Added a c version cos why not lol

Source/c version/main.c

@@ -0,0 +1,16 @@
+#include "wavLoader.h"
+
+#include <ncurses.h>
+
+int main(int argc, char* argv[])
+{
+    struct wavFile * myCoolWav;
+    if(argc > 1){
+        myCoolWav = loadWav(argv[1]);
+    }else{
+        return -1;
+    }
+
+
+return 0;
+}

Source/c version/wavLoader.c

@@ -0,0 +1,78 @@
+#include "wavLoader.h"
+
+unsigned long pullFromBuffer(unsigned char * buffer, int length, int startAddress, int dataType){
+	unsigned long byteSet;
+	if (dataType){ //little endian
+		byteSet = buffer[startAddress+(length-1)];
+		for (int i = length-2; i >= 0; i--){ // -2 offset as need -1 cos array start at 0 and -1 as we already done one above
+			//std::cout << std::hex << byteSet << " " << buffer[startAddress+i] << std::endl;
+			byteSet = byteSet << 8 | buffer[startAddress+i];
+		}
+	}else{
+		byteSet = buffer[startAddress];
+		for (int i = 1; i < length; i++){
+			byteSet = byteSet << 8 | buffer[startAddress+i];
+		}
+	}
+	return byteSet;
+}
+
+struct wavFile * loadWav(char * filePath){
+	//open file and get size
+	FILE *fptr;
+	unsigned long fSize;
+	fptr = fopen(filePath,"rb");
+	if (fptr == NULL){
+		fprintf(stderr, "%s", "File could not be opened!\n");
+		return NULL;
+	}
+	fseek ( fptr , 0 , SEEK_END );
+	fSize = ftell(fptr);
+	rewind(fptr);
+
+	//create file buffer
+	unsigned char * buffer = (unsigned char*) malloc (sizeof(char)*fSize);
+	if (buffer == NULL){
+		fprintf(stderr, "%s", "Memory allocation error will attempt to read from disk!\n");
+		//TODO: disk streaming
+	}
+
+	//populate buffer in 1 byte increments
+	if (fread(buffer,1,fSize,fptr) != fSize){
+		fprintf(stderr, "%s", "File could not be read!\n");
+		return NULL;
+	}else{
+		fclose(fptr);
+	}
+
+	if (!( (pullFromBuffer(buffer, 4, 8, BIG_ENDIAN_TAG) == 0x57415645) && (pullFromBuffer(buffer, 2, 20, LITTLE_ENDIAN_TAG) == 1) )){ //byte 8 is format, byte 20 is audioFormat
+		fprintf(stderr, "%s", "File is not standard wav!\n");
+		return NULL;
+	}
+
+	struct wavFile *wf;
+	//file info
+	wf->Channels = pullFromBuffer(buffer, 2, 22, LITTLE_ENDIAN_TAG);//2 bytes at 20
+	wf->SampleRate = pullFromBuffer(buffer, 2, 24, LITTLE_ENDIAN_TAG);//4 bytes at 24
+	wf->BlockAlign = pullFromBuffer(buffer, 2, 32, LITTLE_ENDIAN_TAG);//2 bytes at 32
+	wf->BitsPerSample = pullFromBuffer(buffer, 2, 34, LITTLE_ENDIAN_TAG);//2 bytes at 34
+
+	//file data
+	unsigned long audioDataSize = pullFromBuffer(buffer, 4, 40, LITTLE_ENDIAN_TAG);
+	wf->AudioData = (unsigned char*) malloc (sizeof(char)*audioDataSize); //4 bytes at 40
+
+	//move the data array
+	memmove(wf->AudioData, buffer+44, audioDataSize);
+	free(buffer);
+
+	printf ("Channels: %d\n", wf->Channels);
+	printf ("Sample Rate: %d\n", wf->SampleRate);
+	printf ("Block Align: %d\n", wf->BlockAlign);
+	printf ("Bits Per Sample: %d\n", wf->BitsPerSample);
+	printf ("Audio Data Size: %lu\n\n", audioDataSize);
+
+	printf ("Audio Data: %x %x %x %x\n", wf->AudioData[0], wf->AudioData[1], wf->AudioData[2], wf->AudioData[3]);
+
+
+	return NULL;
+}

