/// <summary>
///
/// </summary>
/// <param name="sound"></param>
public void LoadStreamOGG(SoundType sound)
{
}
/// <summary>
/// Loads a RIFF WAV
/// </summary>
/// <param name="sound">What soundtype is it</param>
public void LoadWAV(SoundType sound)
{
//AudioContext context = new AudioContext(); // needed for playing sounds!!!
/* OpenAL needs a binaryreadyer to get data from the sound-file.
*
*/
BinaryReader br = new BinaryReader(File.OpenRead(sound.Filename));
//RIFF_Header
string ChunkID;
int ChunkSize; // filesize
string Format;
//WAVE_Format
string subChunk1ID; // fmt
int subChunk1Size; // fmt Size
short audioFormat;
short numChannels;
int SampleRate;
int ByteRate;
short ByteAlign;
short BitsPerSample;
/*int mBytesPerPacket;
int mFramesPerPacket;
int mChannelsPerFrame;
int mBitsPerChannel;*/
//WAVE_Data
string subChunk2ID;
int subChunk2Size;
byte[] data;
// OpenAL Buffer
int ab = sound.Buffer;
/* WAVE
* RIFF-header is 4x4 Bytes
* Format is 2x4, 2x2, 2x4, 2x2 Bytes
* DATA have headers 2x4 Bytes with rest of data as sound info
*/
ChunkID = new string(br.ReadChars(4));
if (ChunkID != "RIFF") // WAV-file header 1
{
throw new Exception("input file not RIFF");
}
ChunkSize = br.ReadInt32(); //BitConverter.ToInt32(br.ReadBytes(4).Reverse().ToArray(), 0); // rest of caf file header, casts error!?
Format = new string(br.ReadChars(4));
if (Format != "WAVE") // WAV-file header 1
{
throw new Exception("input file not WAVE");
}
// HEADER end
// Format start
subChunk1ID = new string(br.ReadChars(4));
subChunk1Size = br.ReadInt32(); //BitConverter.ToInt32(br.ReadBytes(4).ToArray(), 0);
audioFormat = br.ReadInt16(); //BitConverter.ToInt16(br.ReadBytes(2).Reverse().ToArray(), 0);
numChannels = br.ReadInt16(); //BitConverter.ToInt16(br.ReadBytes(2).Reverse().ToArray(), 0);
SampleRate = br.ReadInt32(); //BitConverter.ToInt32(br.ReadBytes(4).Reverse().ToArray(), 0);
ByteRate = br.ReadInt32(); //BitConverter.ToInt32(br.ReadBytes(4).Reverse().ToArray(), 0);
ByteAlign = br.ReadInt16(); //BitConverter.ToInt16(br.ReadBytes(2).Reverse().ToArray(), 0);
BitsPerSample = br.ReadInt16(); //BitConverter.ToInt16(br.ReadBytes(2).Reverse().ToArray(), 0);
ALFormat alf = 0;
if (numChannels == 1) // mono
{
if (BitsPerSample == 8)
{
alf = ALFormat.Mono8;
}
else // say that it is 16 even if not, bad way...
{
alf = ALFormat.Mono16;
}
}
else if (numChannels == 2) // sterio
{
if (BitsPerSample == 8)
{
alf = ALFormat.Stereo8;
}
else // say that it is 16 even if not, bad way...
