in Sources */,
 				17D21CAA0B8BE4BA00D1EBDE /* OutputNode.m in Sources */,
 				17D21CC60B8BE4BA00D1EBDE /* OutputAVFoundation.m in Sources */,
@@ -753,7 +622,6 @@
 				17D21EBE0B8BF44000D1EBDE /* AudioPlayer.m in Sources */,
 				17F94DD60B8D0F7000A34E87 /* PluginController.mm in Sources */,
 				839E56E72879450300DFB5F4 /* HrtfData.cpp in Sources */,
-				831A4FE62865A7DC0049CFE4 /* pffft_double.c in Sources */,
 				17A2D3C60B8D1D37000778C4 /* AudioDecoder.m in Sources */,
 				8328995827CB51B700D7F028 /* SHA256Digest.m in Sources */,
 				17C940240B900909008627D6 /* AudioMetadataReader.m in Sources */,
@@ -789,14 +657,12 @@
 				GCC_OPTIMIZATION_LEVEL = 0;
 				GCC_PRECOMPILE_PREFIX_HEADER = YES;
 				GCC_PREFIX_HEADER = CogAudio_Prefix.pch;
-				GCC_PREPROCESSOR_DEFINITIONS = (
+				GCC_PREPROCESSOR_DEFINITIONS = "DEBUG=1";
-					"DEBUG=1",
+				HEADER_SEARCH_PATHS = ../ThirdParty/soxr/include;
-					"R8B_EXTFFT=1",
-					"R8B_PFFFT_DOUBLE=1",
-				);
 				INFOPLIST_FILE = Info.plist;
 				INSTALL_PATH = "@executable_path/../Frameworks";
 				LD_RUNPATH_SEARCH_PATHS = "@loader_path/Frameworks";
+				LIBRARY_SEARCH_PATHS = ../ThirdParty/soxr/lib;
 				OTHER_LDFLAGS = "";
 				PRODUCT_BUNDLE_IDENTIFIER = org.cogx.cogaudio;
 				PRODUCT_NAME = CogAudio;
@@ -823,13 +689,12 @@
 				GCC_ENABLE_OBJC_EXCEPTIONS = YES;
 				GCC_PRECOMPILE_PREFIX_HEADER = YES;
 				GCC_PREFIX_HEADER = CogAudio_Prefix.pch;
-				GCC_PREPROCESSOR_DEFINITIONS = (
+				GCC_PREPROCESSOR_DEFINITIONS = "";
-					"R8B_EXTFFT=1",
+				HEADER_SEARCH_PATHS = ../ThirdParty/soxr/include;
-					"R8B_PFFFT_DOUBLE=1",
-				);
 				INFOPLIST_FILE = Info.plist;
 				INSTALL_PATH = "@executable_path/../Frameworks";
 				LD_RUNPATH_SEARCH_PATHS = "@loader_path/Frameworks";
+				LIBRARY_SEARCH_PATHS = ../ThirdParty/soxr/lib;
 				OTHER_LDFLAGS = "";
 				PRODUCT_BUNDLE_IDENTIFIER = org.cogx.cogaudio;
 				PRODUCT_NAME = CogAudio;

Audio/Output/HeadphoneFilter.mm

@@ -14,7 +14,7 @@
 
 #import <fstream>
 
-#import "r8bstate.h"
+#import "rsstate.h"
 
 #import "HrtfData.h"
 
@@ -205,24 +205,24 @@ static impulseCache *_sharedController = nil;
 		double sampleRatio = sampleRate / sampleRateOfSource;
 		int resampledCount = (int)ceil((double)sampleCount * sampleRatio);
 
-		void *r8bstate = r8bstate_new(channelCount * 2, 1024, sampleRateOfSource, sampleRate);
+		void *rsstate = rsstate_new(channelCount * 2, sampleRateOfSource, sampleRate);
 
 		float *resampledImpulse = (float *)_memalign_malloc(resampledCount * sizeof(float) * channelCount * 2, 16);
 		if(!resampledImpulse) {
-			r8bstate_delete(r8bstate);
+			rsstate_delete(rsstate);
 			return nil;
 		}
 
 		size_t inputDone = 0;
 		size_t outputDone = 0;
 
-		outputDone = r8bstate_resample(r8bstate, impulseData, sampleCount, &inputDone, resampledImpulse, resampledCount);
+		outputDone = rsstate_resample(rsstate, impulseData, sampleCount, &inputDone, resampledImpulse, resampledCount);
 
 		while(outputDone < resampledCount) {
-			outputDone += r8bstate_flush(r8bstate, resampledImpulse + outputDone * channelCount * 2, resampledCount - outputDone);
+			outputDone += rsstate_flush(rsstate, resampledImpulse + outputDone * channelCount * 2, resampledCount - outputDone);
 		}
 
-		r8bstate_delete(r8bstate);
+		rsstate_delete(rsstate);
 
 		sampleCount = (int)outputDone;
 

Audio/Output/OutputAVFoundation.h

@@ -40,10 +40,10 @@ using std::atomic_long;
 @interface OutputAVFoundation : NSObject {
 	OutputNode *outputController;
 
-	BOOL r8bDone;
+	BOOL rsDone;
-	void *r8bstate, *r8bold;
+	void *rsstate, *rsold;
 
-	void *r8bvis;
+	void *rsvis;
 	double lastVisRate;
 
 	BOOL stopInvoked;
@@ -121,7 +121,7 @@ using std::atomic_long;
 
 	float *samplePtr;
 	float tempBuffer[512 * 32];
-	float r8bTempBuffer[4096 * 32];
+	float rsTempBuffer[4096 * 32];
 	float inputBuffer[4096 * 32]; // 4096 samples times maximum supported channel count
 	float fsurroundBuffer[8192 * 6];
 	float hrtfBuffer[4096 * 2];

Audio/Output/OutputAVFoundation.m

@@ -17,7 +17,7 @@
 
 #import <Accelerate/Accelerate.h>
 
-#import "r8bstate.h"
+#import "rsstate.h"
 
 #import "FSurroundFilter.h"
 
