tahoma2d/toonz/sources/common/flash/FSound.h
Toshihiro Shimizu 890dddabbd first commit
2016-03-19 02:57:51 +09:00

146 lines
3.7 KiB
C++

#ifndef SNDMIX_INCLUDED
#define SNDMIX_INCLUDED
// #include <windows.h>
// #include <mmreg.h>
#include <assert.h>
#include <stdio.h>
#include <vector>
#include "Macromedia.h"
#include "FFixed.h"
struct WaveFormat {
U16 wFormatTag; /* format type */
U16 nChannels; /* number of channels (i.e. mono, stereo...) */
U32 nSamplesPerSec; /* sample rate */
U32 nAvgBytesPerSec; /* for buffer estimation */
U16 nBlockAlign; /* block size of data */
U16 wBitsPerSample; /* number of bits per sample of mono data */
U16 cbSize; /* the count in bytes of the size of */
};
// Our supported sample rates
// Note that sndRate5K is really 5512.5khz
// Use defines instead of an enum since these must be 32 bits on all platforms
#define sndRate5K 5512L
#define sndRate11K 11025L
#define sndRate22K 22050L
#define sndRate44K 44100L
const int sndMono = 0x0;
const int sndStereo = 0x1;
const int snd8Bit = 0x0;
const int snd16Bit = 0x2;
const int snd5K = 0 << 2;
const int snd11K = 1 << 2;
const int snd22K = 2 << 2;
const int snd44K = 3 << 2;
const int sndCompressNone = 0x00; // we could add 14 more compression types here...
const int sndCompressADPCM = 0x10;
const int sndCompressMP3 = 0x20;
const int sndCompressNoneI = 0x30; // save out in intel byte order
const int sndRateMask = 0x3 << 2;
const int sndCompressMask = 0xF0;
enum { // Sound format types
snd5K8Mono = 0,
snd5K8Stereo,
snd5K16Mono,
snd5K16Stereo,
snd11K8Mono,
snd11K8Stereo,
snd11K16Mono,
snd11K16Stereo,
snd22K8Mono,
snd22K8Stereo,
snd22K16Mono,
snd22K16Stereo,
snd44K8Mono,
snd44K8Stereo,
snd44K16Mono,
snd44K16Stereo
};
// This object defines a sound sample
class FSound
{
public:
int format;
S32 nSamples; // the number of samples - duration = nSamples/Rate()
void *samples; // this should probably be a handle on Mac
S32 dataLen; // length in bytes of samples - set only if needed
S32 delay; // MP3 compression has a delay before the real sound data
static const S32 kRateTable[4];
static const int kRateShiftTable[4];
public:
void Init();
S32 Rate() { return kRateTable[(format >> 2) & 0x3]; }
int RateShift() { return kRateShiftTable[(format >> 2) & 0x3]; }
bool Stereo() { return (format & sndStereo) != 0; }
int NChannels() { return (format & sndStereo) ? 2 : 1; }
bool Is8Bit() { return (format & snd16Bit) == 0; }
int BitsPerSample() { return (format & snd16Bit) ? 16 : 8; }
int BytesPerSample() { return (format & snd16Bit) ? 2 : 1; }
int CompressFormat() { return format & sndCompressMask; }
bool Compressed() { return (format & sndCompressMask) != 0; }
// Manage the duration in 44kHz units
S32 GetDuration44() { return nSamples << RateShift(); }
void SetDuration44(S32 d) { nSamples = d >> RateShift(); }
int BytesPerBlock() { return BytesPerSample() * NChannels(); }
S32 SizeBytes() { return nSamples * BytesPerBlock(); }
void Set(WaveFormat *);
};
class FSoundComp
{
private:
BOOL isStereo;
BOOL is8Bit;
int nBits; // number of bits in each sample
S32 nSamples; // samples compressed so far
S32 valpred[2]; // ADPCM state
int index[2];
// The Destination
S32 len; // default is to just calculate the size
// State for writing bits
unsigned int bitBuf;
int bitPos;
public:
FSoundComp(FSound *snd, S32 numberBits); // numberBits from 2-5
~FSoundComp() { assert(bitPos == 0); }
void Compress(void *src, S32 n, std::vector<U8> *stream);
void Flush(std::vector<U8> *stream);
private:
void Compress16(S16 *src, S32 n, std::vector<U8> *stream);
// Write variable width bit fields
void WriteBits(std::vector<U8> *stream); // empty the buffer of whole bytes
void PutBits(S32 v, int n, std::vector<U8> *stream)
{
assert(n <= 16);
if (bitPos + n > 32)
WriteBits(stream);
bitBuf = (bitBuf << n) | (v & ~(0xFFFFFFFFL << n));
bitPos += n;
}
};
#endif