{
alf = ALFormat.Stereo16;
}
}
if (alf == 0)
{
throw new Exception("Wrong number of channels in sound file.");
}
// Format end
// DATA start
subChunk2ID = new string(br.ReadChars(4));
subChunk2Size = br.ReadInt32(); //BitConverter.ToInt32(br.ReadBytes(4).Reverse().ToArray(), 0);
data = new byte[br.BaseStream.Length];
data = br.ReadBytes((int)br.BaseStream.Length);
// DATA end
//IntPtr dataPointer = System.Runtime.InteropServices.Marshal.AllocHGlobal(data.Length);//new IntPtr();
//System.Runtime.InteropServices.Marshal.Copy(data,0,dataPointer, data.Length);
//GCHandle pinnedArray = GCHandle.Alloc(byteArray, GCHandleType.Pinned);
//IntPtr pointer = pinnedArray.AddrOfPinnedObject();
//IntPtr dataPointer = System.Runtime.InteropServices.Marshal.UnsafeAddrOfPinnedArrayElement(data, 0);
AL.BufferData(ab, alf, data, data.Length, SampleRate);
br.Close();
//Add this if you don't want to have large memory allocation, as the allocation sticks to the program until gc can free it
/*data = new byte[1]; // force release of data
GC.Collect(); // well well well, bleh! this releases the memory only if it is top on heap else nothing happens...
*/
//return ab;
}
public void LoadOGG(SoundType sound)
{
// Dirty way of checking that we have ogg-dlls in exe path....
if (!File.Exists("csogg.dll") || !File.Exists("csvorbis.dll"))
{
return; // return 0;
}
int ab = sound.Buffer;
//#define ogg // or activate this, but this is just for this place/file... :D
#if noogg
#warning "Returning 0 on Ogg file, you need to define \"ogg\" in build constants, in project build property."
return ; // quicker upstart
#endif
VorbisFile asd = new VorbisFile(sound.Filename);
System.Diagnostics.Debug.WriteLine(sound.Filename);
Info info = asd.getInfo(-1);
//long samples = asd.pcm_total(-1);
//int streams = asd.streams();
//byte[] data3 = new byte[samples*streams];
byte[] data2 = new byte[4096];
Stream output2 = new MemoryStream();
int readBytes = 0;
while ((readBytes = asd.read(data2, data2.Length, 0 /*Bigendian*/, /*(info.rate < 44100 ? 2 : 2)*/ 2 /*1=byte, 2=16bit*/, 1/*signd*/, null)) > 0)
{
output2.Write(data2, 0, readBytes);
}
data2 = null;
data2 = new byte[output2.Length];
output2.Seek(0, SeekOrigin.Begin);
output2.Read(data2, 0, data2.Length);
output2.Close();
//output2.Dispose();
output2 = null;
//
#region csogg and stuff, commented
// borrow from csogg and csvorbis...
/*using (var input = new FileStream(filename, FileMode.Open, FileAccess.Read))
{
System.Diagnostics.Debug.WriteLine(filename);
bool skipWavHeader = true;
int HEADER_SIZE = 36;
int convsize = 4096 * 2;
byte[] convbuffer = new byte[convsize]; // take 8k out of the data segment, not the stack
Stream output = new MemoryStream();
if (!skipWavHeader)
output.Seek(HEADER_SIZE, SeekOrigin.Begin); // reserve place for WAV header
SyncState oy = new SyncState(); // sync and verify incoming physical bitstream
StreamState os = new StreamState(); // take physical pages, weld into a logical stream of packets
Page og = new Page(); // one Ogg bitstream page. Vorbis packets are inside
Packet op = new Packet(); // one raw packet of data for decode
Info vi = new Info(); // struct that stores all the static vorbis bitstream settings
Comment vc = new Comment(); // struct that stores all the bitstream user comments
DspState vd = new DspState(); // central working state for the packet->PCM decoder
Block vb = new Block(vd); // local working space for packet->PCM decode
byte[] buffer;
int bytes = 0;
// Decode setup
oy.init(); // Now we can read pages
while (true)
{ // we repeat if the bitstream is chained
int eos = 0;
// grab some data at the head of the stream. We want the first page
// (which is guaranteed to be small and only contain the Vorbis
// stream initial header) We need the first page to get the stream
// serialno.