@@ -90,14 +90,14 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 			[currentPtsLock lock];
 			if(formatClipped) {
 				ALog(@"Sample rate clipped to no more than %f Hz!", maxSampleRate);
-				if(r8bstate) {
+				if(rsstate) {
-					r8bold = r8bstate;
+					rsold = rsstate;
-					r8bstate = NULL;
+					rsstate = NULL;
 				}
-				r8bstate = r8bstate_new(format.mChannelsPerFrame, 1024, srcRate, dstRate);
+				rsstate = rsstate_new(format.mChannelsPerFrame, srcRate, dstRate);
-			} else if(r8bstate) {
+			} else if(rsstate) {
-				r8bold = r8bstate;
+				rsold = rsstate;
-				r8bstate = NULL;
+				rsstate = NULL;
 			}
 			[currentPtsLock unlock];
 			newFormat = format;
@@ -110,7 +110,7 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 			visFormat.mBytesPerPacket = visFormat.mBytesPerFrame * visFormat.mFramesPerPacket;
 
 			downmixerForVis = [[DownmixProcessor alloc] initWithInputFormat:format inputConfig:config andOutputFormat:visFormat outputConfig:AudioConfigMono];
-			if(!r8bold) {
+			if(!rsold) {
 				realStreamFormat = format;
 				realStreamChannelConfig = config;
 				streamFormatChanged = YES;
@@ -140,14 +140,14 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 #endif
 		// It should be fine to request up to double, we'll only get downsampled
 		const float *outputPtr = (const float *)[samples bytes];
-		if(r8bstate) {
+		if(rsstate) {
 			size_t inDone = 0;
 			[currentPtsLock lock];
-			size_t framesDone = r8bstate_resample(r8bstate, outputPtr, frameCount, &inDone, &r8bTempBuffer[0], amountToRead);
+			size_t framesDone = rsstate_resample(rsstate, outputPtr, frameCount, &inDone, &rsTempBuffer[0], amountToRead);
 			[currentPtsLock unlock];
 			if(!framesDone) return 0;
 			frameCount = (int)framesDone;
-			outputPtr = &r8bTempBuffer[0];
+			outputPtr = &rsTempBuffer[0];
 			chunkDuration = frameCount / newFormat.mSampleRate;
 		}
 
@@ -159,11 +159,11 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 
 		if(newFormat.mSampleRate != 44100.0) {
 			if(newFormat.mSampleRate != lastVisRate) {
-				if(r8bvis) {
+				if(rsvis) {
 					for(;;) {
 						int samplesFlushed;
 						[currentPtsLock lock];
-						samplesFlushed = (int)r8bstate_flush(r8bvis, &visTemp[0], 8192);
+						samplesFlushed = (int)rsstate_flush(rsvis, &visTemp[0], 8192);
 						[currentPtsLock unlock];
 						if(samplesFlushed) {
 							[visController postVisPCM:visTemp amount:samplesFlushed];
@@ -172,23 +172,23 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 						}
 					}
 					[currentPtsLock lock];
-					r8bstate_delete(r8bvis);
+					rsstate_delete(rsvis);
-					r8bvis = NULL;
+					rsvis = NULL;
 					[currentPtsLock unlock];
 				}
 				lastVisRate = newFormat.mSampleRate;
 				[currentPtsLock lock];
-				r8bvis = r8bstate_new(1, 1024, lastVisRate, 44100.0);
+				rsvis = rsstate_new(1, lastVisRate, 44100.0);
 				[currentPtsLock unlock];
 			}
-			if(r8bvis) {
+			if(rsvis) {
 				int samplesProcessed;
 				size_t totalDone = 0;
 				size_t inDone = 0;
 				size_t visFrameCount = frameCount;
 				{
 					[currentPtsLock lock];
-					samplesProcessed = (int)r8bstate_resample(r8bvis, &visAudio[totalDone], visFrameCount, &inDone, &visTemp[0], 8192);
+					samplesProcessed = (int)rsstate_resample(rsvis, &visAudio[totalDone], visFrameCount, &inDone, &visTemp[0], 8192);
 					[currentPtsLock unlock];
 					if(samplesProcessed) {
 						[visController postVisPCM:&visTemp[0] amount:samplesProcessed];
@@ -197,11 +197,11 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 					visFrameCount -= inDone;
 				} while(samplesProcessed && visFrameCount);
 			}
-		} else if(r8bvis) {
+		} else if(rsvis) {
 			for(;;) {
 				int samplesFlushed;
 				[currentPtsLock lock];
-				samplesFlushed = (int)r8bstate_flush(r8bvis, &visTemp[0], 8192);
+				samplesFlushed = (int)rsstate_flush(rsvis, &visTemp[0], 8192);
 				[currentPtsLock unlock];
 				if(samplesFlushed) {
 					[visController postVisPCM:visTemp amount:samplesFlushed];
@@ -210,8 +210,8 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 				}
 			}
 			[currentPtsLock lock];
-			r8bstate_delete(r8bvis);
+			rsstate_delete(rsvis);
-			r8bvis = NULL;
+			rsvis = NULL;
 			[currentPtsLock unlock];
 			[visController postVisPCM:&visAudio[0] amount:frameCount];
 		} else {
@@ -795,26 +795,26 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 		int samplesRendered;
 
 		if(i == 0) {
-			if(!r8bold) {
+			if(!rsold) {
 				++i;
 				continue;
 			}
 			[currentPtsLock lock];
-			samplesRendered = r8bstate_flush(r8bold, &r8bTempBuffer[0], 4096);
+			samplesRendered = rsstate_flush(rsold, &rsTempBuffer[0], 4096);
 			[currentPtsLock unlock];
 			if(!samplesRendered) {
-				r8bstate_delete(r8bold);
+				rsstate_delete(rsold);
-				r8bold = NULL;
+				rsold = NULL;
-				r8bDone = YES;
+				rsDone = YES;
 				++i;
 				continue;
 			}
-			samplePtr = &r8bTempBuffer[0];
+			samplePtr = &rsTempBuffer[0];
 		} else {
 			samplesRendered = inputRendered;
 			samplePtr = &inputBuffer[0];
-			if(r8bDone) {
+			if(rsDone) {
-				r8bDone = NO;
+				rsDone = NO;
 				realStreamFormat = newFormat;
 				realStreamChannelConfig = newChannelConfig;
 				streamFormatChanged = YES;
@@ -959,11 +959,11 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 
 		downmixerForVis = nil;
 
-		r8bDone = NO;
+		rsDone = NO;
-		r8bstate = NULL;
+		rsstate = NULL;
-		r8bold = NULL;
+		rsold = NULL;
 
-		r8bvis = NULL;
+		rsvis = NULL;
 		lastVisRate = 44100.0;
 
