diff --git a/AssetStudioCLI/AssetStudioCLI.csproj b/AssetStudioCLI/AssetStudioCLI.csproj
index 5f4f735..d8d274c 100644
--- a/AssetStudioCLI/AssetStudioCLI.csproj
+++ b/AssetStudioCLI/AssetStudioCLI.csproj
@@ -39,8 +39,11 @@
         <Copy SourceFiles="$(ProjectDir)Libraries\osx-arm64\libAssetStudioFBXNative.dylib" DestinationFolder="$(TargetDir)runtimes\osx-arm64\native" ContinueOnError="false" />
         <Copy SourceFiles="$(ProjectDir)Libraries\linux-x86\libfmod.so" DestinationFolder="$(TargetDir)runtimes\linux-x86\native" ContinueOnError="false" />
         <Copy SourceFiles="$(ProjectDir)Libraries\linux-x64\libfmod.so" DestinationFolder="$(TargetDir)runtimes\linux-x64\native" ContinueOnError="false" />
+        <Copy SourceFiles="$(ProjectDir)Libraries\linux-arm64\libfmod.so" DestinationFolder="$(TargetDir)runtimes\linux-arm64\native" ContinueOnError="false" />
         <Copy SourceFiles="$(ProjectDir)Libraries\osx-x64\libfmod.dylib" DestinationFolder="$(TargetDir)runtimes\osx-x64\native" ContinueOnError="false" />
         <Copy SourceFiles="$(ProjectDir)Libraries\osx-arm64\libfmod.dylib" DestinationFolder="$(TargetDir)runtimes\osx-arm64\native" ContinueOnError="false" />
+        <Copy SourceFiles="$(ProjectDir)Libraries\win-arm64\fmod.dll" DestinationFolder="$(TargetDir)runtimes\win-arm64\native" ContinueOnError="false" />
+        <Copy SourceFiles="$(ProjectDir)Libraries\linux-arm64\libTexture2DDecoderNative.so" DestinationFolder="$(TargetDir)runtimes\linux-arm64\native" ContinueOnError="false" />
     </Target>
 
     <!-- Publishing an app as framework-dependent produces a cross-platform binary as a dll file, and a platform-specific executable that targets your current platform.
@@ -62,8 +65,11 @@
         <Copy SourceFiles="$(TargetDir)runtimes\osx-arm64\native\libAssetStudioFBXNative.dylib" DestinationFolder="$(PublishDir)runtimes\osx-arm64\native" ContinueOnError="false" />
         <Copy SourceFiles="$(TargetDir)runtimes\linux-x86\native\libfmod.so" DestinationFolder="$(PublishDir)runtimes\linux-x86\native" ContinueOnError="false" />
         <Copy SourceFiles="$(TargetDir)runtimes\linux-x64\native\libfmod.so" DestinationFolder="$(PublishDir)runtimes\linux-x64\native" ContinueOnError="false" />
+        <Copy SourceFiles="$(TargetDir)runtimes\linux-arm64\native\libfmod.so" DestinationFolder="$(PublishDir)runtimes\linux-arm64\native" ContinueOnError="false" />
         <Copy SourceFiles="$(TargetDir)runtimes\osx-x64\native\libfmod.dylib" DestinationFolder="$(PublishDir)runtimes\osx-x64\native" ContinueOnError="false" />
         <Copy SourceFiles="$(TargetDir)runtimes\osx-arm64\native\libfmod.dylib" DestinationFolder="$(PublishDir)runtimes\osx-arm64\native" ContinueOnError="false" />
+        <Copy SourceFiles="$(TargetDir)runtimes\win-arm64\native\fmod.dll" DestinationFolder="$(PublishDir)runtimes\win-arm64\native" ContinueOnError="false" />
+        <Copy SourceFiles="$(TargetDir)runtimes\linux-arm64\native\libTexture2DDecoderNative.so" DestinationFolder="$(PublishDir)runtimes\linux-arm64\native" ContinueOnError="false" />
     </Target>
 
     <Target Name="CopyExtraFilesWin86" AfterTargets="AfterBuild" Condition=" '$(RuntimeIdentifier)' == 'win-x86' ">
@@ -87,6 +93,16 @@
         <Copy SourceFiles="$(TargetDir)\fmod.dll" DestinationFolder="$(PublishDir)" ContinueOnError="false" />
     </Target>
 
+    <Target Name="CopyExtraFilesWinArm64" AfterTargets="AfterBuild" Condition=" '$(RuntimeIdentifier)' == 'win-arm64' ">
+        <Message Text="Copying extra files for $(RuntimeIdentifier)($(TargetFramework))... " Importance="high" />
+        <Copy SourceFiles="$(ProjectDir)Libraries\win-arm64\fmod.dll" DestinationFolder="$(TargetDir)" ContinueOnError="false" />
+    </Target>
+
+    <Target Name="PublishExtraFilesWinArm64" AfterTargets="Publish" Condition=" $(RuntimeIdentifier.Contains('win-arm64')) ">
+        <Message Text="Publishing extra files for $(RuntimeIdentifier)($(TargetFramework))... " Importance="high" />
+        <Copy SourceFiles="$(TargetDir)\fmod.dll" DestinationFolder="$(PublishDir)" ContinueOnError="false" />
+    </Target>
+
     <Target Name="CopyExtraFilesLinux64" AfterTargets="AfterBuild" Condition=" '$(RuntimeIdentifier)' == 'linux-x64' ">
         <Message Text="Copying extra files for $(RuntimeIdentifier)($(TargetFramework))... " Importance="high" />
         <Copy SourceFiles="$(ProjectDir)Libraries\linux-x64\libAssetStudioFBXNative.so" DestinationFolder="$(TargetDir)" ContinueOnError="false" />
@@ -99,6 +115,18 @@
         <Copy SourceFiles="$(TargetDir)\libfmod.so" DestinationFolder="$(PublishDir)" ContinueOnError="false" />
     </Target>
 
+    <Target Name="CopyExtraFilesLinuxArm64" AfterTargets="AfterBuild" Condition=" '$(RuntimeIdentifier)' == 'linux-arm64' ">
+        <Message Text="Copying extra files for $(RuntimeIdentifier)($(TargetFramework))... " Importance="high" />
+        <Copy SourceFiles="$(ProjectDir)Libraries\linux-arm64\libTexture2DDecoderNative.so" DestinationFolder="$(TargetDir)" ContinueOnError="false" />
+        <Copy SourceFiles="$(ProjectDir)Libraries\linux-arm64\libfmod.so" DestinationFolder="$(TargetDir)" ContinueOnError="false" />
+    </Target>
+
+    <Target Name="PublishExtraFilesLinuxArm64" AfterTargets="Publish" Condition=" '$(RuntimeIdentifier)' == 'linux-arm64' ">
+        <Message Text="Publishing extra files for $(RuntimeIdentifier)($(TargetFramework))... " Importance="high" />
+        <Copy SourceFiles="$(TargetDir)\libTexture2DDecoderNative.so" DestinationFolder="$(PublishDir)" ContinueOnError="false" />
+        <Copy SourceFiles="$(TargetDir)\libfmod.so" DestinationFolder="$(PublishDir)" ContinueOnError="false" />
+    </Target>
+
     <Target Name="CopyExtraFilesMac" AfterTargets="AfterBuild" Condition=" $(RuntimeIdentifier.Contains('osx-')) ">
         <Message Text="Copying extra files for $(RuntimeIdentifier)($(TargetFramework))... " Importance="high" />
         <Copy SourceFiles="$(ProjectDir)Libraries\$(RuntimeIdentifier)\libAssetStudioFBXNative.dylib" DestinationFolder="$(TargetDir)" ContinueOnError="false" />
diff --git a/AssetStudioCLI/Libraries/linux-arm64/libTexture2DDecoderNative.so b/AssetStudioCLI/Libraries/linux-arm64/libTexture2DDecoderNative.so
new file mode 100644
index 0000000..40092ae
Binary files /dev/null and b/AssetStudioCLI/Libraries/linux-arm64/libTexture2DDecoderNative.so differ
diff --git a/AssetStudioCLI/Libraries/linux-arm64/libfmod.so b/AssetStudioCLI/Libraries/linux-arm64/libfmod.so
new file mode 100644
index 0000000..47a87fb
Binary files /dev/null and b/AssetStudioCLI/Libraries/linux-arm64/libfmod.so differ
diff --git a/AssetStudioCLI/Libraries/linux-x64/libfmod.so b/AssetStudioCLI/Libraries/linux-x64/libfmod.so
index b1f09cf..60ddc2e 100644
Binary files a/AssetStudioCLI/Libraries/linux-x64/libfmod.so and b/AssetStudioCLI/Libraries/linux-x64/libfmod.so differ
diff --git a/AssetStudioCLI/Libraries/linux-x86/libfmod.so b/AssetStudioCLI/Libraries/linux-x86/libfmod.so
index 6f2e31d..86a1859 100644
Binary files a/AssetStudioCLI/Libraries/linux-x86/libfmod.so and b/AssetStudioCLI/Libraries/linux-x86/libfmod.so differ
diff --git a/AssetStudioCLI/Libraries/osx-arm64/libfmod.dylib b/AssetStudioCLI/Libraries/osx-arm64/libfmod.dylib
index a6476eb..eba525f 100644
Binary files a/AssetStudioCLI/Libraries/osx-arm64/libfmod.dylib and b/AssetStudioCLI/Libraries/osx-arm64/libfmod.dylib differ
diff --git a/AssetStudioCLI/Libraries/osx-x64/libfmod.dylib b/AssetStudioCLI/Libraries/osx-x64/libfmod.dylib
index d605afc..9665d88 100644
Binary files a/AssetStudioCLI/Libraries/osx-x64/libfmod.dylib and b/AssetStudioCLI/Libraries/osx-x64/libfmod.dylib differ
diff --git a/AssetStudioCLI/Libraries/win-arm64/fmod.dll b/AssetStudioCLI/Libraries/win-arm64/fmod.dll
new file mode 100644
index 0000000..153fdbb
Binary files /dev/null and b/AssetStudioCLI/Libraries/win-arm64/fmod.dll differ
diff --git a/AssetStudioCLI/Libraries/win-x64/fmod.dll b/AssetStudioCLI/Libraries/win-x64/fmod.dll
index 22d5d7b..772b35f 100644
Binary files a/AssetStudioCLI/Libraries/win-x64/fmod.dll and b/AssetStudioCLI/Libraries/win-x64/fmod.dll differ
diff --git a/AssetStudioCLI/Libraries/win-x86/fmod.dll b/AssetStudioCLI/Libraries/win-x86/fmod.dll
index b19e554..3a9b0f9 100644
Binary files a/AssetStudioCLI/Libraries/win-x86/fmod.dll and b/AssetStudioCLI/Libraries/win-x86/fmod.dll differ
diff --git a/AssetStudioCLI/ParallelExporter.cs b/AssetStudioCLI/ParallelExporter.cs
index 45af2b4..348c1d7 100644
--- a/AssetStudioCLI/ParallelExporter.cs
+++ b/AssetStudioCLI/ParallelExporter.cs
@@ -96,7 +96,7 @@ namespace AssetStudioCLI
 
         public static bool ExportAudioClip(AssetItem item, string exportPath, out string debugLog)
         {
-            debugLog = "";
+            debugLog = string.Empty;
             string exportFullPath;
             var m_AudioClip = (AudioClip)item.Asset;
             var m_AudioData = BigArrayPool<byte>.Shared.Rent(m_AudioClip.m_AudioData.Size);
@@ -120,11 +120,9 @@ namespace AssetStudioCLI
                         debugLog += GenerateAudioClipInfo(m_AudioClip);
                     }
 
-                    var debugLogConverter = "";
                     var buffer = converter.IsLegacy
-                        ? converter.RawAudioClipToWav(out debugLogConverter)
-                        : converter.ConvertToWav(m_AudioData, out debugLogConverter);
-                    debugLog += debugLogConverter;
+                        ? converter.RawAudioClipToWav(ref debugLog)
+                        : converter.ConvertToWav(m_AudioData, ref debugLog);
                     if (buffer == null)
                     {
                         Logger.Error($"{debugLog}Export error. \"{item.Text}\": Failed to convert fmod audio to Wav");
diff --git a/AssetStudioGUI/AssetStudioGUIForm.cs b/AssetStudioGUI/AssetStudioGUIForm.cs
index 1098eed..9033cda 100644
--- a/AssetStudioGUI/AssetStudioGUIForm.cs
+++ b/AssetStudioGUI/AssetStudioGUIForm.cs
@@ -1133,7 +1133,7 @@ namespace AssetStudioGUI
                         bits;
                 }
                 var errorMsg = result == FMOD.RESULT.ERR_VERSION
-                    ? "Unsupported fmod version. Try to export raw and convert with an external tool instead."
+                    ? "Unsupported version of fmod sound. Try to export raw and convert with an external tool instead."
                     : $"Preview not available, try to export instead. {FMOD.Error.String(result)}";
                 StatusStripUpdate(errorMsg);
                 FMODreset();
@@ -2465,12 +2465,16 @@ namespace AssetStudioGUI
             {
                 //skip
             }
+#pragma warning restore WFO5001
+#endif
             if (isDarkMode)
             {
                 assetListView.GridLines = false;
             }
-#pragma warning restore WFO5001
-#endif
+            else
+            {
+                FMODloopButton.UseVisualStyleBackColor = true;
+            }
         }
 
         private void colorThemeAutoToolStripMenuItem_Click(object sender, EventArgs e)
@@ -2588,10 +2592,8 @@ namespace AssetStudioGUI
             result = system.init(2, FMOD.INITFLAGS.NORMAL, IntPtr.Zero);
             if (ERRCHECK(result)) { return; }
 
-            result = system.getMasterSoundGroup(out masterSoundGroup);
-            if (ERRCHECK(result)) { return; }
-
-            result = masterSoundGroup.setVolume(FMODVolume);
+            _ = system.getMasterChannelGroup(out var channelGroup);
+            result = channelGroup.setVolume(FMODVolume);
             if (ERRCHECK(result)) { return; }
         }
 
@@ -2607,9 +2609,17 @@ namespace AssetStudioGUI
 
             if (sound.hasHandle())
             {
-                var result = sound.release();
+                FMOD.RESULT result;
+                sound.getSubSoundParent(out var parentsound);
+                result = sound.release();
                 ERRCHECK(result);
                 sound.clearHandle();
+                if (parentsound.hasHandle())
+                {
+                    result = parentsound.release();
+                    ERRCHECK(result);
+                    parentsound.clearHandle();
+                }
             }
             if (soundBuff != null)
             {
@@ -2655,7 +2665,6 @@ namespace AssetStudioGUI
                         {
                             if (ERRCHECK(result)) { return; }
                         }
-
                     }
                 }
             }
@@ -2748,16 +2757,20 @@ namespace AssetStudioGUI
                 if (playing || paused)
                 {
                     result = channel.setMode(loopMode);
-                    if (ERRCHECK(result)) { return; }
+                    if ((result != FMOD.RESULT.OK) && (result != FMOD.RESULT.ERR_INVALID_HANDLE))
+                    {
+                        if (ERRCHECK(result)) { return; }
+                    }
                 }
             }
         }
 
         private void FMODvolumeBar_ValueChanged(object sender, EventArgs e)
         {
-            FMODVolume = Convert.ToSingle(FMODvolumeBar.Value) / 10;
+            FMODVolume = FMODvolumeBar.Value / 10f;
 
-            var result = masterSoundGroup.setVolume(FMODVolume);
+            _ = system.getMasterChannelGroup(out var channelGroup);
+            var result = channelGroup.setVolume(FMODVolume);
             if (ERRCHECK(result)) { return; }
         }
 
@@ -2787,7 +2800,6 @@ namespace AssetStudioGUI
                     if (ERRCHECK(result)) { return; }
                 }
 
-
                 result = channel.isPlaying(out var playing);
                 if ((result != FMOD.RESULT.OK) && (result != FMOD.RESULT.ERR_INVALID_HANDLE))
                 {
@@ -2823,6 +2835,12 @@ namespace AssetStudioGUI
                 {
                     ERRCHECK(result);
                 }
+
+                if (!playing)
+                {
+                    timer.Stop();
+                    ms = 0;
+                }
             }
 
             FMODtimerLabel.Text = $"{ms / 1000 / 60:00}:{ms / 1000 % 60:00}.{ms / 10 % 100:00} / {FMODlenms / 1000 / 60:00}:{FMODlenms / 1000 % 60:00}.{FMODlenms / 10 % 100:00}";
diff --git a/AssetStudioGUI/Libraries/x64/fmod.dll b/AssetStudioGUI/Libraries/x64/fmod.dll
index 22d5d7b..772b35f 100644
Binary files a/AssetStudioGUI/Libraries/x64/fmod.dll and b/AssetStudioGUI/Libraries/x64/fmod.dll differ
diff --git a/AssetStudioGUI/Libraries/x86/fmod.dll b/AssetStudioGUI/Libraries/x86/fmod.dll
index b19e554..3a9b0f9 100644
Binary files a/AssetStudioGUI/Libraries/x86/fmod.dll and b/AssetStudioGUI/Libraries/x86/fmod.dll differ
diff --git a/AssetStudioGUI/ParallelExport.cs b/AssetStudioGUI/ParallelExport.cs
index 9274b39..f3fe5e4 100644
--- a/AssetStudioGUI/ParallelExport.cs
+++ b/AssetStudioGUI/ParallelExport.cs
@@ -119,11 +119,9 @@ namespace AssetStudioGUI
                         debugLog += GenerateAudioClipInfo(m_AudioClip);
                     }
 
-                    var debugLogConverter = "";
                     var buffer = converter.IsLegacy
-                        ? converter.RawAudioClipToWav(out debugLogConverter)
-                        : converter.ConvertToWav(m_AudioData, out debugLogConverter);
-                    debugLog += debugLogConverter;
+                        ? converter.RawAudioClipToWav(ref debugLog)
+                        : converter.ConvertToWav(m_AudioData, ref debugLog);
                     if (buffer == null)
                     {
                         Logger.Warning($"{debugLog}Failed to export \"{item.Text}\": Failed to convert fmod audio to Wav");
diff --git a/AssetStudioUtility/AudioClipConverter.cs b/AssetStudioUtility/Audio/AudioClipConverter.cs
similarity index 61%
rename from AssetStudioUtility/AudioClipConverter.cs
rename to AssetStudioUtility/Audio/AudioClipConverter.cs
index 1998d83..d166078 100644
--- a/AssetStudioUtility/AudioClipConverter.cs
+++ b/AssetStudioUtility/Audio/AudioClipConverter.cs
@@ -1,17 +1,16 @@
-using FMOD;
-using System;
+using System;
 using System.Runtime.InteropServices;
-using System.Text;
+using FMOD;
+using WavHelper;
 
 namespace AssetStudio
 {
-    public class AudioClipConverter
+    public sealed class AudioClipConverter
     {
-        public bool IsSupport => m_AudioClip.IsConvertSupport();
-        public bool IsLegacy => m_AudioClip.IsLegacyConvertSupport();
-
         private AudioClip m_AudioClip;
         private static FMOD.System system;
+        public bool IsSupport => m_AudioClip.IsConvertSupport();
+        public bool IsLegacy => m_AudioClip.IsLegacyConvertSupport();
 
         static AudioClipConverter()
         {
@@ -32,70 +31,133 @@ namespace AssetStudio
             m_AudioClip = audioClip;
         }
 
-        public byte[] ConvertToWav(byte[] m_AudioData, out string debugLog)
+        public byte[] ConvertToWav(byte[] m_AudioData, ref string debugLog)
         {
-            debugLog = "";
             var exinfo = new CREATESOUNDEXINFO();
             exinfo.cbsize = Marshal.SizeOf(exinfo);
             exinfo.length = (uint)m_AudioClip.m_Size;
             var result = system.createSound(m_AudioData, MODE.OPENMEMORY | MODE.LOWMEM | MODE.ACCURATETIME, ref exinfo, out var sound);
-            if (ErrorCheck(result, out debugLog))
+            if (ErrorCheck(result, ref debugLog))
                 return null;
+
             result = sound.getNumSubSounds(out var numsubsounds);
-            if (ErrorCheck(result, out debugLog))
+            if (ErrorCheck(result, ref debugLog))
                 return null;
+
             byte[] buff;
             if (numsubsounds > 0)
             {
                 result = sound.getSubSound(0, out var subsound);
-                if (ErrorCheck(result, out debugLog))
+                if (ErrorCheck(result, ref debugLog))
                     return null;
-                buff = SoundToWav(subsound, out debugLog);
+                buff = SoundToWav(subsound, ref debugLog);
                 subsound.release();
                 subsound.clearHandle();
             }
             else
             {
-                buff = SoundToWav(sound, out debugLog);
+                buff = SoundToWav(sound, ref debugLog);
             }
+
             sound.release();
             sound.clearHandle();
             return buff;
         }
 
-        public byte[] SoundToWav(Sound sound, out string debugLog)
+        private byte[] SoundToWav(Sound sound, ref string debugLog)
         {
-            debugLog = "[Fmod] Detecting sound format..\n";
+            var convertToPcm16 = false;
+            var audioFormat = WavAudioFormat.PCM;
+
+            debugLog += "[Fmod] Detecting sound format..\n";
             var result = sound.getFormat(out SOUND_TYPE soundType, out SOUND_FORMAT soundFormat, out int channels, out int bits);
-            if (ErrorCheck(result, out debugLog))
+            if (ErrorCheck(result, ref debugLog))
                 return null;
             debugLog += $"Detected sound type: {soundType}\n" +
                         $"Detected sound format: {soundFormat}\n" +
                         $"Detected channels: {channels}\n" +
                         $"Detected bit depth: {bits}\n";
+            if (soundFormat == SOUND_FORMAT.PCMFLOAT)
+            {
+                switch (m_AudioClip.m_BitsPerSample)
+                {
+                    case 16:
+                        convertToPcm16 = true;
+                        bits = 16;
+                        break;
+                    case 32:
+                        audioFormat = WavAudioFormat.IEEEfloat;
+                        break;
+                }
+            }
             result = sound.getDefaults(out var frequency, out _);
-            if (ErrorCheck(result, out debugLog))
+            if (ErrorCheck(result, ref debugLog))
                 return null;
-            var sampleRate = (int)frequency;
             result = sound.getLength(out var length, TIMEUNIT.PCMBYTES);
-            if (ErrorCheck(result, out debugLog))
+            if (ErrorCheck(result, ref debugLog))
                 return null;
             result = sound.@lock(0, length, out var ptr1, out var ptr2, out var len1, out var len2);
-            if (ErrorCheck(result, out debugLog))
+            if (ErrorCheck(result, ref debugLog))
                 return null;
-            var buffer = new byte[len1 + 44];
-            //添加wav头
-            WriteWavHeader(buffer, len1, sampleRate, channels, bits);
-            Marshal.Copy(ptr1, buffer, 44, (int)len1);
+            var wavDataLength = convertToPcm16
+                ? len1 / 2
+                : len1;
+            var buffer = new byte[wavDataLength + 44];
+            if (convertToPcm16)
+            {
+                ReadAsPcm16(ptr1, buffer, 44, len1, ref debugLog);
+            }
+            else
+            {
+                Marshal.Copy(ptr1, buffer, 44, (int)len1);
+            }
             result = sound.unlock(ptr1, ptr2, len1, len2);
-            if (ErrorCheck(result, out debugLog))
+            if (ErrorCheck(result, ref debugLog))
                 return null;
+            //添加wav头
+            var wavHeader = new WavHeader(wavDataLength, audioFormat, channels, (uint)frequency, bits);
+            wavHeader.WriteToArray(buffer);
             return buffer;
         }
 
-        private static bool ErrorCheck(RESULT result, out string log)
+        public byte[] RawAudioClipToWav(ref string debugLog)
+        {
+            var audioSize = (uint)m_AudioClip.m_Size;
+            var channels = m_AudioClip.m_Channels;
+            var sampleRate = m_AudioClip.m_Frequency;
+            var audioFormat = WavAudioFormat.PCM;
+            var bits = 16;
+
+            debugLog += "[Legacy wav converter] Generating wav header..\n";
+            var buffer = new byte[audioSize + 44];
+            m_AudioClip.m_AudioData.GetData(buffer, out var read, 44);
+            if (read > 0)
+            {
+                var wavHeader = new WavHeader(audioSize, audioFormat, channels, (uint)sampleRate, bits);
+                wavHeader.WriteToArray(buffer);
+            }
+            return buffer;
+        }
+
+        private static void ReadAsPcm16(IntPtr srcPtr, byte[] destBuffer, int offset, uint pcmDataLen, ref string debugLog)
+        {
+            var pcmFloatSample = new byte[4];
+            for (var i = 0; i < pcmDataLen; i += 4)
+            {
+                for (var j = 0; j < 4; j++)
+                {
+                    pcmFloatSample[j] = Marshal.ReadByte(srcPtr, i + j);
+                }
+                var pcm16Sample = (short)(BitConverter.ToSingle(pcmFloatSample, 0) * 32767);
+                destBuffer[offset] = (byte)(pcm16Sample & 255);
+                destBuffer[offset + 1] = (byte)(pcm16Sample >> 8);
+                offset += 2;
+            }
+            debugLog += "Finished PCMFLOAT -> PCM16 converting\n";
+        }
+
+        private static bool ErrorCheck(RESULT result, ref string log)
         {
-            log = string.Empty;
             if (result != RESULT.OK)
             {
                 log += $"FMOD error! {result} - {Error.String(result)}\n";
@@ -104,37 +166,6 @@ namespace AssetStudio
             return false;
         }
 
-        public byte[] RawAudioClipToWav(out string debugLog)
-        {
-            var audioSize = (uint)m_AudioClip.m_Size;
-            var channels = m_AudioClip.m_Channels;
-            var sampleRate = m_AudioClip.m_Frequency;
-            var bits = 16;
-
-            debugLog = "[Legacy wav converter] Generating wav header..\n";
-            var buffer = new byte[audioSize + 44];
-            m_AudioClip.m_AudioData.GetData(buffer, out var read, 44);
-            if (read > 0)
-                WriteWavHeader(buffer, audioSize, sampleRate, channels, bits);
-            return buffer;
-        }
-
-        private static void WriteWavHeader(byte[] buffer, uint size, int sampleRate, int channels, int bits)
-        {
-            Encoding.ASCII.GetBytes("RIFF").CopyTo(buffer, 0);
-            BitConverter.GetBytes(size + 36).CopyTo(buffer, 4);
-            Encoding.ASCII.GetBytes("WAVEfmt ").CopyTo(buffer, 8);
-            BitConverter.GetBytes(16).CopyTo(buffer, 16);
-            BitConverter.GetBytes((short)1).CopyTo(buffer, 20);
-            BitConverter.GetBytes((short)channels).CopyTo(buffer, 22);
-            BitConverter.GetBytes(sampleRate).CopyTo(buffer, 24);
-            BitConverter.GetBytes(sampleRate * channels * bits / 8).CopyTo(buffer, 28);
-            BitConverter.GetBytes((short)(channels * bits / 8)).CopyTo(buffer, 32);
-            BitConverter.GetBytes((short)bits).CopyTo(buffer, 34);
-            Encoding.ASCII.GetBytes("data").CopyTo(buffer, 36);
-            BitConverter.GetBytes(size).CopyTo(buffer, 40);
-        }
-
         public string GetExtensionName()
         {
             if (m_AudioClip.version < 5)
@@ -197,46 +228,4 @@ namespace AssetStudio
             return ".AudioClip";
         }
     }
-
-    public static class AudioClipExtension
-    {
-        public static bool IsConvertSupport(this AudioClip m_AudioClip)
-        {
-            if (m_AudioClip.version < 5)
-            {
-                switch (m_AudioClip.m_Type)
-                {
-                    case FMODSoundType.AIFF:
-                    case FMODSoundType.IT:
-                    case FMODSoundType.MOD:
-                    case FMODSoundType.S3M:
-                    case FMODSoundType.XM:
-                    case FMODSoundType.XMA:
-                    case FMODSoundType.AUDIOQUEUE:
-                        return true;
-                    default:
-                        return false;
-                }
-            }
-            else
-            {
-                switch (m_AudioClip.m_CompressionFormat)
-                {
-                    case AudioCompressionFormat.PCM:
-                    case AudioCompressionFormat.Vorbis:
-                    case AudioCompressionFormat.ADPCM:
-                    case AudioCompressionFormat.MP3:
-                    case AudioCompressionFormat.XMA:
-                        return true;
-                    default:
-                        return false;
-                }
-            }
-        }
-
-        public static bool IsLegacyConvertSupport(this AudioClip m_AudioClip)
-        {
-            return m_AudioClip.version < (2, 6) && m_AudioClip.m_Format != 0x05;
-        }
-    }
 }
diff --git a/AssetStudioUtility/Audio/AudioClipExtension.cs b/AssetStudioUtility/Audio/AudioClipExtension.cs
new file mode 100644
index 0000000..3615cfe
--- /dev/null
+++ b/AssetStudioUtility/Audio/AudioClipExtension.cs
@@ -0,0 +1,44 @@
+namespace AssetStudio
+{
+    public static class AudioClipExtension
+    {
+        public static bool IsConvertSupport(this AudioClip m_AudioClip)
+        {
+            if (m_AudioClip.version < 5)
+            {
+                switch (m_AudioClip.m_Type)
+                {
+                    case FMODSoundType.AIFF:
+                    case FMODSoundType.IT:
+                    case FMODSoundType.MOD:
+                    case FMODSoundType.S3M:
+                    case FMODSoundType.XM:
+                    case FMODSoundType.XMA:
+                    case FMODSoundType.AUDIOQUEUE:
+                        return true;
+                    default:
+                        return false;
+                }
+            }
+            else
+            {
+                switch (m_AudioClip.m_CompressionFormat)
+                {
+                    case AudioCompressionFormat.PCM:
+                    case AudioCompressionFormat.Vorbis:
+                    case AudioCompressionFormat.ADPCM:
+                    case AudioCompressionFormat.MP3:
+                    case AudioCompressionFormat.XMA:
+                        return true;
+                    default:
+                        return false;
+                }
+            }
+        }
+
+        public static bool IsLegacyConvertSupport(this AudioClip m_AudioClip)
+        {
+            return m_AudioClip.version < (2, 6) && m_AudioClip.m_Format != 0x05;
+        }
+    }
+}
diff --git a/AssetStudioUtility/FMOD Studio API/fmod.cs b/AssetStudioUtility/Audio/FMODStudioAPI/fmod.cs
similarity index 73%
rename from AssetStudioUtility/FMOD Studio API/fmod.cs
rename to AssetStudioUtility/Audio/FMODStudioAPI/fmod.cs
index a7040f1..446ec9b 100644
--- a/AssetStudioUtility/FMOD Studio API/fmod.cs	
+++ b/AssetStudioUtility/Audio/FMODStudioAPI/fmod.cs
@@ -1,16 +1,15 @@
 /* ======================================================================================== */
 /* FMOD Core API - C# wrapper.                                                              */
-/* Copyright (c), Firelight Technologies Pty, Ltd. 2004-2020.                               */
+/* Copyright (c), Firelight Technologies Pty, Ltd. 2004-2025.                               */
 /*                                                                                          */
 /* For more detail visit:                                                                   */
-/* https://fmod.com/resources/documentation-api?version=2.0&page=core-api.html              */
+/* https://fmod.com/docs/2.03/api/core-api.html                                             */
 /* ======================================================================================== */
 
 using System;
 using System.Text;
 using System.Runtime.InteropServices;
 using System.Collections.Generic;
-
 #if NETFRAMEWORK
 using AssetStudio.PInvoke;
 #endif
@@ -21,9 +20,9 @@ namespace FMOD
         FMOD version number.  Check this against FMOD::System::getVersion / System_GetVersion
         0xaaaabbcc -> aaaa = major version number.  bb = minor version number.  cc = development version number.
     */
-    public class VERSION
+    public partial class VERSION
     {
-        public const int    number = 0x00020022;
+        public const int    number = 0x00020306;
         public const string dll    = "fmod";
     }
 