Source/c version/wavLoader.h

@@ -0,0 +1,62 @@
+#ifndef WAVLOADER_H
+#define WAVLOADER_H
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#define BIG_ENDIAN_TAG 0
+#define LITTLE_ENDIAN_TAG 1
+
+/*
+wav format
+wav is actually a wrapper for another format which is the riff format
+riff files store everything in juicy lil chunks
+
+every riff chunk has this layout:
+4 byte ChunkID
+4 byte ChunkSize (this excludes the size of chunk id and chunk size)
+the rest of the chunks info
+
+How are Wave files stored:
+Chunk 1 ("Riff Chunk Descriptor"):
+ Endian	|Pos|  Size  |Data Stored		| Descriptions
+
+big	 	| 0 |4 bytes |Chunk ID			| RIFF (0x52494646)
+little 	| 4 |4 bytes |Chunk Size		| 4 + (8 + Chunk2Size) + (8 + Chunk3Size) (size of full file excluding the first 8 bytes)
+big	 	| 8 |4 bytes |Format			| WAVE (0x57415645)
+
+Chunk 2 ("Format chunk"):
+big	 	| 12 |4 bytes |Chunk ID			| fmt (0x666D7420)
+little	| 16 |4 bytes |Chunk Size		| 16 (0x10000000)
+little	| 20 |2 bytes |AudioFormat		| If this is equal to 1 its PCM audio otherwise its got some compression or something
+little	| 22 |2 bytes |NumChannels		| number of channels
+little	| 24 |4 bytes |Sample Rate		| sample rate like 8000 or 44100
+little	| 28 |4 bytes |Byte Rate		| SampleRate * NumChannels * BitsPerSample/8
+little	| 32 |2 bytes |Block Align		| NumChannels * BitsPerSample/8 (num of bytes per one audio sample)
+little	| 34 |2 bytes |Bits Per Sample 	| number of bits per sample
+
+Chunk 2 ("Data Chunk"):
+big	 	| 36 |4 bytes |Chunk ID			| data (0x64617461)
+little	| 40 |4 bytes |Chunk Size		| wav data size
+little	| 44 |X bytes |WAV Data 		| The actual sound data
+
+how is the wav data stored:
+each sample has (Block Align)Bytes, the left channel is the first half of the sample and the right channel is the second half.
+*/
+
+//struct sample??
+
+struct wavFile{
+	int Channels;
+	int SampleRate;
+	int BlockAlign;
+	int BitsPerSample;
+	unsigned char * AudioData;      
+};
+
+struct wavFile * loadWav(char * filePath);
+
+unsigned long pullFromBuffer(unsigned char * buffer, int length, int startAddress, int dataType);
+
+#endif /* WAVLOADER_H */

Source/wavLoader.cpp

@@ -1,9 +1,26 @@
 #include "wavLoader.h"
 
+unsigned long pullFromBuffer(unsigned char * buffer, int length, int startAddress, int dataType){
+	unsigned long byteSet;
+	if (dataType){ //little endian
+		byteSet = buffer[startAddress+(length-1)];
+		for (int i = length-2; i >= 0; i--){ // -2 offset as need -1 cos array start at 0 and -1 as we already done one above
+			//std::cout << std::hex << byteSet << " " << buffer[startAddress+i] << std::endl;
+			byteSet = byteSet << 8 | buffer[startAddress+i];
+		}
+	}else{
+		byteSet = buffer[startAddress];
+		for (int i = 1; i < length; i++){
+			byteSet = byteSet << 8 | buffer[startAddress+i];
+		}
+	}
+	return byteSet;
+}
+
 wavFile * loadWav(char * filePath){
 	//open file and get size
 	FILE *fptr;
-	long fSize;
+	unsigned long fSize;
 	fptr = fopen(filePath,"rb");
 	if (fptr == NULL){
 		fprintf(stderr, "%s", "File could not be opened!\n");
@@ -14,37 +31,48 @@ wavFile * loadWav(char * filePath){
 	rewind(fptr);
 