 		AudioComponentDescription desc;
@@ -1079,9 +1079,9 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 	OutputNode *outputController = self->outputController;
 	double *latencySecondsOut = &self->secondsLatency;
 	VisualizationController *visController = self->visController;
-	void **r8bstate = &self->r8bstate;
+	void **rsstate = &self->rsstate;
-	void **r8bold = &self->r8bold;
+	void **rsold = &self->rsold;
-	void **r8bvis = &self->r8bvis;
+	void **rsvis = &self->rsvis;
 	FSurroundFilter *const *fsurroundtest = &self->fsurround;
 	currentPtsObserver = [renderSynchronizer addPeriodicTimeObserverForInterval:interval
 	                                                                      queue:NULL
@@ -1101,14 +1101,14 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 		                                                                 double latencySeconds = CMTimeGetSeconds(latencyTime);
 		                                                                 double latencyVis = 0.0;
 		                                                                 [lock lock];
-		                                                                 if(*r8bstate) {
+		                                                                 if(*rsstate) {
-			                                                                 latencySeconds += r8bstate_latency(*r8bstate);
+			                                                                 latencySeconds += rsstate_latency(*rsstate);
 		                                                                 }
-		                                                                 if(*r8bold) {
+		                                                                 if(*rsold) {
-			                                                                 latencySeconds += r8bstate_latency(*r8bold);
+			                                                                 latencySeconds += rsstate_latency(*rsold);
 		                                                                 }
-		                                                                 if(*r8bvis) {
+		                                                                 if(*rsvis) {
-			                                                                 latencyVis = r8bstate_latency(*r8bvis);
+			                                                                 latencyVis = rsstate_latency(*rsvis);
 		                                                                 }
 		                                                                 if(*fsurroundtest) {
 			                                                                 latencyVis += 2048.0 / [(*fsurroundtest) srate];
@@ -1268,17 +1268,17 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 #endif
 		outputController = nil;
 		visController = nil;
-		if(r8bstate) {
+		if(rsstate) {
-			r8bstate_delete(r8bstate);
+			rsstate_delete(rsstate);
-			r8bstate = NULL;
+			rsstate = NULL;
 		}
-		if(r8bold) {
+		if(rsold) {
-			r8bstate_delete(r8bold);
+			rsstate_delete(rsold);
-			r8bold = NULL;
+			rsold = NULL;
 		}
-		if(r8bvis) {
+		if(rsvis) {
-			r8bstate_delete(r8bvis);
+			rsstate_delete(rsvis);
-			r8bvis = NULL;
+			rsvis = NULL;
 		}
 		stopCompleted = YES;
 	}

Audio/ThirdParty/r8brain-free-src

@@ -1 +0,0 @@
-Subproject commit 8ce873a7e797c9dc39d5c2603eb4a900d71385cb

Audio/ThirdParty/r8bstate.cpp

@@ -1,31 +0,0 @@
-//
-//  r8bstate.cpp
-//  CogAudio Framework
-//
-//  Created by Christopher Snowhill on 6/24/22.
-//
-
-#include "r8bstate.h"
-#include "r8bstate.hpp"
-
-void *r8bstate_new(int channelCount, int bufferCapacity, double srcRate,
-                   double dstRate) {
-	return (void *)new r8bstate(channelCount, bufferCapacity, srcRate, dstRate);
-}
-
-void r8bstate_delete(void *state) {
-	delete(r8bstate *)state;
-}
-
-double r8bstate_latency(void *state) {
-	return ((r8bstate *)state)->latency();
-}
-
-int r8bstate_resample(void *state, const float *input, size_t inCount, size_t *inDone,
-                      float *output, size_t outMax) {
-	return ((r8bstate *)state)->resample(input, inCount, inDone, output, outMax);
-}
-
-int r8bstate_flush(void *state, float *output, size_t outMax) {
-	return ((r8bstate *)state)->flush(output, outMax);
-}

Audio/ThirdParty/r8bstate.h

@@ -1,33 +0,0 @@
-//
-//  r8bstate.h
-//  CogAudio Framework
-//
-//  Created by Christopher Snowhill on 6/24/22.
-//
-
-#include <stdint.h>
-#include <stdlib.h>
-
-#ifndef r8bstate_h
-#define r8bstate_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-void *r8bstate_new(int channelCount, int bufferCapacity, double srcRate,
-                   double dstRate);
-void r8bstate_delete(void *);
-
-double r8bstate_latency(void *);
-
-int r8bstate_resample(void *, const float *input, size_t inCount, size_t *inDone,
-                      float *output, size_t outMax);
-
-int r8bstate_flush(void *, float *output, size_t outMax);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif /* r8bstate_h */

