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-rw-r--r--sound/decoders/qdm2.cpp3327
-rw-r--r--sound/decoders/qdm2.h51
-rw-r--r--sound/decoders/qdm2data.h531
-rw-r--r--sound/module.mk1
4 files changed, 0 insertions, 3910 deletions
diff --git a/sound/decoders/qdm2.cpp b/sound/decoders/qdm2.cpp
deleted file mode 100644
index aa4eb4b40a..0000000000
--- a/sound/decoders/qdm2.cpp
+++ /dev/null
@@ -1,3327 +0,0 @@
-/* ScummVM - Graphic Adventure Engine
- *
- * ScummVM is the legal property of its developers, whose names
- * are too numerous to list here. Please refer to the COPYRIGHT
- * file distributed with this source distribution.
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
-
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
-
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
- *
- * $URL$
- * $Id$
- *
- */
-
-// Based off ffmpeg's QDM2 decoder
-
-#include "sound/decoders/qdm2.h"
-
-#ifdef SOUND_QDM2_H
-
-#include "sound/audiostream.h"
-#include "sound/decoders/qdm2data.h"
-
-#include "common/array.h"
-#include "common/stream.h"
-#include "common/system.h"
-
-namespace Audio {
-
-enum {
- SOFTCLIP_THRESHOLD = 27600,
- HARDCLIP_THRESHOLD = 35716,
- MPA_MAX_CHANNELS = 2,
- MPA_FRAME_SIZE = 1152,
- FF_INPUT_BUFFER_PADDING_SIZE = 8
-};
-
-typedef int8 sb_int8_array[2][30][64];
-
-/* bit input */
-/* buffer, buffer_end and size_in_bits must be present and used by every reader */
-struct GetBitContext {
- const uint8 *buffer, *bufferEnd;
- int index;
- int sizeInBits;
-};
-
-struct QDM2SubPacket {
- int type;
- unsigned int size;
- const uint8 *data; // pointer to subpacket data (points to input data buffer, it's not a private copy)
-};
-
-struct QDM2SubPNode {
- QDM2SubPacket *packet;
- struct QDM2SubPNode *next; // pointer to next packet in the list, NULL if leaf node
-};
-
-struct QDM2Complex {
- float re;
- float im;
-};
-
-struct FFTTone {
- float level;
- QDM2Complex *complex;
- const float *table;
- int phase;
- int phase_shift;
- int duration;
- short time_index;
- short cutoff;
-};
-
-struct FFTCoefficient {
- int16 sub_packet;
- uint8 channel;
- int16 offset;
- int16 exp;
- uint8 phase;
-};
-
-struct VLC {
- int32 bits;
- int16 (*table)[2]; // code, bits
- int32 table_size;
- int32 table_allocated;
-};
-
-#include "common/pack-start.h"
-struct QDM2FFT {
- QDM2Complex complex[MPA_MAX_CHANNELS][256];
-} PACKED_STRUCT;
-#include "common/pack-end.h"
-
-enum RDFTransformType {
- RDFT,
- IRDFT,
- RIDFT,
- IRIDFT
-};
-
-struct FFTComplex {
- float re, im;
-};
-
-struct FFTContext {
- int nbits;
- int inverse;
- uint16 *revtab;
- FFTComplex *exptab;
- FFTComplex *tmpBuf;
- int mdctSize; // size of MDCT (i.e. number of input data * 2)
- int mdctBits; // n = 2^nbits
- // pre/post rotation tables
- float *tcos;
- float *tsin;
- void (*fftPermute)(struct FFTContext *s, FFTComplex *z);
- void (*fftCalc)(struct FFTContext *s, FFTComplex *z);
- void (*imdctCalc)(struct FFTContext *s, float *output, const float *input);
- void (*imdctHalf)(struct FFTContext *s, float *output, const float *input);
- void (*mdctCalc)(struct FFTContext *s, float *output, const float *input);
- int splitRadix;
- int permutation;
-};
-
-enum {
- FF_MDCT_PERM_NONE = 0,
- FF_MDCT_PERM_INTERLEAVE = 1
-};
-
-struct RDFTContext {
- int nbits;
- int inverse;
- int signConvention;
-
- // pre/post rotation tables