@@ -36,92 +35,92 @@ namespace FMOD
     }
 
     /*
-        FMOD types
+        FMOD core types
     */
     public enum RESULT : int
     {
-        OK,                             /* No errors. */
-        ERR_BADCOMMAND,                 /* Tried to call a function on a data type that does not allow this type of functionality (ie calling Sound::lock on a streaming sound). */
-        ERR_CHANNEL_ALLOC,              /* Error trying to allocate a channel. */
-        ERR_CHANNEL_STOLEN,             /* The specified channel has been reused to play another sound. */
-        ERR_DMA,                        /* DMA Failure.  See debug output for more information. */
-        ERR_DSP_CONNECTION,             /* DSP connection error.  Connection possibly caused a cyclic dependency or connected dsps with incompatible buffer counts. */
-        ERR_DSP_DONTPROCESS,            /* DSP return code from a DSP process query callback.  Tells mixer not to call the process callback and therefore not consume CPU.  Use this to optimize the DSP graph. */
-        ERR_DSP_FORMAT,                 /* DSP Format error.  A DSP unit may have attempted to connect to this network with the wrong format, or a matrix may have been set with the wrong size if the target unit has a specified channel map. */
-        ERR_DSP_INUSE,                  /* DSP is already in the mixer's DSP network. It must be removed before being reinserted or released. */
-        ERR_DSP_NOTFOUND,               /* DSP connection error.  Couldn't find the DSP unit specified. */
-        ERR_DSP_RESERVED,               /* DSP operation error.  Cannot perform operation on this DSP as it is reserved by the system. */
-        ERR_DSP_SILENCE,                /* DSP return code from a DSP process query callback.  Tells mixer silence would be produced from read, so go idle and not consume CPU.  Use this to optimize the DSP graph. */
-        ERR_DSP_TYPE,                   /* DSP operation cannot be performed on a DSP of this type. */
-        ERR_FILE_BAD,                   /* Error loading file. */
-        ERR_FILE_COULDNOTSEEK,          /* Couldn't perform seek operation.  This is a limitation of the medium (ie netstreams) or the file format. */
-        ERR_FILE_DISKEJECTED,           /* Media was ejected while reading. */
-        ERR_FILE_EOF,                   /* End of file unexpectedly reached while trying to read essential data (truncated?). */
-        ERR_FILE_ENDOFDATA,             /* End of current chunk reached while trying to read data. */
-        ERR_FILE_NOTFOUND,              /* File not found. */
-        ERR_FORMAT,                     /* Unsupported file or audio format. */
-        ERR_HEADER_MISMATCH,            /* There is a version mismatch between the FMOD header and either the FMOD Studio library or the FMOD Low Level library. */
-        ERR_HTTP,                       /* A HTTP error occurred. This is a catch-all for HTTP errors not listed elsewhere. */
-        ERR_HTTP_ACCESS,                /* The specified resource requires authentication or is forbidden. */
-        ERR_HTTP_PROXY_AUTH,            /* Proxy authentication is required to access the specified resource. */
-        ERR_HTTP_SERVER_ERROR,          /* A HTTP server error occurred. */
-        ERR_HTTP_TIMEOUT,               /* The HTTP request timed out. */
-        ERR_INITIALIZATION,             /* FMOD was not initialized correctly to support this function. */
-        ERR_INITIALIZED,                /* Cannot call this command after System::init. */
-        ERR_INTERNAL,                   /* An error occurred that wasn't supposed to.  Contact support. */
-        ERR_INVALID_FLOAT,              /* Value passed in was a NaN, Inf or denormalized float. */
-        ERR_INVALID_HANDLE,             /* An invalid object handle was used. */
-        ERR_INVALID_PARAM,              /* An invalid parameter was passed to this function. */
-        ERR_INVALID_POSITION,           /* An invalid seek position was passed to this function. */
-        ERR_INVALID_SPEAKER,            /* An invalid speaker was passed to this function based on the current speaker mode. */
-        ERR_INVALID_SYNCPOINT,          /* The syncpoint did not come from this sound handle. */
-        ERR_INVALID_THREAD,             /* Tried to call a function on a thread that is not supported. */
-        ERR_INVALID_VECTOR,             /* The vectors passed in are not unit length, or perpendicular. */
-        ERR_MAXAUDIBLE,                 /* Reached maximum audible playback count for this sound's soundgroup. */
-        ERR_MEMORY,                     /* Not enough memory or resources. */
-        ERR_MEMORY_CANTPOINT,           /* Can't use FMOD_OPENMEMORY_POINT on non PCM source data, or non mp3/xma/adpcm data if FMOD_CREATECOMPRESSEDSAMPLE was used. */
-        ERR_NEEDS3D,                    /* Tried to call a command on a 2d sound when the command was meant for 3d sound. */
-        ERR_NEEDSHARDWARE,              /* Tried to use a feature that requires hardware support. */
-        ERR_NET_CONNECT,                /* Couldn't connect to the specified host. */
-        ERR_NET_SOCKET_ERROR,           /* A socket error occurred.  This is a catch-all for socket-related errors not listed elsewhere. */
-        ERR_NET_URL,                    /* The specified URL couldn't be resolved. */
-        ERR_NET_WOULD_BLOCK,            /* Operation on a non-blocking socket could not complete immediately. */
-        ERR_NOTREADY,                   /* Operation could not be performed because specified sound/DSP connection is not ready. */
-        ERR_OUTPUT_ALLOCATED,           /* Error initializing output device, but more specifically, the output device is already in use and cannot be reused. */
-        ERR_OUTPUT_CREATEBUFFER,        /* Error creating hardware sound buffer. */
-        ERR_OUTPUT_DRIVERCALL,          /* A call to a standard soundcard driver failed, which could possibly mean a bug in the driver or resources were missing or exhausted. */
-        ERR_OUTPUT_FORMAT,              /* Soundcard does not support the specified format. */
-        ERR_OUTPUT_INIT,                /* Error initializing output device. */
-        ERR_OUTPUT_NODRIVERS,           /* The output device has no drivers installed.  If pre-init, FMOD_OUTPUT_NOSOUND is selected as the output mode.  If post-init, the function just fails. */
-        ERR_PLUGIN,                     /* An unspecified error has been returned from a plugin. */
-        ERR_PLUGIN_MISSING,             /* A requested output, dsp unit type or codec was not available. */
-        ERR_PLUGIN_RESOURCE,            /* A resource that the plugin requires cannot be found. (ie the DLS file for MIDI playback) */
-        ERR_PLUGIN_VERSION,             /* A plugin was built with an unsupported SDK version. */
-        ERR_RECORD,                     /* An error occurred trying to initialize the recording device. */
-        ERR_REVERB_CHANNELGROUP,        /* Reverb properties cannot be set on this channel because a parent channelgroup owns the reverb connection. */
-        ERR_REVERB_INSTANCE,            /* Specified instance in FMOD_REVERB_PROPERTIES couldn't be set. Most likely because it is an invalid instance number or the reverb doesn't exist. */
-        ERR_SUBSOUNDS,                  /* The error occurred because the sound referenced contains subsounds when it shouldn't have, or it doesn't contain subsounds when it should have.  The operation may also not be able to be performed on a parent sound. */
-        ERR_SUBSOUND_ALLOCATED,         /* This subsound is already being used by another sound, you cannot have more than one parent to a sound.  Null out the other parent's entry first. */
-        ERR_SUBSOUND_CANTMOVE,          /* Shared subsounds cannot be replaced or moved from their parent stream, such as when the parent stream is an FSB file. */
-        ERR_TAGNOTFOUND,                /* The specified tag could not be found or there are no tags. */
-        ERR_TOOMANYCHANNELS,            /* The sound created exceeds the allowable input channel count.  This can be increased using the 'maxinputchannels' parameter in System::setSoftwareFormat. */
-        ERR_TRUNCATED,                  /* The retrieved string is too long to fit in the supplied buffer and has been truncated. */
-        ERR_UNIMPLEMENTED,              /* Something in FMOD hasn't been implemented when it should be! contact support! */
-        ERR_UNINITIALIZED,              /* This command failed because System::init or System::setDriver was not called. */
-        ERR_UNSUPPORTED,                /* A command issued was not supported by this object.  Possibly a plugin without certain callbacks specified. */
-        ERR_VERSION,                    /* The version number of this file format is not supported. */
-        ERR_EVENT_ALREADY_LOADED,       /* The specified bank has already been loaded. */
-        ERR_EVENT_LIVEUPDATE_BUSY,      /* The live update connection failed due to the game already being connected. */
-        ERR_EVENT_LIVEUPDATE_MISMATCH,  /* The live update connection failed due to the game data being out of sync with the tool. */
-        ERR_EVENT_LIVEUPDATE_TIMEOUT,   /* The live update connection timed out. */
-        ERR_EVENT_NOTFOUND,             /* The requested event, bus or vca could not be found. */
-        ERR_STUDIO_UNINITIALIZED,       /* The Studio::System object is not yet initialized. */
-        ERR_STUDIO_NOT_LOADED,          /* The specified resource is not loaded, so it can't be unloaded. */
-        ERR_INVALID_STRING,             /* An invalid string was passed to this function. */
-        ERR_ALREADY_LOCKED,             /* The specified resource is already locked. */
-        ERR_NOT_LOCKED,                 /* The specified resource is not locked, so it can't be unlocked. */
-        ERR_RECORD_DISCONNECTED,        /* The specified recording driver has been disconnected. */
-        ERR_TOOMANYSAMPLES,             /* The length provided exceed the allowable limit. */
+        OK,
+        ERR_BADCOMMAND,
+        ERR_CHANNEL_ALLOC,
+        ERR_CHANNEL_STOLEN,
+        ERR_DMA,
+        ERR_DSP_CONNECTION,
+        ERR_DSP_DONTPROCESS,
+        ERR_DSP_FORMAT,
+        ERR_DSP_INUSE,
+        ERR_DSP_NOTFOUND,
+        ERR_DSP_RESERVED,
+        ERR_DSP_SILENCE,
+        ERR_DSP_TYPE,
+        ERR_FILE_BAD,
+        ERR_FILE_COULDNOTSEEK,
+        ERR_FILE_DISKEJECTED,
+        ERR_FILE_EOF,
+        ERR_FILE_ENDOFDATA,
+        ERR_FILE_NOTFOUND,
+        ERR_FORMAT,
+        ERR_HEADER_MISMATCH,
+        ERR_HTTP,
+        ERR_HTTP_ACCESS,
+        ERR_HTTP_PROXY_AUTH,
+        ERR_HTTP_SERVER_ERROR,
+        ERR_HTTP_TIMEOUT,
+        ERR_INITIALIZATION,
+        ERR_INITIALIZED,
+        ERR_INTERNAL,
+        ERR_INVALID_FLOAT,
+        ERR_INVALID_HANDLE,
+        ERR_INVALID_PARAM,
+        ERR_INVALID_POSITION,
+        ERR_INVALID_SPEAKER,
+        ERR_INVALID_SYNCPOINT,
+        ERR_INVALID_THREAD,
+        ERR_INVALID_VECTOR,
+        ERR_MAXAUDIBLE,
+        ERR_MEMORY,
+        ERR_MEMORY_CANTPOINT,
+        ERR_NEEDS3D,
+        ERR_NEEDSHARDWARE,
+        ERR_NET_CONNECT,
+        ERR_NET_SOCKET_ERROR,
+        ERR_NET_URL,
+        ERR_NET_WOULD_BLOCK,
+        ERR_NOTREADY,
+        ERR_OUTPUT_ALLOCATED,
+        ERR_OUTPUT_CREATEBUFFER,
+        ERR_OUTPUT_DRIVERCALL,
+        ERR_OUTPUT_FORMAT,
+        ERR_OUTPUT_INIT,
+        ERR_OUTPUT_NODRIVERS,
+        ERR_PLUGIN,
+        ERR_PLUGIN_MISSING,
+        ERR_PLUGIN_RESOURCE,
+        ERR_PLUGIN_VERSION,
+        ERR_RECORD,
+        ERR_REVERB_CHANNELGROUP,
+        ERR_REVERB_INSTANCE,
+        ERR_SUBSOUNDS,
+        ERR_SUBSOUND_ALLOCATED,
+        ERR_SUBSOUND_CANTMOVE,
+        ERR_TAGNOTFOUND,
+        ERR_TOOMANYCHANNELS,
+        ERR_TRUNCATED,
+        ERR_UNIMPLEMENTED,
+        ERR_UNINITIALIZED,
+        ERR_UNSUPPORTED,
+        ERR_VERSION,
+        ERR_EVENT_ALREADY_LOADED,
+        ERR_EVENT_LIVEUPDATE_BUSY,
+        ERR_EVENT_LIVEUPDATE_MISMATCH,
+        ERR_EVENT_LIVEUPDATE_TIMEOUT,
+        ERR_EVENT_NOTFOUND,
+        ERR_STUDIO_UNINITIALIZED,
+        ERR_STUDIO_NOT_LOADED,
+        ERR_INVALID_STRING,
+        ERR_ALREADY_LOCKED,
+        ERR_NOT_LOCKED,
+        ERR_RECORD_DISCONNECTED,
+        ERR_TOOMANYSAMPLES,
     }
 
     public enum CHANNELCONTROL_TYPE : int
@@ -134,9 +133,9 @@ namespace FMOD
     [StructLayout(LayoutKind.Sequential)]
     public struct VECTOR
     {
-        public float x;        /* X co-ordinate in 3D space. */
-        public float y;        /* Y co-ordinate in 3D space. */
-        public float z;        /* Z co-ordinate in 3D space. */
+        public float x;
+        public float y;
+        public float z;
     }
 
     [StructLayout(LayoutKind.Sequential)]
@@ -148,117 +147,144 @@ namespace FMOD
         public VECTOR up;
     }
 
+    [StructLayout(LayoutKind.Sequential)]
+    public partial struct GUID
+    {
+        public int Data1;
+        public int Data2;
+        public int Data3;
+        public int Data4;
+    }
+
     [StructLayout(LayoutKind.Sequential)]
     public struct ASYNCREADINFO
     {
-        public IntPtr   handle;                     /* [r] The file handle that was filled out in the open callback. */
-        public uint     offset;                     /* [r] Seek position, make sure you read from this file offset. */
-        public uint     sizebytes;                  /* [r] how many bytes requested for read. */
-        public int      priority;                   /* [r] 0 = low importance.  100 = extremely important (ie 'must read now or stuttering may occur') */
+        public IntPtr   handle;
+        public uint     offset;
+        public uint     sizebytes;
+        public int      priority;
 
-        public IntPtr   userdata;                   /* [r] User data pointer. */
-        public IntPtr   buffer;                     /* [w] Buffer to read file data into. */
-        public uint     bytesread;                  /* [w] Fill this in before setting result code to tell FMOD how many bytes were read. */
-        public FILE_ASYNCDONE_FUNC done;            /* [r] FMOD file system wake up function.  Call this when user file read is finished.  Pass result of file read as a parameter. */
+        public IntPtr   userdata;
+        public IntPtr   buffer;
+        public uint     bytesread;
+        public FILE_ASYNCDONE_FUNC done;
     }
 
     public enum OUTPUTTYPE : int
     {
-        AUTODETECT,      /* Picks the best output mode for the platform. This is the default. */
+        AUTODETECT,
 
-        UNKNOWN,         /* All - 3rd party plugin, unknown. This is for use with System::getOutput only. */
-        NOSOUND,         /* All - Perform all mixing but discard the final output. */
-        WAVWRITER,       /* All - Writes output to a .wav file. */
-        NOSOUND_NRT,     /* All - Non-realtime version of FMOD_OUTPUTTYPE_NOSOUND. User can drive mixer with System::update at whatever rate they want. */
-        WAVWRITER_NRT,   /* All - Non-realtime version of FMOD_OUTPUTTYPE_WAVWRITER. User can drive mixer with System::update at whatever rate they want. */
+        UNKNOWN,
+        NOSOUND,
+        WAVWRITER,
+        NOSOUND_NRT,
+        WAVWRITER_NRT,
 
-        WASAPI,          /* Win / Xbox One       - Windows Audio Session API.           (Default on Windows and Xbox One) */
-        ASIO,            /* Win                  - Low latency ASIO 2.0. */
-        PULSEAUDIO,      /* Linux                - Pulse Audio.                         (Default on Linux if available) */
-        ALSA,            /* Linux                - Advanced Linux Sound Architecture.   (Default on Linux if PulseAudio isn't available) */
-        COREAUDIO,       /* Mac / iOS            - Core Audio.                          (Default on Mac and iOS) */
-        AUDIOTRACK,      /* Android              - Java Audio Track.                    (Default on Android 2.2 and below) */
-        OPENSL,          /* Android              - OpenSL ES.                           (Default on Android 2.3 up to 7.1) */
-        AUDIOOUT,        /* PS4                  - Audio Out.                           (Default on PS4) */
-        AUDIO3D,         /* PS4                  - Audio3D. */
-        WEBAUDIO,        /* Web Browser          - JavaScript webaudio output.          (Default on JavaScript) */
-        NNAUDIO,         /* Switch               - nn::audio.                           (Default on Switch) */
-        WINSONIC,        /* Win10 / Xbox One     - Windows Sonic. */
-        AAUDIO,          /* Android              - AAudio                               (Default on Android 8.0 and above) */
+        WASAPI,
+        ASIO,
+        PULSEAUDIO,
+        ALSA,
+        COREAUDIO,
+        AUDIOTRACK,
+        OPENSL,
+        AUDIOOUT,
+        AUDIO3D,
+        WEBAUDIO,
+        NNAUDIO,
+        WINSONIC,
+        AAUDIO,
+        AUDIOWORKLET,
+        PHASE,
+        OHAUDIO,
 
-        MAX,             /* Maximum number of output types supported. */
+        MAX,
+    }
+
+    public enum PORT_TYPE : int
+    {
+        MUSIC,
+        COPYRIGHT_MUSIC,
+        VOICE,
+        CONTROLLER,
+        PERSONAL,
+        VIBRATION,
+        AUX,
+        PASSTHROUGH,
+        VR_VIBRATION,
+
+        MAX
     }
 
     public enum DEBUG_MODE : int
     {
-        TTY,        /* Default log location per platform, i.e. Visual Studio output window, stderr, LogCat, etc */
-        FILE,       /* Write log to specified file path */
-        CALLBACK,   /* Call specified callback with log information */
+        TTY,
+        FILE,
+        CALLBACK,
     }
 
     [Flags]
     public enum DEBUG_FLAGS : uint
     {
-        NONE                    = 0x00000000,   /* Disable all messages */
-        ERROR                   = 0x00000001,   /* Enable only error messages. */
-        WARNING                 = 0x00000002,   /* Enable warning and error messages. */
-        LOG                     = 0x00000004,   /* Enable informational, warning and error messages (default). */
+        NONE                    = 0x00000000,
+        ERROR                   = 0x00000001,
+        WARNING                 = 0x00000002,
+        LOG                     = 0x00000004,
 
-        TYPE_MEMORY             = 0x00000100,   /* Verbose logging for memory operations, only use this if you are debugging a memory related issue. */
-        TYPE_FILE               = 0x00000200,   /* Verbose logging for file access, only use this if you are debugging a file related issue. */
-        TYPE_CODEC              = 0x00000400,   /* Verbose logging for codec initialization, only use this if you are debugging a codec related issue. */
-        TYPE_TRACE              = 0x00000800,   /* Verbose logging for internal errors, use this for tracking the origin of error codes. */
+        TYPE_MEMORY             = 0x00000100,
+        TYPE_FILE               = 0x00000200,
+        TYPE_CODEC              = 0x00000400,
+        TYPE_TRACE              = 0x00000800,
 
-        DISPLAY_TIMESTAMPS      = 0x00010000,   /* Display the time stamp of the log message in milliseconds. */
-        DISPLAY_LINENUMBERS     = 0x00020000,   /* Display the source code file and line number for where the message originated. */
-        DISPLAY_THREAD          = 0x00040000,   /* Display the thread ID of the calling function that generated the message. */
+        DISPLAY_TIMESTAMPS      = 0x00010000,
+        DISPLAY_LINENUMBERS     = 0x00020000,
+        DISPLAY_THREAD          = 0x00040000,
     }
 
     [Flags]
     public enum MEMORY_TYPE : uint
     {
-        NORMAL             = 0x00000000,       /* Standard memory. */
-        STREAM_FILE        = 0x00000001,       /* Stream file buffer, size controllable with System::setStreamBufferSize. */
-        STREAM_DECODE      = 0x00000002,       /* Stream decode buffer, size controllable with FMOD_CREATESOUNDEXINFO::decodebuffersize. */
-        SAMPLEDATA         = 0x00000004,       /* Sample data buffer.  Raw audio data, usually PCM/MPEG/ADPCM/XMA data. */
-        DSP_BUFFER         = 0x00000008,       /* DSP memory block allocated when more than 1 output exists on a DSP node. */
-        PLUGIN             = 0x00000010,       /* Memory allocated by a third party plugin. */
-        PERSISTENT         = 0x00200000,       /* Persistent memory. Memory will be freed when System::release is called. */
-        ALL                = 0xFFFFFFFF
+        NORMAL                  = 0x00000000,
+        STREAM_FILE             = 0x00000001,
+        STREAM_DECODE           = 0x00000002,
+        SAMPLEDATA              = 0x00000004,
+        DSP_BUFFER              = 0x00000008,
+        PLUGIN                  = 0x00000010,
+        PERSISTENT              = 0x00200000,
+        ALL                     = 0xFFFFFFFF
     }
 
     public enum SPEAKERMODE : int
     {
-        DEFAULT,          /* Default speaker mode for the chosen output mode which will resolve after System::init. */
-        RAW,              /* Assume there is no special mapping from a given channel to a speaker, channels map 1:1 in order. Use System::setSoftwareFormat to specify the speaker count. */
-        MONO,             /*  1 speaker setup (monaural). */
-        STEREO,           /*  2 speaker setup (stereo) front left, front right. */
-        QUAD,             /*  4 speaker setup (4.0)    front left, front right, surround left, surround right. */
-        SURROUND,         /*  5 speaker setup (5.0)    front left, front right, center, surround left, surround right. */
-        _5POINT1,         /*  6 speaker setup (5.1)    front left, front right, center, low frequency, surround left, surround right. */
-        _7POINT1,         /*  8 speaker setup (7.1)    front left, front right, center, low frequency, surround left, surround right, back left, back right. */
-        _7POINT1POINT4,   /* 12 speaker setup (7.1.4)  front left, front right, center, low frequency, surround left, surround right, back left, back right, top front left, top front right, top back left, top back right. */
+        DEFAULT,
+        RAW,
+        MONO,
+        STEREO,
+        QUAD,
+        SURROUND,
+        _5POINT1,
+        _7POINT1,
+        _7POINT1POINT4,
 
-        MAX,              /* Maximum number of speaker modes supported. */
+        MAX,
     }
-     
+
     public enum SPEAKER : int
     {
-        NONE = -1,         /* No speaker */
-        FRONT_LEFT,        /* The front left speaker */
-        FRONT_RIGHT,       /* The front right speaker */
-        FRONT_CENTER,      /* The front center speaker */
-        LOW_FREQUENCY,     /* The LFE or 'subwoofer' speaker */
-        SURROUND_LEFT,     /* The surround left (usually to the side) speaker */
-        SURROUND_RIGHT,    /* The surround right (usually to the side) speaker */
-        BACK_LEFT,         /* The back left speaker */
-        BACK_RIGHT,        /* The back right speaker */
-        TOP_FRONT_LEFT,    /* The top front left speaker */
-        TOP_FRONT_RIGHT,   /* The top front right speaker */
-        TOP_BACK_LEFT,     /* The top back left speaker */
-        TOP_BACK_RIGHT,    /* The top back right speaker */
+        NONE = -1,
+        FRONT_LEFT,
+        FRONT_RIGHT,
+        FRONT_CENTER,
+        LOW_FREQUENCY,
+        SURROUND_LEFT,
+        SURROUND_RIGHT,
+        BACK_LEFT,
+        BACK_RIGHT,
+        TOP_FRONT_LEFT,
+        TOP_FRONT_RIGHT,
+        TOP_BACK_LEFT,
+        TOP_BACK_RIGHT,
 
-        MAX,               /* Maximum number of speaker types supported. */
+        MAX,
     }
 
     [Flags]
@@ -287,23 +313,23 @@ namespace FMOD
 
     public enum CHANNELORDER : int
     {
-        DEFAULT,              /* Left, Right, Center, LFE, Surround Left, Surround Right, Back Left, Back Right (see FMOD_SPEAKER enumeration)   */
-        WAVEFORMAT,           /* Left, Right, Center, LFE, Back Left, Back Right, Surround Left, Surround Right (as per Microsoft .wav WAVEFORMAT structure master order) */
-        PROTOOLS,             /* Left, Center, Right, Surround Left, Surround Right, LFE */
-        ALLMONO,              /* Mono, Mono, Mono, Mono, Mono, Mono, ... (each channel all the way up to 32 channels are treated as if they were mono) */
-        ALLSTEREO,            /* Left, Right, Left, Right, Left, Right, ... (each pair of channels is treated as stereo all the way up to 32 channels) */
-        ALSA,                 /* Left, Right, Surround Left, Surround Right, Center, LFE (as per Linux ALSA channel order) */
+        DEFAULT,
+        WAVEFORMAT,
+        PROTOOLS,
+        ALLMONO,
+        ALLSTEREO,
+        ALSA,
 
-        MAX,                  /* Maximum number of channel orderings supported. */
+        MAX,
     }
 
     public enum PLUGINTYPE : int
     {
-        OUTPUT,          /* The plugin type is an output module.  FMOD mixed audio will play through one of these devices */
-        CODEC,           /* The plugin type is a file format codec.  FMOD will use these codecs to load file formats for playback. */
-        DSP,             /* The plugin type is a DSP unit.  FMOD will use these plugins as part of its DSP network to apply effects to output or generate sound in realtime. */
+        OUTPUT,
+        CODEC,
+        DSP,
 
-        MAX,             /* Maximum number of plugin types supported. */
+        MAX,
     }
 
     [StructLayout(LayoutKind.Sequential)]
@@ -316,137 +342,138 @@ namespace FMOD
     [Flags]
     public enum INITFLAGS : uint
     {
-        NORMAL                     = 0x00000000, /* Initialize normally */
-        STREAM_FROM_UPDATE         = 0x00000001, /* No stream thread is created internally.  Streams are driven from System::update.  Mainly used with non-realtime outputs. */
-        MIX_FROM_UPDATE            = 0x00000002, /* FMOD Mixer thread is woken up to do a mix when System::update is called rather than waking periodically on its own timer. */
-        _3D_RIGHTHANDED            = 0x00000004, /* FMOD will treat +X as right, +Y as up and +Z as backwards (towards you). */
-        CHANNEL_LOWPASS            = 0x00000100, /* All FMOD_3D based voices will add a software lowpass filter effect into the DSP chain which is automatically used when Channel::set3DOcclusion is used or the geometry API.   This also causes sounds to sound duller when the sound goes behind the listener, as a fake HRTF style effect.  Use System::setAdvancedSettings to disable or adjust cutoff frequency for this feature. */
-        CHANNEL_DISTANCEFILTER     = 0x00000200, /* All FMOD_3D based voices will add a software lowpass and highpass filter effect into the DSP chain which will act as a distance-automated bandpass filter. Use System::setAdvancedSettings to adjust the center frequency. */
-        PROFILE_ENABLE             = 0x00010000, /* Enable TCP/IP based host which allows FMOD Designer or FMOD Profiler to connect to it, and view memory, CPU and the DSP network graph in real-time. */
-        VOL0_BECOMES_VIRTUAL       = 0x00020000, /* Any sounds that are 0 volume will go virtual and not be processed except for having their positions updated virtually.  Use System::setAdvancedSettings to adjust what volume besides zero to switch to virtual at. */
-        GEOMETRY_USECLOSEST        = 0x00040000, /* With the geometry engine, only process the closest polygon rather than accumulating all polygons the sound to listener line intersects. */
-        PREFER_DOLBY_DOWNMIX       = 0x00080000, /* When using FMOD_SPEAKERMODE_5POINT1 with a stereo output device, use the Dolby Pro Logic II downmix algorithm instead of the default stereo downmix algorithm. */
-        THREAD_UNSAFE              = 0x00100000, /* Disables thread safety for API calls. Only use this if FMOD low level is being called from a single thread, and if Studio API is not being used! */
-        PROFILE_METER_ALL          = 0x00200000, /* Slower, but adds level metering for every single DSP unit in the graph.  Use DSP::setMeteringEnabled to turn meters off individually. */
-        MEMORY_TRACKING            = 0x00400000, /* Enable detailed memory allocation tracking. Only useful when using the Studio API. */
+        NORMAL                     = 0x00000000,
+        STREAM_FROM_UPDATE         = 0x00000001,
+        MIX_FROM_UPDATE            = 0x00000002,
+        _3D_RIGHTHANDED            = 0x00000004,
+        CLIP_OUTPUT                = 0x00000008,
+        CHANNEL_LOWPASS            = 0x00000100,
+        CHANNEL_DISTANCEFILTER     = 0x00000200,
+        PROFILE_ENABLE             = 0x00010000,
+        VOL0_BECOMES_VIRTUAL       = 0x00020000,
+        GEOMETRY_USECLOSEST        = 0x00040000,
+        PREFER_DOLBY_DOWNMIX       = 0x00080000,
+        THREAD_UNSAFE              = 0x00100000,
+        PROFILE_METER_ALL          = 0x00200000,
+        MEMORY_TRACKING            = 0x00400000,
     }
 
     public enum SOUND_TYPE : int
     {
-        UNKNOWN,         /* 3rd party / unknown plugin format. */
-        AIFF,            /* AIFF. */
-        ASF,             /* Microsoft Advanced Systems Format (ie WMA/ASF/WMV). */
-        DLS,             /* Sound font / downloadable sound bank. */
-        FLAC,            /* FLAC lossless codec. */
-        FSB,             /* FMOD Sample Bank. */
-        IT,              /* Impulse Tracker. */
-        MIDI,            /* MIDI. extracodecdata is a pointer to an FMOD_MIDI_EXTRACODECDATA structure. */
-        MOD,             /* Protracker / Fasttracker MOD. */
-        MPEG,            /* MP2/MP3 MPEG. */
-        OGGVORBIS,       /* Ogg vorbis. */
-        PLAYLIST,        /* Information only from ASX/PLS/M3U/WAX playlists */
-        RAW,             /* Raw PCM data. */
-        S3M,             /* ScreamTracker 3. */
-        USER,            /* User created sound. */
-        WAV,             /* Microsoft WAV. */
-        XM,              /* FastTracker 2 XM. */
-        XMA,             /* Xbox One XMA */
-        AUDIOQUEUE,      /* iPhone hardware decoder, supports AAC, ALAC and MP3. extracodecdata is a pointer to an FMOD_AUDIOQUEUE_EXTRACODECDATA structure. */
-        AT9,             /* PS4 ATRAC 9 format */
-        VORBIS,          /* Vorbis */
-        MEDIA_FOUNDATION,/* Windows Store Application built in system codecs */
-        MEDIACODEC,      /* Android MediaCodec */
-        FADPCM,          /* FMOD Adaptive Differential Pulse Code Modulation */
-        OPUS,            /* Opus */
+        UNKNOWN,
+        AIFF,
+        ASF,
+        DLS,
+        FLAC,
+        FSB,
+        IT,
+        MIDI,
+        MOD,
+        MPEG,
+        OGGVORBIS,
+        PLAYLIST,
+        RAW,
+        S3M,
+        USER,
+        WAV,
+        XM,
+        XMA,
+        AUDIOQUEUE,
+        AT9,
+        VORBIS,
+        MEDIA_FOUNDATION,
+        MEDIACODEC,
+        FADPCM,
+        OPUS,
 
