Formatting Fix

core/odin/parser/parser.odin

@@ -1276,28 +1276,28 @@ parse_unrolled_for_loop :: proc(p: ^Parser, inline_tok: tokenizer.Token) -> ^ast
 			args = make([dynamic]^ast.Expr)
 			for p.curr_tok.kind != .Close_Paren &&
 			    p.curr_tok.kind != .EOF {
-			    	arg := parse_value(p)
+				arg := parse_value(p)
 
-			    	if p.curr_tok.kind == .Eq {
-			    		eq := expect_token(p, .Eq)
-			    		if arg != nil {
-			    			if _, ok := arg.derived.(^ast.Ident); !ok {
-			    				error(p, arg.pos, "expected an identifier for 'key=value'")
-			    			}
-			    		}
-			    		value := parse_value(p)
-			    		fv := ast.new(ast.Field_Value, arg.pos, value)
-			    		fv.field = arg
-			    		fv.sep   = eq.pos
-			    		fv.value = value
+				if p.curr_tok.kind == .Eq {
+					eq := expect_token(p, .Eq)
+					if arg != nil {
+						if _, ok := arg.derived.(^ast.Ident); !ok {
+							error(p, arg.pos, "expected an identifier for 'key=value'")
+						}
+					}
+					value := parse_value(p)
+					fv := ast.new(ast.Field_Value, arg.pos, value)
+					fv.field = arg
+					fv.sep   = eq.pos
+					fv.value = value
 
-			    		arg = fv
-			    	}
+					arg = fv
+				}
 
-			    	append(&args, arg)
+				append(&args, arg)
 
 				allow_token(p, .Comma) or_break
-			    }
+			}
 		}
 
 		p.expr_level -= 1

examples/all/all_vendor_windows.odin

@@ -3,8 +3,10 @@ package all
 import wgpu "vendor:wgpu"
 import b2 "vendor:box2d"
 import game_input "vendor:windows/GameInput"
+import XAudio2 "vendor:windows/XAudio2"
 
 _ :: wgpu
 _ :: b2
 _ :: game_input
+_ :: XAudio2
 

vendor/windows/XAudio2/x3daudio.odin

@@ -1,7 +1,7 @@
 #+build windows
 /* NOTES:
 	1.  Definition of terms:
-	    LFE: Low Frequency Effect -- always omnidirectional.
+	    LFE: Low Frequency Effect -- always omnidirectional.
 	    LPF: Low Pass Filter, divided into two classifications:
 		 Direct -- Applied to the direct signal path,
 			   used for obstruction/occlusion effects.
@@ -9,7 +9,7 @@
 			   used for occlusion effects only.
 
 	2.  Volume level is expressed as a linear amplitude scaler:
-	1.0f represents no attenuation applied to the original signal,
+	1.0f represents no attenuation applied to the original signal,
 	0.5f denotes an attenuation of 6dB, and 0.0f results in silence.
 	Amplification (volume > 1.0f) is also allowed, and is not clamped.
 
@@ -17,7 +17,7 @@
 	to 0.0f which results in silence as all frequencies are filtered out.
 
 	3.  X3DAudio uses a left-handed Cartesian coordinate system with values
-	on the x-axis increasing from left to right, on the y-axis from
+	on the x-axis increasing from left to right, on the y-axis from
 	bottom to top, and on the z-axis from near to far.
 	Azimuths are measured clockwise from a given reference direction.
 
@@ -29,7 +29,7 @@
 	Distance is calculated using the Euclidean norm formula.
 
 	4.  Only real values are permissible with functions using 32-bit
-	float parameters -- NAN and infinite values are not accepted.
+	float parameters -- NAN and infinite values are not accepted.
 	All computation occurs in 32-bit precision mode.                    */
 
 
@@ -100,7 +100,7 @@ CALCULATE_FLAG :: enum u32 {
 
 //--------------<D-A-T-A---T-Y-P-E-S>---------------------------------------//
 VECTOR :: [3]f32 // float 3D vector
-
+
 // instance handle of precalculated constants
 HANDLE :: distinct [HANDLE_BYTESIZE]byte
 
@@ -123,22 +123,22 @@ DISTANCE_CURVE_POINT :: struct #packed {
 // For all points in the distance curve where 1 <= k < PointCount.
 DISTANCE_CURVE :: struct #packed {
 	pPoints:    [^]DISTANCE_CURVE_POINT `fmt:"v,PointCount"`, // distance curve point array, must have at least PointCount elements with no duplicates and be sorted in ascending order with respect to Distance
-	PointCount: u32,                        // number of distance curve points, must be >= 2 as all distance curves must have at least two endpoints, defining DSP settings at 0.0f and 1.0f normalized distance
-}
+	PointCount: u32,                                          // number of distance curve points, must be >= 2 as all distance curves must have at least two endpoints, defining DSP settings at 0.0f and 1.0f normalized distance
+}
 Default_LinearCurvePoints := [2]DISTANCE_CURVE_POINT{{0.0, 1.0}, {1.0, 0.0}}
 Default_LinearCurve       := DISTANCE_CURVE{&Default_LinearCurvePoints[0], 2}
 
