Add support for 24 and 32 bit .wav's

Apparently those .wav's are always format_code 3?

The conversion is SLOW on long files (like songs).
It requires going through every byte to do the conversion.
So if performance is important, GDScript is just not fast enough for this.

Author: L4Vo5

GDScriptAudioImport.gd

@@ -56,15 +56,15 @@ func loadfile(filepath):
 		return AudioStreamSample.new()
 
 	var bytes = file.get_buffer(file.get_len())
-
 	# if File is wav
 	if filepath.ends_with(".wav"):
-
 		var newstream = AudioStreamSample.new()
 
 		#---------------------------
 		#parrrrseeeeee!!! :D
-
+		
+		var bits_per_sample = 0
+		
 		for i in range(0, 100):
 			var those4bytes = str(char(bytes[i])+char(bytes[i+1])+char(bytes[i+2])+char(bytes[i+3]))
 
@@ -90,6 +90,9 @@ func loadfile(filepath):
 				if format_code == 0: format_name = "8_BITS"
 				elif format_code == 1: format_name = "16_BITS"
 				elif format_code == 2: format_name = "IMA_ADPCM"
+				else: 
+					format_name = "UNKNOWN (trying to interpret as 16_BITS)"
+					format_code = 1
 				print ("Format: " + str(format_code) + " " + format_name)
 				#assign format to our AudioStreamSample
 				newstream.format = format_code
@@ -114,20 +117,25 @@ func loadfile(filepath):
 				var bits_sample_channel = bytes[fsc0+12] + (bytes[fsc0+13] << 8)
 				print ("BitsPerSample * Channel / 8: " + str(bits_sample_channel))
 
-				#aaaand bits per sample/bitrate [Bytes 14-15] - TODO: Handle different bitrates
-				var bits_per_sample = bytes[fsc0+14] + (bytes[fsc0+15] << 8)
+				#aaaand bits per sample/bitrate [Bytes 14-15]
+				bits_per_sample = bytes[fsc0+14] + (bytes[fsc0+15] << 8)
 				print ("Bits per sample: " + str(bits_per_sample))
 
-				
 			if those4bytes == "data":
+				assert(bits_per_sample != 0)
+				
 				var audio_data_size = bytes[i+4] + (bytes[i+5] << 8) + (bytes[i+6] << 16) + (bytes[i+7] << 24)
 				print ("Audio data/stream size is " + str(audio_data_size) + " bytes")
 
 				var data_entry_point = (i+8)
 				print ("Audio data starts at byte " + str(data_entry_point))
 
-				newstream.data = bytes.subarray(data_entry_point, data_entry_point+audio_data_size-1)
+				var data = bytes.subarray(data_entry_point, data_entry_point+audio_data_size-1)
 
+				if bits_per_sample in [24, 32]:
+					newstream.data = convert_to_16bit(data, bits_per_sample)
+				else:
+					newstream.data = data
 			# end of parsing
 			#---------------------------
 
@@ -155,6 +163,44 @@ func loadfile(filepath):
 		print ("ERROR: Wrong filetype or format")
 	file.close()
 
+# Converts .wav data from 24 or 32 bits to 16
+#
+# These conversions are SLOW in GDScript
+# on my one test song, 32 -> 16 was around 3x slower than 24 -> 16
+#
+# I couldn't get threads to help very much
+# They made the 24bit case about 2x faster in my test file
+# And the 32bit case abour 50% slower
+# I don't wanna risk it always being slower on other files
+# And really, the solution would be to handle it in a low-level language
+func convert_to_16bit(data: PoolByteArray, from: int) -> PoolByteArray:
+	print("converting to 16-bit from %d" % from)
+	var time = OS.get_ticks_msec()
+	# 24 bit .wav's are typically stored as integers
+	# so we just grab the 2 most significant bytes and ignore the other
+	if from == 24:
+		var j = 0
+		for i in range(0, data.size(), 3):
+			data[j] = data[i+1]
+			data[j+1] = data[i+2]
+			j += 2
+		data.resize(data.size() * 2 / 3)
+	# 32 bit .wav's are typically stored as floating point numbers
+	# so we need to grab all 4 bytes and interpret them as a float first
+	if from == 32:
+		var spb := StreamPeerBuffer.new()
+		var single_float: float
+		var value: int
+		for i in range(0, data.size(), 4):
+			spb.data_array = data.subarray(i, i+3)
+			single_float = spb.get_float()
+			value = single_float * 32768
+			data[i/2] = value
+			data[i/2+1] = value >> 8
+		data.resize(data.size() / 2)
+	print("Took %f seconds for slow conversion" % ((OS.get_ticks_msec() - time) / 1000.0))
+	return data
+
 
 # ---------- REFERENCE ---------------
 # note: typical values doesn't always match