Audio/ThirdParty/r8bstate.hpp

@@ -1,165 +0,0 @@
-//
-//  r8bstate.hpp
-//  CogAudio Framework
-//
-//  Created by Christopher Snowhill on 3/3/22.
-//
-
-#ifndef r8bstate_hpp
-#define r8bstate_hpp
-
-#include <Accelerate/Accelerate.h>
-
-#include "r8bbase.h"
-
-#include "CDSPResampler.h"
-
-struct r8bstate {
-	int channelCount;
-	int bufferCapacity;
-	size_t remainder;
-	uint64_t inProcessed;
-	uint64_t outProcessed;
-	double sampleRatio;
-	double dstRate;
-	r8b::CFixedBuffer<double> InBuf;
-	r8b::CFixedBuffer<double> *OutBufs;
-	r8b::CDSPResampler24 **Resamps;
-	r8bstate(int _channelCount, int _bufferCapacity, double srcRate, double _dstRate)
-	: channelCount(_channelCount), bufferCapacity(_bufferCapacity), inProcessed(0), outProcessed(0), remainder(0), dstRate(_dstRate) {
-		InBuf.alloc(bufferCapacity);
-		OutBufs = new r8b::CFixedBuffer<double>[channelCount];
-		Resamps = new r8b::CDSPResampler24 *[channelCount];
-		for(int i = 0; i < channelCount; ++i) {
-			Resamps[i] = new r8b::CDSPResampler24(srcRate, dstRate, bufferCapacity);
-		}
-		sampleRatio = dstRate / srcRate;
-	}
-
-	~r8bstate() {
-		delete[] OutBufs;
-		for(int i = 0; i < channelCount; ++i) {
-			delete Resamps[i];
-		}
-		delete[] Resamps;
-	}
-
-	double latency() {
-		return (((double)inProcessed * sampleRatio) - (double)outProcessed) / dstRate;
-	}
-
-	int resample(const float *input, size_t inCount, size_t *inDone, float *output, size_t outMax) {
-		int ret = 0;
-		int i;
-		if(inDone) *inDone = 0;
-		while(remainder > 0) {
-			size_t blockCount = remainder;
-			if(blockCount > outMax)
-				blockCount = outMax;
-			for(i = 0; i < channelCount; ++i) {
-				vDSP_vdpsp(&OutBufs[i][0], 1, output + i, channelCount, blockCount);
-			}
-			remainder -= blockCount;
-			if(remainder > 0) {
-				for(i = 0; i < channelCount; ++i) {
-					memmove(&OutBufs[i][0], &OutBufs[i][blockCount], remainder * sizeof(double));
-				}
-			}
-			output += channelCount * blockCount;
-			outProcessed += blockCount;
-			outMax -= blockCount;
-			ret += blockCount;
-			if(!outMax)
-				return ret;
-		}
-		while(inCount > 0) {
-			size_t blockCount = inCount;
-			if(blockCount > bufferCapacity)
-				blockCount = bufferCapacity;
-			int outputDone = 0;
-			for(i = 0; i < channelCount; ++i) {
-				double *outputPointer;
-				vDSP_vspdp(input + i, channelCount, &InBuf[0], 1, blockCount);
-				outputDone = Resamps[i]->process(InBuf, (int)blockCount, outputPointer);
-				if(outputDone) {
-					if(outputDone > outMax) {
-						vDSP_vdpsp(outputPointer, 1, output + i, channelCount, outMax);
-						remainder = outputDone - outMax;
-						OutBufs[i].alloc((int)remainder);
-						memcpy(&OutBufs[i][0], outputPointer + outMax, remainder);
-					} else {
-						vDSP_vdpsp(outputPointer, 1, output + i, channelCount, outputDone);
-					}
-				}
-			}
-			size_t outputActual = outputDone - remainder;
-			input += channelCount * blockCount;
-			output += channelCount * outputActual;
-			inCount -= blockCount;
-			if(inDone) *inDone += blockCount;
-			inProcessed += blockCount;
-			outProcessed += outputActual;
-			outMax -= outputActual;
-			ret += outputActual;
-			if(remainder)
-				break;
-		}
-		return ret;
-	}
-
-	int flush(float *output, size_t outMax) {
-		int ret = 0;
-		int i;
-		if(remainder > 0) {
-			size_t blockCount = remainder;
-			if(blockCount > outMax)
-				blockCount = outMax;
-			for(i = 0; i < channelCount; ++i) {
-				vDSP_vdpsp(&OutBufs[i][0], 1, output + i, channelCount, blockCount);
-			}
-			remainder -= blockCount;
-			if(remainder > 0) {
-				for(i = 0; i < channelCount; ++i) {
-					memmove(&OutBufs[i][0], &OutBufs[i][blockCount], remainder * sizeof(double));
-				}
-			}
-			output += channelCount * blockCount;
-			outProcessed += blockCount;
-			outMax -= blockCount;
-			ret += blockCount;
-			if(!outMax)
-				return ret;
-		}
-		uint64_t outputWanted = ceil(inProcessed * sampleRatio);
-		memset(&InBuf[0], 0, sizeof(double) * bufferCapacity);
-		while(outProcessed < outputWanted) {
-			int outputDone = 0;
-			for(int i = 0; i < channelCount; ++i) {
-				double *outputPointer;
-				outputDone = Resamps[i]->process(InBuf, bufferCapacity, outputPointer);
-				if(outputDone) {
-					if(outputDone > (outputWanted - outProcessed))
-						outputDone = (int)(outputWanted - outProcessed);
-					if(outputDone > outMax) {
-						vDSP_vdpsp(outputPointer, 1, output + i, channelCount, outMax);
-						remainder = outputDone - outMax;
-						OutBufs[i].alloc((int)remainder);
-						memcpy(&OutBufs[i][0], outputPointer + outMax, remainder);
-					} else {
-						vDSP_vdpsp(outputPointer, 1, output + i, channelCount, outputDone);
-					}
-				}
-			}
-			size_t outputActual = outputDone - remainder;
-			outProcessed += outputActual;
-			output += channelCount * outputActual;
-			outMax -= outputActual;
-			ret += outputActual;
-			if(remainder)
-				break;
-		}
-		return ret;
-	}
-};
-
-#endif /* r8bstate_h */

Audio/ThirdParty/rsstate.cpp

@@ -0,0 +1,30 @@
+//
+//  rsstate.cpp
+//  CogAudio Framework
+//
+//  Created by Christopher Snowhill on 2/4/23.
+//
+
+#include "rsstate.h"
+#include "rsstate.hpp"
+
+void *rsstate_new(int channelCount, double srcRate, double dstRate) {
+	return (void *)new rsstate(channelCount, srcRate, dstRate);
+}
+
+void rsstate_delete(void *state) {
+	delete(rsstate *)state;
+}
+
+double rsstate_latency(void *state) {
+	return ((rsstate *)state)->latency();
+}
+
+int rsstate_resample(void *state, const float *input, size_t inCount, size_t *inDone,
+                      float *output, size_t outMax) {
+	return ((rsstate *)state)->resample(input, inCount, inDone, output, outMax);
+}
+
+int rsstate_flush(void *state, float *output, size_t outMax) {
+	return ((rsstate *)state)->flush(output, outMax);
+}

Audio/ThirdParty/rsstate.h

@@ -0,0 +1,32 @@
+//
+//  rsstate.h
+//  CogAudio Framework
+//
+//  Created by Christopher Snowhill on 2/4/23.
+//
+
+#include <stdint.h>
+#include <stdlib.h>
+
+#ifndef rsstate_h
+#define rsstate_h
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+void *rsstate_new(int channelCount, double srcRate, double dstRate);
+void rsstate_delete(void *);
+
+double rsstate_latency(void *);
+
+int rsstate_resample(void *, const float *input, size_t inCount, size_t *inDone,
+                      float *output, size_t outMax);
+
+int rsstate_flush(void *, float *output, size_t outMax);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* rsstate_h */

