Add AudioBufferDataFormat

project/include/media/AudioBuffer.h

@@ -35,6 +35,7 @@ namespace lime {
 		int bitsPerSample;
 		int channels;
 		ArrayBufferView* data;
+		int dataFormat;
 		int sampleRate;
 
 		vdynamic* __srcAudio;

project/src/ExternalInterface.cpp

@@ -4098,7 +4098,7 @@ namespace lime {
 
 	#define _TARRAYBUFFER _TBYTES
 	#define _TARRAYBUFFERVIEW _OBJ (_I32 _TARRAYBUFFER _I32 _I32 _I32 _I32)
-	#define _TAUDIOBUFFER _OBJ (_I32 _I32 _TARRAYBUFFERVIEW _I32 _DYN _DYN _DYN _DYN _DYN _TVORBISFILE)
+	#define _TAUDIOBUFFER _OBJ (_I32 _I32 _TARRAYBUFFERVIEW _I32 _I32 _DYN _DYN _DYN _DYN _DYN _TVORBISFILE)
 	#define _TIMAGEBUFFER _OBJ (_I32 _TARRAYBUFFERVIEW _I32 _I32 _BOOL _BOOL _I32 _DYN _DYN _DYN _DYN _DYN _DYN)
 	#define _TIMAGE _OBJ (_TIMAGEBUFFER _BOOL _I32 _I32 _I32 _TRECTANGLE _ENUM _I32 _I32 _F64 _F64)
 

project/src/media/AudioBuffer.cpp

@@ -7,6 +7,7 @@ namespace lime {
 	static int id_bitsPerSample;
 	static int id_channels;
 	static int id_data;
+	static int id_dataFormat;
 	static int id_sampleRate;
 	static bool init = false;
 
@@ -18,6 +19,7 @@ namespace lime {
 			id_bitsPerSample = val_id ("bitsPerSample");
 			id_channels = val_id ("channels");
 			id_data = val_id ("data");
+			id_dataFormat = val_id ("dataFormat");
 			id_sampleRate = val_id ("sampleRate");
 			init = true;
 
@@ -28,13 +30,15 @@ namespace lime {
 			bitsPerSample = val_int (val_field (audioBuffer, id_bitsPerSample));
 			channels = val_int (val_field (audioBuffer, id_channels));
 			data = new ArrayBufferView (val_field (audioBuffer, id_data));
+			dataFormat = val_int (val_field (audioBuffer, id_dataFormat));
 			sampleRate = val_int (val_field (audioBuffer, id_sampleRate));
 
 		} else {
 
 			bitsPerSample = 0;
 			channels = 0;
 			// data = new ArrayBufferView ();
+			dataFormat = 0;
 			sampleRate = 0;
 
 		}
@@ -69,6 +73,7 @@ namespace lime {
 			id_bitsPerSample = val_id ("bitsPerSample");
 			id_channels = val_id ("channels");
 			id_data = val_id ("data");
+			id_dataFormat = val_id ("dataFormat");
 			id_sampleRate = val_id ("sampleRate");
 			init = true;
 
@@ -77,6 +82,7 @@ namespace lime {
 		alloc_field (audioBuffer, id_bitsPerSample, alloc_int (bitsPerSample));
 		alloc_field (audioBuffer, id_channels, alloc_int (channels));
 		alloc_field (audioBuffer, id_data, data ? data->Value (val_field (audioBuffer, id_data)) : alloc_null ());
+		alloc_field (audioBuffer, id_dataFormat, alloc_int (dataFormat));
 		alloc_field (audioBuffer, id_sampleRate, alloc_int (sampleRate));
 		return audioBuffer;
 

project/src/media/containers/OGG.cpp

@@ -55,6 +55,7 @@ namespace lime {
 		audioBuffer->sampleRate = pInfo->rate;
 
 		audioBuffer->bitsPerSample = 16;
+		audioBuffer->dataFormat = 1;
 