// submit a 4k block to libvorbis' Ogg layer
int index = oy.buffer(4096);
buffer = oy.data;
try
{
bytes = input.Read(buffer, index, 4096);
}
catch (Exception e)
{
System.Diagnostics.Debug.WriteLine(e);
}
oy.wrote(bytes);
// Get the first page.
if (oy.pageout(og) != 1)
{
// have we simply run out of data? If so, we're done.
if (bytes < 4096) break;
// error case. Must not be Vorbis data
System.Diagnostics.Debug.WriteLine("Input does not appear to be an Ogg bitstream.");
}
// Get the serial number and set up the rest of decode.
// serialno first; use it to set up a logical stream
os.init(og.serialno());
// extract the initial header from the first page and verify that the
// Ogg bitstream is in fact Vorbis data
// I handle the initial header first instead of just having the code
// read all three Vorbis headers at once because reading the initial
// header is an easy way to identify a Vorbis bitstream and it's
// useful to see that functionality seperated out.
vi.init();
vc.init();
if (os.pagein(og) < 0)
{
// error; stream version mismatch perhaps
System.Diagnostics.Debug.WriteLine("Error reading first page of Ogg bitstream data.");
}
if (os.packetout(op) != 1)
{
// no page? must not be vorbis
System.Diagnostics.Debug.WriteLine("Error reading initial header packet.");
}
if (vi.synthesis_headerin(vc, op) < 0)
{
// error case; not a vorbis header
System.Diagnostics.Debug.WriteLine("This Ogg bitstream does not contain Vorbis audio data.");
}
// At this point, we're sure we're Vorbis. We've set up the logical
// (Ogg) bitstream decoder. Get the comment and codebook headers and
// set up the Vorbis decoder
// The next two packets in order are the comment and codebook headers.
// They're likely large and may span multiple pages. Thus we reead
// and submit data until we get our two pacakets, watching that no
// pages are missing. If a page is missing, error out; losing a
// header page is the only place where missing data is fatal.
int i = 0;
while (i < 2)
{
while (i < 2)
{
int result = oy.pageout(og);
if (result == 0) break; // Need more data
// Don't complain about missing or corrupt data yet. We'll
// catch it at the packet output phase
if (result == 1)
{
os.pagein(og); // we can ignore any errors here
// as they'll also become apparent
// at packetout
while (i < 2)
{
result = os.packetout(op);
if (result == 0) break;
if (result == -1)
{
// Uh oh; data at some point was corrupted or missing!
// We can't tolerate that in a header. Die.
System.Diagnostics.Debug.WriteLine("Corrupt secondary header. Exiting.");
}
vi.synthesis_headerin(vc, op);
i++;
}
}
}
// no harm in not checking before adding more
index = oy.buffer(4096);
buffer = oy.data;
try
{
bytes = input.Read(buffer, index, 4096);
}
catch (Exception e)
{
System.Diagnostics.Debug.WriteLine(e);
}
if (bytes == 0 && i < 2)
{
System.Diagnostics.Debug.WriteLine("End of file before finding all Vorbis headers!");
}
oy.wrote(bytes);
}
// Throw the comments plus a few lines about the bitstream we're decoding
{
byte[][] ptr = vc.user_comments;
for (int j = 0; j < vc.user_comments.Length; j++)
{
if (ptr[j] == null) break;
System.Diagnostics.Debug.WriteLine(vc.getComment(j));
}
System.Diagnostics.Debug.WriteLine("\nBitstream is " + vi.channels + " channel, " + vi.rate + "Hz");
System.Diagnostics.Debug.WriteLine("Encoded by: " + vc.getVendor() + "\n");
} // comment this on release...
convsize = 4096 / vi.channels;
// OK, got and parsed all three headers. Initialize the Vorbis
// packet->PCM decoder.