Audio/ThirdParty/rsstate.hpp

@@ -0,0 +1,80 @@
+//
+//  rsstate.hpp
+//  CogAudio Framework
+//
+//  Created by Christopher Snowhill on 2/3/23.
+//
+
+#ifndef rsstate_hpp
+#define rsstate_hpp
+
+#include "soxr.h"
+
+#include <cmath>
+#include <vector>
+
+struct rsstate {
+	int channelCount;
+	int bufferCapacity;
+	size_t remainder;
+	uint64_t inProcessed;
+	uint64_t outProcessed;
+	double sampleRatio;
+	double dstRate;
+	std::vector<float> SilenceBuf;
+	soxr_t Resampler;
+	rsstate(int _channelCount, double srcRate, double _dstRate)
+	: channelCount(_channelCount), inProcessed(0), outProcessed(0), remainder(0), dstRate(_dstRate) {
+		SilenceBuf.resize(1024 * channelCount);
+		memset(&SilenceBuf[0], 0, 1024 * channelCount * sizeof(float));
+		Resampler = soxr_create(srcRate, dstRate, channelCount, NULL, NULL, NULL, NULL);
+		sampleRatio = dstRate / srcRate;
+	}
+
+	~rsstate() {
+		soxr_delete(Resampler);
+	}
+
+	double latency() {
+		return (((double)inProcessed * sampleRatio) - (double)outProcessed) / dstRate;
+	}
+
+	int resample(const float *input, size_t inCount, size_t *inDone, float *output, size_t outMax) {
+		size_t outDone = 0;
+		soxr_error_t errmsg = soxr_process(Resampler, (soxr_in_t)input, inCount, inDone, (soxr_out_t)output, outMax, &outDone);
+		if(!errmsg) {
+			inProcessed += *inDone;
+			outProcessed += outDone;
+			return (int)outDone;
+		} else {
+			return 0;
+		}
+	}
+
+	int flush(float *output, size_t outMax) {
+		size_t outTotal = 0;
+		uint64_t outputWanted = std::ceil(inProcessed * sampleRatio);
+		while(outProcessed < outputWanted) {
+			size_t outWanted = outputWanted - outProcessed;
+			if(outWanted > outMax) {
+				outWanted = outMax;
+			}
+			size_t outDone = 0;
+			size_t inDone = 0;
+			soxr_error_t errmsg = soxr_process(Resampler, (soxr_in_t)(&SilenceBuf[0]), 1024, &inDone, (soxr_out_t)output, outWanted, &outDone);
+			if(!errmsg) {
+				outProcessed += outDone;
+				outTotal += outDone;
+				output += outDone * channelCount;
+				outMax -= outDone;
+				if(!outMax || outProcessed == outputWanted) {
+					return (int)outTotal;
+				}
+			} else {
+				return 0;
+			}
+		}
+	}
+};
+
+#endif /* r8bstate_h */

Cog.xcodeproj/project.pbxproj

@@ -132,6 +132,7 @@
 		835FAC7F27BCDF5B00BA8562 /* libavif.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 835FAC7D27BCDF5B00BA8562 /* libavif.a */; };
 		8360EF6D17F92E56005208A4 /* HighlyComplete.bundle in CopyFiles */ = {isa = PBXBuildFile; fileRef = 8360EF0517F92B24005208A4 /* HighlyComplete.bundle */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
 		836D28A818086386005B7299 /* MiniModeMenuTitleTransformer.m in Sources */ = {isa = PBXBuildFile; fileRef = 836D28A718086386005B7299 /* MiniModeMenuTitleTransformer.m */; };
+		836DF617298F6F1700CD0580 /* libsoxr.0.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836DF616298F6EC400CD0580 /* libsoxr.0.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
 		836EF0C827BB91E600BF35B2 /* libogg.0.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0C427BB919300BF35B2 /* libogg.0.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
 		836EF0C927BB91E900BF35B2 /* libvorbisfile.3.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0C627BB91AB00BF35B2 /* libvorbisfile.3.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
 		836EF0CA27BB91EB00BF35B2 /* libvorbis.0.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0C527BB91AB00BF35B2 /* libvorbis.0.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
@@ -743,6 +744,7 @@
 			dstPath = "";
 			dstSubfolderSpec = 10;
 			files = (
+				836DF617298F6F1700CD0580 /* libsoxr.0.dylib in CopyFiles */,
 				836EF0E127BB98AB00BF35B2 /* libopusfile.0.dylib in CopyFiles */,
 				836EF0E027BB98A800BF35B2 /* libopus.0.dylib in CopyFiles */,
 				836EF0CB27BB91EE00BF35B2 /* libFLAC.12.dylib in CopyFiles */,
@@ -992,6 +994,7 @@
 		8360EF0017F92B23005208A4 /* HighlyComplete.xcodeproj */ = {isa = PBXFileReference; lastKnownFileType = "wrapper.pb-project"; name = HighlyComplete.xcodeproj; path = Plugins/HighlyComplete/HighlyComplete.xcodeproj; sourceTree = "<group>"; };
 		836D28A618086386005B7299 /* MiniModeMenuTitleTransformer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = MiniModeMenuTitleTransformer.h; path = Window/MiniModeMenuTitleTransformer.h; sourceTree = "<group>"; };
 		836D28A718086386005B7299 /* MiniModeMenuTitleTransformer.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; name = MiniModeMenuTitleTransformer.m; path = Window/MiniModeMenuTitleTransformer.m; sourceTree = "<group>"; };
+		836DF616298F6EC400CD0580 /* libsoxr.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libsoxr.0.dylib; path = ThirdParty/soxr/lib/libsoxr.0.dylib; sourceTree = "<group>"; };
 		836EF0C427BB919300BF35B2 /* libogg.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libogg.0.dylib; path = ThirdParty/ogg/lib/libogg.0.dylib; sourceTree = "<group>"; };
 		836EF0C527BB91AB00BF35B2 /* libvorbis.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libvorbis.0.dylib; path = ThirdParty/vorbis/lib/libvorbis.0.dylib; sourceTree = "<group>"; };
 		836EF0C627BB91AB00BF35B2 /* libvorbisfile.3.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libvorbisfile.3.dylib; path = ThirdParty/vorbis/lib/libvorbisfile.3.dylib; sourceTree = "<group>"; };
@@ -1211,6 +1214,7 @@
 		1058C7A2FEA54F0111CA2CBB /* Other Frameworks */ = {
 			isa = PBXGroup;
 			children = (
+				836DF616298F6EC400CD0580 /* libsoxr.0.dylib */,
 				83256B672866617F0036D9C0 /* libmpg123.0.dylib */,
 				835FAC7C27BCDF5B00BA8562 /* libaom.a */,
 				835FAC7D27BCDF5B00BA8562 /* libavif.a */,
@@ -3095,6 +3099,7 @@
 					"$(inherited)",
 					"$(PROJECT_DIR)/ThirdParty/avif/lib",
 					"$(PROJECT_DIR)/ThirdParty/mpg123/lib",
+					"$(PROJECT_DIR)/ThirdParty/soxr/lib",
 				);
 				MACOSX_DEPLOYMENT_TARGET = 10.13;
 				OTHER_CFLAGS = (
@@ -3151,6 +3156,7 @@
 					"$(inherited)",
 					"$(PROJECT_DIR)/ThirdParty/avif/lib",
 					"$(PROJECT_DIR)/ThirdParty/mpg123/lib",
+					"$(PROJECT_DIR)/ThirdParty/soxr/lib",
 				);
 				MACOSX_DEPLOYMENT_TARGET = 10.13;
 				OTHER_CFLAGS = (