 		int dataLength = ov_pcm_total (oggFile, -1) * audioBuffer->channels * audioBuffer->bitsPerSample / 8;
 		audioBuffer->data->Resize (dataLength);

project/src/media/containers/WAV.cpp

@@ -204,6 +204,7 @@ namespace lime {
 		audioBuffer->sampleRate = (int)wave_format.sampleRate;
 		audioBuffer->channels = wave_format.numChannels;
 		audioBuffer->bitsPerSample = wave_format.bitsPerSample;
+		audioBuffer->dataFormat = wave_format.audioFormat;
 
 		return true;
 

src/lime/media/AudioBuffer.hx

@@ -33,10 +33,10 @@ import flash.net.URLRequest;
 #end
 
 /**
-	The `AudioBuffer` class represents a buffer of audio data that can be played back using an `AudioSource`. 
+	The `AudioBuffer` class represents a buffer of audio data that can be played back using an `AudioSource`.
 	It supports a variety of audio formats and platforms, providing a consistent API for loading and managing audio data.
 
-	Depending on the platform, the audio backend may differ, but the class provides a unified interface for accessing 
+	Depending on the platform, the audio backend may differ, but the class provides a unified interface for accessing
 	audio data, whether it's stored in memory, loaded from a file, or streamed.
 
 	@see lime.media.AudioSource
@@ -58,6 +58,11 @@ class AudioBuffer
 	**/
 	public var data:UInt8Array;
 
+	/**
+		The format the raw audio data is stored in.
+	**/
+	public var dataFormat:AudioBufferDataFormat;
+
 	/**
 		The sample rate of the audio data, in Hz.
 	**/
@@ -144,6 +149,7 @@ class AudioBuffer
 			audioBuffer.bitsPerSample = data.bitsPerSample;
 			audioBuffer.channels = data.channels;
 			audioBuffer.data = new UInt8Array(@:privateAccess new Bytes(data.data.length, data.data.b));
+			audioBuffer.dataFormat = data.dataFormat;
 			audioBuffer.sampleRate = data.sampleRate;
 			return audioBuffer;
 		}
@@ -183,6 +189,7 @@ class AudioBuffer
 			audioBuffer.bitsPerSample = data.bitsPerSample;
 			audioBuffer.channels = data.channels;
 			audioBuffer.data = new UInt8Array(@:privateAccess new Bytes(data.data.length, data.data.b));
+			audioBuffer.dataFormat = data.dataFormat;
 			audioBuffer.sampleRate = data.sampleRate;
 			return audioBuffer;
 		}
@@ -238,6 +245,7 @@ class AudioBuffer
 			audioBuffer.bitsPerSample = data.bitsPerSample;
 			audioBuffer.channels = data.channels;
 			audioBuffer.data = new UInt8Array(@:privateAccess new Bytes(data.data.length, data.data.b));
+			audioBuffer.dataFormat = data.dataFormat;
 			audioBuffer.sampleRate = data.sampleRate;
 			return audioBuffer;
 		}
@@ -287,7 +295,7 @@ class AudioBuffer
 		@return An `AudioBuffer` instance with the decoded audio data.
 	**/
 	#if lime_vorbis
-		
+
 	public static function fromVorbisFile(vorbisFile:VorbisFile):AudioBuffer
 	{
 		if (vorbisFile == null) return null;
@@ -298,6 +306,7 @@ class AudioBuffer
 		audioBuffer.channels = info.channels;
 		audioBuffer.sampleRate = info.rate;
 		audioBuffer.bitsPerSample = 16;
+		audioBuffer.dataFormat = PCM;
 		audioBuffer.__srcVorbisFile = vorbisFile;
 
 		return audioBuffer;

src/lime/media/AudioBufferDataFormat.hx

@@ -0,0 +1,7 @@
+package lime.media;
+
+#if (haxe_ver >= 4.0) enum #else @:enum #end abstract AudioBufferDataFormat(Int) from Int to Int
+{
+	var PCM = 1;
+	var IEEEFloat = 3;
+}