-        MAX,             /* Maximum number of sound types supported. */
+        MAX,
     }
 
     public enum SOUND_FORMAT : int
     {
-        NONE,       /* Unitialized / unknown */
-        PCM8,       /* 8bit integer PCM data */
-        PCM16,      /* 16bit integer PCM data  */
-        PCM24,      /* 24bit integer PCM data  */
-        PCM32,      /* 32bit integer PCM data  */
-        PCMFLOAT,   /* 32bit floating point PCM data  */
-        BITSTREAM,  /* Sound data is in its native compressed format. */
+        NONE,
+        PCM8,
+        PCM16,
+        PCM24,
+        PCM32,
+        PCMFLOAT,
+        BITSTREAM,
 
-        MAX         /* Maximum number of sound formats supported. */
+        MAX
     }
 
     [Flags]
     public enum MODE : uint
     {
-        DEFAULT                = 0x00000000,    /* Default for all modes listed below. FMOD_LOOP_OFF, FMOD_2D, FMOD_3D_WORLDRELATIVE, FMOD_3D_INVERSEROLLOFF */
-        LOOP_OFF               = 0x00000001,    /* For non looping sounds. (default).  Overrides FMOD_LOOP_NORMAL / FMOD_LOOP_BIDI. */
-        LOOP_NORMAL            = 0x00000002,    /* For forward looping sounds. */
-        LOOP_BIDI              = 0x00000004,    /* For bidirectional looping sounds. (only works on software mixed static sounds). */
-        _2D                    = 0x00000008,    /* Ignores any 3d processing. (default). */
-        _3D                    = 0x00000010,    /* Makes the sound positionable in 3D.  Overrides FMOD_2D. */
-        CREATESTREAM           = 0x00000080,    /* Decompress at runtime, streaming from the source provided (standard stream).  Overrides FMOD_CREATESAMPLE. */
-        CREATESAMPLE           = 0x00000100,    /* Decompress at loadtime, decompressing or decoding whole file into memory as the target sample format. (standard sample). */
-        CREATECOMPRESSEDSAMPLE = 0x00000200,    /* Load MP2, MP3, IMAADPCM or XMA into memory and leave it compressed.  During playback the FMOD software mixer will decode it in realtime as a 'compressed sample'.  Can only be used in combination with FMOD_SOFTWARE. */
-        OPENUSER               = 0x00000400,    /* Opens a user created static sample or stream. Use FMOD_CREATESOUNDEXINFO to specify format and/or read callbacks.  If a user created 'sample' is created with no read callback, the sample will be empty.  Use FMOD_Sound_Lock and FMOD_Sound_Unlock to place sound data into the sound if this is the case. */
-        OPENMEMORY             = 0x00000800,    /* "name_or_data" will be interpreted as a pointer to memory instead of filename for creating sounds. */
-        OPENMEMORY_POINT       = 0x10000000,    /* "name_or_data" will be interpreted as a pointer to memory instead of filename for creating sounds.  Use FMOD_CREATESOUNDEXINFO to specify length.  This differs to FMOD_OPENMEMORY in that it uses the memory as is, without duplicating the memory into its own buffers.  Cannot be freed after open, only after Sound::release.   Will not work if the data is compressed and FMOD_CREATECOMPRESSEDSAMPLE is not used. */
-        OPENRAW                = 0x00001000,    /* Will ignore file format and treat as raw pcm.  User may need to declare if data is FMOD_SIGNED or FMOD_UNSIGNED */
-        OPENONLY               = 0x00002000,    /* Just open the file, dont prebuffer or read.  Good for fast opens for info, or when sound::readData is to be used. */
-        ACCURATETIME           = 0x00004000,    /* For FMOD_CreateSound - for accurate FMOD_Sound_GetLength / FMOD_Channel_SetPosition on VBR MP3, AAC and MOD/S3M/XM/IT/MIDI files.  Scans file first, so takes longer to open. FMOD_OPENONLY does not affect this. */
-        MPEGSEARCH             = 0x00008000,    /* For corrupted / bad MP3 files.  This will search all the way through the file until it hits a valid MPEG header.  Normally only searches for 4k. */
-        NONBLOCKING            = 0x00010000,    /* For opening sounds and getting streamed subsounds (seeking) asyncronously.  Use Sound::getOpenState to poll the state of the sound as it opens or retrieves the subsound in the background. */
-        UNIQUE                 = 0x00020000,    /* Unique sound, can only be played one at a time */
-        _3D_HEADRELATIVE       = 0x00040000,    /* Make the sound's position, velocity and orientation relative to the listener. */
-        _3D_WORLDRELATIVE      = 0x00080000,    /* Make the sound's position, velocity and orientation absolute (relative to the world). (DEFAULT) */
-        _3D_INVERSEROLLOFF     = 0x00100000,    /* This sound will follow the inverse rolloff model where mindistance = full volume, maxdistance = where sound stops attenuating, and rolloff is fixed according to the global rolloff factor.  (DEFAULT) */
-        _3D_LINEARROLLOFF      = 0x00200000,    /* This sound will follow a linear rolloff model where mindistance = full volume, maxdistance = silence.  */
-        _3D_LINEARSQUAREROLLOFF= 0x00400000,    /* This sound will follow a linear-square rolloff model where mindistance = full volume, maxdistance = silence.  Rolloffscale is ignored. */
-        _3D_INVERSETAPEREDROLLOFF = 0x00800000, /* This sound will follow the inverse rolloff model at distances close to mindistance and a linear-square rolloff close to maxdistance. */
-        _3D_CUSTOMROLLOFF      = 0x04000000,    /* This sound will follow a rolloff model defined by Sound::set3DCustomRolloff / Channel::set3DCustomRolloff.  */
-        _3D_IGNOREGEOMETRY     = 0x40000000,    /* Is not affect by geometry occlusion.  If not specified in Sound::setMode, or Channel::setMode, the flag is cleared and it is affected by geometry again. */
-        IGNORETAGS             = 0x02000000,    /* Skips id3v2/asf/etc tag checks when opening a sound, to reduce seek/read overhead when opening files (helps with CD performance). */
-        LOWMEM                 = 0x08000000,    /* Removes some features from samples to give a lower memory overhead, like Sound::getName. */
-        VIRTUAL_PLAYFROMSTART  = 0x80000000     /* For sounds that start virtual (due to being quiet or low importance), instead of swapping back to audible, and playing at the correct offset according to time, this flag makes the sound play from the start. */
+        DEFAULT                     = 0x00000000,
+        LOOP_OFF                    = 0x00000001,
+        LOOP_NORMAL                 = 0x00000002,
+        LOOP_BIDI                   = 0x00000004,
+        _2D                         = 0x00000008,
+        _3D                         = 0x00000010,
+        CREATESTREAM                = 0x00000080,
+        CREATESAMPLE                = 0x00000100,
+        CREATECOMPRESSEDSAMPLE      = 0x00000200,
+        OPENUSER                    = 0x00000400,
+        OPENMEMORY                  = 0x00000800,
+        OPENMEMORY_POINT            = 0x10000000,
+        OPENRAW                     = 0x00001000,
+        OPENONLY                    = 0x00002000,
+        ACCURATETIME                = 0x00004000,
+        MPEGSEARCH                  = 0x00008000,
+        NONBLOCKING                 = 0x00010000,
+        UNIQUE                      = 0x00020000,
+        _3D_HEADRELATIVE            = 0x00040000,
+        _3D_WORLDRELATIVE           = 0x00080000,
+        _3D_INVERSEROLLOFF          = 0x00100000,
+        _3D_LINEARROLLOFF           = 0x00200000,
+        _3D_LINEARSQUAREROLLOFF     = 0x00400000,
+        _3D_INVERSETAPEREDROLLOFF   = 0x00800000,
+        _3D_CUSTOMROLLOFF           = 0x04000000,
+        _3D_IGNOREGEOMETRY          = 0x40000000,
+        IGNORETAGS                  = 0x02000000,
+        LOWMEM                      = 0x08000000,
+        VIRTUAL_PLAYFROMSTART       = 0x80000000
     }
 
     public enum OPENSTATE : int
     {
-        READY = 0,       /* Opened and ready to play */
-        LOADING,         /* Initial load in progress */
-        ERROR,           /* Failed to open - file not found, out of memory etc.  See return value of Sound::getOpenState for what happened. */
-        CONNECTING,      /* Connecting to remote host (internet sounds only) */
-        BUFFERING,       /* Buffering data */
-        SEEKING,         /* Seeking to subsound and re-flushing stream buffer. */
-        PLAYING,         /* Ready and playing, but not possible to release at this time without stalling the main thread. */
-        SETPOSITION,     /* Seeking within a stream to a different position. */
+        READY = 0,
+        LOADING,
+        ERROR,
+        CONNECTING,
+        BUFFERING,
+        SEEKING,
+        PLAYING,
+        SETPOSITION,
 
-        MAX,             /* Maximum number of open state types. */
+        MAX,
     }
 
     public enum SOUNDGROUP_BEHAVIOR : int
     {
-        BEHAVIOR_FAIL,              /* Any sound played that puts the sound count over the SoundGroup::setMaxAudible setting, will simply fail during System::playSound. */
-        BEHAVIOR_MUTE,              /* Any sound played that puts the sound count over the SoundGroup::setMaxAudible setting, will be silent, then if another sound in the group stops the sound that was silent before becomes audible again. */
-        BEHAVIOR_STEALLOWEST,       /* Any sound played that puts the sound count over the SoundGroup::setMaxAudible setting, will steal the quietest / least important sound playing in the group. */
+        BEHAVIOR_FAIL,
+        BEHAVIOR_MUTE,
+        BEHAVIOR_STEALLOWEST,
 
-        MAX,                        /* Maximum number of sound group behaviors. */
+        MAX,
     }
 
     public enum CHANNELCONTROL_CALLBACK_TYPE : int
     {
-        END,                  /* Called when a sound ends. */
-        VIRTUALVOICE,         /* Called when a voice is swapped out or swapped in. */
-        SYNCPOINT,            /* Called when a syncpoint is encountered.  Can be from wav file markers. */
-        OCCLUSION,            /* Called when the channel has its geometry occlusion value calculated.  Can be used to clamp or change the value. */
+        END,
+        VIRTUALVOICE,
+        SYNCPOINT,
+        OCCLUSION,
 
-        MAX,                  /* Maximum number of callback types supported. */
+        MAX,
     }
 
     public struct CHANNELCONTROL_DSP_INDEX
     {
-        public const int HEAD    = -1;         /* Head of the DSP chain.   Equivalent of index 0. */
-        public const int FADER   = -2;         /* Built in fader DSP. */
-        public const int TAIL    = -3;         /* Tail of the DSP chain.  Equivalent of the number of dsps minus 1. */
+        public const int HEAD    = -1;
+        public const int FADER   = -2;
+        public const int TAIL    = -3;
     }
 
     public enum ERRORCALLBACK_INSTANCETYPE : int
@@ -475,72 +502,102 @@ namespace FMOD
     [StructLayout(LayoutKind.Sequential)]
     public struct ERRORCALLBACK_INFO
     {
-        public  RESULT                      result;                     /* Error code result */
-        public  ERRORCALLBACK_INSTANCETYPE  instancetype;               /* Type of instance the error occurred on */
-        public  IntPtr                      instance;                   /* Instance pointer */
-        public  StringWrapper               functionname;               /* Function that the error occurred on */
-        public  StringWrapper               functionparams;             /* Function parameters that the error ocurred on */
+        public  RESULT                      result;
+        public  ERRORCALLBACK_INSTANCETYPE  instancetype;
+        public  IntPtr                      instance;
+        public  StringWrapper               functionname;
+        public  StringWrapper               functionparams;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct CPU_USAGE
+    {
+        public float    dsp;                    /* DSP mixing CPU usage. */
+        public float    stream;                 /* Streaming engine CPU usage. */
+        public float    geometry;               /* Geometry engine CPU usage. */
+        public float    update;                 /* System::update CPU usage. */
+        public float    convolution1;           /* Convolution reverb processing thread #1 CPU usage */
+        public float    convolution2;           /* Convolution reverb processing thread #2 CPU usage */
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_DATA_PARAMETER_INFO
+    {
+        public IntPtr   data;
+        public uint     length;
+        public int      index;
     }
 
     [Flags]
     public enum SYSTEM_CALLBACK_TYPE : uint
     {
-        DEVICELISTCHANGED      = 0x00000001,  /* Called from System::update when the enumerated list of devices has changed. */
-        DEVICELOST             = 0x00000002,  /* Called from System::update when an output device has been lost due to control panel parameter changes and FMOD cannot automatically recover. */
-        MEMORYALLOCATIONFAILED = 0x00000004,  /* Called directly when a memory allocation fails somewhere in FMOD.  (NOTE - 'system' will be NULL in this callback type.)*/
-        THREADCREATED          = 0x00000008,  /* Called directly when a thread is created. (NOTE - 'system' will be NULL in this callback type.) */
-        BADDSPCONNECTION       = 0x00000010,  /* Called when a bad connection was made with DSP::addInput. Usually called from mixer thread because that is where the connections are made.  */
-        PREMIX                 = 0x00000020,  /* Called each tick before a mix update happens. */
-        POSTMIX                = 0x00000040,  /* Called each tick after a mix update happens. */
-        ERROR                  = 0x00000080,  /* Called when each API function returns an error code, including delayed async functions. */
-        MIDMIX                 = 0x00000100,  /* Called each tick in mix update after clocks have been updated before the main mix occurs. */
-        THREADDESTROYED        = 0x00000200,  /* Called directly when a thread is destroyed. */
-        PREUPDATE              = 0x00000400,  /* Called at start of System::update function. */
-        POSTUPDATE             = 0x00000800,  /* Called at end of System::update function. */
-        RECORDLISTCHANGED      = 0x00001000,  /* Called from System::update when the enumerated list of recording devices has changed. */
-        ALL                    = 0xFFFFFFFF,  /* Pass this mask to System::setCallback to receive all callback types.  */
+        DEVICELISTCHANGED      = 0x00000001,
+        DEVICELOST             = 0x00000002,
+        MEMORYALLOCATIONFAILED = 0x00000004,
+        THREADCREATED          = 0x00000008,
+        BADDSPCONNECTION       = 0x00000010,
+        PREMIX                 = 0x00000020,
+        POSTMIX                = 0x00000040,
+        ERROR                  = 0x00000080,
+        THREADDESTROYED        = 0x00000100,
+        PREUPDATE              = 0x00000200,
+        POSTUPDATE             = 0x00000400,
+        RECORDLISTCHANGED      = 0x00000800,
+        BUFFEREDNOMIX          = 0x00001000,
+        DEVICEREINITIALIZE     = 0x00002000,
+        OUTPUTUNDERRUN         = 0x00004000,
+        RECORDPOSITIONCHANGED  = 0x00008000,
+        ALL                    = 0xFFFFFFFF,
     }
 
     /*
         FMOD Callbacks
     */
-    public delegate RESULT DEBUG_CALLBACK           (DEBUG_FLAGS flags, StringWrapper file, int line, StringWrapper func, StringWrapper message);
+    public delegate RESULT DEBUG_CALLBACK           (DEBUG_FLAGS flags, IntPtr file, int line, IntPtr func, IntPtr message);
     public delegate RESULT SYSTEM_CALLBACK          (IntPtr system, SYSTEM_CALLBACK_TYPE type, IntPtr commanddata1, IntPtr commanddata2, IntPtr userdata);
-    public delegate RESULT CHANNELCONTROL_CALLBACK(IntPtr channelcontrol, CHANNELCONTROL_TYPE controltype, CHANNELCONTROL_CALLBACK_TYPE callbacktype, IntPtr commanddata1, IntPtr commanddata2);
+    public delegate RESULT CHANNELCONTROL_CALLBACK  (IntPtr channelcontrol, CHANNELCONTROL_TYPE controltype, CHANNELCONTROL_CALLBACK_TYPE callbacktype, IntPtr commanddata1, IntPtr commanddata2);
+    public delegate RESULT DSP_CALLBACK             (IntPtr dsp, DSP_CALLBACK_TYPE type, IntPtr data);
     public delegate RESULT SOUND_NONBLOCK_CALLBACK  (IntPtr sound, RESULT result);
     public delegate RESULT SOUND_PCMREAD_CALLBACK   (IntPtr sound, IntPtr data, uint datalen);
     public delegate RESULT SOUND_PCMSETPOS_CALLBACK (IntPtr sound, int subsound, uint position, TIMEUNIT postype);
-    public delegate RESULT FILE_OPEN_CALLBACK       (StringWrapper name, ref uint filesize, ref IntPtr handle, IntPtr userdata);
+    public delegate RESULT FILE_OPEN_CALLBACK       (IntPtr name, ref uint filesize, ref IntPtr handle, IntPtr userdata);
     public delegate RESULT FILE_CLOSE_CALLBACK      (IntPtr handle, IntPtr userdata);
     public delegate RESULT FILE_READ_CALLBACK       (IntPtr handle, IntPtr buffer, uint sizebytes, ref uint bytesread, IntPtr userdata);
     public delegate RESULT FILE_SEEK_CALLBACK       (IntPtr handle, uint pos, IntPtr userdata);
     public delegate RESULT FILE_ASYNCREAD_CALLBACK  (IntPtr info, IntPtr userdata);
     public delegate RESULT FILE_ASYNCCANCEL_CALLBACK(IntPtr info, IntPtr userdata);
-    public delegate RESULT FILE_ASYNCDONE_FUNC      (IntPtr info, RESULT result);
-    public delegate IntPtr MEMORY_ALLOC_CALLBACK    (uint size, MEMORY_TYPE type, StringWrapper sourcestr);
-    public delegate IntPtr MEMORY_REALLOC_CALLBACK  (IntPtr ptr, uint size, MEMORY_TYPE type, StringWrapper sourcestr);
-    public delegate void   MEMORY_FREE_CALLBACK     (IntPtr ptr, MEMORY_TYPE type, StringWrapper sourcestr);
-    public delegate float  CB_3D_ROLLOFF_CALLBACK      (IntPtr channelcontrol, float distance);
+    public delegate void   FILE_ASYNCDONE_FUNC      (IntPtr info, RESULT result);
+    public delegate IntPtr MEMORY_ALLOC_CALLBACK    (uint size, MEMORY_TYPE type, IntPtr sourcestr);
+    public delegate IntPtr MEMORY_REALLOC_CALLBACK  (IntPtr ptr, uint size, MEMORY_TYPE type, IntPtr sourcestr);
+    public delegate void   MEMORY_FREE_CALLBACK     (IntPtr ptr, MEMORY_TYPE type, IntPtr sourcestr);
+    public delegate float  CB_3D_ROLLOFF_CALLBACK   (IntPtr channelcontrol, float distance);
 
     public enum DSP_RESAMPLER : int
     {
-        DEFAULT,         /* Default interpolation method.  Currently equal to FMOD_DSP_RESAMPLER_LINEAR. */
-        NOINTERP,        /* No interpolation.  High frequency aliasing hiss will be audible depending on the sample rate of the sound. */
-        LINEAR,          /* Linear interpolation (default method).  Fast and good quality, causes very slight lowpass effect on low frequency sounds. */
-        CUBIC,           /* Cubic interpolation.  Slower than linear interpolation but better quality. */
-        SPLINE,          /* 5 point spline interpolation.  Slowest resampling method but best quality. */
+        DEFAULT,
+        NOINTERP,
+        LINEAR,
+        CUBIC,
+        SPLINE,
 
-        MAX,             /* Maximum number of resample methods supported. */
+        MAX,
+    }
+
+    public enum DSP_CALLBACK_TYPE : int
+    {
+        DATAPARAMETERRELEASE,
+
+        MAX,
     }
 
     public enum DSPCONNECTION_TYPE : int
     {
-        STANDARD,          /* Default connection type.         Audio is mixed from the input to the output DSP's audible buffer.  */
-        SIDECHAIN,         /* Sidechain connection type.       Audio is mixed from the input to the output DSP's sidechain buffer.  */
-        SEND,              /* Send connection type.            Audio is mixed from the input to the output DSP's audible buffer, but the input is NOT executed, only copied from.  A standard connection or sidechain needs to make an input execute to generate data. */
-        SEND_SIDECHAIN,    /* Send sidechain connection type.  Audio is mixed from the input to the output DSP's sidechain buffer, but the input is NOT executed, only copied from.  A standard connection or sidechain needs to make an input execute to generate data. */
+        STANDARD,
+        SIDECHAIN,
+        SEND,
+        SEND_SIDECHAIN,
 
-        MAX,               /* Maximum number of DSP connection types supported. */
+        MAX,
     }
 
     public enum TAGTYPE : int
@@ -557,7 +614,7 @@ namespace FMOD
         FMOD,
         USER,
 
-        MAX                /* Maximum number of tag types supported. */
+        MAX
     }
 
     public enum TAGDATATYPE : int
@@ -570,31 +627,31 @@ namespace FMOD
         STRING_UTF16BE,
         STRING_UTF8,
 
-        MAX                /* Maximum number of tag datatypes supported. */
+        MAX
     }
 
     [StructLayout(LayoutKind.Sequential)]
     public struct TAG
     {
-        public  TAGTYPE           type;         /* [r] The type of this tag. */
-        public  TAGDATATYPE       datatype;     /* [r] The type of data that this tag contains */
-        public  StringWrapper     name;         /* [r] The name of this tag i.e. "TITLE", "ARTIST" etc. */
-        public  IntPtr            data;         /* [r] Pointer to the tag data - its format is determined by the datatype member */
-        public  uint              datalen;      /* [r] Length of the data contained in this tag */
-        public  bool              updated;      /* [r] True if this tag has been updated since last being accessed with Sound::getTag */
+        public  TAGTYPE           type;
+        public  TAGDATATYPE       datatype;
+        public  StringWrapper     name;
+        public  IntPtr            data;
+        public  uint              datalen;
+        public  bool              updated;
     }
 
     [Flags]
     public enum TIMEUNIT : uint
     {
-        MS          = 0x00000001,  /* Milliseconds. */
-        PCM         = 0x00000002,  /* PCM Samples, related to milliseconds * samplerate / 1000. */
-        PCMBYTES    = 0x00000004,  /* Bytes, related to PCM samples * channels * datawidth (ie 16bit = 2 bytes). */
-        RAWBYTES    = 0x00000008,  /* Raw file bytes of (compressed) sound data (does not include headers).  Only used by Sound::getLength and Channel::getPosition. */
-        PCMFRACTION = 0x00000010,  /* Fractions of 1 PCM sample.  Unsigned int range 0 to 0xFFFFFFFF.  Used for sub-sample granularity for DSP purposes. */
-        MODORDER    = 0x00000100,  /* MOD/S3M/XM/IT.  Order in a sequenced module format.  Use Sound::getFormat to determine the format. */
-        MODROW      = 0x00000200,  /* MOD/S3M/XM/IT.  Current row in a sequenced module format.  Cannot use with Channel::setPosition.  Sound::getLength will return the number if rows in the currently playing or seeked to pattern. */
-        MODPATTERN  = 0x00000400,  /* MOD/S3M/XM/IT.  Current pattern in a sequenced module format.  Cannot use with Channel::setPosition.  Sound::getLength will return the number of patterns in the song and Channel::getPosition will return the currently playing pattern. */
+        MS          = 0x00000001,
+        PCM         = 0x00000002,
+        PCMBYTES    = 0x00000004,
+        RAWBYTES    = 0x00000008,
+        PCMFRACTION = 0x00000010,
+        MODORDER    = 0x00000100,
+        MODROW      = 0x00000200,
+        MODPATTERN  = 0x00000400,
     }
 
     public struct PORT_INDEX
@@ -605,62 +662,109 @@ namespace FMOD
     [StructLayout(LayoutKind.Sequential)]
     public struct CREATESOUNDEXINFO
     {
-        public int                         cbsize;                 /* [w]   Size of this structure.  This is used so the structure can be expanded in the future and still work on older versions of FMOD Ex. */
-        public uint                        length;                 /* [w]   Optional. Specify 0 to ignore. Size in bytes of file to load, or sound to create (in this case only if FMOD_OPENUSER is used).  Required if loading from memory.  If 0 is specified, then it will use the size of the file (unless loading from memory then an error will be returned). */
-        public uint                        fileoffset;             /* [w]   Optional. Specify 0 to ignore. Offset from start of the file to start loading from.  This is useful for loading files from inside big data files. */
-        public int                         numchannels;            /* [w]   Optional. Specify 0 to ignore. Number of channels in a sound specified only if OPENUSER is used. */
-        public int                         defaultfrequency;       /* [w]   Optional. Specify 0 to ignore. Default frequency of sound in a sound specified only if OPENUSER is used.  Other formats use the frequency determined by the file format. */
-        public SOUND_FORMAT                format;                 /* [w]   Optional. Specify 0 or SOUND_FORMAT_NONE to ignore. Format of the sound specified only if OPENUSER is used.  Other formats use the format determined by the file format.   */
-        public uint                        decodebuffersize;       /* [w]   Optional. Specify 0 to ignore. For streams.  This determines the size of the double buffer (in PCM samples) that a stream uses.  Use this for user created streams if you want to determine the size of the callback buffer passed to you.  Specify 0 to use FMOD's default size which is currently equivalent to 400ms of the sound format created/loaded. */
-        public int                         initialsubsound;        /* [w]   Optional. Specify 0 to ignore. In a multi-sample file format such as .FSB/.DLS/.SF2, specify the initial subsound to seek to, only if CREATESTREAM is used. */
-        public int                         numsubsounds;           /* [w]   Optional. Specify 0 to ignore or have no subsounds.  In a user created multi-sample sound, specify the number of subsounds within the sound that are accessable with Sound::getSubSound / SoundGetSubSound. */
-        public IntPtr                      inclusionlist;          /* [w]   Optional. Specify 0 to ignore. In a multi-sample format such as .FSB/.DLS/.SF2 it may be desirable to specify only a subset of sounds to be loaded out of the whole file.  This is an array of subsound indicies to load into memory when created. */
-        public int                         inclusionlistnum;       /* [w]   Optional. Specify 0 to ignore. This is the number of integers contained within the */
-        public SOUND_PCMREAD_CALLBACK      pcmreadcallback;        /* [w]   Optional. Specify 0 to ignore. Callback to 'piggyback' on FMOD's read functions and accept or even write PCM data while FMOD is opening the sound.  Used for user sounds created with OPENUSER or for capturing decoded data as FMOD reads it. */
-        public SOUND_PCMSETPOS_CALLBACK    pcmsetposcallback;      /* [w]   Optional. Specify 0 to ignore. Callback for when the user calls a seeking function such as Channel::setPosition within a multi-sample sound, and for when it is opened.*/
-        public SOUND_NONBLOCK_CALLBACK     nonblockcallback;       /* [w]   Optional. Specify 0 to ignore. Callback for successful completion, or error while loading a sound that used the FMOD_NONBLOCKING flag.*/
-        public IntPtr                      dlsname;                /* [w]   Optional. Specify 0 to ignore. Filename for a DLS or SF2 sample set when loading a MIDI file.   If not specified, on windows it will attempt to open /windows/system32/drivers/gm.dls, otherwise the MIDI will fail to open.  */
-        public IntPtr                      encryptionkey;          /* [w]   Optional. Specify 0 to ignore. Key for encrypted FSB file.  Without this key an encrypted FSB file will not load. */
-        public int                         maxpolyphony;           /* [w]   Optional. Specify 0 to ingore. For sequenced formats with dynamic channel allocation such as .MID and .IT, this specifies the maximum voice count allowed while playing.  .IT defaults to 64.  .MID defaults to 32. */
-        public IntPtr                      userdata;               /* [w]   Optional. Specify 0 to ignore. This is user data to be attached to the sound during creation.  Access via Sound::getUserData. */
-        public SOUND_TYPE                  suggestedsoundtype;     /* [w]   Optional. Specify 0 or FMOD_SOUND_TYPE_UNKNOWN to ignore.  Instead of scanning all codec types, use this to speed up loading by making it jump straight to this codec. */
-        public FILE_OPEN_CALLBACK          fileuseropen;           /* [w]   Optional. Specify 0 to ignore. Callback for opening this file. */
-        public FILE_CLOSE_CALLBACK         fileuserclose;          /* [w]   Optional. Specify 0 to ignore. Callback for closing this file. */
-        public FILE_READ_CALLBACK          fileuserread;           /* [w]   Optional. Specify 0 to ignore. Callback for reading from this file. */
-        public FILE_SEEK_CALLBACK          fileuserseek;           /* [w]   Optional. Specify 0 to ignore. Callback for seeking within this file. */
-        public FILE_ASYNCREAD_CALLBACK     fileuserasyncread;      /* [w]   Optional. Specify 0 to ignore. Callback for asyncronously reading from this file. */
-        public FILE_ASYNCCANCEL_CALLBACK   fileuserasynccancel;    /* [w]   Optional. Specify 0 to ignore. Callback for cancelling an asyncronous read. */
-        public IntPtr                      fileuserdata;           /* [w]   Optional. Specify 0 to ignore. User data to be passed into the file callbacks. */
-        public int                         filebuffersize;         /* [w]   Optional. Specify 0 to ignore. Buffer size for reading the file, -1 to disable buffering, or 0 for system default. */
-        public CHANNELORDER                channelorder;           /* [w]   Optional. Specify 0 to ignore. Use this to differ the way fmod maps multichannel sounds to speakers.  See FMOD_CHANNELORDER for more. */
-        public IntPtr                      initialsoundgroup;      /* [w]   Optional. Specify 0 to ignore. Specify a sound group if required, to put sound in as it is created. */
-        public uint                        initialseekposition;    /* [w]   Optional. Specify 0 to ignore. For streams. Specify an initial position to seek the stream to. */
-        public TIMEUNIT                    initialseekpostype;     /* [w]   Optional. Specify 0 to ignore. For streams. Specify the time unit for the position set in initialseekposition. */
-        public int                         ignoresetfilesystem;    /* [w]   Optional. Specify 0 to ignore. Set to 1 to use fmod's built in file system. Ignores setFileSystem callbacks and also FMOD_CREATESOUNEXINFO file callbacks.  Useful for specific cases where you don't want to use your own file system but want to use fmod's file system (ie net streaming). */
-        public uint                        audioqueuepolicy;       /* [w]   Optional. Specify 0 or FMOD_AUDIOQUEUE_CODECPOLICY_DEFAULT to ignore. Policy used to determine whether hardware or software is used for decoding, see FMOD_AUDIOQUEUE_CODECPOLICY for options (iOS >= 3.0 required, otherwise only hardware is available) */
-        public uint                        minmidigranularity;     /* [w]   Optional. Specify 0 to ignore. Allows you to set a minimum desired MIDI mixer granularity. Values smaller than 512 give greater than default accuracy at the cost of more CPU and vise versa. Specify 0 for default (512 samples). */
-        public int                         nonblockthreadid;       /* [w]   Optional. Specify 0 to ignore. Specifies a thread index to execute non blocking load on.  Allows for up to 5 threads to be used for loading at once.  This is to avoid one load blocking another.  Maximum value = 4. */
-        public IntPtr                      fsbguid;                /* [r/w] Optional. Specify 0 to ignore. Allows you to provide the GUID lookup for cached FSB header info. Once loaded the GUID will be written back to the pointer. This is to avoid seeking and reading the FSB header. */
+        public int                         cbsize;
+        public uint                        length;
+        public uint                        fileoffset;
+        public int                         numchannels;
+        public int                         defaultfrequency;
+        public SOUND_FORMAT                format;
+        public uint                        decodebuffersize;
+        public int                         initialsubsound;
+        public int                         numsubsounds;
+        public IntPtr                      inclusionlist;
+        public int                         inclusionlistnum;
+        public IntPtr                      pcmreadcallback_internal;
+        public IntPtr                      pcmsetposcallback_internal;
+        public IntPtr                      nonblockcallback_internal;
+        public IntPtr                      dlsname;
+        public IntPtr                      encryptionkey;
+        public int                         maxpolyphony;
+        public IntPtr                      userdata;
+        public SOUND_TYPE                  suggestedsoundtype;
+        public IntPtr                      fileuseropen_internal;
+        public IntPtr                      fileuserclose_internal;
+        public IntPtr                      fileuserread_internal;
+        public IntPtr                      fileuserseek_internal;
+        public IntPtr                      fileuserasyncread_internal;
+        public IntPtr                      fileuserasynccancel_internal;
+        public IntPtr                      fileuserdata;
+        public int                         filebuffersize;
+        public CHANNELORDER                channelorder;
+        public IntPtr                      initialsoundgroup;
+        public uint                        initialseekposition;
+        public TIMEUNIT                    initialseekpostype;
+        public int                         ignoresetfilesystem;
+        public uint                        audioqueuepolicy;
+        public uint                        minmidigranularity;
+        public int                         nonblockthreadid;
+        public IntPtr                      fsbguid;
+
+        public SOUND_PCMREAD_CALLBACK pcmreadcallback
+        {
+            set { pcmreadcallback_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return pcmreadcallback_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<SOUND_PCMREAD_CALLBACK>(pcmreadcallback_internal); }
+        }
+        public SOUND_PCMSETPOS_CALLBACK pcmsetposcallback
+        {
+            set { pcmsetposcallback_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return pcmsetposcallback_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<SOUND_PCMSETPOS_CALLBACK>(pcmsetposcallback_internal); }
+        }
+        public SOUND_NONBLOCK_CALLBACK nonblockcallback
+        {
+            set { nonblockcallback_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return nonblockcallback_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<SOUND_NONBLOCK_CALLBACK>(nonblockcallback_internal); }
+        }
+        public FILE_OPEN_CALLBACK fileuseropen
+        {
+            set { fileuseropen_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return fileuseropen_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<FILE_OPEN_CALLBACK>(fileuseropen_internal); }
+        }
+        public FILE_CLOSE_CALLBACK fileuserclose
+        {
+            set { fileuserclose_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return fileuserclose_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<FILE_CLOSE_CALLBACK>(fileuserclose_internal); }
+        }
+        public FILE_READ_CALLBACK fileuserread
+        {
+            set { fileuserread_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return fileuserread_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<FILE_READ_CALLBACK>(fileuserread_internal); }
+        }
+        public FILE_SEEK_CALLBACK fileuserseek
+        {
+            set { fileuserseek_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return fileuserseek_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<FILE_SEEK_CALLBACK>(fileuserseek_internal); }
+        }
+        public FILE_ASYNCREAD_CALLBACK fileuserasyncread
+        {
+            set { fileuserasyncread_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return fileuserasyncread_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<FILE_ASYNCREAD_CALLBACK>(fileuserasyncread_internal); }
+        }
+        public FILE_ASYNCCANCEL_CALLBACK fileuserasynccancel
+        {
+            set { fileuserasynccancel_internal = (value == null ? IntPtr.Zero : Marshal.GetFunctionPointerForDelegate(value)); }
+            get { return fileuserasynccancel_internal == IntPtr.Zero ? null : Marshal.GetDelegateForFunctionPointer<FILE_ASYNCCANCEL_CALLBACK>(fileuserasynccancel_internal); }
+        }
+
     }
 
