static uint WriteHeader(FileStream SubSoundStream, FMOD.Sound SubSound, string FSBFileStr)
{
uint Milliseconds = 0;
uint RAWBytes = 0;
uint PCMBytes = 0;
uint PCMSamples = 0;
uint Length = 0;
SubSound.getLength(ref Milliseconds, TIMEUNIT.MS); // Get the length of the current wave file
SubSound.getLength(ref RAWBytes, TIMEUNIT.RAWBYTES); // Get the length of the current wave file
SubSound.getLength(ref PCMSamples, TIMEUNIT.PCM); // Get the length of the current wave file
SubSound.getLength(ref PCMBytes, TIMEUNIT.PCMBYTES); // Get the length of the current wave file
// Seek to the beginning of our newly created output file
SubSoundStream.Seek(0, SeekOrigin.Begin);
float Frequency = 0f; // Set some default values that aren't very
int DefPriority = 0; // important because they'll get overwritten
float DefPan = 0;
float DefVolume = 0;
FMOD.SOUND_TYPE FModSoundType = new SOUND_TYPE();
FMOD.SOUND_FORMAT FModSoundFormat = new SOUND_FORMAT();
int Channels = 0;
int BitsPerSample = 0;
// Get the default Frequency (important), and other stuff (not important)
SubSound.getDefaults(ref Frequency, ref DefVolume, ref DefPan, ref DefPriority);
// Get the actual sound format. We pretty much ignore this data since we know its going to be WAV, RIFF, PCM
SubSound.getFormat(ref FModSoundType, ref FModSoundFormat, ref Channels, ref BitsPerSample);
//Console.WriteLine(" MS={0}\tRAWBytes={1}\tPCMSamples={2}\tPCMBytes={3}", Milliseconds, RAWBytes, PCMSamples, PCMBytes);
//Console.WriteLine(" Freq={0}\tChan={1}", Frequency, Channels);
if (Switches.Contains("/doublefreq")) {
if (Channels == 1) { Frequency = Frequency * 2; }
}
int BlockAlign = Channels * (BitsPerSample / 8);
float DataRate = Channels * Frequency * (BitsPerSample / 8);
PCMSamples = Convert.ToUInt32(Milliseconds * Frequency / 1000);
// This is more of a sloppy hack. I don't know if I am just reading the file incorrectly, or using the wrong
// version of the API, or the FSB files are incorrectly structured, but the metadata that comes from the file
// seems to be incorrect. According to the WAV/RIFF specification and the FMOD API Documentation the
// following line should be the correct computation. If you use just the following line you get a random
// blip at the end of the sounds for the voice overs and the SFX don't play at all. A annoying
// trial-and-error process revealed that the logic below works properly. I wish there was a better way to
// identify which file we're parsing other than "SFX" in the name.
// "Correct" Computation: Length = Convert.ToUInt32(PCMSamples * Channels * (BitsPerSample / 8));
if (Path.GetFileName(FSBFileStr).Contains("VOBank")) {
// If we're parsing the VOBank FSB
if (Channels == 2) { // Works for the Character Speech FSBs
Length = Convert.ToUInt32(PCMSamples * Channels * (BitsPerSample / 8));
} else {
Length = PCMBytes * 2;
}
} else {
// If we're parsing a SFX FSB
if (Channels == 2) { // Works for the SFX FSBs
Length = PCMBytes;
} else {
//Length = Convert.ToUInt32(PCMSamples * Channels * (BitsPerSample / 8));
//Console.WriteLine(" CustomLen={0}\tPCMBytes={1}\tDiff={2}", Convert.ToUInt32(PCMSamples * Channels * (BitsPerSample / 8)), PCMBytes, (Convert.ToUInt32(PCMSamples * Channels * (BitsPerSample / 8)) - PCMBytes) * 1.0 / PCMBytes * 1.0);
if (Switches.Contains("/doublelen")) {
Length = PCMBytes * 2;
} else {
Length = PCMBytes;
}
}
}
// Some debugging outputs:
//Console.WriteLine(" MS={0}\tRAWBytes={1}\tPCMSamples={2}\tPCMBytes={3}", Milliseconds, RAWBytes, PCMSamples, PCMBytes);
//Console.WriteLine(" Freq={0}\tChan={1}\tBits/Sam={2}\tDataRate={3}", Frequency, Channels, BitsPerSample, DataRate);
// WAVE RIFF Format: http://www.topherlee.com/software/pcm-tut-wavformat.html
// Write the actual specification to the header of the file.
SubSoundStream.Write(BitConverter.GetBytes('R'), 0, 1); // 0 - Letter R
SubSoundStream.Write(BitConverter.GetBytes('I'), 0, 1); // 1 - Letter I
SubSoundStream.Write(BitConverter.GetBytes('F'), 0, 1); // 2 - Letter F
SubSoundStream.Write(BitConverter.GetBytes('F'), 0, 1); // 3 - Letter F
SubSoundStream.Write(BitConverter.GetBytes((long)0), 0, 4); // 4,5,6,7 - Length of entire file minus 8
SubSoundStream.Write(BitConverter.GetBytes('W'), 0, 1); // 8 - Letter W
SubSoundStream.Write(BitConverter.GetBytes('A'), 0, 1); // 9 - Letter A
SubSoundStream.Write(BitConverter.GetBytes('V'), 0, 1); // 10 - Letter V
SubSoundStream.Write(BitConverter.GetBytes('E'), 0, 1); // 11 - Letter E
SubSoundStream.Write(BitConverter.GetBytes('f'), 0, 1); // 12 - Letter f
SubSoundStream.Write(BitConverter.GetBytes('m'), 0, 1); // 13 - Letter m
SubSoundStream.Write(BitConverter.GetBytes('t'), 0, 1); // 14 - Letter t
SubSoundStream.Write(BitConverter.GetBytes(' '), 0, 1); // 15 - Space
SubSoundStream.Write(BitConverter.GetBytes((long)16), 0, 4); // 16,17,18,19 - Size of following subchunk
SubSoundStream.Write(BitConverter.GetBytes((int)1), 0, 2); // 20,21 - Format code (PCM)
SubSoundStream.Write(BitConverter.GetBytes((int)Channels), 0, 2); // 22,23 - Number of channels
SubSoundStream.Write(BitConverter.GetBytes((long)Frequency), 0, 4); // 24,25,26,27 - Sampling Rate (Blocks Per Second)
SubSoundStream.Write(BitConverter.GetBytes((int)DataRate), 0, 4); // 28,29,30,31 - Data rate
SubSoundStream.Write(BitConverter.GetBytes(BlockAlign), 0, 2); // 32,33 - Data block size (bytes)
SubSoundStream.Write(BitConverter.GetBytes(BitsPerSample), 0, 2); // 34,35 - Bits per sample
SubSoundStream.Write(BitConverter.GetBytes('d'), 0, 1); // 36 - Letter d
SubSoundStream.Write(BitConverter.GetBytes('a'), 0, 1); // 37 - Letter a
SubSoundStream.Write(BitConverter.GetBytes('t'), 0, 1); // 38 - Letter t
SubSoundStream.Write(BitConverter.GetBytes('a'), 0, 1); // 39 - Letter a
SubSoundStream.Write(BitConverter.GetBytes((long)Length), 0, 4); // 40,41,42,43 - Size of Data Section
return Length;
}