- float *tcos;
- float *tsin;
- FFTContext fft;
-};
-
-class QDM2Stream : public Audio::AudioStream {
-public:
- QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
- ~QDM2Stream();
-
- bool isStereo() const { return _channels == 2; }
- bool endOfData() const { return ((_stream->pos() == _stream->size()) && (_outputSamples.size() == 0)); }
- int getRate() const { return _sampleRate; }
- int readBuffer(int16 *buffer, const int numSamples);
-
-private:
- Common::SeekableReadStream *_stream;
-
- // Parameters from codec header, do not change during playback
- uint8 _channels;
- uint16 _sampleRate;
- uint16 _bitRate;
- uint16 _blockSize; // Group
- uint16 _frameSize; // FFT
- uint16 _packetSize; // Checksum
-
- // Parameters built from header parameters, do not change during playback
- int _groupOrder; // order of frame group
- int _fftOrder; // order of FFT (actually fft order+1)
- int _fftFrameSize; // size of fft frame, in components (1 comples = re + im)
- int _sFrameSize; // size of data frame
- int _frequencyRange;
- int _subSampling; // subsampling: 0=25%, 1=50%, 2=100% */
- int _coeffPerSbSelect; // selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
- int _cmTableSelect; // selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
-
- // Packets and packet lists
- QDM2SubPacket _subPackets[16]; // the packets themselves
- QDM2SubPNode _subPacketListA[16]; // list of all packets
- QDM2SubPNode _subPacketListB[16]; // FFT packets B are on list
- int _subPacketsB; // number of packets on 'B' list
- QDM2SubPNode _subPacketListC[16]; // packets with errors?
- QDM2SubPNode _subPacketListD[16]; // DCT packets
-
- // FFT and tones
- FFTTone _fftTones[1000];
- int _fftToneStart;
- int _fftToneEnd;
- FFTCoefficient _fftCoefs[1000];
- int _fftCoefsIndex;
- int _fftCoefsMinIndex[5];
- int _fftCoefsMaxIndex[5];
- int _fftLevelExp[6];
- //RDFTContext _rdftCtx;
- QDM2FFT _fft;
-
- // I/O data
- uint8 *_compressedData;
- float _outputBuffer[1024];
- Common::Array<int16> _outputSamples;
-
- // Synthesis filter
- int16 ff_mpa_synth_window[512];
- int16 _synthBuf[MPA_MAX_CHANNELS][512*2];
- int _synthBufOffset[MPA_MAX_CHANNELS];
- int32 _sbSamples[MPA_MAX_CHANNELS][128][32];
-
- // Mixed temporary data used in decoding
- float _toneLevel[MPA_MAX_CHANNELS][30][64];
- int8 _codingMethod[MPA_MAX_CHANNELS][30][64];
- int8 _quantizedCoeffs[MPA_MAX_CHANNELS][10][8];
- int8 _toneLevelIdxBase[MPA_MAX_CHANNELS][30][8];
- int8 _toneLevelIdxHi1[MPA_MAX_CHANNELS][3][8][8];
- int8 _toneLevelIdxMid[MPA_MAX_CHANNELS][26][8];
- int8 _toneLevelIdxHi2[MPA_MAX_CHANNELS][26];
- int8 _toneLevelIdx[MPA_MAX_CHANNELS][30][64];
- int8 _toneLevelIdxTemp[MPA_MAX_CHANNELS][30][64];
-
- // Flags
- bool _hasErrors; // packet has errors
- int _superblocktype_2_3; // select fft tables and some algorithm based on superblock type
- int _doSynthFilter; // used to perform or skip synthesis filter
-
- uint8 _subPacket; // 0 to 15
- int _noiseIdx; // index for dithering noise table
-
- byte _emptyBuffer[FF_INPUT_BUFFER_PADDING_SIZE];
-
- VLC _vlcTabLevel;
- VLC _vlcTabDiff;
- VLC _vlcTabRun;
- VLC _fftLevelExpAltVlc;
- VLC _fftLevelExpVlc;
- VLC _fftStereoExpVlc;
- VLC _fftStereoPhaseVlc;
- VLC _vlcTabToneLevelIdxHi1;
- VLC _vlcTabToneLevelIdxMid;
- VLC _vlcTabToneLevelIdxHi2;
- VLC _vlcTabType30;
- VLC _vlcTabType34;
- VLC _vlcTabFftToneOffset[5];