 #pragma warning disable 414
     [StructLayout(LayoutKind.Sequential)]
     public struct REVERB_PROPERTIES
-    {                            /*        MIN     MAX    DEFAULT   DESCRIPTION */
-        public float DecayTime;         /* [r/w]  0.0    20000.0 1500.0  Reverberation decay time in ms                                        */
-        public float EarlyDelay;        /* [r/w]  0.0    300.0   7.0     Initial reflection delay time                                         */
-        public float LateDelay;         /* [r/w]  0.0    100     11.0    Late reverberation delay time relative to initial reflection          */
-        public float HFReference;       /* [r/w]  20.0   20000.0 5000    Reference high frequency (hz)                                         */
-        public float HFDecayRatio;      /* [r/w]  10.0   100.0   50.0    High-frequency to mid-frequency decay time ratio                      */
-        public float Diffusion;         /* [r/w]  0.0    100.0   100.0   Value that controls the echo density in the late reverberation decay. */
-        public float Density;           /* [r/w]  0.0    100.0   100.0   Value that controls the modal density in the late reverberation decay */
-        public float LowShelfFrequency; /* [r/w]  20.0   1000.0  250.0   Reference low frequency (hz)                                          */
-        public float LowShelfGain;      /* [r/w]  -36.0  12.0    0.0     Relative room effect level at low frequencies                         */
-        public float HighCut;           /* [r/w]  20.0   20000.0 20000.0 Relative room effect level at high frequencies                        */
-        public float EarlyLateMix;      /* [r/w]  0.0    100.0   50.0    Early reflections level relative to room effect                       */
-        public float WetLevel;          /* [r/w]  -80.0  20.0    -6.0    Room effect level (at mid frequencies)
-                                  * */
-#region wrapperinternal
+    {
+        public float DecayTime;
+        public float EarlyDelay;
+        public float LateDelay;
+        public float HFReference;
+        public float HFDecayRatio;
+        public float Diffusion;
+        public float Density;
+        public float LowShelfFrequency;
+        public float LowShelfGain;
+        public float HighCut;
+        public float EarlyLateMix;
+        public float WetLevel;
+
+        #region wrapperinternal
         public REVERB_PROPERTIES(float decayTime, float earlyDelay, float lateDelay, float hfReference,
             float hfDecayRatio, float diffusion, float density, float lowShelfFrequency, float lowShelfGain,
             float highCut, float earlyLateMix, float wetLevel)
@@ -678,13 +782,12 @@ namespace FMOD
             EarlyLateMix = earlyLateMix;
             WetLevel = wetLevel;
         }
-#endregion
+        #endregion
     }
 #pragma warning restore 414
 
     public class PRESET
     {
-        /*                                                                                  Instance  Env   Diffus  Room   RoomHF  RmLF DecTm   DecHF  DecLF   Refl  RefDel   Revb  RevDel  ModTm  ModDp   HFRef    LFRef   Diffus  Densty  FLAGS */
         public static REVERB_PROPERTIES OFF()                 { return new REVERB_PROPERTIES(  1000,    7,  11, 5000, 100, 100, 100, 250, 0,    20,  96, -80.0f );}
         public static REVERB_PROPERTIES GENERIC()             { return new REVERB_PROPERTIES(  1500,    7,  11, 5000,  83, 100, 100, 250, 0, 14500,  96,  -8.0f );}
         public static REVERB_PROPERTIES PADDEDCELL()          { return new REVERB_PROPERTIES(   170,    1,   2, 5000,  10, 100, 100, 250, 0,   160,  84,  -7.8f );}
@@ -714,43 +817,154 @@ namespace FMOD
     [StructLayout(LayoutKind.Sequential)]
     public struct ADVANCEDSETTINGS
     {
-        public int                 cbSize;                     /* [w]   Size of this structure.  Use sizeof(FMOD_ADVANCEDSETTINGS) */
-        public int                 maxMPEGCodecs;              /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only.  MPEG   codecs consume 30,528 bytes per instance and this number will determine how many MPEG   channels can be played simultaneously. Default = 32. */
-        public int                 maxADPCMCodecs;             /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only.  ADPCM  codecs consume  3,128 bytes per instance and this number will determine how many ADPCM  channels can be played simultaneously. Default = 32. */
-        public int                 maxXMACodecs;               /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only.  XMA    codecs consume 14,836 bytes per instance and this number will determine how many XMA    channels can be played simultaneously. Default = 32. */
-        public int                 maxVorbisCodecs;            /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only.  Vorbis codecs consume 23,256 bytes per instance and this number will determine how many Vorbis channels can be played simultaneously. Default = 32. */    
-        public int                 maxAT9Codecs;               /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only.  AT9    codecs consume  8,720 bytes per instance and this number will determine how many AT9    channels can be played simultaneously. Default = 32. */    
-        public int                 maxFADPCMCodecs;            /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only.  This number will determine how many FADPCM channels can be played simultaneously. Default = 32. */
-        public int                 maxPCMCodecs;               /* [r/w] Unsupported. */
-        public int                 ASIONumChannels;            /* [r/w] Optional. Specify 0 to ignore. Number of channels available on the ASIO device. */
-        public IntPtr              ASIOChannelList;            /* [r/w] Optional. Specify 0 to ignore. Pointer to an array of strings (number of entries defined by ASIONumChannels) with ASIO channel names. */
-        public IntPtr              ASIOSpeakerList;            /* [r/w] Optional. Specify 0 to ignore. Pointer to a list of speakers that the ASIO channels map to.  This can be called after System::init to remap ASIO output. */
-        public float               vol0virtualvol;             /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_INIT_VOL0_BECOMES_VIRTUAL.  If this flag is used, and the volume is below this, then the sound will become virtual.  Use this value to raise the threshold to a different point where a sound goes virtual. */
-        public uint                defaultDecodeBufferSize;    /* [r/w] Optional. Specify 0 to ignore. For streams. This determines the default size of the double buffer (in milliseconds) that a stream uses.  Default = 400ms */
-        public ushort              profilePort;                /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_INIT_PROFILE_ENABLE.  Specify the port to listen on for connections by the profiler application. */
-        public uint                geometryMaxFadeTime;        /* [r/w] Optional. Specify 0 to ignore. The maximum time in miliseconds it takes for a channel to fade to the new level when its occlusion changes. */
-        public float               distanceFilterCenterFreq;   /* [r/w] Optional. Specify 0 to ignore. For use with FMOD_INIT_DISTANCE_FILTERING.  The default center frequency in Hz for the distance filtering effect. Default = 1500.0. */
-        public int                 reverb3Dinstance;           /* [r/w] Optional. Specify 0 to ignore. Out of 0 to 3, 3d reverb spheres will create a phyical reverb unit on this instance slot.  See FMOD_REVERB_PROPERTIES. */
-        public int                 DSPBufferPoolSize;          /* [r/w] Optional. Specify 0 to ignore. Number of buffers in DSP buffer pool.  Each buffer will be DSPBlockSize * sizeof(float) * SpeakerModeChannelCount.  ie 7.1 @ 1024 DSP block size = 8 * 1024 * 4 = 32kb.  Default = 8. */
-        public uint                stackSizeStream;            /* [r/w] Optional. Specify 0 to ignore. Specify the stack size for the FMOD Stream thread in bytes.  Useful for custom codecs that use excess stack.  Default 49,152 (48kb) */
-        public uint                stackSizeNonBlocking;       /* [r/w] Optional. Specify 0 to ignore. Specify the stack size for the FMOD_NONBLOCKING loading thread.  Useful for custom codecs that use excess stack.  Default 65,536 (64kb) */
-        public uint                stackSizeMixer;             /* [r/w] Optional. Specify 0 to ignore. Specify the stack size for the FMOD mixer thread.  Useful for custom dsps that use excess stack.  Default 49,152 (48kb) */
-        public DSP_RESAMPLER       resamplerMethod;            /* [r/w] Optional. Specify 0 to ignore. Resampling method used with fmod's software mixer.  See FMOD_DSP_RESAMPLER for details on methods. */
-        public uint                commandQueueSize;           /* [r/w] Optional. Specify 0 to ignore. Specify the command queue size for thread safe processing.  Default 2048 (2kb) */
-        public uint                randomSeed;                 /* [r/w] Optional. Specify 0 to ignore. Seed value that FMOD will use to initialize its internal random number generators. */
+        public int                 cbSize;
+        public int                 maxMPEGCodecs;
+        public int                 maxADPCMCodecs;
+        public int                 maxXMACodecs;
+        public int                 maxVorbisCodecs;
+        public int                 maxAT9Codecs;
+        public int                 maxFADPCMCodecs;
+        public int                 maxOpusCodecs;
+        public int                 ASIONumChannels;
+        public IntPtr              ASIOChannelList;
+        public IntPtr              ASIOSpeakerList;
+        public float               vol0virtualvol;
+        public uint                defaultDecodeBufferSize;
+        public ushort              profilePort;
+        public uint                geometryMaxFadeTime;
+        public float               distanceFilterCenterFreq;
+        public int                 reverb3Dinstance;
+        public int                 DSPBufferPoolSize;
+        public DSP_RESAMPLER       resamplerMethod;
+        public uint                randomSeed;
+        public int                 maxConvolutionThreads;
+        public int                 maxSpatialObjects;
     }
 
     [Flags]
     public enum DRIVER_STATE : uint
     {
-        CONNECTED = 0x00000001, /* Device is currently plugged in. */
-        DEFAULT   = 0x00000002, /* Device is the users preferred choice. */
+        CONNECTED = 0x00000001,
+        DEFAULT   = 0x00000002,
+    }
+
+    public enum THREAD_PRIORITY : int
+    {
+        /* Platform specific priority range */
+        PLATFORM_MIN        = -32 * 1024,
+        PLATFORM_MAX        =  32 * 1024,
+
+        /* Platform agnostic priorities, maps internally to platform specific value */
+        DEFAULT             = PLATFORM_MIN - 1,
+        LOW                 = PLATFORM_MIN - 2,
+        MEDIUM              = PLATFORM_MIN - 3,
+        HIGH                = PLATFORM_MIN - 4,
+        VERY_HIGH           = PLATFORM_MIN - 5,
+        EXTREME             = PLATFORM_MIN - 6,
+        CRITICAL            = PLATFORM_MIN - 7,
+
+        /* Thread defaults */
+        MIXER               = EXTREME,
+        FEEDER              = CRITICAL,
+        STREAM              = VERY_HIGH,
+        FILE                = HIGH,
+        NONBLOCKING         = HIGH,
+        RECORD              = HIGH,
+        GEOMETRY            = LOW,
+        PROFILER            = MEDIUM,
+        STUDIO_UPDATE       = MEDIUM,
+        STUDIO_LOAD_BANK    = MEDIUM,
+        STUDIO_LOAD_SAMPLE  = MEDIUM,
+        CONVOLUTION1        = VERY_HIGH,
+        CONVOLUTION2        = VERY_HIGH
+
+    }
+
+    public enum THREAD_STACK_SIZE : uint
+    {
+        DEFAULT             = 0,
+        MIXER               = 80  * 1024,
+        FEEDER              = 16  * 1024,
+        STREAM              = 96  * 1024,
+        FILE                = 64  * 1024,
+        NONBLOCKING         = 112 * 1024,
+        RECORD              = 16  * 1024,
+        GEOMETRY            = 48  * 1024,
+        PROFILER            = 128 * 1024,
+        STUDIO_UPDATE       = 96  * 1024,
+        STUDIO_LOAD_BANK    = 96  * 1024,
+        STUDIO_LOAD_SAMPLE  = 96  * 1024,
+        CONVOLUTION1        = 16  * 1024,
+        CONVOLUTION2        = 16  * 1024
+    }
+
+    [Flags]
+    public enum THREAD_AFFINITY : long
+    {
+        /* Platform agnostic thread groupings */
+        GROUP_DEFAULT       = 0x4000000000000000,
+        GROUP_A             = 0x4000000000000001,
+        GROUP_B             = 0x4000000000000002,
+        GROUP_C             = 0x4000000000000003,
+
+        /* Thread defaults */
+        MIXER               = GROUP_A,
+        FEEDER              = GROUP_C,
+        STREAM              = GROUP_C,
+        FILE                = GROUP_C,
+        NONBLOCKING         = GROUP_C,
+        RECORD              = GROUP_C,
+        GEOMETRY            = GROUP_C,
+        PROFILER            = GROUP_C,
+        STUDIO_UPDATE       = GROUP_B,
+        STUDIO_LOAD_BANK    = GROUP_C,
+        STUDIO_LOAD_SAMPLE  = GROUP_C,
+        CONVOLUTION1        = GROUP_C,
+        CONVOLUTION2        = GROUP_C,
+
+        /* Core mask, valid up to 1 << 61 */
+        CORE_ALL            = 0,
+        CORE_0              = 1 << 0,
+        CORE_1              = 1 << 1,
+        CORE_2              = 1 << 2,
+        CORE_3              = 1 << 3,
+        CORE_4              = 1 << 4,
+        CORE_5              = 1 << 5,
+        CORE_6              = 1 << 6,
+        CORE_7              = 1 << 7,
+        CORE_8              = 1 << 8,
+        CORE_9              = 1 << 9,
+        CORE_10             = 1 << 10,
+        CORE_11             = 1 << 11,
+        CORE_12             = 1 << 12,
+        CORE_13             = 1 << 13,
+        CORE_14             = 1 << 14,
+        CORE_15             = 1 << 15
+    }
+
+    public enum THREAD_TYPE : int
+    {
+        MIXER,
+        FEEDER,
+        STREAM,
+        FILE,
+        NONBLOCKING,
+        RECORD,
+        GEOMETRY,
+        PROFILER,
+        STUDIO_UPDATE,
+        STUDIO_LOAD_BANK,
+        STUDIO_LOAD_SAMPLE,
+        CONVOLUTION1,
+        CONVOLUTION2,
+
+        MAX
     }
 
     /*
         FMOD System factory functions.  Use this to create an FMOD System Instance.  below you will see System init/close to get started.
     */
-    public struct Factory
+        public struct Factory
     {
 #if NETFRAMEWORK
         static Factory()
@@ -758,19 +972,21 @@ namespace FMOD
             DllLoader.PreloadDll(VERSION.dll);
         }
 #endif
-
         public static RESULT System_Create(out System system)
         {
-            return FMOD5_System_Create(out system.handle);
+            return FMOD5_System_Create(out system.handle, VERSION.number);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
-        private static extern RESULT FMOD5_System_Create(out IntPtr system);
+        private static extern RESULT FMOD5_System_Create(out IntPtr system, uint headerversion);
 
-#endregion
+        #endregion
     }
 
+    /*
+        FMOD global system functions (optional).
+    */
     public struct Memory
     {
         public static RESULT Initialize(IntPtr poolmem, int poollen, MEMORY_ALLOC_CALLBACK useralloc, MEMORY_REALLOC_CALLBACK userrealloc, MEMORY_FREE_CALLBACK userfree, MEMORY_TYPE memtypeflags = MEMORY_TYPE.ALL)
@@ -783,13 +999,13 @@ namespace FMOD
             return FMOD5_Memory_GetStats(out currentalloced, out maxalloced, blocking);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Memory_Initialize(IntPtr poolmem, int poollen, MEMORY_ALLOC_CALLBACK useralloc, MEMORY_REALLOC_CALLBACK userrealloc, MEMORY_FREE_CALLBACK userfree, MEMORY_TYPE memtypeflags);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Memory_GetStats  (out int currentalloced, out int maxalloced, bool blocking);
 
-#endregion
+        #endregion
     }
 
     public struct Debug
@@ -802,11 +1018,24 @@ namespace FMOD
             }
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Debug_Initialize(DEBUG_FLAGS flags, DEBUG_MODE mode, DEBUG_CALLBACK callback, byte[] filename);
 
-#endregion
+        #endregion
+    }
+
+    public struct Thread
+    {
+        public static RESULT SetAttributes(THREAD_TYPE type, THREAD_AFFINITY affinity = THREAD_AFFINITY.GROUP_DEFAULT, THREAD_PRIORITY priority = THREAD_PRIORITY.DEFAULT, THREAD_STACK_SIZE stacksize = THREAD_STACK_SIZE.DEFAULT)
+        {
+            return FMOD5_Thread_SetAttributes(type, affinity, priority, stacksize);
+        }
+
+        #region importfunctions
+        [DllImport(VERSION.dll)]
+        private static extern RESULT FMOD5_Thread_SetAttributes(THREAD_TYPE type, THREAD_AFFINITY affinity, THREAD_PRIORITY priority, THREAD_STACK_SIZE stacksize);
+        #endregion
     }
 
     /*
@@ -819,7 +1048,7 @@ namespace FMOD
             return FMOD5_System_Release(this.handle);
         }
 
-        // Pre-init functions.
+        // Setup functions.
         public RESULT setOutput(OUTPUTTYPE output)
         {
             return FMOD5_System_SetOutput(this.handle, output);
@@ -891,12 +1120,12 @@ namespace FMOD
         }
         public RESULT setAdvancedSettings(ref ADVANCEDSETTINGS settings)
         {
-            settings.cbSize = Marshal.SizeOf(settings);
+            settings.cbSize = Marshal.SizeOf<ADVANCEDSETTINGS>();
             return FMOD5_System_SetAdvancedSettings(this.handle, ref settings);
         }
         public RESULT getAdvancedSettings(ref ADVANCEDSETTINGS settings)
         {
-            settings.cbSize = Marshal.SizeOf(settings);
+            settings.cbSize = Marshal.SizeOf<ADVANCEDSETTINGS>();
             return FMOD5_System_GetAdvancedSettings(this.handle, ref settings);
         }
         public RESULT setCallback(SYSTEM_CALLBACK callback, SYSTEM_CALLBACK_TYPE callbackmask = SYSTEM_CALLBACK_TYPE.ALL)
@@ -972,24 +1201,10 @@ namespace FMOD
         {
             return FMOD5_System_GetDSPInfoByPlugin(this.handle, handle, out description);
         }
-        /*
-        public RESULT registerCodec          (ref CODEC_DESCRIPTION description, out uint handle, uint priority)
-        public RESULT registerCodec          (ref CODEC_DESCRIPTION description, out uint handle, uint priority = 0)
-        public RESULT registerCodec(ref CODEC_DESCRIPTION description, out uint handle, uint priority)
-        {
-            return FMOD5_System_RegisterCodec(this.handle, ref description, out handle, priority);
-        }
-        */
         public RESULT registerDSP(ref DSP_DESCRIPTION description, out uint handle)
         {
             return FMOD5_System_RegisterDSP(this.handle, ref description, out handle);
         }
-        /*
-        public RESULT registerOutput(ref OUTPUT_DESCRIPTION description, out uint handle)
-        {
-            return FMOD5_System_RegisterOutput(this.handle, ref description, out handle);
-        }
-        */
 
         // Init/Close.
         public RESULT init(int maxchannels, INITFLAGS flags, IntPtr extradriverdata)
@@ -1070,7 +1285,12 @@ namespace FMOD
         // System information functions.
         public RESULT getVersion(out uint version)
         {
-            return FMOD5_System_GetVersion(this.handle, out version);
+            uint buildnumber;
+            return getVersion(out version, out buildnumber);
+        }
+        public RESULT getVersion(out uint version, out uint buildnumber)
+        {
+            return FMOD5_System_GetVersion(this.handle, out version, out buildnumber);
         }
         public RESULT getOutputHandle(out IntPtr handle)
         {
@@ -1084,9 +1304,9 @@ namespace FMOD
         {
             return FMOD5_System_GetChannelsPlaying(this.handle, out channels, out realchannels);
         }
-        public RESULT getCPUUsage(out float dsp, out float stream, out float geometry, out float update, out float total)
+        public RESULT getCPUUsage(out CPU_USAGE usage)
         {
-            return FMOD5_System_GetCPUUsage(this.handle, out dsp, out stream, out geometry, out update, out total);
+            return FMOD5_System_GetCPUUsage(this.handle, out usage);
         }
         public RESULT getFileUsage(out Int64 sampleBytesRead, out Int64 streamBytesRead, out Int64 otherBytesRead)
         {
@@ -1112,7 +1332,7 @@ namespace FMOD
         public RESULT createSound(string name, MODE mode, out Sound sound)
         {
             CREATESOUNDEXINFO exinfo = new CREATESOUNDEXINFO();
-            exinfo.cbsize = Marshal.SizeOf(exinfo);
+            exinfo.cbsize = Marshal.SizeOf<CREATESOUNDEXINFO>();
 
             return createSound(name, mode, ref exinfo, out sound);
         }
@@ -1134,7 +1354,7 @@ namespace FMOD
         public RESULT createStream(string name, MODE mode, out Sound sound)
         {
             CREATESOUNDEXINFO exinfo = new CREATESOUNDEXINFO();
-            exinfo.cbsize = Marshal.SizeOf(exinfo);
+            exinfo.cbsize = Marshal.SizeOf<CREATESOUNDEXINFO>();
 
             return createStream(name, mode, ref exinfo, out sound);
         }
@@ -1176,6 +1396,10 @@ namespace FMOD
         {
             return FMOD5_System_GetChannel(this.handle, channelid, out channel.handle);
         }
+        public RESULT getDSPInfoByType(DSP_TYPE type, out IntPtr description)
+        {
+            return FMOD5_System_GetDSPInfoByType(this.handle, type, out description);
+        }
         public RESULT getMasterChannelGroup(out ChannelGroup channelgroup)
         {
             return FMOD5_System_GetMasterChannelGroup(this.handle, out channelgroup.handle);
@@ -1186,7 +1410,7 @@ namespace FMOD
         }
 
         // Routing to ports.
-        public RESULT attachChannelGroupToPort(uint portType, ulong portIndex, ChannelGroup channelgroup, bool passThru = false)
+        public RESULT attachChannelGroupToPort(PORT_TYPE portType, ulong portIndex, ChannelGroup channelgroup, bool passThru = false)
         {
             return FMOD5_System_AttachChannelGroupToPort(this.handle, portType, portIndex, channelgroup.handle, passThru);
         }
@@ -1317,7 +1541,7 @@ namespace FMOD
             return FMOD5_System_GetUserData(this.handle, out userdata);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_Release                   (IntPtr system);
         [DllImport(VERSION.dll)]
@@ -1378,12 +1602,8 @@ namespace FMOD
         private static extern RESULT FMOD5_System_CreateDSPByPlugin         (IntPtr system, uint handle, out IntPtr dsp);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetDSPInfoByPlugin        (IntPtr system, uint handle, out IntPtr description);
-        //[DllImport(VERSION.dll)]
-        //private static extern RESULT FMOD5_System_RegisterCodec           (IntPtr system, out CODEC_DESCRIPTION description, out uint handle, uint priority);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_RegisterDSP               (IntPtr system, ref DSP_DESCRIPTION description, out uint handle);
-        //[DllImport(VERSION.dll)]
-        //private static extern RESULT FMOD5_System_RegisterOutput          (IntPtr system, ref OUTPUT_DESCRIPTION description, out uint handle);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_Init                      (IntPtr system, int maxchannels, INITFLAGS flags, IntPtr extradriverdata);
         [DllImport(VERSION.dll)]
@@ -1421,7 +1641,7 @@ namespace FMOD
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetSpeakerModeChannels    (IntPtr system, SPEAKERMODE mode, out int channels);
         [DllImport(VERSION.dll)]
-        private static extern RESULT FMOD5_System_GetVersion                (IntPtr system, out uint version);
+        private static extern RESULT FMOD5_System_GetVersion                (IntPtr system, out uint version, out uint buildnumber);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetOutputHandle           (IntPtr system, out IntPtr handle);
         [DllImport(VERSION.dll)]
@@ -1429,7 +1649,7 @@ namespace FMOD
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetChannelsPlaying        (IntPtr system, out int channels, out int realchannels);
         [DllImport(VERSION.dll)]
-        private static extern RESULT FMOD5_System_GetCPUUsage               (IntPtr system, out float dsp, out float stream, out float geometry, out float update, out float total);
+        private static extern RESULT FMOD5_System_GetCPUUsage               (IntPtr system, out CPU_USAGE usage);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetFileUsage              (IntPtr system, out Int64 sampleBytesRead, out Int64 streamBytesRead, out Int64 otherBytesRead);
         [DllImport(VERSION.dll)]
@@ -1457,11 +1677,13 @@ namespace FMOD
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetChannel                (IntPtr system, int channelid, out IntPtr channel);
         [DllImport(VERSION.dll)]
+        private static extern RESULT FMOD5_System_GetDSPInfoByType          (IntPtr system, DSP_TYPE type, out IntPtr description);
+        [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetMasterChannelGroup     (IntPtr system, out IntPtr channelgroup);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetMasterSoundGroup       (IntPtr system, out IntPtr soundgroup);
         [DllImport(VERSION.dll)]
-        private static extern RESULT FMOD5_System_AttachChannelGroupToPort  (IntPtr system, uint portType, ulong portIndex, IntPtr channelgroup, bool passThru);
+        private static extern RESULT FMOD5_System_AttachChannelGroupToPort  (IntPtr system, PORT_TYPE portType, ulong portIndex, IntPtr channelgroup, bool passThru);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_DetachChannelGroupFromPort(IntPtr system, IntPtr channelgroup);
         [DllImport(VERSION.dll)]
@@ -1506,16 +1728,17 @@ namespace FMOD
         private static extern RESULT FMOD5_System_SetUserData               (IntPtr system, IntPtr userdata);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_System_GetUserData               (IntPtr system, out IntPtr userdata);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
+        public System(IntPtr ptr)   { this.handle = ptr; }
         public bool hasHandle()     { return this.handle != IntPtr.Zero; }
         public void clearHandle()   { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
 