 	//create file buffer
-	char * buffer = (char*) malloc (sizeof(char)*fSize);
+	unsigned char * buffer = (unsigned char*) malloc (sizeof(char)*fSize);
 	if (buffer == NULL){
 		fprintf(stderr, "%s", "Memory allocation error will attempt to read from disk!\n");
+		//TODO: disk streaming
 	}
 
 	//populate buffer in 1 byte increments
-	if (fread(buffer,1,fSize,fptr) !=fSize){
+	if (fread(buffer,1,fSize,fptr) != fSize){
 		fprintf(stderr, "%s", "File could not be read!\n");
 		return NULL;
+	}else{
+		fclose(fptr);
 	}
 
-
-	unsigned int fmt;
-	memcpy(&fmt, buffer, sizeof(unsigned int)); //this boy gonna be backwards cos lil endian
-	std::cout << std::hex << fmt << std::endl;
-	std::cout << std::hex << (int)buffer[8] << std::endl;
-
-	/*if (!( (fmt == WAVE) && (audioFMT == 1) )){ //byte 8 is format, byte 18 is audioFormat
+	if (!( (pullFromBuffer(buffer, 4, 8, BIG_ENDIAN_TAG) == 0x57415645) && (pullFromBuffer(buffer, 2, 20, LITTLE_ENDIAN_TAG) == 1) )){ //byte 8 is format, byte 20 is audioFormat
 		fprintf(stderr, "%s", "File is not standard wav!\n");
-		return null;
+		return NULL;
 	}
 
+	wavFile *wf;
 	//file info
-
-	wf.Channels = ;//2 bytes at 20
-	wf.SampleRate = ;//4 bytes at 24
-	wf.BitsPerSample = ;//2 bytes at 32
+	wf->Channels = pullFromBuffer(buffer, 2, 22, LITTLE_ENDIAN_TAG);//2 bytes at 20
+	wf->SampleRate = pullFromBuffer(buffer, 2, 24, LITTLE_ENDIAN_TAG);//4 bytes at 24
+	wf->BlockAlign = pullFromBuffer(buffer, 2, 32, LITTLE_ENDIAN_TAG);//2 bytes at 32
+	wf->BitsPerSample = pullFromBuffer(buffer, 2, 34, LITTLE_ENDIAN_TAG);//2 bytes at 34
 
 	//file data
-	wf.AudioData = ;//memcpy from address 44 for address 40 bytes
-	*/
-	fclose(fptr);
+	unsigned long audioDataSize = pullFromBuffer(buffer, 4, 40, LITTLE_ENDIAN_TAG);
+	wf->AudioData = (unsigned char*) malloc (sizeof(char)*audioDataSize); //4 bytes at 40
+
+	//move the data array
+	memmove(wf->AudioData, buffer+44, audioDataSize);
+	free(buffer);
+
+	printf ("Channels: %d\n", wf->Channels);
+	printf ("Sample Rate: %d\n", wf->SampleRate);
+	printf ("Block Align: %d\n", wf->BlockAlign);
+	printf ("Bits Per Sample: %d\n", wf->BitsPerSample);
+	printf ("Audio Data Size: %lu\n\n", audioDataSize);
+
+	printf ("Audio Data: %x %x %x %x\n", wf->AudioData[0], wf->AudioData[1], wf->AudioData[2], wf->AudioData[3]);
+
+
 	return NULL;
 }