vd.synthesis_init(vi); // central decode state
vb.init(vd); // local state for most of the decode
// so multiple block decodes can
// proceed in parallel. We could init
// multiple vorbis_block structures
// for vd here
float[][][] _pcm = new float[1][][];
int[] _index = new int[vi.channels];
// The rest is just a straight decode loop until end of stream
while (eos == 0)
{
while (eos == 0)
{
int result = oy.pageout(og);
if (result == 0) break; // need more data
if (result == -1)
{ // missing or corrupt data at this page position
System.Diagnostics.Debug.WriteLine("Corrupt or missing data in bitstream; continuing...");
}
else
{
os.pagein(og); // can safely ignore errors at
// this point
while (true)
{
result = os.packetout(op);
if (result == 0) break; // need more data
if (result == -1)
{ // missing or corrupt data at this page position
// no reason to complain; already complained above
}
else
{
// we have a packet. Decode it
int samples;
if (vb.synthesis(op) == 0)
{ // test for success!
vd.synthesis_blockin(vb);
}
// **pcm is a multichannel float vector. In stereo, for
// example, pcm[0] is left, and pcm[1] is right. samples is
// the size of each channel. Convert the float values
// (-1.<=range<=1.) to whatever PCM format and write it out
while ((samples = vd.synthesis_pcmout(_pcm, _index)) > 0)
{
float[][] pcm = _pcm[0];
bool clipflag = false;
int bout = (samples < convsize ? samples : convsize);
// convert floats to 16 bit signed ints (host order) and
// interleave
for (i = 0; i < vi.channels; i++)
{
int ptr = i * 2;
//int ptr=i;
int mono = _index[i];
for (int j = 0; j < bout; j++)
{
int val = (int)(pcm[i][mono + j] * 32767.0);
// short val=(short)(pcm[i][mono+j]*32767.);
// int val=(int)Math.round(pcm[i][mono+j]*32767.);
// might as well guard against clipping
if (val > 32767)
{
val = 32767;
clipflag = true;
}
if (val < -32768)
{
val = -32768;
clipflag = true;
}
if (val < 0) val = val | 0x8000;
convbuffer[ptr] = (byte)(val);
convbuffer[ptr + 1] = (byte)((uint)val >> 8);
ptr += 2 * (vi.channels);
}
}
if (clipflag)
{
//s_err.WriteLine("Clipping in frame "+vd.sequence);
}
output.Write(convbuffer, 0, 2 * vi.channels * bout);
vd.synthesis_read(bout); // tell libvorbis how
// many samples we
// actually consumed
}
}
}
if (og.eos() != 0) eos = 1;
}
}
if (eos == 0)
{
index = oy.buffer(4096);
buffer = oy.data;
try
{
bytes = input.Read(buffer, index, 4096);
}
catch (Exception e)
{
System.Diagnostics.Debug.WriteLine(e);
}
oy.wrote(bytes);
if (bytes == 0) eos = 1;
}
}
// clean up this logical bitstream; before exit we see if we're
// followed by another [chained]
os.clear();
// ogg_page and ogg_packet structs always point to storage in
// libvorbis. They're never freed or manipulated directly
vb.clear();
vd.clear();
vi.clear(); // must be called last
}
// OK, clean up the framer
oy.clear();
System.Diagnostics.Debug.WriteLine("Done.");
output.Seek(0, SeekOrigin.Begin);
if (!skipWavHeader)
{
WriteHeader(output, (int)(output.Length - HEADER_SIZE), vi.rate, (ushort)16, (ushort)vi.channels);
output.Seek(0, SeekOrigin.Begin);
}
*/
#endregion
//
ALFormat alf = 0;
if (/*vi*/info.channels == 1) // mono
{
alf = ALFormat.Mono16;
}
else if (/*vi*/info.channels == 2) // sterio
{
alf = ALFormat.Stereo16;
}
else if (alf == 0)
{
throw new Exception("Wrong number of channels in sound file.");
}
/*BinaryReader bw = new BinaryReader(output);
byte[] data = bw.ReadBytes((int)output.Length);
bw.Close();
bw.Dispose();*/
/*byte[] data = new byte[(int)output.Length];
output.Read(data, 0, data.Length);
output.Close();
output.Dispose();*/
AL.BufferData(ab, alf, data2, data2.Length, /*vi.rate*/ info.rate);
//data = null;
//}
//return ab;
}