@@ -1614,7 +1837,7 @@ namespace FMOD
         }
         public RESULT getTag(string name, int index, out TAG tag)
         {
-             using (StringHelper.ThreadSafeEncoding encoder = StringHelper.GetFreeHelper())
+            using (StringHelper.ThreadSafeEncoding encoder = StringHelper.GetFreeHelper())
             {
                 return FMOD5_Sound_GetTag(this.handle, encoder.byteFromStringUTF8(name), index, out tag);
             }
@@ -1623,9 +1846,13 @@ namespace FMOD
         {
             return FMOD5_Sound_GetOpenState(this.handle, out openstate, out percentbuffered, out starving, out diskbusy);
         }
-        public RESULT readData(IntPtr buffer, uint length, out uint read)
+        public RESULT readData(byte[] buffer)
         {
-            return FMOD5_Sound_ReadData(this.handle, buffer, length, out read);
+            return FMOD5_Sound_ReadData(this.handle, buffer, (uint)buffer.Length, IntPtr.Zero);
+        }
+        public RESULT readData(byte[] buffer, out uint read)
+        {
+            return FMOD5_Sound_ReadData(this.handle, buffer, (uint)buffer.Length, out read);
         }
         public RESULT seekData(uint pcm)
         {
@@ -1736,7 +1963,7 @@ namespace FMOD
             return FMOD5_Sound_GetUserData(this.handle, out userdata);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Sound_Release                 (IntPtr sound);
         [DllImport(VERSION.dll)]
@@ -1780,7 +2007,9 @@ namespace FMOD
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Sound_GetOpenState            (IntPtr sound, out OPENSTATE openstate, out uint percentbuffered, out bool starving, out bool diskbusy);
         [DllImport(VERSION.dll)]
-        private static extern RESULT FMOD5_Sound_ReadData                (IntPtr sound, IntPtr buffer, uint length, out uint read);
+        private static extern RESULT FMOD5_Sound_ReadData                (IntPtr sound, byte[] buffer, uint length, IntPtr zero);
+        [DllImport(VERSION.dll)]
+        private static extern RESULT FMOD5_Sound_ReadData                (IntPtr sound, byte[] buffer, uint length, out uint read);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Sound_SeekData                (IntPtr sound, uint pcm);
         [DllImport(VERSION.dll)]
@@ -1823,16 +2052,17 @@ namespace FMOD
         private static extern RESULT FMOD5_Sound_SetUserData             (IntPtr sound, IntPtr userdata);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Sound_GetUserData             (IntPtr sound, out IntPtr userdata);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
+        public Sound(IntPtr ptr)    { this.handle = ptr; }
         public bool hasHandle()     { return this.handle != IntPtr.Zero; }
         public void clearHandle()   { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
     /*
@@ -1864,7 +2094,7 @@ namespace FMOD
         RESULT setCallback                  (CHANNELCONTROL_CALLBACK callback);
         RESULT isPlaying                    (out bool isplaying);
 
-        // Panning and level adjustment.
+        // Note all 'set' functions alter a final matrix, this is why the only get function is getMixMatrix, to avoid other get functions returning incorrect/obsolete values.
         RESULT setPan                       (float pan);
         RESULT setMixLevelsOutput           (float frontleft, float frontright, float center, float lfe, float surroundleft, float surroundright, float backleft, float backright);
         RESULT setMixLevelsInput            (float[] levels, int numlevels);
@@ -1984,6 +2214,7 @@ namespace FMOD
         {
             return FMOD5_Channel_GetIndex(this.handle, out index);
         }
+
         public RESULT getSystemObject(out System system)
         {
             return FMOD5_Channel_GetSystemObject(this.handle, out system.handle);
@@ -2071,7 +2302,7 @@ namespace FMOD
             return FMOD5_Channel_IsPlaying(this.handle, out isplaying);
         }
 
-        // Panning and level adjustment.
+        // Note all 'set' functions alter a final matrix, this is why the only get function is getMixMatrix, to avoid other get functions returning incorrect/obsolete values.
         public RESULT setPan(float pan)
         {
             return FMOD5_Channel_SetPan(this.handle, pan);
@@ -2245,7 +2476,7 @@ namespace FMOD
             return FMOD5_Channel_GetUserData(this.handle, out userdata);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Channel_SetFrequency         (IntPtr channel, float frequency);
         [DllImport(VERSION.dll)]
@@ -2400,16 +2631,17 @@ namespace FMOD
         private static extern RESULT FMOD5_Channel_SetUserData          (IntPtr channel, IntPtr userdata);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Channel_GetUserData          (IntPtr channel, out IntPtr userdata);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
+        public Channel(IntPtr ptr)  { this.handle = ptr; }
         public bool hasHandle()     { return this.handle != IntPtr.Zero; }
         public void clearHandle()   { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
     /*
@@ -2554,7 +2786,7 @@ namespace FMOD
             return FMOD5_ChannelGroup_IsPlaying(this.handle, out isplaying);
         }
 
-        // Panning and level adjustment.
+        // Note all 'set' functions alter a final matrix, this is why the only get function is getMixMatrix, to avoid other get functions returning incorrect/obsolete values.
         public RESULT setPan(float pan)
         {
             return FMOD5_ChannelGroup_SetPan(this.handle, pan);
@@ -2728,7 +2960,7 @@ namespace FMOD
             return FMOD5_ChannelGroup_GetUserData(this.handle, out userdata);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_ChannelGroup_Release             (IntPtr channelgroup);
         [DllImport(VERSION.dll)]
@@ -2871,16 +3103,17 @@ namespace FMOD
         private static extern RESULT FMOD5_ChannelGroup_SetUserData         (IntPtr channelgroup, IntPtr userdata);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_ChannelGroup_GetUserData         (IntPtr channelgroup, out IntPtr userdata);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
-        public bool hasHandle()     { return this.handle != IntPtr.Zero; }
-        public void clearHandle()   { this.handle = IntPtr.Zero; }
+        public ChannelGroup(IntPtr ptr) { this.handle = ptr; }
+        public bool hasHandle()         { return this.handle != IntPtr.Zero; }
+        public void clearHandle()       { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
     /*
@@ -2973,7 +3206,7 @@ namespace FMOD
             return FMOD5_SoundGroup_GetUserData(this.handle, out userdata);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_SoundGroup_Release               (IntPtr soundgroup);
         [DllImport(VERSION.dll)]
@@ -3008,16 +3241,17 @@ namespace FMOD
         private static extern RESULT FMOD5_SoundGroup_SetUserData           (IntPtr soundgroup, IntPtr userdata);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_SoundGroup_GetUserData           (IntPtr soundgroup, out IntPtr userdata);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
-        public bool hasHandle()     { return this.handle != IntPtr.Zero; }
-        public void clearHandle()   { this.handle = IntPtr.Zero; }
+        public SoundGroup(IntPtr ptr) { this.handle = ptr; }
+        public bool hasHandle()       { return this.handle != IntPtr.Zero; }
+        public void clearHandle()     { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
     /*
@@ -3109,6 +3343,10 @@ namespace FMOD
         {
             return FMOD5_DSP_Reset(this.handle);
         }
+        public RESULT setCallback(DSP_CALLBACK callback)
+        {
+            return FMOD5_DSP_SetCallback(this.handle, callback);
+        }
 
         // DSP parameter control.
         public RESULT setParameterFloat(int index, float value)
@@ -3151,11 +3389,7 @@ namespace FMOD
         {
             IntPtr descPtr;
             RESULT result = FMOD5_DSP_GetParameterInfo(this.handle, index, out descPtr);
-#if (UNITY_2017_4_OR_NEWER) && !NET_4_6
-            desc = (DSP_PARAMETER_DESC)Marshal.PtrToStructure(descPtr, typeof(DSP_PARAMETER_DESC));
-#else
-            desc = Marshal.PtrToStructure<DSP_PARAMETER_DESC>(descPtr);
-#endif // (UNITY_2017_4_OR_NEWER) && !NET_4_6
+            desc = (DSP_PARAMETER_DESC)Marshal.PtrToStructure<DSP_PARAMETER_DESC>(descPtr);
             return result;
         }
         public RESULT getDataParameterIndex(int datatype, out int index)
@@ -3231,7 +3465,7 @@ namespace FMOD
             return FMOD5_DSP_GetCPUUsage(this.handle, out exclusive, out inclusive);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_DSP_Release                   (IntPtr dsp);
         [DllImport(VERSION.dll)]
@@ -3273,6 +3507,8 @@ namespace FMOD
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_DSP_Reset                     (IntPtr dsp);
         [DllImport(VERSION.dll)]
+        private static extern RESULT FMOD5_DSP_SetCallback               (IntPtr dsp, DSP_CALLBACK callback);
+        [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_DSP_SetParameterFloat         (IntPtr dsp, int index, float value);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_DSP_SetParameterInt           (IntPtr dsp, int index, int value);
@@ -3318,16 +3554,17 @@ namespace FMOD
         public static extern RESULT FMOD5_DSP_GetMeteringInfo            (IntPtr dsp, out DSP_METERING_INFO inputInfo, out DSP_METERING_INFO outputInfo);
         [DllImport(VERSION.dll)]
         public static extern RESULT FMOD5_DSP_GetCPUUsage                (IntPtr dsp, out uint exclusive, out uint inclusive);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
+        public DSP(IntPtr ptr)      { this.handle = ptr; }
         public bool hasHandle()     { return this.handle != IntPtr.Zero; }
         public void clearHandle()   { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
     /*
@@ -3374,7 +3611,7 @@ namespace FMOD
             return FMOD5_DSPConnection_GetUserData(this.handle, out userdata);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_DSPConnection_GetInput        (IntPtr dspconnection, out IntPtr input);
         [DllImport(VERSION.dll)]
@@ -3393,16 +3630,17 @@ namespace FMOD
         private static extern RESULT FMOD5_DSPConnection_SetUserData     (IntPtr dspconnection, IntPtr userdata);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_DSPConnection_GetUserData     (IntPtr dspconnection, out IntPtr userdata);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
-        public bool hasHandle()     { return this.handle != IntPtr.Zero; }
-        public void clearHandle()   { this.handle = IntPtr.Zero; }
+        public DSPConnection(IntPtr ptr) { this.handle = ptr; }
+        public bool hasHandle()          { return this.handle != IntPtr.Zero; }
+        public void clearHandle()        { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
     /*
@@ -3497,7 +3735,7 @@ namespace FMOD
             return FMOD5_Geometry_GetUserData(this.handle, out userdata);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Geometry_Release              (IntPtr geometry);
         [DllImport(VERSION.dll)]
@@ -3538,16 +3776,17 @@ namespace FMOD
         private static extern RESULT FMOD5_Geometry_SetUserData          (IntPtr geometry, IntPtr userdata);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Geometry_GetUserData          (IntPtr geometry, out IntPtr userdata);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
+        public Geometry(IntPtr ptr) { this.handle = ptr; }
         public bool hasHandle()     { return this.handle != IntPtr.Zero; }
         public void clearHandle()   { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
     /*
@@ -3596,7 +3835,7 @@ namespace FMOD
             return FMOD5_Reverb3D_GetUserData(this.handle, out userdata);
         }
 
-#region importfunctions
+        #region importfunctions
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Reverb3D_Release             (IntPtr reverb3d);
         [DllImport(VERSION.dll)]
@@ -3615,24 +3854,30 @@ namespace FMOD
         private static extern RESULT FMOD5_Reverb3D_SetUserData         (IntPtr reverb3d, IntPtr userdata);
         [DllImport(VERSION.dll)]
         private static extern RESULT FMOD5_Reverb3D_GetUserData         (IntPtr reverb3d, out IntPtr userdata);
-#endregion
+        #endregion
 
-#region wrapperinternal
+        #region wrapperinternal
 
         public IntPtr handle;
 
+        public Reverb3D(IntPtr ptr) { this.handle = ptr; }
         public bool hasHandle()     { return this.handle != IntPtr.Zero; }
         public void clearHandle()   { this.handle = IntPtr.Zero; }
 
-#endregion
+        #endregion
     }
 
-#region String Helper Functions
+    #region Helper Functions
     [StructLayout(LayoutKind.Sequential)]
     public struct StringWrapper
     {
         IntPtr nativeUtf8Ptr;
 
+        public StringWrapper(IntPtr ptr)
+        {
+            nativeUtf8Ptr = ptr;
+        }
+
         public static implicit operator string(StringWrapper fstring)
         {
             using (StringHelper.ThreadSafeEncoding encoder = StringHelper.GetFreeHelper())
@@ -3640,6 +3885,47 @@ namespace FMOD
                 return encoder.stringFromNative(fstring.nativeUtf8Ptr);
             }
         }
+
+        public bool StartsWith(byte[] prefix)
+        {
+            if (nativeUtf8Ptr == IntPtr.Zero)
+            {
+                return false;
+            }
+
+            for (int i = 0; i < prefix.Length; i++)
+            {
+                if (Marshal.ReadByte(nativeUtf8Ptr, i) != prefix[i])
+                {
+                    return false;
+                }
+            }
+
+            return true;
+        }
+
+        public bool Equals(byte[] comparison)
+        {
+            if (nativeUtf8Ptr == IntPtr.Zero)
+            {
+                return false;
+            }
+
+            for (int i = 0; i < comparison.Length; i++)
+            {
+                if (Marshal.ReadByte(nativeUtf8Ptr, i) != comparison[i])
+                {
+                    return false;
+                }
+            }
+
+            if (Marshal.ReadByte(nativeUtf8Ptr, comparison.Length) != 0)
+            {
+                return false;
+            }
+
+            return true;
+        }
     }
 
     static class StringHelper
@@ -3781,5 +4067,5 @@ namespace FMOD
         }
     }
 
-#endregion
+    #endregion
 }
diff --git a/AssetStudioUtility/Audio/FMODStudioAPI/fmod_dsp.cs b/AssetStudioUtility/Audio/FMODStudioAPI/fmod_dsp.cs
new file mode 100644
index 0000000..02820ee
--- /dev/null
+++ b/AssetStudioUtility/Audio/FMODStudioAPI/fmod_dsp.cs
@@ -0,0 +1,1007 @@
+/* ======================================================================================== */
+/* FMOD Core API - DSP header file.                                                         */
+/* Copyright (c), Firelight Technologies Pty, Ltd. 2004-2025.                               */
+/*                                                                                          */
+/* Use this header if you are wanting to develop your own DSP plugin to use with FMODs      */
+/* dsp system.  With this header you can make your own DSP plugin that FMOD can             */
+/* register and use.  See the documentation and examples on how to make a working plugin.   */
+/*                                                                                          */
+/* For more detail visit:                                                                   */
+/* https://fmod.com/docs/2.03/api/plugin-api-dsp.html                                       */
+/* =========================================================================================*/
+
+using System;
+using System.Text;
+using System.Runtime.InteropServices;
+
+namespace FMOD
+{
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_BUFFER_ARRAY
+    {
+        public int              numbuffers;
+        public IntPtr           buffernumchannels;
+        public IntPtr           bufferchannelmask;
+        public IntPtr           buffers;
+        public SPEAKERMODE      speakermode;
+
+        /*
+            These properties take advantage of the fact that numbuffers is always zero or one
+        */
+
+        public int numchannels
+        {
+            get 
+            {
+                if (buffernumchannels != IntPtr.Zero && numbuffers != 0)
+                    return Marshal.ReadInt32(buffernumchannels);
+
+                return 0;
+            }
+            set
+            {
+                if (buffernumchannels != IntPtr.Zero && numbuffers != 0)
+                    Marshal.WriteInt32(buffernumchannels, value);
+            }
+        }
+
+        public IntPtr buffer
+        {
+            get
+            {
+                if (buffers != IntPtr.Zero && numbuffers != 0)
+                    return Marshal.ReadIntPtr(buffers);
+
+                return IntPtr.Zero;
+            }
+            set
+            {
+                if (buffers != IntPtr.Zero && numbuffers != 0)
+                    Marshal.WriteIntPtr(buffers, value);
+            }
+        }
+    }
+
+    public enum DSP_PROCESS_OPERATION
+    {
+        PROCESS_PERFORM = 0,
+        PROCESS_QUERY
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct COMPLEX
+    {
+        public float real;
+        public float imag;
+    }
+
+    public enum DSP_PAN_SURROUND_FLAGS
+    {
+        DEFAULT = 0,
+        ROTATION_NOT_BIASED = 1,
+    }
+
+
+    /*
+        DSP callbacks
+    */
+    public delegate RESULT DSP_CREATE_CALLBACK                  (ref DSP_STATE dsp_state);
+    public delegate RESULT DSP_RELEASE_CALLBACK                 (ref DSP_STATE dsp_state);
+    public delegate RESULT DSP_RESET_CALLBACK                   (ref DSP_STATE dsp_state);
+    public delegate RESULT DSP_SETPOSITION_CALLBACK             (ref DSP_STATE dsp_state, uint pos);
+    public delegate RESULT DSP_READ_CALLBACK                    (ref DSP_STATE dsp_state, IntPtr inbuffer, IntPtr outbuffer, uint length, int inchannels, ref int outchannels);
+    public delegate RESULT DSP_SHOULDIPROCESS_CALLBACK          (ref DSP_STATE dsp_state, bool inputsidle, uint length, CHANNELMASK inmask, int inchannels, SPEAKERMODE speakermode);
+    public delegate RESULT DSP_PROCESS_CALLBACK                 (ref DSP_STATE dsp_state, uint length, ref DSP_BUFFER_ARRAY inbufferarray, ref DSP_BUFFER_ARRAY outbufferarray, bool inputsidle, DSP_PROCESS_OPERATION op);
+    public delegate RESULT DSP_SETPARAM_FLOAT_CALLBACK          (ref DSP_STATE dsp_state, int index, float value);
+    public delegate RESULT DSP_SETPARAM_INT_CALLBACK            (ref DSP_STATE dsp_state, int index, int value);
+    public delegate RESULT DSP_SETPARAM_BOOL_CALLBACK           (ref DSP_STATE dsp_state, int index, bool value);
+    public delegate RESULT DSP_SETPARAM_DATA_CALLBACK           (ref DSP_STATE dsp_state, int index, IntPtr data, uint length);
+    public delegate RESULT DSP_GETPARAM_FLOAT_CALLBACK          (ref DSP_STATE dsp_state, int index, ref float value, IntPtr valuestr);
+    public delegate RESULT DSP_GETPARAM_INT_CALLBACK            (ref DSP_STATE dsp_state, int index, ref int value, IntPtr valuestr);
+    public delegate RESULT DSP_GETPARAM_BOOL_CALLBACK           (ref DSP_STATE dsp_state, int index, ref bool value, IntPtr valuestr);
+    public delegate RESULT DSP_GETPARAM_DATA_CALLBACK           (ref DSP_STATE dsp_state, int index, ref IntPtr data, ref uint length, IntPtr valuestr);
+    public delegate RESULT DSP_SYSTEM_REGISTER_CALLBACK         (ref DSP_STATE dsp_state);
+    public delegate RESULT DSP_SYSTEM_DEREGISTER_CALLBACK       (ref DSP_STATE dsp_state);
+    public delegate RESULT DSP_SYSTEM_MIX_CALLBACK              (ref DSP_STATE dsp_state, int stage);
+
+
+    /*
+        DSP functions
+    */
+    public delegate IntPtr DSP_ALLOC_FUNC                         (uint size, MEMORY_TYPE type, IntPtr sourcestr);
+    public delegate IntPtr DSP_REALLOC_FUNC                       (IntPtr ptr, uint size, MEMORY_TYPE type, IntPtr sourcestr);
+    public delegate void   DSP_FREE_FUNC                          (IntPtr ptr, MEMORY_TYPE type, IntPtr sourcestr);
+    public delegate void   DSP_LOG_FUNC                           (DEBUG_FLAGS level, IntPtr file, int line, IntPtr function, IntPtr str);
+    public delegate RESULT DSP_GETSAMPLERATE_FUNC                 (ref DSP_STATE dsp_state, ref int rate);
+    public delegate RESULT DSP_GETBLOCKSIZE_FUNC                  (ref DSP_STATE dsp_state, ref uint blocksize);
+    public delegate RESULT DSP_GETSPEAKERMODE_FUNC                (ref DSP_STATE dsp_state, ref int speakermode_mixer, ref int speakermode_output);
+    public delegate RESULT DSP_GETCLOCK_FUNC                      (ref DSP_STATE dsp_state, out ulong clock, out uint offset, out uint length);
+    public delegate RESULT DSP_GETLISTENERATTRIBUTES_FUNC         (ref DSP_STATE dsp_state, ref int numlisteners, IntPtr attributes);
+    public delegate RESULT DSP_GETUSERDATA_FUNC                   (ref DSP_STATE dsp_state, out IntPtr userdata);
+    public delegate RESULT DSP_DFT_FFTREAL_FUNC                   (ref DSP_STATE dsp_state, int size, IntPtr signal, IntPtr dft, IntPtr window, int signalhop);
+    public delegate RESULT DSP_DFT_IFFTREAL_FUNC                  (ref DSP_STATE dsp_state, int size, IntPtr dft, IntPtr signal, IntPtr window, int signalhop);
+    public delegate RESULT DSP_PAN_SUMMONOMATRIX_FUNC             (ref DSP_STATE dsp_state, int sourceSpeakerMode, float lowFrequencyGain, float overallGain, IntPtr matrix);
+    public delegate RESULT DSP_PAN_SUMSTEREOMATRIX_FUNC           (ref DSP_STATE dsp_state, int sourceSpeakerMode, float pan, float lowFrequencyGain, float overallGain, int matrixHop, IntPtr matrix);
+    public delegate RESULT DSP_PAN_SUMSURROUNDMATRIX_FUNC         (ref DSP_STATE dsp_state, int sourceSpeakerMode, int targetSpeakerMode, float direction, float extent, float rotation, float lowFrequencyGain, float overallGain, int matrixHop, IntPtr matrix, DSP_PAN_SURROUND_FLAGS flags);
+    public delegate RESULT DSP_PAN_SUMMONOTOSURROUNDMATRIX_FUNC   (ref DSP_STATE dsp_state, int targetSpeakerMode, float direction, float extent, float lowFrequencyGain, float overallGain, int matrixHop, IntPtr matrix);
+    public delegate RESULT DSP_PAN_SUMSTEREOTOSURROUNDMATRIX_FUNC (ref DSP_STATE dsp_state, int targetSpeakerMode, float direction, float extent, float rotation, float lowFrequencyGain, float overallGain, int matrixHop, IntPtr matrix);