 CONE :: struct #packed {
 	InnerAngle:  f32, // inner cone angle in radians, must be within [0.0f, TAU]
-	OuterAngle:  f32,   // outer cone angle in radians, must be within [InnerAngle, TAU]
-
+	OuterAngle:  f32, // outer cone angle in radians, must be within [InnerAngle, TAU]
+
 	InnerVolume: f32, // volume level scaler on/within inner cone, used only for matrix calculations, must be within [0.0f, 2.0f] when used
-	OuterVolume: f32,   // volume level scaler on/beyond outer cone, used only for matrix calculations, must be within [0.0f, 2.0f] when used
-	InnerLPF:    f32,   // LPF (both direct and reverb paths) coefficient subtrahend on/within inner cone, used only for LPF (both direct and reverb paths) calculations, must be within [0.0f, 1.0f] when used
-	OuterLPF:    f32,   // LPF (both direct and reverb paths) coefficient subtrahend on/beyond outer cone, used only for LPF (both direct and reverb paths) calculations, must be within [0.0f, 1.0f] when used
-	InnerReverb: f32,   // reverb send level scaler on/within inner cone, used only for reverb calculations, must be within [0.0f, 2.0f] when used
-	OuterReverb: f32,   // reverb send level scaler on/beyond outer cone, used only for reverb calculations, must be within [0.0f, 2.0f] when used
-}
+	OuterVolume: f32, // volume level scaler on/beyond outer cone, used only for matrix calculations, must be within [0.0f, 2.0f] when used
+	InnerLPF:    f32, // LPF (both direct and reverb paths) coefficient subtrahend on/within inner cone, used only for LPF (both direct and reverb paths) calculations, must be within [0.0f, 1.0f] when used
+	OuterLPF:    f32, // LPF (both direct and reverb paths) coefficient subtrahend on/beyond outer cone, used only for LPF (both direct and reverb paths) calculations, must be within [0.0f, 1.0f] when used
+	InnerReverb: f32, // reverb send level scaler on/within inner cone, used only for reverb calculations, must be within [0.0f, 2.0f] when used
+	OuterReverb: f32, // reverb send level scaler on/beyond outer cone, used only for reverb calculations, must be within [0.0f, 2.0f] when used
+}
 Default_DirectionalCone := CONE{math.PI / 2, math.PI, 1.0, 0.708, 0.0, 0.25, 0.708, 1.0}
 
 // Listener:
@@ -147,13 +147,13 @@ Default_DirectionalCone := CONE{math.PI / 2, math.PI, 1.0, 0.708, 0.0, 0.25, 0.7
 // The cone is directed by the listener's front orientation.
 LISTENER :: struct #packed {
 	OrientFront: VECTOR, // orientation of front direction, used only for matrix and delay calculations or listeners with cones for matrix, LPF (both direct and reverb paths), and reverb calculations, must be normalized when used
-	OrientTop:   VECTOR,   // orientation of top direction, used only for matrix and delay calculations, must be orthonormal with OrientFront when used
-
+	OrientTop:   VECTOR, // orientation of top direction, used only for matrix and delay calculations, must be orthonormal with OrientFront when used
+
 	Position:    VECTOR, // position in user-defined world units, does not affect Velocity
-	Velocity: VECTOR,      // velocity vector in user-defined world units/second, used only for doppler calculations, does not affect Position
-
+	Velocity:    VECTOR, // velocity vector in user-defined world units/second, used only for doppler calculations, does not affect Position
+
 	pCone:       ^CONE,  // sound cone, used only for matrix, LPF (both direct and reverb paths), and reverb calculations, NULL specifies omnidirectionality
-}
+}
 
 // Emitter:
 // Defines a 3D audio source, divided into two classifications:
@@ -187,9 +187,9 @@ EMITTER :: struct #packed {
 	InnerRadiusAngle: f32,    // inner radius angle, must be within [0.0f, PI/4.0)
 
 	ChannelCount:     u32,                           // number of sound channels, must be > 0
-	ChannelRadius:    f32,       // channel radius, used only with multi-channel emitters for matrix calculations, must be >= 0.0f when used
-	pChannelAzimuths: [^]f32,    // channel azimuth array, used only with multi-channel emitters for matrix calculations, contains positions of each channel expressed in radians along the channel radius with respect to the front orientation vector in the plane orthogonal to the top orientation vector, or TAU to specify an LFE channel, must have at least ChannelCount elements, all within [0.0f, TAU] when used
-
+	ChannelRadius:    f32,                           // channel radius, used only with multi-channel emitters for matrix calculations, must be >= 0.0f when used
+	pChannelAzimuths: [^]f32 `fmt:"v,ChannelCount"`, // channel azimuth array, used only with multi-channel emitters for matrix calculations, contains positions of each channel expressed in radians along the channel radius with respect to the front orientation vector in the plane orthogonal to the top orientation vector, or TAU to specify an LFE channel, must have at least ChannelCount elements, all within [0.0f, TAU] when used
+
 	pVolumeCurve:    ^DISTANCE_CURVE,    // volume level distance curve, used only for matrix calculations, NULL specifies a default curve that conforms to the inverse square law, calculated in user-defined world units with distances <= CurveDistanceScaler clamped to no attenuation
 	pLFECurve:       ^DISTANCE_CURVE,    // LFE level distance curve, used only for matrix calculations, NULL specifies a default curve that conforms to the inverse square law, calculated in user-defined world units with distances <= CurveDistanceScaler clamped to no attenuation
 	pLPFDirectCurve: ^DISTANCE_CURVE,    // LPF direct-path coefficient distance curve, used only for LPF direct-path calculations, NULL specifies the default curve: [0.0f,1.0f], [1.0f,0.75f]