ThirdParty/soxr/README.md

@@ -0,0 +1,35 @@
+Build with CMake, using the following options for x86_64:
+
+```
+arch -x86_64 (x86_64 version of cmake) .. -DCMAKE_OSX_ARCHITECTURES="x86_64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.13" -DBUILD_SHARED_LIBS=ON -DCMAKE_BUILD_TYPE=Release|Debug
+```
+
+And these for ARM64:
+
+```
+cmake .. -DCMAKE_OSX_ARCHITECTURES="arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="11.0" -DBUILD_SHARED_LIBS=ON -DCMAKE_BUILD_TYPE=Release|Debug
+```
+
+(Release|Debug) means one or the other for the separate release/debug builds
+
+And some minor tweaks with `install_name_tool -id` to make sure that the
+resulting libsoxr.0.dylib imported libavutil properly, as it seems to want
+to use that for CPU detection on ARM, even though the only feature being
+detected is NEON, which is mandatory on the supported Apple Silicon platforms.
+
+Version 0.1.3 was retrieved from:
+
+https://downloads.sourceforge.net/project/soxr/soxr-0.1.3-Source.tar.xz
+
+With the following sha256 hash:
+
+b111c15fdc8c029989330ff559184198c161100a59312f5dc19ddeb9b5a15889
+
+
+And a patch from Homebrew was applied to fix ARM64 building:
+
+https://raw.githubusercontent.com/Homebrew/formula-patches/76868b36263be42440501d3692fd3a258f507d82/libsoxr/arm64_defines.patch
+
+With the following sha256 hash:
+
+9df5737a21b9ce70cc136c302e195fad9f9f6c14418566ad021f14bb34bb022c