+    public delegate RESULT DSP_PAN_GETROLLOFFGAIN_FUNC            (ref DSP_STATE dsp_state, DSP_PAN_3D_ROLLOFF_TYPE rolloff, float distance, float mindistance, float maxdistance, out float gain);
+
+
+    public enum DSP_TYPE : int
+    {
+        UNKNOWN,
+        MIXER,
+        OSCILLATOR,
+        LOWPASS,
+        ITLOWPASS,
+        HIGHPASS,
+        ECHO,
+        FADER,
+        FLANGE,
+        DISTORTION,
+        NORMALIZE,
+        LIMITER,
+        PARAMEQ,
+        PITCHSHIFT,
+        CHORUS,
+        ITECHO,
+        COMPRESSOR,
+        SFXREVERB,
+        LOWPASS_SIMPLE,
+        DELAY,
+        TREMOLO,
+        SEND,
+        RETURN,
+        HIGHPASS_SIMPLE,
+        PAN,
+        THREE_EQ,
+        FFT,
+        LOUDNESS_METER,
+        CONVOLUTIONREVERB,
+        CHANNELMIX,
+        TRANSCEIVER,
+        OBJECTPAN,
+        MULTIBAND_EQ,
+        MULTIBAND_DYNAMICS,
+        MAX
+    }
+
+    public enum DSP_PARAMETER_TYPE
+    {
+        FLOAT = 0,
+        INT,
+        BOOL,
+        DATA,
+        MAX
+    }
+
+    public enum DSP_PARAMETER_FLOAT_MAPPING_TYPE
+    {
+        DSP_PARAMETER_FLOAT_MAPPING_TYPE_LINEAR = 0,
+        DSP_PARAMETER_FLOAT_MAPPING_TYPE_AUTO,
+        DSP_PARAMETER_FLOAT_MAPPING_TYPE_PIECEWISE_LINEAR,
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_FLOAT_MAPPING_PIECEWISE_LINEAR
+    {
+        public int numpoints;
+        public IntPtr pointparamvalues;
+        public IntPtr pointpositions;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_FLOAT_MAPPING
+    {
+        public DSP_PARAMETER_FLOAT_MAPPING_TYPE type;
+        public DSP_PARAMETER_FLOAT_MAPPING_PIECEWISE_LINEAR piecewiselinearmapping;
+    }
+
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_DESC_FLOAT
+    {
+        public float                     min;
+        public float                     max;
+        public float                     defaultval;
+        public DSP_PARAMETER_FLOAT_MAPPING mapping;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_DESC_INT
+    {
+        public int                       min;
+        public int                       max;
+        public int                       defaultval;
+        public bool                      goestoinf;
+        public IntPtr                    valuenames;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_DESC_BOOL
+    {
+        public bool                      defaultval;
+        public IntPtr                    valuenames;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_DESC_DATA
+    {
+        public int                       datatype;
+    }
+
+    [StructLayout(LayoutKind.Explicit)]
+    public struct DSP_PARAMETER_DESC_UNION
+    {
+        [FieldOffset(0)]
+        public DSP_PARAMETER_DESC_FLOAT   floatdesc;
+        [FieldOffset(0)]
+        public DSP_PARAMETER_DESC_INT     intdesc;
+        [FieldOffset(0)]
+        public DSP_PARAMETER_DESC_BOOL    booldesc;
+        [FieldOffset(0)]
+        public DSP_PARAMETER_DESC_DATA    datadesc;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_DESC
+    {
+        public DSP_PARAMETER_TYPE   type;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 16)]
+        public byte[]               name;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 16)]
+        public byte[]               label;
+        public string               description;
+
+        public DSP_PARAMETER_DESC_UNION desc;
+    }
+
+    public enum DSP_PARAMETER_DATA_TYPE
+    {
+        DSP_PARAMETER_DATA_TYPE_USER =                       0,
+        DSP_PARAMETER_DATA_TYPE_OVERALLGAIN =               -1,
+        DSP_PARAMETER_DATA_TYPE_3DATTRIBUTES =              -2,
+        DSP_PARAMETER_DATA_TYPE_SIDECHAIN =                 -3,
+        DSP_PARAMETER_DATA_TYPE_FFT =                       -4,
+        DSP_PARAMETER_DATA_TYPE_3DATTRIBUTES_MULTI =        -5,
+        DSP_PARAMETER_DATA_TYPE_ATTENUATION_RANGE =         -6,
+        DSP_PARAMETER_DATA_TYPE_DYNAMIC_RESPONSE =          -7
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_OVERALLGAIN
+    {
+        public float linear_gain;
+        public float linear_gain_additive;
+    }
+    
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_3DATTRIBUTES
+    {
+        public ATTRIBUTES_3D relative;
+        public ATTRIBUTES_3D absolute;
+    }
+    
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_3DATTRIBUTES_MULTI
+    {
+        public int            numlisteners;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 8)]
+        public ATTRIBUTES_3D[] relative;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 8)]
+        public float[] weight;
+        public ATTRIBUTES_3D absolute;
+    }
+    
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_SIDECHAIN
+    {
+        public int sidechainenable;
+    }
+    
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_FFT
+    {
+        public int     length;
+        public int     numchannels;
+        
+        [MarshalAs(UnmanagedType.ByValArray,SizeConst=32)]
+        private IntPtr[] spectrum_internal;
+
+        public float[][] spectrum
+        {
+            get
+            {
+                var buffer = new float[numchannels][];
+                
+                for (int i = 0; i < numchannels; ++i)
+                {
+                    buffer[i] = new float[length];
+                    Marshal.Copy(spectrum_internal[i], buffer[i], 0, length);
+                }
+                
+                return buffer;
+            }
+        }
+
+        public void getSpectrum(ref float[][] buffer)
+        {
+            int bufferLength = Math.Min(buffer.Length, numchannels);
+            for (int i = 0; i < bufferLength; ++i)
+            {
+                getSpectrum(i, ref buffer[i]);
+            }
+        }
+
+        public void getSpectrum(int channel, ref float[] buffer)
+        {
+            int bufferLength = Math.Min(buffer.Length, length);
+            Marshal.Copy(spectrum_internal[channel], buffer, 0, bufferLength);
+        }
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_DYNAMIC_RESPONSE
+    {
+        public int numchannels;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 32)]
+        public float[] rms;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_LOUDNESS_METER_INFO_TYPE
+    {
+        public float momentaryloudness;
+        public float shorttermloudness;
+        public float integratedloudness;
+        public float loudness10thpercentile;
+        public float loudness95thpercentile;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 66)]
+        public float[] loudnesshistogram;
+        public float maxtruepeak;
+        public float maxmomentaryloudness;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_LOUDNESS_METER_WEIGHTING_TYPE
+    {
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 32)]
+        public float[] channelweight;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_PARAMETER_ATTENUATION_RANGE
+    {
+        public float min;
+        public float max;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_DESCRIPTION
+    {
+        public uint                           pluginsdkversion;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 32)]
+        public byte[]                         name;
+        public uint                           version;
+        public int                            numinputbuffers;
+        public int                            numoutputbuffers;
+        public DSP_CREATE_CALLBACK            create;
+        public DSP_RELEASE_CALLBACK           release;
+        public DSP_RESET_CALLBACK             reset;
+        public DSP_READ_CALLBACK              read;
+        public DSP_PROCESS_CALLBACK           process;
+        public DSP_SETPOSITION_CALLBACK       setposition;
+
+        public int                            numparameters;
+        public IntPtr                         paramdesc;
+        public DSP_SETPARAM_FLOAT_CALLBACK    setparameterfloat;
+        public DSP_SETPARAM_INT_CALLBACK      setparameterint;
+        public DSP_SETPARAM_BOOL_CALLBACK     setparameterbool;
+        public DSP_SETPARAM_DATA_CALLBACK     setparameterdata;
+        public DSP_GETPARAM_FLOAT_CALLBACK    getparameterfloat;
+        public DSP_GETPARAM_INT_CALLBACK      getparameterint;
+        public DSP_GETPARAM_BOOL_CALLBACK     getparameterbool;
+        public DSP_GETPARAM_DATA_CALLBACK     getparameterdata;
+        public DSP_SHOULDIPROCESS_CALLBACK    shouldiprocess;
+        public IntPtr                         userdata;
+
+        public DSP_SYSTEM_REGISTER_CALLBACK   sys_register;
+        public DSP_SYSTEM_DEREGISTER_CALLBACK sys_deregister;
+        public DSP_SYSTEM_MIX_CALLBACK        sys_mix;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_STATE_DFT_FUNCTIONS
+    {
+        public DSP_DFT_FFTREAL_FUNC  fftreal;
+        public DSP_DFT_IFFTREAL_FUNC inversefftreal;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_STATE_PAN_FUNCTIONS
+    {
+        public DSP_PAN_SUMMONOMATRIX_FUNC             summonomatrix;
+        public DSP_PAN_SUMSTEREOMATRIX_FUNC           sumstereomatrix;
+        public DSP_PAN_SUMSURROUNDMATRIX_FUNC         sumsurroundmatrix;
+        public DSP_PAN_SUMMONOTOSURROUNDMATRIX_FUNC   summonotosurroundmatrix;
+        public DSP_PAN_SUMSTEREOTOSURROUNDMATRIX_FUNC sumstereotosurroundmatrix;
+        public DSP_PAN_GETROLLOFFGAIN_FUNC            getrolloffgain;
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_STATE_FUNCTIONS
+    {
+        public DSP_ALLOC_FUNC                  alloc;
+        public DSP_REALLOC_FUNC                realloc;
+        public DSP_FREE_FUNC                   free;
+        public DSP_GETSAMPLERATE_FUNC          getsamplerate;
+        public DSP_GETBLOCKSIZE_FUNC           getblocksize;
+        public IntPtr                          dft_internal;
+        public IntPtr                          pan_internal;
+        public DSP_GETSPEAKERMODE_FUNC         getspeakermode;
+        public DSP_GETCLOCK_FUNC               getclock;
+        public DSP_GETLISTENERATTRIBUTES_FUNC  getlistenerattributes;
+        public DSP_LOG_FUNC                    log;
+        public DSP_GETUSERDATA_FUNC            getuserdata;
+        public DSP_STATE_DFT_FUNCTIONS dft
+        {
+            get { return Marshal.PtrToStructure<FMOD.DSP_STATE_DFT_FUNCTIONS>(dft_internal); }
+        }
+        public DSP_STATE_PAN_FUNCTIONS pan
+        {
+            get { return Marshal.PtrToStructure<FMOD.DSP_STATE_PAN_FUNCTIONS>(pan_internal); }
+        }
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_STATE
+    {
+        public IntPtr     instance;
+        public IntPtr     plugindata;
+        public uint       channelmask;
+        public int        source_speakermode;
+        public IntPtr     sidechaindata;
+        public int        sidechainchannels;
+        private IntPtr    functions_internal;
+        public int        systemobject;
+
+        public DSP_STATE_FUNCTIONS functions
+        {
+            get { return Marshal.PtrToStructure<FMOD.DSP_STATE_FUNCTIONS>(functions_internal); }
+        }
+    }
+
+    [StructLayout(LayoutKind.Sequential)]
+    public struct DSP_METERING_INFO
+    {
+        public int   numsamples;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst=32)]
+        public float[] peaklevel;
+        [MarshalAs(UnmanagedType.ByValArray, SizeConst=32)]
+        public float[] rmslevel;
+        public short numchannels;
+    }
+
+    /*
+        ==============================================================================================================
+
+        FMOD built in effect parameters.
+        Use DSP::setParameter with these enums for the 'index' parameter.
+
+        ==============================================================================================================
+    */
+
+    public enum DSP_OSCILLATOR : int
+    {
+        TYPE,
+        RATE
+    }
+
+    public enum DSP_LOWPASS : int
+    {
+        CUTOFF,
+        RESONANCE
+    }
+
+    public enum DSP_ITLOWPASS : int
+    {
+        CUTOFF,
+        RESONANCE
+    }
+
+    public enum DSP_HIGHPASS : int
+    {
+        CUTOFF,
+        RESONANCE
+    }
+
+    public enum DSP_ECHO : int
+    {
+        DELAY,
+        FEEDBACK,
+        DRYLEVEL,
+        WETLEVEL,
+        DELAYCHANGEMODE
+    }
+
+    public enum DSP_ECHO_DELAYCHANGEMODE_TYPE : int
+    {
+        FADE,
+        LERP,
+        NONE
+    }
+
+    public enum DSP_FADER : int
+    {
+        GAIN,
+        OVERALL_GAIN,
+    }
+
+    public enum DSP_DELAY : int
+    {
+        CH0,
+        CH1,
+        CH2,
+        CH3,
+        CH4,
+        CH5,
+        CH6,
+        CH7,
+        CH8,
+        CH9,
+        CH10,
+        CH11,
+        CH12,
+        CH13,
+        CH14,
+        CH15,
+        MAXDELAY,
+    }
+
+    public enum DSP_FLANGE : int
+    {
+        MIX,
+        DEPTH,
+        RATE
+    }
+
+    public enum DSP_TREMOLO : int
+    {
+        FREQUENCY,
+        DEPTH,
+        SHAPE,
+        SKEW,
+        DUTY,
+        SQUARE,
+        PHASE,
+        SPREAD
+    }
+
+    public enum DSP_DISTORTION : int
+    {
+        LEVEL
+    }
+
+    public enum DSP_NORMALIZE : int
+    {
+        FADETIME,
+        THRESHOLD,
+        MAXAMP
+    }
+
+    public enum DSP_LIMITER : int
+    {
+        RELEASETIME,
+        CEILING,
+        MAXIMIZERGAIN,
+        MODE,
+    }
+
+    public enum DSP_PARAMEQ : int
+    {
+        CENTER,
+        BANDWIDTH,
+        GAIN
+    }
+
+    public enum DSP_MULTIBAND_EQ : int
+    {
+        A_FILTER,
+        A_FREQUENCY,
+        A_Q,
+        A_GAIN,
+        B_FILTER,
+        B_FREQUENCY,
+        B_Q,
+        B_GAIN,
+        C_FILTER,
+        C_FREQUENCY,
+        C_Q,
+        C_GAIN,
+        D_FILTER,
+        D_FREQUENCY,
+        D_Q,
+        D_GAIN,
+        E_FILTER,
+        E_FREQUENCY,
+        E_Q,
+        E_GAIN,
+    }
+
+    public enum DSP_MULTIBAND_EQ_FILTER_TYPE : int
+    {
+        DISABLED,
+        LOWPASS_12DB,
+        LOWPASS_24DB,
+        LOWPASS_48DB,
+        HIGHPASS_12DB,
+        HIGHPASS_24DB,
+        HIGHPASS_48DB,
+        LOWSHELF,
+        HIGHSHELF,
+        PEAKING,
+        BANDPASS,
+        NOTCH,
+        ALLPASS,
+        LOWPASS_6DB,
+        HIGHPASS_6DB,
+    }
+
+    public enum DSP_MULTIBAND_DYNAMICS : int
+    {
+        LOWER_FREQUENCY,
+        UPPER_FREQUENCY,
+        LINKED,
+        USE_SIDECHAIN,
+        A_MODE,
+        A_GAIN,
+        A_THRESHOLD,
+        A_RATIO,
+        A_ATTACK,
+        A_RELEASE,
+        A_GAIN_MAKEUP,
+        A_RESPONSE_DATA,
+        B_MODE,
+        B_GAIN,
+        B_THRESHOLD,
+        B_RATIO,
+        B_ATTACK,
+        B_RELEASE,
+        B_GAIN_MAKEUP,
+        B_RESPONSE_DATA,
+        C_MODE,
+        C_GAIN,
+        C_THRESHOLD,
+        C_RATIO,
+        C_ATTACK,
+        C_RELEASE,
+        C_GAIN_MAKEUP,
+        C_RESPONSE_DATA,
+    }
+
+    public enum DSP_MULTIBAND_DYNAMICS_MODE_TYPE : int
+    {
+        DISABLED,
+        COMPRESS_UP,
+        COMPRESS_DOWN,
+        EXPAND_UP,
+        EXPAND_DOWN
+    }
+
+    public enum DSP_PITCHSHIFT : int
+    {
+        PITCH,
+        FFTSIZE,
+        OVERLAP,
+        MAXCHANNELS
+    }
+
+    public enum DSP_CHORUS : int
+    {
+        MIX,
+        RATE,
+        DEPTH,
+    }
+
+    public enum DSP_ITECHO : int
+    {
+        WETDRYMIX,
+        FEEDBACK,
+        LEFTDELAY,
+        RIGHTDELAY,
+        PANDELAY
+    }
+
+    public enum DSP_COMPRESSOR : int
+    {
+        THRESHOLD,
+        RATIO,
+        ATTACK,
+        RELEASE,
+        GAINMAKEUP,
+        USESIDECHAIN,
+        LINKED
+    }
+
+    public enum DSP_SFXREVERB : int
+    {
+        DECAYTIME,
+        EARLYDELAY,
+        LATEDELAY,
+        HFREFERENCE,
+        HFDECAYRATIO,
+        DIFFUSION,
+        DENSITY,
+        LOWSHELFFREQUENCY,
+        LOWSHELFGAIN,
+        HIGHCUT,
+        EARLYLATEMIX,
+        WETLEVEL,
+        DRYLEVEL
+    }
+
+    public enum DSP_LOWPASS_SIMPLE : int
+    {
+        CUTOFF
+    }
+
+    public enum DSP_SEND : int
+    {
+        RETURNID,
+        LEVEL,
+    }
+
+    public enum DSP_RETURN : int
+    {
+        ID,
+        INPUT_SPEAKER_MODE
+    }
+
+    public enum DSP_HIGHPASS_SIMPLE : int
+    {
+        CUTOFF
+    }
+
+    public enum DSP_PAN_2D_STEREO_MODE_TYPE : int
+    {
+        DISTRIBUTED,
+        DISCRETE
+    }
+
+    public enum DSP_PAN_MODE_TYPE : int
+    {
+        MONO,
+        STEREO,
+        SURROUND
+    }
+
+    public enum DSP_PAN_3D_ROLLOFF_TYPE : int
+    {
+        LINEARSQUARED,
+        LINEAR,
+        INVERSE,
+        INVERSETAPERED,
+        CUSTOM
+    }
+
+    public enum DSP_PAN_3D_EXTENT_MODE_TYPE : int
+    {
+        AUTO,
+        USER,
+        OFF
+    }
+
+    public enum DSP_PAN : int
+    {
+        MODE,
+        _2D_STEREO_POSITION,
+        _2D_DIRECTION,
+        _2D_EXTENT,
+        _2D_ROTATION,
+        _2D_LFE_LEVEL,
+        _2D_STEREO_MODE,
+        _2D_STEREO_SEPARATION,
+        _2D_STEREO_AXIS,
+        ENABLED_SPEAKERS,
+        _3D_POSITION,
+        _3D_ROLLOFF,
+        _3D_MIN_DISTANCE,
+        _3D_MAX_DISTANCE,
+        _3D_EXTENT_MODE,
+        _3D_SOUND_SIZE,
+        _3D_MIN_EXTENT,
+        _3D_PAN_BLEND,
+        LFE_UPMIX_ENABLED,
+        OVERALL_GAIN,
+        SURROUND_SPEAKER_MODE,
+        _2D_HEIGHT_BLEND,
+        ATTENUATION_RANGE,
+        OVERRIDE_RANGE
+    }
+
+    public enum DSP_THREE_EQ_CROSSOVERSLOPE_TYPE : int
+    {
+        _12DB,
+        _24DB,
+        _48DB
+    }
+
+    public enum DSP_THREE_EQ : int
+    {
+        LOWGAIN,
+        MIDGAIN,
+        HIGHGAIN,
+        LOWCROSSOVER,
+        HIGHCROSSOVER,
+        CROSSOVERSLOPE
+    }
+
+    public enum DSP_FFT_WINDOW_TYPE : int
+    {
+        RECT,
+        TRIANGLE,
+        HAMMING,
+        HANNING,
+        BLACKMAN,
+        BLACKMANHARRIS
+    }
+
+    public enum DSP_FFT_DOWNMIX_TYPE : int
+    {
+        NONE,
+        MONO
+    }
+
+    public enum DSP_FFT : int
+    {
+        WINDOWSIZE,
+        WINDOW,
+        BAND_START_FREQ,
+        BAND_STOP_FREQ,
+        SPECTRUMDATA,
+        RMS,
+        SPECTRAL_CENTROID,
+        IMMEDIATE_MODE,
+        DOWNMIX,
+        CHANNEL
+    }
+
+
+    public enum DSP_LOUDNESS_METER : int
+    {
+        STATE,
+        WEIGHTING,
+        INFO
+    }
+
+
+    public enum DSP_LOUDNESS_METER_STATE_TYPE : int
+    {
+        RESET_INTEGRATED = -3,
+        RESET_MAXPEAK = -2,
+        RESET_ALL = -1,
+        PAUSED = 0,
+        ANALYZING = 1
+    }
+
+    public enum DSP_CONVOLUTION_REVERB : int
+    {
+        IR,
+        WET,
+        DRY,
+        LINKED
+    }
+
+    public enum DSP_CHANNELMIX_OUTPUT : int
+    {
+        DEFAULT,
+        ALLMONO,
+        ALLSTEREO,
+        ALLQUAD,
+        ALL5POINT1,
+        ALL7POINT1,
+        ALLLFE,
+        ALL7POINT1POINT4
+    }
+
+    public enum DSP_CHANNELMIX : int
+    {
+        OUTPUTGROUPING,
+        GAIN_CH0,
+        GAIN_CH1,
+        GAIN_CH2,
+        GAIN_CH3,
+        GAIN_CH4,
+        GAIN_CH5,
+        GAIN_CH6,
+        GAIN_CH7,
+        GAIN_CH8,
+        GAIN_CH9,
+        GAIN_CH10,
+        GAIN_CH11,
+        GAIN_CH12,
+        GAIN_CH13,
+        GAIN_CH14,
+        GAIN_CH15,
+        GAIN_CH16,
+        GAIN_CH17,
+        GAIN_CH18,
+        GAIN_CH19,
+        GAIN_CH20,
+        GAIN_CH21,
+        GAIN_CH22,
+        GAIN_CH23,
+        GAIN_CH24,
+        GAIN_CH25,
+        GAIN_CH26,
+        GAIN_CH27,
+        GAIN_CH28,
+        GAIN_CH29,
+        GAIN_CH30,
+        GAIN_CH31,
+        OUTPUT_CH0,
+        OUTPUT_CH1,
+        OUTPUT_CH2,
+        OUTPUT_CH3,
+        OUTPUT_CH4,
+        OUTPUT_CH5,
+        OUTPUT_CH6,
+        OUTPUT_CH7,
+        OUTPUT_CH8,
+        OUTPUT_CH9,
+        OUTPUT_CH10,
+        OUTPUT_CH11,
+        OUTPUT_CH12,
+        OUTPUT_CH13,
+        OUTPUT_CH14,
+        OUTPUT_CH15,
+        OUTPUT_CH16,
+        OUTPUT_CH17,
+        OUTPUT_CH18,
+        OUTPUT_CH19,
+        OUTPUT_CH20,
+        OUTPUT_CH21,
+        OUTPUT_CH22,
+        OUTPUT_CH23,
+        OUTPUT_CH24,
+        OUTPUT_CH25,
+        OUTPUT_CH26,
+        OUTPUT_CH27,
+        OUTPUT_CH28,
+        OUTPUT_CH29,
+        OUTPUT_CH30,
+        OUTPUT_CH31,
+    }
+
+    public enum DSP_TRANSCEIVER_SPEAKERMODE : int
+    {
+        AUTO = -1,
+        MONO = 0,
+        STEREO,
+        SURROUND,
+    }
+
+    public enum DSP_TRANSCEIVER : int
+    {
+        TRANSMIT,
+        GAIN,
+        CHANNEL,
+        TRANSMITSPEAKERMODE
+    }
+
+    public enum DSP_OBJECTPAN : int
+    {
+        _3D_POSITION,
+        _3D_ROLLOFF,
+        _3D_MIN_DISTANCE,
+        _3D_MAX_DISTANCE,
+        _3D_EXTENT_MODE,
+        _3D_SOUND_SIZE,
+        _3D_MIN_EXTENT,
+        OVERALL_GAIN,
+        OUTPUTGAIN,
+        ATTENUATION_RANGE,
+        OVERRIDE_RANGE
+    }
+}
diff --git a/AssetStudioUtility/FMOD Studio API/fmod_errors.cs b/AssetStudioUtility/Audio/FMODStudioAPI/fmod_errors.cs
similarity index 95%
rename from AssetStudioUtility/FMOD Studio API/fmod_errors.cs
rename to AssetStudioUtility/Audio/FMODStudioAPI/fmod_errors.cs
index b50ca5e..88f5ae2 100644
--- a/AssetStudioUtility/FMOD Studio API/fmod_errors.cs	
+++ b/AssetStudioUtility/Audio/FMODStudioAPI/fmod_errors.cs
@@ -1,12 +1,12 @@
 /* ==============================================================================================  */
 /* FMOD Core / Studio API - Error string header file.                                              */
-/* Copyright (c), Firelight Technologies Pty, Ltd. 2004-2020.                                      */
+/* Copyright (c), Firelight Technologies Pty, Ltd. 2004-2025.                                      */
 /*                                                                                                 */
 /* Use this header if you want to store or display a string version / english explanation          */
 /* of the FMOD error codes.                                                                        */
 /*                                                                                                 */
 /* For more detail visit:                                                                          */
-/* https://fmod.com/resources/documentation-api?version=2.0&page=core-api-common.html#fmod_result  */
+/* https://fmod.com/docs/2.03/api/core-api-common.html#fmod_result                                 */
 /* =============================================================================================== */
 