vendor/windows/XAudio2/xapo.odin

@@ -1,7 +1,7 @@
 #+build windows
 /* NOTES:
 	1.  Definition of terms:
-	    DSP: Digital Signal Processing.
+	    DSP: Digital Signal Processing.
 
 	    CBR: Constant BitRate -- DSP that consumes a constant number of
 		 input samples to produce an output sample.
@@ -27,7 +27,7 @@
 		      when processing in-place.
 
 	2.  XAPO member variables are divided into three classifications:
-	    Immutable: Set once via IXAPO.Initialize and remain
+	    Immutable: Set once via IXAPO.Initialize and remain
 		       constant during the lifespan of the XAPO.
 
 	    Locked: May change before the XAPO is locked via
@@ -48,27 +48,27 @@
 	best performance, maintainability, and ease of use.
 
 	3.  To minimize glitches, the realtime audio processing thread must
-	not block.  XAPO methods called by the realtime thread are commented
+	not block.  XAPO methods called by the realtime thread are commented
 	as non-blocking and therefore should not use blocking synchronization,
 	allocate memory, access the disk, etc.  The XAPO interfaces were
 	designed to allow an effect implementer to move such operations
 	into other methods called on an application controlled thread.
 
 	4.  Extending functionality is accomplished through the addition of new
-	COM interfaces.  For example, if a new member is added to a parameter
+	COM interfaces.  For example, if a new member is added to a parameter
 	structure, a new interface using the new structure should be added,
 	leaving the original interface unchanged.
 	This ensures consistent communication between future versions of
 	XAudio2 and various versions of XAPOs that may exist in an application.
 
 	5.  All audio data is interleaved in XAudio2.
-	The default audio format for an effect chain is WAVE_FORMAT_IEEE_FLOAT.
+	The default audio format for an effect chain is WAVE_FORMAT_IEEE_FLOAT.
 
 	6.  User-defined XAPOs should assume all input and output buffers are
-	16-byte aligned.
+	16-byte aligned.
 
 	7.  See XAPOBase.odin for an XAPO base class which provides a default
-	implementation for most of the interface methods defined below.     */
+	implementation for most of the interface methods defined below.     */
 
 package windows_xaudio2
 

vendor/windows/XAudio2/xapofx.odin

@@ -110,7 +110,7 @@ FXMASTERINGLIMITER_PARAMETERS :: struct #packed {
 	Diffusion: f32,  // diffusion
 	RoomSize:  f32,  // room size
 }
-
+
 
 // Echo initialization data, used with CreateFX:
 // Use of this structure is optional, the default MaxDelay is FXECHO_DEFAULT_DELAY.

vendor/windows/XAudio2/xaudio2.odin

@@ -11,12 +11,10 @@ package windows_xaudio2
 import win "core:sys/windows"
 import "core:math"
 
-foreign import xa2 "system:xaudio2.lib"
-
-HRESULT :: win.HRESULT
-IUnknown :: win.IUnknown
+HRESULT         :: win.HRESULT
+IUnknown        :: win.IUnknown
 IUnknown_VTable :: win.IUnknown_VTable
-WAVEFORMATEX :: win.WAVEFORMATEX
+WAVEFORMATEX    :: win.WAVEFORMATEX
 
 /**************************************************************************
  *

vendor/windows/XAudio2/xaudio2fx.odin

@@ -32,9 +32,9 @@ foreign xa2 {
 // The volume meter does not support SetEffectParameters().
 VOLUMEMETER_LEVELS :: struct #packed {
 	pPeakLevels:  [^]f32 `fmt:"v,ChannelCount"`, // Peak levels table: receives maximum absolute level for each channel over a processing pass, may be NULL if pRMSLevls != NULL, otherwise must have at least ChannelCount elements.
-	pRMSLevels:   [^]f32,   // Root mean square levels table: receives RMS level for each channel over a processing pass, may be NULL if pPeakLevels != NULL, otherwise must have at least ChannelCount elements.
-	ChannelCount: u32, 	    // Number of channels being processed by the volume meter APO
-}
+	pRMSLevels:   [^]f32 `fmt:"v,ChannelCount"`, // Root mean square levels table: receives RMS level for each channel over a processing pass, may be NULL if pPeakLevels != NULL, otherwise must have at least ChannelCount elements.
+	ChannelCount: u32,                           // Number of channels being processed by the volume meter APO
+}
 
 /**************************************************************************
  *