ThirdParty/soxr/include/soxr.h

@@ -0,0 +1,344 @@
+/* SoX Resampler Library      Copyright (c) 2007-18 robs@users.sourceforge.net
+ *
+ * This library is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 2.1 of the License, or (at
+ * your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU Lesser
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this library; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
+ */
+
+
+
+/* -------------------------------- Gubbins --------------------------------- */
+
+#if !defined soxr_included
+#define soxr_included
+
+
+#if defined __cplusplus
+  #include <cstddef>
+  extern "C" {
+#else
+  #include <stddef.h>
+#endif
+
+#if defined SOXR_DLL
+  #if defined soxr_EXPORTS
+    #define SOXR __declspec(dllexport)
+  #else
+    #define SOXR __declspec(dllimport)
+  #endif
+#elif defined SOXR_VISIBILITY && defined __GNUC__ && (__GNUC__ > 4 || __GNUC__ == 4 && __GNUC_MINOR__ >= 1)
+  #define SOXR __attribute__ ((visibility("default")))
+#else
+  #define SOXR
+#endif
+
+typedef struct soxr_io_spec soxr_io_spec_t;
+typedef struct soxr_quality_spec soxr_quality_spec_t;
+typedef struct soxr_runtime_spec soxr_runtime_spec_t;
+
+
+
+/* ---------------------------- API conventions --------------------------------
+
+Buffer lengths (and occupancies) are expressed as the number of contained
+samples per channel.
+
+Parameter names for buffer lengths have the suffix `len'.
+
+A single-character `i' or 'o' is often used in names to give context as
+input or output (e.g. ilen, olen).                                            */
+
+
+
+/* --------------------------- Version management --------------------------- */
+
+/* E.g. #if SOXR_THIS_VERSION >= SOXR_VERSION(0,1,1) ...                      */
+
+#define SOXR_VERSION(x,y,z)     (((x)<<16)|((y)<<8)|(z))
+#define SOXR_THIS_VERSION       SOXR_VERSION(0,1,3)
+#define SOXR_THIS_VERSION_STR               "0.1.3"
+
+
+
+/* --------------------------- Type declarations ---------------------------- */
+
+typedef struct soxr * soxr_t;          /* A resampler for 1 or more channels. */
+typedef char const * soxr_error_t;                /* 0:no-error; non-0:error. */
+
+typedef void       * soxr_buf_t;  /* 1 buffer of channel-interleaved samples. */
+typedef void const * soxr_cbuf_t;                        /* Ditto; read-only. */
+
+typedef soxr_buf_t const  * soxr_bufs_t;/* Or, a separate buffer for each ch. */
+typedef soxr_cbuf_t const * soxr_cbufs_t;                /* Ditto; read-only. */
+
+typedef void const * soxr_in_t;      /* Either a soxr_cbuf_t or soxr_cbufs_t,
+                                        depending on itype in soxr_io_spec_t. */
+typedef void       * soxr_out_t;     /* Either a soxr_buf_t or soxr_bufs_t,
+                                        depending on otype in soxr_io_spec_t. */
+
+
+
+/* --------------------------- API main functions --------------------------- */
+
+SOXR char const * soxr_version(void);  /* Query library version: "libsoxr-x.y.z" */
+
+#define soxr_strerror(e)               /* Soxr counterpart to strerror. */     \
+    ((e)?(e):"no error")
+
+
+/* Create a stream resampler: */
+
+SOXR soxr_t soxr_create(
+    double      input_rate,      /* Input sample-rate. */
+    double      output_rate,     /* Output sample-rate. */
+    unsigned    num_channels,    /* Number of channels to be used. */
+        /* All following arguments are optional (may be set to NULL). */
+    soxr_error_t *,              /* To report any error during creation. */
+    soxr_io_spec_t const *,      /* To specify non-default I/O formats. */
+    soxr_quality_spec_t const *, /* To specify non-default resampling quality.*/
+    soxr_runtime_spec_t const *);/* To specify non-default runtime resources.
+
+    Default io_spec      is per soxr_io_spec(SOXR_FLOAT32_I, SOXR_FLOAT32_I)
+    Default quality_spec is per soxr_quality_spec(SOXR_HQ, 0)
+    Default runtime_spec is per soxr_runtime_spec(1)                          */
+
+
+
+/* If not using an app-supplied input function, after creating a stream
+ * resampler, repeatedly call: */
+
+SOXR soxr_error_t soxr_process(
+    soxr_t      resampler,      /* As returned by soxr_create. */
+                            /* Input (to be resampled): */
+    soxr_in_t   in,             /* Input buffer(s); may be NULL (see below). */
+    size_t      ilen,           /* Input buf. length (samples per channel). */
+    size_t      * idone,        /* To return actual # samples used (<= ilen). */
+                            /* Output (resampled): */
+    soxr_out_t  out,            /* Output buffer(s).*/
+    size_t      olen,           /* Output buf. length (samples per channel). */
+    size_t      * odone);       /* To return actual # samples out (<= olen).
+
+    Note that no special meaning is associated with ilen or olen equal to
+    zero.  End-of-input (i.e. no data is available nor shall be available)
+    may be indicated by seting `in' to NULL.                                  */
+
+
+
+/* If using an app-supplied input function, it must look and behave like this:*/
+
+typedef size_t /* data_len */
+  (* soxr_input_fn_t)(         /* Supply data to be resampled. */
+    void * input_fn_state,     /* As given to soxr_set_input_fn (below). */
+    soxr_in_t * data,          /* Returned data; see below. N.B. ptr to ptr(s)*/
+    size_t requested_len);     /* Samples per channel, >= returned data_len.
+
+  data_len  *data     Indicates    Meaning
+   ------- -------   ------------  -------------------------
+     !=0     !=0       Success     *data contains data to be
+                                   input to the resampler.
+      0    !=0 (or   End-of-input  No data is available nor
+           not set)                shall be available.
+      0       0        Failure     An error occurred whilst trying to
+                                   source data to be input to the resampler.  */
+
+/* and be registered with a previously created stream resampler using: */
+
+SOXR soxr_error_t soxr_set_input_fn(/* Set (or reset) an input function.*/
+    soxr_t resampler,            /* As returned by soxr_create. */
+    soxr_input_fn_t,             /* Function to supply data to be resampled.*/
+    void * input_fn_state,       /* If needed by the input function. */
+    size_t max_ilen);            /* Maximum value for input fn. requested_len.*/
+
+/* then repeatedly call: */
+
+SOXR size_t /*odone*/ soxr_output(/* Resample and output a block of data.*/
+    soxr_t resampler,            /* As returned by soxr_create. */
+    soxr_out_t data,             /* App-supplied buffer(s) for resampled data.*/
+    size_t olen);                /* Amount of data to output; >= odone. */
+
+
+
+/* Common stream resampler operations: */
+
+SOXR soxr_error_t soxr_error(soxr_t);   /* Query error status. */
+SOXR size_t   * soxr_num_clips(soxr_t); /* Query int. clip counter (for R/W). */
+SOXR double     soxr_delay(soxr_t);  /* Query current delay in output samples.*/
+SOXR char const * soxr_engine(soxr_t);  /* Query resampling engine name. */
+
+SOXR soxr_error_t soxr_clear(soxr_t); /* Ready for fresh signal, same config. */
+SOXR void         soxr_delete(soxr_t);  /* Free resources. */
+
+
+
+/* `Short-cut', single call to resample a (probably short) signal held entirely
+ * in memory.  See soxr_create and soxr_process above for parameter details.
+ * Note that unlike soxr_create however, the default quality spec. for
+ * soxr_oneshot is per soxr_quality_spec(SOXR_LQ, 0). */
+
+SOXR soxr_error_t soxr_oneshot(
+    double         input_rate,
+    double         output_rate,
+    unsigned       num_channels,
+    soxr_in_t    in , size_t ilen, size_t * idone,
+    soxr_out_t   out, size_t olen, size_t * odone,
+    soxr_io_spec_t const *,
+    soxr_quality_spec_t const *,
+    soxr_runtime_spec_t const *);
+
+
+
+/* For variable-rate resampling. See example # 5 for how to create a
+ * variable-rate resampler and how to use this function. */
+
+SOXR soxr_error_t soxr_set_io_ratio(soxr_t, double io_ratio, size_t slew_len);
+
+
+
+/* -------------------------- API type definitions -------------------------- */
+
+typedef enum {          /* Datatypes supported for I/O to/from the resampler: */
+  /* Internal; do not use: */
+  SOXR_FLOAT32, SOXR_FLOAT64, SOXR_INT32, SOXR_INT16, SOXR_SPLIT = 4,
+
+  /* Use for interleaved channels: */
+  SOXR_FLOAT32_I = SOXR_FLOAT32, SOXR_FLOAT64_I, SOXR_INT32_I, SOXR_INT16_I,
+
+  /* Use for split channels: */
+  SOXR_FLOAT32_S = SOXR_SPLIT  , SOXR_FLOAT64_S, SOXR_INT32_S, SOXR_INT16_S
+
+} soxr_datatype_t;
+
+#define soxr_datatype_size(x)  /* Returns `sizeof' a soxr_datatype_t sample. */\
+  ((unsigned char *)"\4\10\4\2")[(x)&3]
+
+
+
+struct soxr_io_spec {                                            /* Typically */
+  soxr_datatype_t itype;     /* Input datatype.                SOXR_FLOAT32_I */
+  soxr_datatype_t otype;     /* Output datatype.               SOXR_FLOAT32_I */
+  double scale;              /* Linear gain to apply during resampling.  1    */
+  void * e;                  /* Reserved for internal use                0    */
+  unsigned long flags;       /* Per the following #defines.              0    */
+};
+
+#define SOXR_TPDF              0     /* Applicable only if otype is INT16. */
+#define SOXR_NO_DITHER         8u    /* Disable the above. */
+
+
+
+struct soxr_quality_spec {                                       /* Typically */
+  double precision;         /* Conversion precision (in bits).           20   */
+  double phase_response;    /* 0=minimum, ... 50=linear, ... 100=maximum 50   */
+  double passband_end;      /* 0dB pt. bandwidth to preserve; nyquist=1  0.913*/
+  double stopband_begin;    /* Aliasing/imaging control; > passband_end   1   */
+  void * e;                 /* Reserved for internal use.                 0   */
+  unsigned long flags;      /* Per the following #defines.                0   */
+};
+
+#define SOXR_ROLLOFF_SMALL     0u    /* <= 0.01 dB */
+#define SOXR_ROLLOFF_MEDIUM    1u    /* <= 0.35 dB */
+#define SOXR_ROLLOFF_NONE      2u    /* For Chebyshev bandwidth. */
+
+#define SOXR_HI_PREC_CLOCK     8u  /* Increase `irrational' ratio accuracy. */
+#define SOXR_DOUBLE_PRECISION 16u  /* Use D.P. calcs even if precision <= 20. */
+#define SOXR_VR               32u  /* Variable-rate resampling. */
+
+
+
+struct soxr_runtime_spec {                                       /* Typically */
+  unsigned log2_min_dft_size;   /* For DFT efficiency. [8,15]           10    */
+  unsigned log2_large_dft_size; /* For DFT efficiency. [8,20]           17    */
+  unsigned coef_size_kbytes;    /* For SOXR_COEF_INTERP_AUTO (below).   400   */
+  unsigned num_threads;         /* 0: per OMP_NUM_THREADS; 1: 1 thread.  1    */
+  void * e;                     /* Reserved for internal use.            0    */
+  unsigned long flags;          /* Per the following #defines.           0    */
+};
+                                   /* For `irrational' ratios only: */
+#define SOXR_COEF_INTERP_AUTO  0u    /* Auto select coef. interpolation. */
+#define SOXR_COEF_INTERP_LOW   2u    /* Man. select: less CPU, more memory. */
+#define SOXR_COEF_INTERP_HIGH  3u    /* Man. select: more CPU, less memory. */
+
+
+
+/* -------------------------- API type constructors ------------------------- */
+
+/* These functions allow setting of the most commonly-used structure
+ * parameters, with other parameters being given default values.  The default
+ * values may then be overridden, directly in the structure, if needed.  */
+
+SOXR soxr_quality_spec_t soxr_quality_spec(
+    unsigned long recipe,       /* Per the #defines immediately below. */
+    unsigned long flags);       /* As soxr_quality_spec_t.flags. */
+
+                                  /* The 5 standard qualities found in SoX: */
+#define SOXR_QQ                 0   /* 'Quick' cubic interpolation. */
+#define SOXR_LQ                 1   /* 'Low' 16-bit with larger rolloff. */
+#define SOXR_MQ                 2   /* 'Medium' 16-bit with medium rolloff. */
+#define SOXR_HQ                 SOXR_20_BITQ /* 'High quality'. */
+#define SOXR_VHQ                SOXR_28_BITQ /* 'Very high quality'. */
+
+#define SOXR_16_BITQ            3
+#define SOXR_20_BITQ            4
+#define SOXR_24_BITQ            5
+#define SOXR_28_BITQ            6
+#define SOXR_32_BITQ            7
+                                /* Reserved for internal use (to be removed): */
+#define SOXR_LSR0Q              8     /* 'Best sinc'. */
+#define SOXR_LSR1Q              9     /* 'Medium sinc'. */
+#define SOXR_LSR2Q              10    /* 'Fast sinc'. */
+
+#define SOXR_LINEAR_PHASE       0x00
+#define SOXR_INTERMEDIATE_PHASE 0x10
+#define SOXR_MINIMUM_PHASE      0x30
+
+#define SOXR_STEEP_FILTER       0x40
+
+
+
+SOXR soxr_runtime_spec_t soxr_runtime_spec(
+    unsigned num_threads);
+
+
+
+SOXR soxr_io_spec_t soxr_io_spec(
+    soxr_datatype_t itype,
+    soxr_datatype_t otype);
+
+
+
+/* --------------------------- Advanced use only ---------------------------- */
+
+/* For new designs, the following functions/usage will probably not be needed.
+ * They might be useful when adding soxr into an existing design where values
+ * for the resampling-rate and/or number-of-channels parameters to soxr_create
+ * are not available when that function will be called.  In such cases, the
+ * relevant soxr_create parameter(s) can be given as 0, then one or both of the
+ * following (as appropriate) later invoked (but prior to calling soxr_process
+ * or soxr_output):
+ *
+ * soxr_set_error(soxr, soxr_set_io_ratio(soxr, io_ratio, 0));
+ * soxr_set_error(soxr, soxr_set_num_channels(soxr, num_channels));
+ */
+
+SOXR soxr_error_t soxr_set_error(soxr_t, soxr_error_t);
+SOXR soxr_error_t soxr_set_num_channels(soxr_t, unsigned);
+
+
+
+#undef SOXR
+
+#if defined __cplusplus
+}
+#endif
+
+#endif