 namespace FMOD
@@ -45,7 +45,7 @@ namespace FMOD
                 case FMOD.RESULT.ERR_HTTP_TIMEOUT:              return "The HTTP request timed out.";
                 case FMOD.RESULT.ERR_INITIALIZATION:            return "FMOD was not initialized correctly to support this function.";
                 case FMOD.RESULT.ERR_INITIALIZED:               return "Cannot call this command after System::init.";
-                case FMOD.RESULT.ERR_INTERNAL:                  return "An error occurred that wasn't supposed to.  Contact support.";
+                case FMOD.RESULT.ERR_INTERNAL:                  return "An error occured in the FMOD system. Use the logging version of FMOD for more information.";
                 case FMOD.RESULT.ERR_INVALID_FLOAT:             return "Value passed in was a NaN, Inf or denormalized float.";
                 case FMOD.RESULT.ERR_INVALID_HANDLE:            return "An invalid object handle was used.";
                 case FMOD.RESULT.ERR_INVALID_PARAM:             return "An invalid parameter was passed to this function.";
@@ -72,7 +72,7 @@ namespace FMOD
                 case FMOD.RESULT.ERR_OUTPUT_NODRIVERS:          return "The output device has no drivers installed.  If pre-init, FMOD_OUTPUT_NOSOUND is selected as the output mode.  If post-init, the function just fails.";
                 case FMOD.RESULT.ERR_PLUGIN:                    return "An unspecified error has been returned from a plugin.";
                 case FMOD.RESULT.ERR_PLUGIN_MISSING:            return "A requested output, dsp unit type or codec was not available.";
-                case FMOD.RESULT.ERR_PLUGIN_RESOURCE:           return "A resource that the plugin requires cannot be found. (ie the DLS file for MIDI playback)";
+                case FMOD.RESULT.ERR_PLUGIN_RESOURCE:           return "A resource that the plugin requires cannot be allocated or found. (ie the DLS file for MIDI playback)";
                 case FMOD.RESULT.ERR_PLUGIN_VERSION:            return "A plugin was built with an unsupported SDK version.";
                 case FMOD.RESULT.ERR_RECORD:                    return "An error occurred trying to initialize the recording device.";
                 case FMOD.RESULT.ERR_REVERB_CHANNELGROUP:       return "Reverb properties cannot be set on this channel because a parent channelgroup owns the reverb connection.";
@@ -83,7 +83,7 @@ namespace FMOD
                 case FMOD.RESULT.ERR_TAGNOTFOUND:               return "The specified tag could not be found or there are no tags.";
                 case FMOD.RESULT.ERR_TOOMANYCHANNELS:           return "The sound created exceeds the allowable input channel count.  This can be increased using the 'maxinputchannels' parameter in System::setSoftwareFormat.";
                 case FMOD.RESULT.ERR_TRUNCATED:                 return "The retrieved string is too long to fit in the supplied buffer and has been truncated.";
-                case FMOD.RESULT.ERR_UNIMPLEMENTED:             return "Something in FMOD hasn't been implemented when it should be! contact support!";
+                case FMOD.RESULT.ERR_UNIMPLEMENTED:             return "Something in FMOD hasn't been implemented when it should be. Contact support.";
                 case FMOD.RESULT.ERR_UNINITIALIZED:             return "This command failed because System::init or System::setDriver was not called.";
                 case FMOD.RESULT.ERR_UNSUPPORTED:               return "A command issued was not supported by this object.  Possibly a plugin without certain callbacks specified.";
                 case FMOD.RESULT.ERR_VERSION:                   return "The version number of this file format is not supported.";
@@ -97,8 +97,8 @@ namespace FMOD
                 case FMOD.RESULT.ERR_INVALID_STRING:            return "An invalid string was passed to this function.";
                 case FMOD.RESULT.ERR_ALREADY_LOCKED:            return "The specified resource is already locked.";
                 case FMOD.RESULT.ERR_NOT_LOCKED:                return "The specified resource is not locked, so it can't be unlocked.";
-				case FMOD.RESULT.ERR_RECORD_DISCONNECTED:       return "The specified recording driver has been disconnected.";
-				case FMOD.RESULT.ERR_TOOMANYSAMPLES:            return "The length provided exceed the allowable limit.";
+                case FMOD.RESULT.ERR_RECORD_DISCONNECTED:       return "The specified recording driver has been disconnected.";
+                case FMOD.RESULT.ERR_TOOMANYSAMPLES:            return "The length provided exceed the allowable limit.";
                 default:                                        return "Unknown error.";
             }
         }
diff --git a/AssetStudioUtility/Audio/WavHelper/AudioFormat.cs b/AssetStudioUtility/Audio/WavHelper/AudioFormat.cs
new file mode 100644
index 0000000..58c7626
--- /dev/null
+++ b/AssetStudioUtility/Audio/WavHelper/AudioFormat.cs
@@ -0,0 +1,8 @@
+namespace WavHelper
+{
+    public enum WavAudioFormat : ushort
+    {
+        PCM = 1,
+        IEEEfloat = 3,
+    }
+}
diff --git a/AssetStudioUtility/Audio/WavHelper/WavHeader.cs b/AssetStudioUtility/Audio/WavHelper/WavHeader.cs
new file mode 100644
index 0000000..ba0641b
--- /dev/null
+++ b/AssetStudioUtility/Audio/WavHelper/WavHeader.cs
@@ -0,0 +1,57 @@
+using System.IO;
+using System.Text;
+
+namespace WavHelper
+{
+    public sealed class WavHeader
+    {
+        private static byte[] ChunkId = Encoding.ASCII.GetBytes("RIFF");
+        private uint ChunkSize;
+        private static byte[] Format = Encoding.ASCII.GetBytes("WAVE");
+        private static byte[] FmtChunkId = Encoding.ASCII.GetBytes("fmt ");
+        private static uint FmtChunkSize = 16;
+        private WavAudioFormat AudioFormat;
+        private ushort NumChannels;
+        private uint SampleRate;
+        private uint ByteRate;
+        private ushort BlockAlign;
+        private ushort BitsPerSample;
+        private static byte[] DataChunkId = Encoding.ASCII.GetBytes("data");
+        private uint DataChunkSize;
+
+        public WavHeader(uint dataSize, WavAudioFormat audioFormat, int channels, uint sampleRate, int bits)
+        {
+            ChunkSize = dataSize + 36;
+            AudioFormat = audioFormat;
+            NumChannels = (ushort)channels;
+            SampleRate = sampleRate;
+            BitsPerSample = (ushort)bits;
+            ByteRate = SampleRate * NumChannels * BitsPerSample / 8;
+            BlockAlign = (ushort)(NumChannels * BitsPerSample / 8);
+            DataChunkSize = dataSize;
+        }
+
+        public void WriteToArray(byte[] buffer)
+        {
+            using (var stream = new MemoryStream(buffer))
+            {
+                using (var writer = new BinaryWriter(stream))
+                {
+                    writer.Write(ChunkId);
+                    writer.Write(ChunkSize);
+                    writer.Write(Format);
+                    writer.Write(FmtChunkId);
+                    writer.Write(FmtChunkSize);
+                    writer.Write((ushort)AudioFormat);
+                    writer.Write(NumChannels);
+                    writer.Write(SampleRate);
+                    writer.Write(ByteRate);
+                    writer.Write(BlockAlign);
+                    writer.Write(BitsPerSample);
+                    writer.Write(DataChunkId);
+                    writer.Write(DataChunkSize);
+                }
+            }
+        }
+    }
+}
diff --git a/AssetStudioUtility/FMOD Studio API/fmod_dsp.cs b/AssetStudioUtility/FMOD Studio API/fmod_dsp.cs
deleted file mode 100644
index 113ebbc..0000000
--- a/AssetStudioUtility/FMOD Studio API/fmod_dsp.cs	
+++ /dev/null
@@ -1,846 +0,0 @@
-/* ======================================================================================== */
-/* FMOD Core API - DSP header file.                                                         */
-/* Copyright (c), Firelight Technologies Pty, Ltd. 2004-2020.                               */
-/*                                                                                          */
-/* Use this header if you are wanting to develop your own DSP plugin to use with FMODs      */
-/* dsp system.  With this header you can make your own DSP plugin that FMOD can             */
-/* register and use.  See the documentation and examples on how to make a working plugin.   */
-/*                                                                                          */
-/* For more detail visit:                                                                   */
-/* https://fmod.com/resources/documentation-api?version=2.0&page=plugin-api-dsp.html        */
-/* =========================================================================================*/
-
-using System;
-using System.Text;
-using System.Runtime.InteropServices;
-
-namespace FMOD
-{
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_BUFFER_ARRAY
-    {
-        public int              numbuffers;              /* [r/w] number of buffers */
-        public int[]            buffernumchannels;       /* [r/w] array of number of channels for each buffer */
-        public CHANNELMASK[]    bufferchannelmask;       /* [r/w] array of channel masks for each buffer */
-        public IntPtr[]         buffers;                 /* [r/w] array of buffers */
-        public SPEAKERMODE      speakermode;             /* [r/w] speaker mode for all buffers in the array */
-    }
-
-    public enum DSP_PROCESS_OPERATION
-    {
-        PROCESS_PERFORM = 0,               /* Process the incoming audio in 'inbufferarray' and output to 'outbufferarray'. */
-        PROCESS_QUERY                      /* The DSP is being queried for the expected output format and whether it needs to process audio or should be bypassed.  The function should return any value other than FMOD_OK if audio can pass through unprocessed. If audio is to be processed, 'outbufferarray' must be filled with the expected output format, channel count and mask. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct COMPLEX
-    {
-        public float real; /* Real component */
-        public float imag; /* Imaginary component */
-    }
-
-    public enum DSP_PAN_SURROUND_FLAGS
-    {
-        DEFAULT = 0,
-        ROTATION_NOT_BIASED = 1,
-    }
-
-
-    /*
-        DSP callbacks
-    */
-    public delegate RESULT DSP_CREATECALLBACK                   (ref DSP_STATE dsp_state);
-    public delegate RESULT DSP_RELEASECALLBACK                  (ref DSP_STATE dsp_state);
-    public delegate RESULT DSP_RESETCALLBACK                    (ref DSP_STATE dsp_state);
-    public delegate RESULT DSP_SETPOSITIONCALLBACK              (ref DSP_STATE dsp_state, uint pos);
-    public delegate RESULT DSP_READCALLBACK                     (ref DSP_STATE dsp_state, IntPtr inbuffer, IntPtr outbuffer, uint length, int inchannels, ref int outchannels);
-    public delegate RESULT DSP_SHOULDIPROCESS_CALLBACK          (ref DSP_STATE dsp_state, bool inputsidle, uint length, CHANNELMASK inmask, int inchannels, SPEAKERMODE speakermode);
-    public delegate RESULT DSP_PROCESS_CALLBACK                 (ref DSP_STATE dsp_state, uint length, ref DSP_BUFFER_ARRAY inbufferarray, ref DSP_BUFFER_ARRAY outbufferarray, bool inputsidle, DSP_PROCESS_OPERATION op);
-    public delegate RESULT DSP_SETPARAM_FLOAT_CALLBACK          (ref DSP_STATE dsp_state, int index, float value);
-    public delegate RESULT DSP_SETPARAM_INT_CALLBACK            (ref DSP_STATE dsp_state, int index, int value);
-    public delegate RESULT DSP_SETPARAM_BOOL_CALLBACK           (ref DSP_STATE dsp_state, int index, bool value);
-    public delegate RESULT DSP_SETPARAM_DATA_CALLBACK           (ref DSP_STATE dsp_state, int index, IntPtr data, uint length);
-    public delegate RESULT DSP_GETPARAM_FLOAT_CALLBACK          (ref DSP_STATE dsp_state, int index, ref float value, IntPtr valuestr);
-    public delegate RESULT DSP_GETPARAM_INT_CALLBACK            (ref DSP_STATE dsp_state, int index, ref int value, IntPtr valuestr);
-    public delegate RESULT DSP_GETPARAM_BOOL_CALLBACK           (ref DSP_STATE dsp_state, int index, ref bool value, IntPtr valuestr);
-    public delegate RESULT DSP_GETPARAM_DATA_CALLBACK           (ref DSP_STATE dsp_state, int index, ref IntPtr data, ref uint length, IntPtr valuestr);
-    public delegate RESULT DSP_SYSTEM_REGISTER_CALLBACK         (ref DSP_STATE dsp_state);
-    public delegate RESULT DSP_SYSTEM_DEREGISTER_CALLBACK       (ref DSP_STATE dsp_state);
-    public delegate RESULT DSP_SYSTEM_MIX_CALLBACK              (ref DSP_STATE dsp_state, int stage);
-
-
-    /*
-        DSP functions
-    */
-    public delegate IntPtr DSP_ALLOC_FUNC                         (uint size, MEMORY_TYPE type, StringWrapper sourcestr);
-    public delegate IntPtr DSP_REALLOC_FUNC                       (IntPtr ptr, uint size, MEMORY_TYPE type, StringWrapper sourcestr);
-    public delegate void   DSP_FREE_FUNC                          (IntPtr ptr, MEMORY_TYPE type, StringWrapper sourcestr);
-    public delegate void   DSP_LOG_FUNC                           (DEBUG_FLAGS level, StringWrapper file, int line, StringWrapper function, StringWrapper format);
-    public delegate RESULT DSP_GETSAMPLERATE_FUNC                 (ref DSP_STATE dsp_state, ref int rate);
-    public delegate RESULT DSP_GETBLOCKSIZE_FUNC                  (ref DSP_STATE dsp_state, ref uint blocksize);
-    public delegate RESULT DSP_GETSPEAKERMODE_FUNC                (ref DSP_STATE dsp_state, ref int speakermode_mixer, ref int speakermode_output);
-    public delegate RESULT DSP_GETCLOCK_FUNC                      (ref DSP_STATE dsp_state, out ulong clock, out uint offset, out uint length);
-    public delegate RESULT DSP_GETLISTENERATTRIBUTES_FUNC         (ref DSP_STATE dsp_state, ref int numlisteners, IntPtr attributes);
-    public delegate RESULT DSP_GETUSERDATA_FUNC                   (ref DSP_STATE dsp_state, out IntPtr userdata);
-    public delegate RESULT DSP_DFT_FFTREAL_FUNC                   (ref DSP_STATE dsp_state, int size, IntPtr signal, IntPtr dft, IntPtr window, int signalhop);
-    public delegate RESULT DSP_DFT_IFFTREAL_FUNC                  (ref DSP_STATE dsp_state, int size, IntPtr dft, IntPtr signal, IntPtr window, int signalhop);
-    public delegate RESULT DSP_PAN_SUMMONOMATRIX_FUNC             (ref DSP_STATE dsp_state, int sourceSpeakerMode, float lowFrequencyGain, float overallGain, IntPtr matrix);
-    public delegate RESULT DSP_PAN_SUMSTEREOMATRIX_FUNC           (ref DSP_STATE dsp_state, int sourceSpeakerMode, float pan, float lowFrequencyGain, float overallGain, int matrixHop, IntPtr matrix);
-    public delegate RESULT DSP_PAN_SUMSURROUNDMATRIX_FUNC         (ref DSP_STATE dsp_state, int sourceSpeakerMode, int targetSpeakerMode, float direction, float extent, float rotation, float lowFrequencyGain, float overallGain, int matrixHop, IntPtr matrix, DSP_PAN_SURROUND_FLAGS flags);
-    public delegate RESULT DSP_PAN_SUMMONOTOSURROUNDMATRIX_FUNC   (ref DSP_STATE dsp_state, int targetSpeakerMode, float direction, float extent, float lowFrequencyGain, float overallGain, int matrixHop, IntPtr matrix);
-    public delegate RESULT DSP_PAN_SUMSTEREOTOSURROUNDMATRIX_FUNC (ref DSP_STATE dsp_state, int targetSpeakerMode, float direction, float extent, float rotation, float lowFrequencyGain, float overallGain, int matrixHop, IntPtr matrix);
-    public delegate RESULT DSP_PAN_GETROLLOFFGAIN_FUNC            (ref DSP_STATE dsp_state, DSP_PAN_3D_ROLLOFF_TYPE rolloff, float distance, float mindistance, float maxdistance, out float gain);
-    
-
-    public enum DSP_TYPE : int
-    {
-        UNKNOWN,            /* This unit was created via a non FMOD plugin so has an unknown purpose. */
-        MIXER,              /* This unit does nothing but take inputs and mix them together then feed the result to the soundcard unit. */
-        OSCILLATOR,         /* This unit generates sine/square/saw/triangle or noise tones. */
-        LOWPASS,            /* This unit filters sound using a high quality, resonant lowpass filter algorithm but consumes more CPU time. */
-        ITLOWPASS,          /* This unit filters sound using a resonant lowpass filter algorithm that is used in Impulse Tracker, but with limited cutoff range (0 to 8060hz). */
-        HIGHPASS,           /* This unit filters sound using a resonant highpass filter algorithm. */
-        ECHO,               /* This unit produces an echo on the sound and fades out at the desired rate. */
-        FADER,              /* This unit pans and scales the volume of a unit. */
-        FLANGE,             /* This unit produces a flange effect on the sound. */
-        DISTORTION,         /* This unit distorts the sound. */
-        NORMALIZE,          /* This unit normalizes or amplifies the sound to a certain level. */
-        LIMITER,            /* This unit limits the sound to a certain level. */
-        PARAMEQ,            /* This unit attenuates or amplifies a selected frequency range. */
-        PITCHSHIFT,         /* This unit bends the pitch of a sound without changing the speed of playback. */
-        CHORUS,             /* This unit produces a chorus effect on the sound. */
-        VSTPLUGIN,          /* This unit allows the use of Steinberg VST plugins */
-        WINAMPPLUGIN,       /* This unit allows the use of Nullsoft Winamp plugins */
-        ITECHO,             /* This unit produces an echo on the sound and fades out at the desired rate as is used in Impulse Tracker. */
-        COMPRESSOR,         /* This unit implements dynamic compression (linked multichannel, wideband) */
-        SFXREVERB,          /* This unit implements SFX reverb */
-        LOWPASS_SIMPLE,     /* This unit filters sound using a simple lowpass with no resonance, but has flexible cutoff and is fast. */
-        DELAY,              /* This unit produces different delays on individual channels of the sound. */
-        TREMOLO,            /* This unit produces a tremolo / chopper effect on the sound. */
-        LADSPAPLUGIN,       /* This unit allows the use of LADSPA standard plugins. */
-        SEND,               /* This unit sends a copy of the signal to a return DSP anywhere in the DSP tree. */
-        RETURN,             /* This unit receives signals from a number of send DSPs. */
-        HIGHPASS_SIMPLE,    /* This unit filters sound using a simple highpass with no resonance, but has flexible cutoff and is fast. */
-        PAN,                /* This unit pans the signal, possibly upmixing or downmixing as well. */
-        THREE_EQ,           /* This unit is a three-band equalizer. */
-        FFT,                /* This unit simply analyzes the signal and provides spectrum information back through getParameter. */
-        LOUDNESS_METER,     /* This unit analyzes the loudness and true peak of the signal. */
-        ENVELOPEFOLLOWER,   /* This unit tracks the envelope of the input/sidechain signal. Deprecated and will be removed in a future release. */
-        CONVOLUTIONREVERB,  /* This unit implements convolution reverb. */
-        CHANNELMIX,         /* This unit provides per signal channel gain, and output channel mapping to allow 1 multichannel signal made up of many groups of signals to map to a single output signal. */
-        TRANSCEIVER,        /* This unit 'sends' and 'receives' from a selection of up to 32 different slots.  It is like a send/return but it uses global slots rather than returns as the destination.  It also has other features.  Multiple transceivers can receive from a single channel, or multiple transceivers can send to a single channel, or a combination of both. */
-        OBJECTPAN,          /* This unit sends the signal to a 3d object encoder like Dolby Atmos. */
-        MULTIBAND_EQ,       /* This unit is a flexible five band parametric equalizer. */
-        MAX
-    }
-
-    public enum DSP_PARAMETER_TYPE
-    {
-        FLOAT = 0,
-        INT,
-        BOOL,
-        DATA,
-        MAX
-    }
-
-    public enum DSP_PARAMETER_FLOAT_MAPPING_TYPE
-    {
-        DSP_PARAMETER_FLOAT_MAPPING_TYPE_LINEAR = 0,          /* Values mapped linearly across range. */
-        DSP_PARAMETER_FLOAT_MAPPING_TYPE_AUTO,                /* A mapping is automatically chosen based on range and units.  See remarks. */
-        DSP_PARAMETER_FLOAT_MAPPING_TYPE_PIECEWISE_LINEAR,    /* Values mapped in a piecewise linear fashion defined by FMOD_DSP_PARAMETER_DESC_FLOAT::mapping.piecewiselinearmapping. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_FLOAT_MAPPING_PIECEWISE_LINEAR
-    {
-        public int numpoints;                       /* [w] The number of <position, value> pairs in the piecewise mapping (at least 2). */
-        public IntPtr pointparamvalues;             /* [w] The values in the parameter's units for each point */
-        public IntPtr pointpositions;               /* [w] The positions along the control's scale (e.g. dial angle) corresponding to each parameter value.  The range of this scale is arbitrary and all positions will be relative to the minimum and maximum values (e.g. [0,1,3] is equivalent to [1,2,4] and [2,4,8]).  If this array is zero, pointparamvalues will be distributed with equal spacing. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_FLOAT_MAPPING
-    {
-        public DSP_PARAMETER_FLOAT_MAPPING_TYPE type;
-        public DSP_PARAMETER_FLOAT_MAPPING_PIECEWISE_LINEAR piecewiselinearmapping;    /* [w] Only required for FMOD_DSP_PARAMETER_FLOAT_MAPPING_TYPE_PIECEWISE_LINEAR type mapping. */
-    }
-
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_DESC_FLOAT
-    {
-        public float                     min;                      /* [w] Minimum parameter value. */
-        public float                     max;                      /* [w] Maximum parameter value. */
-        public float                     defaultval;               /* [w] Default parameter value. */
-        public DSP_PARAMETER_FLOAT_MAPPING mapping;           /* [w] How the values are distributed across dials and automation curves (e.g. linearly, exponentially etc). */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_DESC_INT
-    {
-        public int                       min;                      /* [w] Minimum parameter value. */
-        public int                       max;                      /* [w] Maximum parameter value. */
-        public int                       defaultval;               /* [w] Default parameter value. */
-        public bool                      goestoinf;                /* [w] Whether the last value represents infiniy. */
-        public IntPtr                    valuenames;               /* [w] Names for each value.  There should be as many strings as there are possible values (max - min + 1).  Optional. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_DESC_BOOL
-    {
-        public bool                      defaultval;               /* [w] Default parameter value. */
-        public IntPtr                    valuenames;               /* [w] Names for false and true, respectively.  There should be two strings.  Optional. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_DESC_DATA
-    {
-        public int                       datatype;                 /* [w] The type of data for this parameter.  Use 0 or above for custom types or set to one of the FMOD_DSP_PARAMETER_DATA_TYPE values. */
-    }
-
-    [StructLayout(LayoutKind.Explicit)]
-    public struct DSP_PARAMETER_DESC_UNION
-    {
-        [FieldOffset(0)]
-        public DSP_PARAMETER_DESC_FLOAT   floatdesc;  /* [w] Struct containing information about the parameter in floating point format.  Use when type is FMOD_DSP_PARAMETER_TYPE_FLOAT. */
-        [FieldOffset(0)]
-        public DSP_PARAMETER_DESC_INT     intdesc;    /* [w] Struct containing information about the parameter in integer format.  Use when type is FMOD_DSP_PARAMETER_TYPE_INT. */
-        [FieldOffset(0)]
-        public DSP_PARAMETER_DESC_BOOL    booldesc;   /* [w] Struct containing information about the parameter in boolean format.  Use when type is FMOD_DSP_PARAMETER_TYPE_BOOL. */
-        [FieldOffset(0)]
-        public DSP_PARAMETER_DESC_DATA    datadesc;   /* [w] Struct containing information about the parameter in data format.  Use when type is FMOD_DSP_PARAMETER_TYPE_DATA. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_DESC
-    {
-        public DSP_PARAMETER_TYPE   type;            /* [w] Type of this parameter. */
-        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 16)]
-        public byte[]               name;            /* [w] Name of the parameter to be displayed (ie "Cutoff frequency"). */
-        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 16)]
-        public byte[]               label;           /* [w] Short string to be put next to value to denote the unit type (ie "hz"). */
-        public string               description;     /* [w] Description of the parameter to be displayed as a help item / tooltip for this parameter. */
-
-        public DSP_PARAMETER_DESC_UNION desc;
-    }
-
-    public enum DSP_PARAMETER_DATA_TYPE
-    {
-        DSP_PARAMETER_DATA_TYPE_USER = 0,                /* The default data type.  All user data types should be 0 or above. */
-        DSP_PARAMETER_DATA_TYPE_OVERALLGAIN = -1,        /* The data type for FMOD_DSP_PARAMETER_OVERALLGAIN parameters.  There should a maximum of one per DSP. */
-        DSP_PARAMETER_DATA_TYPE_3DATTRIBUTES = -2,       /* The data type for FMOD_DSP_PARAMETER_3DATTRIBUTES parameters.  There should a maximum of one per DSP. */
-        DSP_PARAMETER_DATA_TYPE_SIDECHAIN = -3,          /* The data type for FMOD_DSP_PARAMETER_SIDECHAIN parameters.  There should a maximum of one per DSP. */
-        DSP_PARAMETER_DATA_TYPE_FFT = -4,                /* The data type for FMOD_DSP_PARAMETER_FFT parameters.  There should a maximum of one per DSP. */
-        DSP_PARAMETER_DATA_TYPE_3DATTRIBUTES_MULTI = -5  /* The data type for FMOD_DSP_PARAMETER_3DATTRIBUTES_MULTI parameters.  There should a maximum of one per DSP. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_OVERALLGAIN
-    {
-        public float linear_gain;                                  /* [r] The overall linear gain of the effect on the direct signal path */
-        public float linear_gain_additive;                         /* [r] Additive gain, for parallel signal paths */
-    }
-    
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_3DATTRIBUTES
-    {
-        public ATTRIBUTES_3D relative;                        /* [w] The position of the sound relative to the listener. */
-        public ATTRIBUTES_3D absolute;                        /* [w] The position of the sound in world coordinates. */
-    }
-    
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_3DATTRIBUTES_MULTI
-    {
-        public int            numlisteners;                    /* [w] The number of listeners. */
-        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 8)]
-        public ATTRIBUTES_3D[] relative;                       /* [w] The position of the sound relative to the listeners. */
-        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 8)]
-        public float[] weight;                                 /* [w] The weighting of the listeners where 0 means listener has no contribution and 1 means full contribution. */
-        public ATTRIBUTES_3D absolute;                         /* [w] The position of the sound in world coordinates. */
-    }
-    
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_SIDECHAIN
-    {
-        public int sidechainenable;                               /* [r/w] Whether sidechains are enabled. */
-    }
-    
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_PARAMETER_FFT
-    {
-        public int     length;                                    /* [r] Number of entries in this spectrum window.   Divide this by the output rate to get the hz per entry. */
-        public int     numchannels;                               /* [r] Number of channels in spectrum. */
-        
-        [MarshalAs(UnmanagedType.ByValArray,SizeConst=32)]
-        private IntPtr[] spectrum_internal;                           /* [r] Per channel spectrum arrays.  See remarks for more. */
-
-        public float[][] spectrum
-        {
-            get
-            {
-                var buffer = new float[numchannels][];
-                
-                for (int i = 0; i < numchannels; ++i)
-                {
-                    buffer[i] = new float[length];
-                    Marshal.Copy(spectrum_internal[i], buffer[i], 0, length);
-                }
-                
-                return buffer;
-            }
-        }
-
-        public void getSpectrum(ref float[][] buffer)
-        {
-            int bufferLength = Math.Min(buffer.Length, numchannels);
-            for (int i = 0; i < bufferLength; ++i)
-            {
-                getSpectrum(i, ref buffer[i]);
-            }
-        }
-
-        public void getSpectrum(int channel, ref float[] buffer)
-        {
-            int bufferLength = Math.Min(buffer.Length, length);
-            Marshal.Copy(spectrum_internal[channel], buffer, 0, bufferLength);
-        }
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_DESCRIPTION
-    {
-        public uint                           pluginsdkversion;   /* [w] The plugin SDK version this plugin is built for.  set to this to FMOD_PLUGIN_SDK_VERSION defined above. */
-        [MarshalAs(UnmanagedType.ByValArray, SizeConst = 32)]
-        public char[]                         name;               /* [w] Name of the unit to be displayed in the network. */
-        public uint                           version;            /* [w] Plugin writer's version number. */
-        public int                            numinputbuffers;    /* [w] Number of input buffers to process.  Use 0 for DSPs that only generate sound and 1 for effects that process incoming sound. */
-        public int                            numoutputbuffers;   /* [w] Number of audio output buffers.  Only one output buffer is currently supported. */
-        public DSP_CREATECALLBACK             create;             /* [w] Create callback.  This is called when DSP unit is created.  Can be null. */
-        public DSP_RELEASECALLBACK            release;            /* [w] Release callback.  This is called just before the unit is freed so the user can do any cleanup needed for the unit.  Can be null. */
-        public DSP_RESETCALLBACK              reset;              /* [w] Reset callback.  This is called by the user to reset any history buffers that may need resetting for a filter, when it is to be used or re-used for the first time to its initial clean state.  Use to avoid clicks or artifacts. */
-        public DSP_READCALLBACK               read;               /* [w] Read callback.  Processing is done here.  Can be null. */
-        public DSP_PROCESS_CALLBACK           process;            /* [w] Process callback.  Can be specified instead of the read callback if any channel format changes occur between input and output.  This also replaces shouldiprocess and should return an error if the effect is to be bypassed.  Can be null. */
-        public DSP_SETPOSITIONCALLBACK        setposition;        /* [w] Setposition callback.  This is called if the unit wants to update its position info but not process data.  Can be null. */
-
-        public int                            numparameters;      /* [w] Number of parameters used in this filter.  The user finds this with DSP::getNumParameters */
-        public IntPtr                         paramdesc;          /* [w] Variable number of parameter structures. */
-        public DSP_SETPARAM_FLOAT_CALLBACK    setparameterfloat;  /* [w] This is called when the user calls DSP.setParameterFloat. Can be null. */
-        public DSP_SETPARAM_INT_CALLBACK      setparameterint;    /* [w] This is called when the user calls DSP.setParameterInt.   Can be null. */
-        public DSP_SETPARAM_BOOL_CALLBACK     setparameterbool;   /* [w] This is called when the user calls DSP.setParameterBool.  Can be null. */
-        public DSP_SETPARAM_DATA_CALLBACK     setparameterdata;   /* [w] This is called when the user calls DSP.setParameterData.  Can be null. */
-        public DSP_GETPARAM_FLOAT_CALLBACK    getparameterfloat;  /* [w] This is called when the user calls DSP.getParameterFloat. Can be null. */
-        public DSP_GETPARAM_INT_CALLBACK      getparameterint;    /* [w] This is called when the user calls DSP.getParameterInt.   Can be null. */
-        public DSP_GETPARAM_BOOL_CALLBACK     getparameterbool;   /* [w] This is called when the user calls DSP.getParameterBool.  Can be null. */
-        public DSP_GETPARAM_DATA_CALLBACK     getparameterdata;   /* [w] This is called when the user calls DSP.getParameterData.  Can be null. */
-        public DSP_SHOULDIPROCESS_CALLBACK    shouldiprocess;     /* [w] This is called before processing.  You can detect if inputs are idle and return FMOD_OK to process, or any other error code to avoid processing the effect.  Use a count down timer to allow effect tails to process before idling! */
-        public IntPtr                         userdata;           /* [w] Optional. Specify 0 to ignore. This is user data to be attached to the DSP unit during creation.  Access via FMOD_DSP_STATE_FUNCTIONS::getuserdata. */
-
-        public DSP_SYSTEM_REGISTER_CALLBACK   sys_register;       /* [w] Register callback.  This is called when DSP unit is loaded/registered.  Useful for 'global'/per system object init for plugin.  Can be null. */
-        public DSP_SYSTEM_DEREGISTER_CALLBACK sys_deregister;     /* [w] Deregister callback.  This is called when DSP unit is unloaded/deregistered.  Useful as 'global'/per system object shutdown for plugin.  Can be null. */
-        public DSP_SYSTEM_MIX_CALLBACK        sys_mix;            /* [w] System mix stage callback.  This is called when the mixer starts to execute or is just finishing executing.  Useful for 'global'/per system object once a mix update calls for a plugin.  Can be null. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_STATE_DFT_FUNCTIONS
-    {
-        public DSP_DFT_FFTREAL_FUNC  fftreal;        /* [r] Function for performing an FFT on a real signal. */
-        public DSP_DFT_IFFTREAL_FUNC inversefftreal; /* [r] Function for performing an inverse FFT to get a real signal. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_STATE_PAN_FUNCTIONS
-    {
-        public DSP_PAN_SUMMONOMATRIX_FUNC             summonomatrix;             /* [r] TBD. */
-        public DSP_PAN_SUMSTEREOMATRIX_FUNC           sumstereomatrix;           /* [r] TBD. */
-        public DSP_PAN_SUMSURROUNDMATRIX_FUNC         sumsurroundmatrix;         /* [r] TBD. */
-        public DSP_PAN_SUMMONOTOSURROUNDMATRIX_FUNC   summonotosurroundmatrix;   /* [r] TBD. */
-        public DSP_PAN_SUMSTEREOTOSURROUNDMATRIX_FUNC sumstereotosurroundmatrix; /* [r] TBD. */
-        public DSP_PAN_GETROLLOFFGAIN_FUNC            getrolloffgain;            /* [r] TBD. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_STATE_FUNCTIONS
-    {
-        public DSP_ALLOC_FUNC                  alloc;                  /* [r] Memory allocation callback. Use this for all dynamic memory allocation within the plugin. */
-        public DSP_REALLOC_FUNC                realloc;                /* [r] Memory reallocation callback. */
-        public DSP_FREE_FUNC                   free;                   /* [r] Memory free callback. */
-        public DSP_GETSAMPLERATE_FUNC          getsamplerate;          /* [r] Callback for getting the system samplerate. */
-        public DSP_GETBLOCKSIZE_FUNC           getblocksize;           /* [r] Callback for getting the system's block size.  DSPs will be requested to process blocks of varying length up to this size.*/
-        public IntPtr                          dft;                    /* [r] Struct containing callbacks for performing FFTs and inverse FFTs. */
-        public IntPtr                          pan;                    /* [r] Pointer to a structure of callbacks for calculating pan, up-mix and down-mix matrices. */
-        public DSP_GETSPEAKERMODE_FUNC         getspeakermode;         /* [r] Callback for getting the system's speaker modes.  One is the mixer's default speaker mode, the other is the output mode the system is downmixing or upmixing to.*/
-        public DSP_GETCLOCK_FUNC               getclock;               /* [r] Callback for getting the clock of the current DSP, as well as the subset of the input buffer that contains the signal */
-        public DSP_GETLISTENERATTRIBUTES_FUNC  getlistenerattributes;  /* [r] Callback for getting the absolute listener attributes set via the API (returned as left-handed co-ordinates). */
-        public DSP_LOG_FUNC                    log;                    /* [r] Function to write to the FMOD logging system. */
-        public DSP_GETUSERDATA_FUNC            getuserdata;            /* [r] Function to get the user data attached to this DSP. See FMOD_DSP_DESCRIPTION::userdata. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_STATE
-    {
-        public IntPtr     instance;            /* [r] Handle to the DSP hand the user created.  Not to be modified.  C++ users cast to FMOD::DSP to use.  */
-        public IntPtr     plugindata;          /* [r/w] Plugin writer created data the output author wants to attach to this object. */