Source/wavLoader.h

@@ -1,8 +1,14 @@
+#ifndef WAVLOADER_H
+#define WAVLOADER_H
+
 #include <iostream>
 #include <stdio.h>
 #include <stdlib.h>
 #include <string>
 
+#define BIG_ENDIAN_TAG 0
+#define LITTLE_ENDIAN_TAG 1
+
 /*
 wav format
 wav is actually a wrapper for another format which is the riff format
@@ -15,37 +21,46 @@ the rest of the chunks info
 
 How are Wave files stored:
 Chunk 1 ("Riff Chunk Descriptor"):
-0 |4 bytes |Chunk ID		| RIFF (0x52494646)
-4 |4 bytes |Chunk Size		| 4 + (8 + Chunk2Size) + (8 + Chunk3Size) (size of full file excluding the first 8 bytes)
-8 |4 bytes |Format			| WAVE (0x57415645)
+ Endian	|Pos|  Size  |Data Stored		| Descriptions
+
+big	 	| 0 |4 bytes |Chunk ID			| RIFF (0x52494646)
+little 	| 4 |4 bytes |Chunk Size		| 4 + (8 + Chunk2Size) + (8 + Chunk3Size) (size of full file excluding the first 8 bytes)
+big	 	| 8 |4 bytes |Format			| WAVE (0x57415645)
 
 Chunk 2 ("Format chunk"):
-12|4 bytes |Chunk ID		| fmt (0x666D7420)
-16|4 bytes |Chunk Size		| 16 (0x10000000)
-18|2 bytes |AudioFormat		| If this is equal to 1 its PCM audio otherwise its got some compression or something
-20|2 bytes |NumChannels		| number of channels
-24|4 bytes |Sample Rate		| sample rate like 8000 or 44100
-28|4 bytes |Byte Rate		| SampleRate * NumChannels * BitsPerSample/8
-32|2 bytes |Block Align		| NumChannels * BitsPerSample/8 (num of bytes per one audio sample)
-34|2 bytes |Bits Per Sample | number of bits per sample
+big	 	| 12 |4 bytes |Chunk ID			| fmt (0x666D7420)
+little	| 16 |4 bytes |Chunk Size		| 16 (0x10000000)
+little	| 20 |2 bytes |AudioFormat		| If this is equal to 1 its PCM audio otherwise its got some compression or something
+little	| 22 |2 bytes |NumChannels		| number of channels
+little	| 24 |4 bytes |Sample Rate		| sample rate like 8000 or 44100
+little	| 28 |4 bytes |Byte Rate		| SampleRate * NumChannels * BitsPerSample/8
+little	| 32 |2 bytes |Block Align		| NumChannels * BitsPerSample/8 (num of bytes per one audio sample)
+little	| 34 |2 bytes |Bits Per Sample 	| number of bits per sample
 
 Chunk 2 ("Data Chunk"):
-36|4 bytes |Chunk ID		| data (0x64617461)
-40|4 bytes |Chunk Size		| wav data size
-44|X bytes |WAV Data 		| The actual sound data
+big	 	| 36 |4 bytes |Chunk ID			| data (0x64617461)
+little	| 40 |4 bytes |Chunk Size		| wav data size
+little	| 44 |X bytes |WAV Data 		| The actual sound data
 
 how is the wav data stored:
 each sample has (Block Align)Bytes, the left channel is the first half of the sample and the right channel is the second half.
 */
 
-class wavFile{
+//struct sample??
+
+struct wavFile{
     public:
     	int Channels;
         int SampleRate;
+        int BlockAlign;
         int BitsPerSample;
-        int * AudioData;
+        unsigned char * AudioData;
     private:
 
 };
 
 wavFile * loadWav(char * filePath);
+
+unsigned long pullFromBuffer(unsigned char * buffer, int length, int startAddress, int dataType);
+
+#endif /* WAVLOADER_H */