en.lproj/Credits.html

@@ -35,7 +35,6 @@
         <br><br>
         <em>This program has been made possible through contributions from users like you.</em>
         <br><br> All Cog code is copyrighted by me, and is licensed under the <a href="http://www.gnu.org/licenses/old-licenses/gpl-2.0.txt">GPL</a>. Cog contains bits of other code from third parties that are under their own licenses.
-        <br><br> Sample rate converter designed by Aleksey Vaneev of Voxengo.
     </div>
 </body>
 

es.lproj/Credits.html

@@ -36,7 +36,6 @@
         <br><br>
         <em>Este programa ha sido posible gracias a las contribuciones de usuarios y usuarias como tú.</em>
         <br><br> Todo el código de Cog es de mi propiedad, y está licenciado bajo la <a href="http://www.gnu.org/licenses/old-licenses/gpl-2.0.txt">GPL</a>. Cog contiene código de terceros que están bajo sus propias licencias.
-        <br><br> Convertidor de frecuencia de muestreo diseñado por Aleksey Vaneev de Voxengo.
     </div>
 </body>
 

pl.lproj/Credits.html

@@ -35,7 +35,6 @@
         <br><br>
         <em>This program has been made possible through contributions from users like you.</em>
         <br><br> All Cog code is copyrighted by me, and is licensed under the <a href="http://www.gnu.org/licenses/old-licenses/gpl-2.0.txt">GPL</a>. Cog contains bits of other code from third parties that are under their own licenses.
-        <br><br> Sample rate converter designed by Aleksey Vaneev of Voxengo.
     </div>
 </body>
 

ru.lproj/Credits.html

@@ -35,7 +35,6 @@
         <br><br>
         <em>Создние этой программы стало возможным благодаря вкладу таких пользователей, как вы.</em>
         <br><br> Весь код Cog защищен моим авторским правом и распространяется под лицензией <a href="http://www.gnu.org/licenses/old-licenses/gpl-2.0.txt">GPL</a>. Cog содержит фрагменты другого кода от третьих сторон, который находятся под их собственными лицензиями.
-        <br><br> Преобразователь частоты дискретизации, разработан Aleksey Vaneev из Voxengo.
     </div>
 </body>