-        public uint       channelmask;         /* [r] Specifies which speakers the DSP effect is active on */
-        public int        source_speakermode;  /* [r] Specifies which speaker mode the signal originated for information purposes, ie in case panning needs to be done differently. */
-        public IntPtr     sidechaindata;       /* [r] The mixed result of all incoming sidechains is stored at this pointer address. */
-        public int        sidechainchannels;   /* [r] The number of channels of pcm data stored within the sidechain buffer. */
-        public IntPtr     functions;           /* [r] Struct containing callbacks for system level functionality. */
-        public int        systemobject;        /* [r] FMOD::System object index, relating to the System object that created this DSP. */
-    }
-
-    [StructLayout(LayoutKind.Sequential)]
-    public struct DSP_METERING_INFO
-    {
-        public int   numsamples;        /* [r] The number of samples considered for this metering info. */
-        [MarshalAs(UnmanagedType.ByValArray, SizeConst=32)]
-        public float[] peaklevel;       /* [r] The peak level per channel. */
-        [MarshalAs(UnmanagedType.ByValArray, SizeConst=32)]
-        public float[] rmslevel;        /* [r] The rms level per channel. */
-        public short numchannels;       /* [r] Number of channels. */
-    }
-
-    /*
-        ==============================================================================================================
-
-        FMOD built in effect parameters.
-        Use DSP::setParameter with these enums for the 'index' parameter.
-
-        ==============================================================================================================
-    */
-
-    public enum DSP_OSCILLATOR : int
-    {
-        TYPE,   /* Waveform type.  0 = sine.  1 = square. 2 = sawup. 3 = sawdown. 4 = triangle. 5 = noise.  */
-        RATE    /* Frequency of the sinewave in hz.  1.0 to 22000.0.  Default = 220.0. */
-    }
-
-    public enum DSP_LOWPASS : int
-    {
-        CUTOFF,    /* Lowpass cutoff frequency in hz.   1.0 to 22000.0.  Default = 5000.0. */
-        RESONANCE  /* Lowpass resonance Q value. 1.0 to 10.0.  Default = 1.0. */
-    }
-
-    public enum DSP_ITLOWPASS : int
-    {
-        CUTOFF,    /* Lowpass cutoff frequency in hz.  1.0 to 22000.0.  Default = 5000.0/ */
-        RESONANCE  /* Lowpass resonance Q value.  0.0 to 127.0.  Default = 1.0. */
-    }
-
-    public enum DSP_HIGHPASS : int
-    {
-        CUTOFF,    /* (Type:float) - Highpass cutoff frequency in hz.  1.0 to output 22000.0.  Default = 5000.0. */
-        RESONANCE  /* (Type:float) - Highpass resonance Q value.  1.0 to 10.0.  Default = 1.0. */
-    }
-
-    public enum DSP_ECHO : int
-    {
-        DELAY,       /* (Type:float) - Echo delay in ms.  10  to 5000.  Default = 500. */
-        FEEDBACK,    /* (Type:float) - Echo decay per delay.  0 to 100.  100.0 = No decay, 0.0 = total decay (ie simple 1 line delay).  Default = 50.0. */
-        DRYLEVEL,    /* (Type:float) - Original sound volume in dB.  -80.0 to 10.0.  Default = 0. */
-        WETLEVEL     /* (Type:float) - Volume of echo signal to pass to output in dB.  -80.0 to 10.0.  Default = 0. */
-    }
-
-    public enum DSP_FADER : int
-    {
-        GAIN,           /* (Type:float) - Signal gain in dB. -80.0 to 10.0. Default = 0.0. */
-        OVERALL_GAIN,   /* (Type:data)  - Overall gain. For information only, not set by user. Data of type FMOD_DSP_PARAMETER_DATA_TYPE_OVERALLGAIN to provide to FMOD, to allow FMOD to know the DSP is scaling the signal for virtualization purposes. */
-    }
-
-    public enum DSP_DELAY : int
-    {
-        CH0,      /* Channel #0 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH1,      /* Channel #1 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH2,      /* Channel #2 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH3,      /* Channel #3 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH4,      /* Channel #4 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH5,      /* Channel #5 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH6,      /* Channel #6 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH7,      /* Channel #7 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH8,      /* Channel #8 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH9,      /* Channel #9 Delay in ms.   0  to 10000.  Default = 0.  */
-        CH10,     /* Channel #10 Delay in ms.  0  to 10000.  Default = 0.  */
-        CH11,     /* Channel #11 Delay in ms.  0  to 10000.  Default = 0.  */
-        CH12,     /* Channel #12 Delay in ms.  0  to 10000.  Default = 0.  */
-        CH13,     /* Channel #13 Delay in ms.  0  to 10000.  Default = 0.  */
-        CH14,     /* Channel #14 Delay in ms.  0  to 10000.  Default = 0.  */
-        CH15,     /* Channel #15 Delay in ms.  0  to 10000.  Default = 0.  */
-        MAXDELAY, /* Maximum delay in ms.      0  to 1000.   Default = 10. */
-    }
-
-    public enum DSP_FLANGE : int
-    {
-        MIX,         /* (Type:float) - Percentage of wet signal in mix.  0 to 100. Default = 50. */
-        DEPTH,       /* (Type:float) - Flange depth (percentage of 40ms delay).  0.01 to 1.0.  Default = 1.0. */
-        RATE         /* (Type:float) - Flange speed in hz.  0.0 to 20.0.  Default = 0.1. */
-    }
-
-    public enum DSP_TREMOLO : int
-    {
-        FREQUENCY,     /* LFO frequency in Hz.  0.1 to 20.  Default = 4. */
-        DEPTH,         /* Tremolo depth.  0 to 1.  Default = 0. */
-        SHAPE,         /* LFO shape morph between triangle and sine.  0 to 1.  Default = 0. */
-        SKEW,          /* Time-skewing of LFO cycle.  -1 to 1.  Default = 0. */
-        DUTY,          /* LFO on-time.  0 to 1.  Default = 0.5. */
-        SQUARE,        /* Flatness of the LFO shape.  0 to 1.  Default = 0. */
-        PHASE,         /* Instantaneous LFO phase.  0 to 1.  Default = 0. */
-        SPREAD         /* Rotation / auto-pan effect.  -1 to 1.  Default = 0. */
-    }
-
-    public enum DSP_DISTORTION : int
-    {
-        LEVEL    /* Distortion value.  0.0 to 1.0.  Default = 0.5. */
-    }
-
-    public enum DSP_NORMALIZE : int
-    {
-        FADETIME,    /* Time to ramp the silence to full in ms.  0.0 to 20000.0. Default = 5000.0. */
-        THRESHHOLD,  /* Lower volume range threshold to ignore.  0.0 to 1.0.  Default = 0.1.  Raise higher to stop amplification of very quiet signals. */
-        MAXAMP       /* Maximum amplification allowed.  1.0 to 100000.0.  Default = 20.0.  1.0 = no amplifaction, higher values allow more boost. */
-    }
-
-    public enum DSP_LIMITER : int
-    {
-        RELEASETIME,   /* (Type:float) - Time to ramp the silence to full in ms.  1.0 to 1000.0. Default = 10.0. */
-        CEILING,       /* (Type:float) - Maximum level of the output signal in dB.  -12.0 to 0.0.  Default = 0.0. */
-        MAXIMIZERGAIN, /* (Type:float) - Maximum amplification allowed in dB.  0.0 to 12.0.  Default = 0.0. 0.0 = no amplifaction, higher values allow more boost. */
-        MODE,          /* (Type:float) - Channel processing mode. 0 or 1. Default = 0. 0 = Independent (limiter per channel), 1 = Linked. */
-    }
-    
-    public enum DSP_PARAMEQ : int
-    {
-        CENTER,     /* Frequency center.  20.0 to 22000.0.  Default = 8000.0. */
-        BANDWIDTH,  /* Octave range around the center frequency to filter.  0.2 to 5.0.  Default = 1.0. */
-        GAIN        /* Frequency Gain.  0.05 to 3.0.  Default = 1.0.  */
-    }
-
-    public enum DSP_MULTIBAND_EQ : int
-    {
-        A_FILTER,    /* (Type:int)   - Band A: FMOD_DSP_MULTIBAND_EQ_FILTER_TYPE used to interpret the behavior of the remaining parameters. Default = FMOD_DSP_MULTIBAND_EQ_FILTER_LOWPASS_12DB */
-        A_FREQUENCY, /* (Type:float) - Band A: Significant frequency in Hz, cutoff [low/high pass, low/high shelf], center [notch, peaking, band-pass], phase transition point [all-pass]. 20 to 24000. Default = 8000. */
-        A_Q,         /* (Type:float) - Band A: Quality factory, resonance [low/high pass], bandwidth [notch, peaking, band-pass], phase transition sharpness [all-pass], unused [low/high shelf]. 0.1 to 10.0. Default = 0.707. */
-        A_GAIN,      /* (Type:float) - Band A: Boost or attenuation in dB [peaking, high/low shelf only]. -30 to 30. Default = 0. */
-        B_FILTER,    /* (Type:int)   - Band B: See Band A. Default = FMOD_DSP_MULTIBAND_EQ_FILTER_DISABLED */
-        B_FREQUENCY, /* (Type:float) - Band B: See Band A */
-        B_Q,         /* (Type:float) - Band B: See Band A */
-        B_GAIN,      /* (Type:float) - Band B: See Band A */
-        C_FILTER,    /* (Type:int)   - Band C: See Band A. Default = FMOD_DSP_MULTIBAND_EQ_FILTER_DISABLED */
-        C_FREQUENCY, /* (Type:float) - Band C: See Band A. */
-        C_Q,         /* (Type:float) - Band C: See Band A. */
-        C_GAIN,      /* (Type:float) - Band C: See Band A. */
-        D_FILTER,    /* (Type:int)   - Band D: See Band A. Default = FMOD_DSP_MULTIBAND_EQ_FILTER_DISABLED */
-        D_FREQUENCY, /* (Type:float) - Band D: See Band A. */
-        D_Q,         /* (Type:float) - Band D: See Band A. */
-        D_GAIN,      /* (Type:float) - Band D: See Band A. */
-        E_FILTER,    /* (Type:int)   - Band E: See Band A. Default = FMOD_DSP_MULTIBAND_EQ_FILTER_DISABLED */
-        E_FREQUENCY, /* (Type:float) - Band E: See Band A. */
-        E_Q,         /* (Type:float) - Band E: See Band A. */
-        E_GAIN,      /* (Type:float) - Band E: See Band A. */
-    }
-
-    public enum DSP_MULTIBAND_EQ_FILTER_TYPE : int
-    {
-        DISABLED,       /* Disabled filter, no processing. */
-        LOWPASS_12DB,   /* Resonant low-pass filter, attenuates frequencies (12dB per octave) above a given point (with specificed resonance) while allowing the rest to pass. */
-        LOWPASS_24DB,   /* Resonant low-pass filter, attenuates frequencies (24dB per octave) above a given point (with specificed resonance) while allowing the rest to pass. */
-        LOWPASS_48DB,   /* Resonant low-pass filter, attenuates frequencies (48dB per octave) above a given point (with specificed resonance) while allowing the rest to pass. */
-        HIGHPASS_12DB,  /* Resonant low-pass filter, attenuates frequencies (12dB per octave) below a given point (with specificed resonance) while allowing the rest to pass. */
-        HIGHPASS_24DB,  /* Resonant low-pass filter, attenuates frequencies (24dB per octave) below a given point (with specificed resonance) while allowing the rest to pass. */
-        HIGHPASS_48DB,  /* Resonant low-pass filter, attenuates frequencies (48dB per octave) below a given point (with specificed resonance) while allowing the rest to pass. */
-        LOWSHELF,       /* Low-shelf filter, boosts or attenuates frequencies (with specified gain) below a given point while allowing the rest to pass. */
-        HIGHSHELF,      /* High-shelf filter, boosts or attenuates frequencies (with specified gain) above a given point while allowing the rest to pass. */
-        PEAKING,        /* Peaking filter, boosts or attenuates frequencies (with specified gain) at a given point (with specificed bandwidth) while allowing the rest to pass. */
-        BANDPASS,       /* Band-pass filter, allows frequencies at a given point (with specificed bandwidth) to pass while attenuating frequencies outside this range. */
-        NOTCH,          /* Notch or band-reject filter, attenuates frequencies at a given point (with specificed bandwidth) while allowing frequencies outside this range to pass. */
-        ALLPASS,        /* All-pass filter, allows all frequencies to pass, but changes the phase response at a given point (with specified sharpness). */
-    }
-
-    public enum DSP_PITCHSHIFT : int
-    {
-        PITCH,       /* Pitch value.  0.5 to 2.0.  Default = 1.0. 0.5 = one octave down, 2.0 = one octave up.  1.0 does not change the pitch. */
-        FFTSIZE,     /* FFT window size.  256, 512, 1024, 2048, 4096.  Default = 1024.  Increase this to reduce 'smearing'.  This effect is a warbling sound similar to when an mp3 is encoded at very low bitrates. */
-        OVERLAP,     /* Window overlap.  1 to 32.  Default = 4.  Increase this to reduce 'tremolo' effect.  Increasing it by a factor of 2 doubles the CPU usage. */
-        MAXCHANNELS  /* Maximum channels supported.  0 to 16.  0 = same as fmod's default output polyphony, 1 = mono, 2 = stereo etc.  See remarks for more.  Default = 0.  It is suggested to leave at 0! */
-    }
-
-    public enum DSP_CHORUS : int
-    {
-        MIX,      /* (Type:float) - Volume of original signal to pass to output.  0.0 to 100.0. Default = 50.0. */
-        RATE,     /* (Type:float) - Chorus modulation rate in Hz.  0.0 to 20.0.  Default = 0.8 Hz. */
-        DEPTH,    /* (Type:float) - Chorus modulation depth.  0.0 to 100.0.  Default = 3.0. */
-    }
-
-    public enum DSP_ITECHO : int
-    {
-        WETDRYMIX,      /* (Type:float) - Ratio of wet (processed) signal to dry (unprocessed) signal. Must be in the range from 0.0 through 100.0 (all wet).  Default = 50. */
-        FEEDBACK,       /* (Type:float) - Percentage of output fed back into input, in the range from 0.0 through 100.0.  Default = 50. */
-        LEFTDELAY,      /* (Type:float) - Delay for left channel, in milliseconds, in the range from 1.0 through 2000.0.  Default = 500 ms. */
-        RIGHTDELAY,     /* (Type:float) - Delay for right channel, in milliseconds, in the range from 1.0 through 2000.0.  Default = 500 ms. */
-        PANDELAY        /* (Type:float) - Value that specifies whether to swap left and right delays with each successive echo.  Ranges from 0.0 (equivalent to FALSE) to 1.0 (equivalent to TRUE), meaning no swap.  Default = 0.  CURRENTLY NOT SUPPORTED. */
-    }
-
-    public enum DSP_COMPRESSOR : int
-    {
-        THRESHOLD,   /* (Type:float) - Threshold level (dB) in the range from -80 through 0. The default value is 0. */ 
-        RATIO,       /* (Type:float) - Compression Ratio (dB/dB) in the range from 1 to 50. The default value is 2.5. */ 
-        ATTACK,      /* (Type:float) - Attack time (milliseconds), in the range from 0.1 through 1000. The default value is 20. */
-        RELEASE,     /* (Type:float) - Release time (milliseconds), in the range from 10 through 5000. The default value is 100 */
-        GAINMAKEUP,  /* (Type:float) - Make-up gain (dB) applied after limiting, in the range from 0 through 30. The default value is 0. */
-        USESIDECHAIN,/* (Type:bool)  - Whether to analyse the sidechain signal instead of the input signal. The default value is false */
-        LINKED       /* (Type:bool)  - FALSE = Independent (compressor per channel), TRUE = Linked.  The default value is TRUE. */
-    }
-
-    public enum DSP_SFXREVERB : int
-    {
-        DECAYTIME,           /* (Type:float) - Decay Time       : Reverberation decay time at low-frequencies in milliseconds.  Ranges from 100.0 to 20000.0. Default is 1500. */
-        EARLYDELAY,          /* (Type:float) - Early Delay      : Delay time of first reflection in milliseconds.  Ranges from 0.0 to 300.0.  Default is 20. */
-        LATEDELAY,           /* (Type:float) - Reverb Delay     : Late reverberation delay time relative to first reflection in milliseconds.  Ranges from 0.0 to 100.0.  Default is 40. */
-        HFREFERENCE,         /* (Type:float) - HF Reference     : Reference frequency for high-frequency decay in Hz.  Ranges from 20.0 to 20000.0. Default is 5000. */
-        HFDECAYRATIO,        /* (Type:float) - Decay HF Ratio   : High-frequency decay time relative to decay time in percent.  Ranges from 10.0 to 100.0. Default is 50. */
-        DIFFUSION,           /* (Type:float) - Diffusion        : Reverberation diffusion (echo density) in percent.  Ranges from 0.0 to 100.0.  Default is 100. */
-        DENSITY,             /* (Type:float) - Density          : Reverberation density (modal density) in percent.  Ranges from 0.0 to 100.0.  Default is 100. */
-        LOWSHELFFREQUENCY,   /* (Type:float) - Low Shelf Frequency : Transition frequency of low-shelf filter in Hz.  Ranges from 20.0 to 1000.0. Default is 250. */
-        LOWSHELFGAIN,        /* (Type:float) - Low Shelf Gain   : Gain of low-shelf filter in dB.  Ranges from -36.0 to 12.0.  Default is 0. */
-        HIGHCUT,             /* (Type:float) - High Cut         : Cutoff frequency of low-pass filter in Hz.  Ranges from 20.0 to 20000.0. Default is 20000. */
-        EARLYLATEMIX,        /* (Type:float) - Early/Late Mix   : Blend ratio of late reverb to early reflections in percent.  Ranges from 0.0 to 100.0.  Default is 50. */
-        WETLEVEL,            /* (Type:float) - Wet Level        : Reverb signal level in dB.  Ranges from -80.0 to 20.0.  Default is -6. */
-        DRYLEVEL             /* (Type:float) - Dry Level        : Dry signal level in dB.  Ranges from -80.0 to 20.0.  Default is 0. */
-    }
-
-    public enum DSP_LOWPASS_SIMPLE : int
-    {
-        CUTOFF     /* Lowpass cutoff frequency in hz.  10.0 to 22000.0.  Default = 5000.0 */
-    }
-
-    public enum DSP_SEND : int
-    {
-        RETURNID,     /* (Type:int) - ID of the Return DSP this send is connected to (integer values only). -1 indicates no connected Return DSP. Default = -1. */
-        LEVEL,        /* (Type:float) - Send level. 0.0 to 1.0. Default = 1.0 */
-    }
-
-    public enum DSP_RETURN : int
-    {
-        ID,                 /* (Type:int) - ID of this Return DSP. Read-only.  Default = -1. */
-        INPUT_SPEAKER_MODE  /* (Type:int) - Input speaker mode of this return.  Default = FMOD_SPEAKERMODE_DEFAULT. */
-    }
-
-    public enum DSP_HIGHPASS_SIMPLE : int
-    {
-        CUTOFF     /* (Type:float) - Highpass cutoff frequency in hz.  10.0 to 22000.0.  Default = 1000.0 */
-    }
-
-    public enum DSP_PAN_2D_STEREO_MODE_TYPE : int
-    {
-        DISTRIBUTED,        /* The parts of a stereo sound are spread around desination speakers based on FMOD_DSP_PAN_2D_EXTENT / FMOD_DSP_PAN_2D_DIRECTION */
-        DISCRETE            /* The L/R parts of a stereo sound are rotated around a circle based on FMOD_DSP_PAN_2D_STEREO_AXIS / FMOD_DSP_PAN_2D_STEREO_SEPARATION. */
-    }
-
-    public enum DSP_PAN_MODE_TYPE : int
-    {
-        MONO,
-        STEREO,
-        SURROUND
-    }
-
-    public enum DSP_PAN_3D_ROLLOFF_TYPE : int
-    {
-        LINEARSQUARED,
-        LINEAR,
-        INVERSE,
-        INVERSETAPERED,
-        CUSTOM
-    }
-
-    public enum DSP_PAN_3D_EXTENT_MODE_TYPE : int
-    {
-        AUTO,
-        USER,
-        OFF
-    }
-
-    public enum DSP_PAN : int
-    {
-        MODE,                          /* (Type:int)   - Panner mode.               FMOD_DSP_PAN_MODE_MONO for mono down-mix, FMOD_DSP_PAN_MODE_STEREO for stereo panning or FMOD_DSP_PAN_MODE_SURROUND for surround panning.  Default = FMOD_DSP_PAN_MODE_SURROUND */
-        _2D_STEREO_POSITION,           /* (Type:float) - 2D Stereo pan position.    -100.0 to 100.0.  Default = 0.0. */
-        _2D_DIRECTION,                 /* (Type:float) - 2D Surround pan direction. Direction from center point of panning circle. -180.0 (degrees) to 180.0 (degrees).  0 = front center, -180 or +180 = rear speakers center point. Default = 0.0. */
-        _2D_EXTENT,                    /* (Type:float) - 2D Surround pan extent.    Distance from center point of panning circle.  0.0 (degrees) to 360.0 (degrees).  Default = 360.0. */
-        _2D_ROTATION,                  /* (Type:float) - 2D Surround pan rotation.  -180.0 (degrees) to 180.0 (degrees).  Default = 0.0. */
-        _2D_LFE_LEVEL,                 /* (Type:float) - 2D Surround pan LFE level. 2D LFE level in dB.  -80.0 (db) to 20.0 (db).  Default = 0.0. */
-        _2D_STEREO_MODE,               /* (Type:int)   - Stereo-To-Surround Mode.   FMOD_DSP_PAN_2D_STEREO_MODE_DISTRIBUTED to FMOD_DSP_PAN_2D_STEREO_MODE_DISCRETE.  Default = FMOD_DSP_PAN_2D_STEREO_MODE_DISCRETE.*/
-        _2D_STEREO_SEPARATION,         /* (Type:float) - Stereo-To-Surround Stereo  For FMOD_DSP_PAN_2D_STEREO_MODE_DISCRETE mode.  Separation/width of L/R parts of stereo sound. -180.0 (degrees) to +180.0 (degrees).  Default = 60.0. */
-        _2D_STEREO_AXIS,               /* (Type:float) - Stereo-To-Surround Stereo  For FMOD_DSP_PAN_2D_STEREO_MODE_DISCRETE mode.  Axis/rotation of L/R parts of stereo sound. -180.0 (degrees) to +180.0 (degrees).  Default = 0.0. */
-        ENABLED_SPEAKERS,              /* (Type:int)   - Speakers Enabled.          Bitmask for each speaker from 0 to 32 to be considered by panner.  Use to disable speakers from being panned to.  0 to 0xFFF.  Default = 0xFFF (All on).  */
-        _3D_POSITION,                  /* (Type:data)  - 3D Position.               Data of type FMOD_DSP_PARAMETER_3DATTRIBUTES_MULTI.  See remarks on what to fill out. */
-        _3D_ROLLOFF,                   /* (Type:int)   - 3D Rolloff.                FMOD_DSP_PAN_3D_ROLLOFF_LINEARSQUARED to FMOD_DSP_PAN_3D_ROLLOFF_CUSTOM.  Default = FMOD_DSP_PAN_3D_ROLLOFF_LINEARSQUARED. */
-        _3D_MIN_DISTANCE,              /* (Type:float) - 3D Min Distance.           0.0 to 1e+18f.  Default = 1.0. */
-        _3D_MAX_DISTANCE,              /* (Type:float) - 3D Max Distance.           0.0 to 1e+18f.  Default = 20.0. */
-        _3D_EXTENT_MODE,               /* (Type:int)   - 3D Extent Mode.            FMOD_DSP_PAN_3D_EXTENT_MODE_AUTO to FMOD_DSP_PAN_3D_EXTENT_MODE_OFF.  Default = FMOD_DSP_PAN_3D_EXTENT_MODE_AUTO. */
-        _3D_SOUND_SIZE,                /* (Type:float) - 3D Sound Size.             0.0 to 1e+18f.  Default = 0.0. */
-        _3D_MIN_EXTENT,                /* (Type:float) - 3D Min Extent.             0.0 (degrees) to 360.0 (degrees).  Default = 0.0. */
-        _3D_PAN_BLEND,                 /* (Type:float) - 3D Pan Blend.              0.0 (fully 2D) to 1.0 (fully 3D).  Default = 0.0. */
-        LFE_UPMIX_ENABLED,             /* (Type:int)   - LFE Upmix Enabled.         Determines whether non-LFE source channels should mix to the LFE or leave it alone.  0 (off) to 1 (on).  Default = 0 (off). */
-        OVERALL_GAIN,                  /* (Type:data)  - Overall gain.              For information only, not set by user.  Data of type FMOD_DSP_PARAMETER_DATA_TYPE_OVERALLGAIN to provide to FMOD, to allow FMOD to know the DSP is scaling the signal for virtualization purposes. */
-        SURROUND_SPEAKER_MODE,         /* (Type:int)   - Surround speaker mode.     Target speaker mode for surround panning.  Default = FMOD_SPEAKERMODE_DEFAULT. */
-        _2D_HEIGHT_BLEND,              /* (Type:float) - 2D Height blend.           When FMOD_DSP_PAN_SURROUND_SPEAKER_MODE has height speakers, control the blend between ground and height. 0.0 (ground speakers only) to 1.0 (top speakers only). Default = 0.0. */
-    }
-
-    public enum DSP_THREE_EQ_CROSSOVERSLOPE_TYPE : int
-    {
-        _12DB,
-        _24DB,
-        _48DB
-    }
-
-    public enum DSP_THREE_EQ : int
-    {
-        LOWGAIN,       /* (Type:float) - Low frequency gain in dB.  -80.0 to 10.0.  Default = 0. */
-        MIDGAIN,       /* (Type:float) - Mid frequency gain in dB.  -80.0 to 10.0.  Default = 0. */
-        HIGHGAIN,      /* (Type:float) - High frequency gain in dB.  -80.0 to 10.0.  Default = 0. */
-        LOWCROSSOVER,  /* (Type:float) - Low-to-mid crossover frequency in Hz.  10.0 to 22000.0.  Default = 400.0. */
-        HIGHCROSSOVER, /* (Type:float) - Mid-to-high crossover frequency in Hz.  10.0 to 22000.0.  Default = 4000.0. */
-        CROSSOVERSLOPE /* (Type:int)   - Crossover Slope.  0 = 12dB/Octave, 1 = 24dB/Octave, 2 = 48dB/Octave.  Default = 1 (24dB/Octave). */
-    }
-
-    public enum DSP_FFT_WINDOW : int
-    {
-        RECT,            /* w[n] = 1.0                                                                                            */
-        TRIANGLE,        /* w[n] = TRI(2n/N)                                                                                      */
-        HAMMING,         /* w[n] = 0.54 - (0.46 * COS(n/N) )                                                                      */
-        HANNING,         /* w[n] = 0.5 *  (1.0  - COS(n/N) )                                                                      */
-        BLACKMAN,        /* w[n] = 0.42 - (0.5  * COS(n/N) ) + (0.08 * COS(2.0 * n/N) )                                           */
-        BLACKMANHARRIS   /* w[n] = 0.35875 - (0.48829 * COS(1.0 * n/N)) + (0.14128 * COS(2.0 * n/N)) - (0.01168 * COS(3.0 * n/N)) */
-    }
-
-    public enum DSP_FFT : int
-    {
-        WINDOWSIZE,            /*  (Type:int)   - [r/w] Must be a power of 2 between 128 and 16384.  128, 256, 512, 1024, 2048, 4096, 8192, 16384 are accepted.  Default = 2048. */
-        WINDOWTYPE,            /*  (Type:int)   - [r/w] Refer to FMOD_DSP_FFT_WINDOW enumeration.  Default = FMOD_DSP_FFT_WINDOW_HAMMING. */
-        SPECTRUMDATA,          /*  (Type:data)  - [r]   Returns the current spectrum values between 0 and 1 for each 'fft bin'.  Cast data to FMOD_DSP_PARAMETER_DATA_TYPE_FFT.  Divide the niquist rate by the window size to get the hz value per entry. */
-        DOMINANT_FREQ          /*  (Type:float) - [r]   Returns the dominant frequencies for each channel. */
-    }
-
-    public enum DSP_ENVELOPEFOLLOWER : int
-    {
-        ATTACK,      /* (Type:float) - Attack time (milliseconds), in the range from 0.1 through 1000. The default value is 20. */
-        RELEASE,     /* (Type:float) - Release time (milliseconds), in the range from 10 through 5000. The default value is 100 */
-        ENVELOPE,    /* (Type:float) - Current value of the envelope, in the range 0 to 1. Read-only. */
-        USESIDECHAIN /* (Type:bool)  - Whether to analyse the sidechain signal instead of the input signal. The default value is false */
-    }
-
-    public enum DSP_CONVOLUTION_REVERB : int
-    {
-        IR,       /* (Type:data)  - [w]   16-bit reverb IR (short*) with an extra sample prepended to the start which specifies the number of channels. */
-        WET,      /* (Type:float) - [r/w] Volume of echo signal to pass to output in dB.  -80.0 to 10.0.  Default = 0. */
-        DRY,      /* (Type:float) - [r/w] Original sound volume in dB.  -80.0 to 10.0.  Default = 0. */
-        LINKED    /* (Type:bool)  - [r/w] Linked - channels are mixed together before processing through the reverb.  Default = TRUE. */
-    }
-
-    public enum DSP_CHANNELMIX_OUTPUT : int
-    {
-        DEFAULT,            /*  Output channel count = input channel count.  Mapping: See FMOD_SPEAKER enumeration. */
-        ALLMONO,            /*  Output channel count = 1.  Mapping: Mono, Mono, Mono, Mono, Mono, Mono, ... (each channel all the way up to FMOD_MAX_CHANNEL_WIDTH channels are treated as if they were mono) */
-        ALLSTEREO,          /*  Output channel count = 2.  Mapping: Left, Right, Left, Right, Left, Right, ... (each pair of channels is treated as stereo all the way up to FMOD_MAX_CHANNEL_WIDTH channels) */
-        ALLQUAD,            /*  Output channel count = 4.  Mapping: Repeating pattern of Front Left, Front Right, Surround Left, Surround Right. */
-        ALL5POINT1,         /*  Output channel count = 6.  Mapping: Repeating pattern of Front Left, Front Right, Center, LFE, Surround Left, Surround Right. */
-        ALL7POINT1,         /*  Output channel count = 8.  Mapping: Repeating pattern of Front Left, Front Right, Center, LFE, Surround Left, Surround Right, Back Left, Back Right.  */
-        ALLLFE,             /*  Output channel count = 6.  Mapping: Repeating pattern of LFE in a 5.1 output signal.  */
-        ALL7POINT1POINT4    /*  Output channel count = 12. Mapping: Repeating pattern of Front Left, Front Right, Center, LFE, Surround Left, Surround Right, Back Left, Back Right, Top Front Left, Top Front Right, Top Back Left, Top Back Right.  */
-    }
-
-    public enum DSP_CHANNELMIX : int
-    {
-        OUTPUTGROUPING,     /* (Type:int)   - Refer to FMOD_DSP_CHANNELMIX_OUTPUT enumeration.  Default = FMOD_DSP_CHANNELMIX_OUTPUT_DEFAULT.  See remarks. */
-        GAIN_CH0,           /* (Type:float) - Channel  #0 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH1,           /* (Type:float) - Channel  #1 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH2,           /* (Type:float) - Channel  #2 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH3,           /* (Type:float) - Channel  #3 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH4,           /* (Type:float) - Channel  #4 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH5,           /* (Type:float) - Channel  #5 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH6,           /* (Type:float) - Channel  #6 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH7,           /* (Type:float) - Channel  #7 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH8,           /* (Type:float) - Channel  #8 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH9,           /* (Type:float) - Channel  #9 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH10,          /* (Type:float) - Channel #10 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH11,          /* (Type:float) - Channel #11 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH12,          /* (Type:float) - Channel #12 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH13,          /* (Type:float) - Channel #13 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH14,          /* (Type:float) - Channel #14 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH15,          /* (Type:float) - Channel #15 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH16,          /* (Type:float) - Channel #16 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH17,          /* (Type:float) - Channel #17 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH18,          /* (Type:float) - Channel #18 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH19,          /* (Type:float) - Channel #19 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH20,          /* (Type:float) - Channel #20 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH21,          /* (Type:float) - Channel #21 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH22,          /* (Type:float) - Channel #22 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH23,          /* (Type:float) - Channel #23 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH24,          /* (Type:float) - Channel #24 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH25,          /* (Type:float) - Channel #25 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH26,          /* (Type:float) - Channel #26 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH27,          /* (Type:float) - Channel #27 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH28,          /* (Type:float) - Channel #28 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH29,          /* (Type:float) - Channel #29 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH30,          /* (Type:float) - Channel #30 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        GAIN_CH31,          /* (Type:float) - Channel #31 gain in dB.  -80.0 to 10.0.  Default = 0. */
-        OUTPUT_CH0,         /* (Type:int)   - Input channel #0  Output channel.   0 to 31.  Default = 0. */
-        OUTPUT_CH1,         /* (Type:int)   - Input channel #1  Output channel.   0 to 31.  Default = 1. */
-        OUTPUT_CH2,         /* (Type:int)   - Input channel #2  Output channel.   0 to 31.  Default = 2. */
-        OUTPUT_CH3,         /* (Type:int)   - Input channel #3  Output channel.   0 to 31.  Default = 3. */
-        OUTPUT_CH4,         /* (Type:int)   - Input channel #4  Output channel.   0 to 31.  Default = 4. */
-        OUTPUT_CH5,         /* (Type:int)   - Input channel #5  Output channel.   0 to 31.  Default = 5. */
-        OUTPUT_CH6,         /* (Type:int)   - Input channel #6  Output channel.   0 to 31.  Default = 6. */
-        OUTPUT_CH7,         /* (Type:int)   - Input channel #7  Output channel.   0 to 31.  Default = 7. */
-        OUTPUT_CH8,         /* (Type:int)   - Input channel #8  Output channel.   0 to 31.  Default = 8. */
-        OUTPUT_CH9,         /* (Type:int)   - Input channel #9  Output channel.   0 to 31.  Default = 9. */
-        OUTPUT_CH10,        /* (Type:int)   - Input channel #10 Output channel.   0 to 31.  Default = 10. */
-        OUTPUT_CH11,        /* (Type:int)   - Input channel #11 Output channel.   0 to 31.  Default = 11. */
-        OUTPUT_CH12,        /* (Type:int)   - Input channel #12 Output channel.   0 to 31.  Default = 12. */
-        OUTPUT_CH13,        /* (Type:int)   - Input channel #13 Output channel.   0 to 31.  Default = 13. */
-        OUTPUT_CH14,        /* (Type:int)   - Input channel #14 Output channel.   0 to 31.  Default = 14. */
-        OUTPUT_CH15,        /* (Type:int)   - Input channel #15 Output channel.   0 to 31.  Default = 15. */
-        OUTPUT_CH16,        /* (Type:int)   - Input channel #16 Output channel.   0 to 31.  Default = 16. */
-        OUTPUT_CH17,        /* (Type:int)   - Input channel #17 Output channel.   0 to 31.  Default = 17. */
-        OUTPUT_CH18,        /* (Type:int)   - Input channel #18 Output channel.   0 to 31.  Default = 18. */
-        OUTPUT_CH19,        /* (Type:int)   - Input channel #19 Output channel.   0 to 31.  Default = 19. */
-        OUTPUT_CH20,        /* (Type:int)   - Input channel #20 Output channel.   0 to 31.  Default = 20. */
-        OUTPUT_CH21,        /* (Type:int)   - Input channel #21 Output channel.   0 to 31.  Default = 21. */
-        OUTPUT_CH22,        /* (Type:int)   - Input channel #22 Output channel.   0 to 31.  Default = 22. */
-        OUTPUT_CH23,        /* (Type:int)   - Input channel #23 Output channel.   0 to 31.  Default = 23. */
-        OUTPUT_CH24,        /* (Type:int)   - Input channel #24 Output channel.   0 to 31.  Default = 24. */
-        OUTPUT_CH25,        /* (Type:int)   - Input channel #25 Output channel.   0 to 31.  Default = 25. */
-        OUTPUT_CH26,        /* (Type:int)   - Input channel #26 Output channel.   0 to 31.  Default = 26. */
-        OUTPUT_CH27,        /* (Type:int)   - Input channel #27 Output channel.   0 to 31.  Default = 27. */
-        OUTPUT_CH28,        /* (Type:int)   - Input channel #28 Output channel.   0 to 31.  Default = 28. */
-        OUTPUT_CH29,        /* (Type:int)   - Input channel #29 Output channel.   0 to 31.  Default = 29. */
-        OUTPUT_CH30,        /* (Type:int)   - Input channel #30 Output channel.   0 to 31.  Default = 30. */
-        OUTPUT_CH31,        /* (Type:int)   - Input channel #31 Output channel.   0 to 31.  Default = 31. */
-    }
-
-    public enum DSP_TRANSCEIVER_SPEAKERMODE : int
-    {
-        AUTO = -1,     /* A transmitter will use whatever signal channel count coming in to the transmitter, to determine which speaker mode is allocated for the transceiver channel. */
-        MONO = 0,      /* A transmitter will always downmix to a mono channel buffer. */
-        STEREO,        /* A transmitter will always upmix or downmix to a stereo channel buffer. */
-        SURROUND,      /* A transmitter will always upmix or downmix to a surround channel buffer.   Surround is the speaker mode of the system above stereo, so could be quad/surround/5.1/7.1. */
-    }
-
-    public enum DSP_TRANSCEIVER : int
-    {
-        TRANSMIT,            /* (Type:bool)  - [r/w] - FALSE = Transceiver is a 'receiver' (like a return) and accepts data from a channel.  TRUE = Transceiver is a 'transmitter' (like a send).  Default = FALSE. */
-        GAIN,                /* (Type:float) - [r/w] - Gain to receive or transmit at in dB.  -80.0 to 10.0.  Default = 0. */
-        CHANNEL,             /* (Type:int)   - [r/w] - Integer to select current global slot, shared by all Transceivers, that can be transmitted to or received from.  0 to 31.  Default = 0.*/
-        TRANSMITSPEAKERMODE  /* (Type:int)   - [r/w] - Speaker mode (transmitter mode only).  Specifies either 0 (Auto) Default = 0.*/
-    }
-
-    public enum DSP_OBJECTPAN : int
-    {
-        _3D_POSITION,        /* (Type:data)  - 3D Position.               data of type FMOD_DSP_PARAMETER_3DATTRIBUTES_MULTI */
-        _3D_ROLLOFF,         /* (Type:int)   - 3D Rolloff.                FMOD_DSP_PAN_3D_ROLLOFF_LINEARSQUARED to FMOD_DSP_PAN_3D_ROLLOFF_CUSTOM.  Default = FMOD_DSP_PAN_3D_ROLLOFF_LINEARSQUARED. */
-        _3D_MIN_DISTANCE,    /* (Type:float) - 3D Min Distance.           0.0 to 1e+19f.  Default = 1.0. */
-        _3D_MAX_DISTANCE,    /* (Type:float) - 3D Max Distance.           0.0 to 1e+19f.  Default = 20.0. */
-        _3D_EXTENT_MODE,     /* (Type:int)   - 3D Extent Mode.            FMOD_DSP_PAN_3D_EXTENT_MODE_AUTO to FMOD_DSP_PAN_3D_EXTENT_MODE_OFF.  Default = FMOD_DSP_PAN_3D_EXTENT_MODE_AUTO. */
-        _3D_SOUND_SIZE,      /* (Type:float) - 3D Sound Size.             0.0 to 1e+19f.  Default = 0.0. */
-        _3D_MIN_EXTENT,      /* (Type:float) - 3D Min Extent.             0.0 (degrees) to 360.0 (degrees).  Default = 0.0. */
-        OVERALL_GAIN,        /* (Type:data)  - Overall gain.              For information only, not set by user.  Data of type FMOD_DSP_PARAMETER_DATA_TYPE_OVERALLGAIN to provide to FMOD, to allow FMOD to know the DSP is scaling the signal for virtualization purposes. */
-        OUTPUTGAIN           /* (Type:float) - Output gain level.         0.0 to 1.0 linear scale.  For the user to scale the output of the object panner's signal. */
-    }
-}