- bool _vlcsInitialized;
- void initVlc(void);
-
- uint16 _softclipTable[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
- void softclipTableInit(void);
-
- float _noiseTable[4096];
- byte _randomDequantIndex[256][5];
- byte _randomDequantType24[128][3];
- void rndTableInit(void);
-
- float _noiseSamples[128];
- void initNoiseSamples(void);
-
- RDFTContext _rdftCtx;
-
- void average_quantized_coeffs(void);
- void build_sb_samples_from_noise(int sb);
- void fix_coding_method_array(int sb, int channels, sb_int8_array coding_method);
- void fill_tone_level_array(int flag);
- void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
- sb_int8_array coding_method, int nb_channels,
- int c, int superblocktype_2_3, int cm_table_select);
- void synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max);
- void init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length);
- void init_tone_level_dequantization(GetBitContext *gb, int length);
- void process_subpacket_9(QDM2SubPNode *node);
- void process_subpacket_10(QDM2SubPNode *node, int length);
- void process_subpacket_11(QDM2SubPNode *node, int length);
- void process_subpacket_12(QDM2SubPNode *node, int length);
- void process_synthesis_subpackets(QDM2SubPNode *list);
- void qdm2_decode_super_block(void);
- void qdm2_fft_init_coefficient(int sub_packet, int offset, int duration,
- int channel, int exp, int phase);
- void qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b);
- void qdm2_decode_fft_packets(void);
- void qdm2_fft_generate_tone(FFTTone *tone);
- void qdm2_fft_tone_synthesizer(uint8 sub_packet);
- void qdm2_calculate_fft(int channel);
- void qdm2_synthesis_filter(uint8 index);
- int qdm2_decodeFrame(Common::SeekableReadStream *in);
-};
-
-// Fix compilation for non C99-compliant compilers, like MSVC
-#ifndef int64_t
-typedef signed long long int int64_t;
-#endif
-
-// Integer log2 function. This is much faster than invoking
-// double precision C99 log2 math functions or equivalent, since
-// this is only used to determine maximum number of bits needed
-// i.e. only non-fractional part is needed. Also, the double
-// version is incorrect for exact cases due to floating point
-// rounding errors.
-static inline int scummvm_log2(int n) {
- int ret = -1;
- while(n != 0) {
- n /= 2;
- ret++;
- }
- return ret;
-}
-
-#define QDM2_LIST_ADD(list, size, packet) \
- do { \
- if (size > 0) \
- list[size - 1].next = &list[size]; \
- list[size].packet = packet; \
- list[size].next = NULL; \
- size++; \
- } while(0)
-
-// Result is 8, 16 or 30
-#define QDM2_SB_USED(subSampling) (((subSampling) >= 2) ? 30 : 8 << (subSampling))
-
-#define FIX_NOISE_IDX(noiseIdx) \
- if ((noiseIdx) >= 3840) \
- (noiseIdx) -= 3840 \
-
-#define SB_DITHERING_NOISE(sb, noiseIdx) (_noiseTable[(noiseIdx)++] * sb_noise_attenuation[(sb)])
-
-static inline void initGetBits(GetBitContext *s, const uint8 *buffer, int bitSize) {
- int bufferSize = (bitSize + 7) >> 3;
-
- debug(1, "void initGetBits(GetBitContext *s, const uint8 *buffer, int bitSize)");
-
- if (bufferSize < 0 || bitSize < 0) {
- bufferSize = bitSize = 0;
- buffer = NULL;
- }
-
- s->buffer = buffer;
- s->sizeInBits = bitSize;
- s->bufferEnd = buffer + bufferSize;
- s->index = 0;
-}
-
-static inline int getBitsCount(GetBitContext *s) {
- debug(1, "int getBitsCount(GetBitContext *s)");
- return s->index;
-}
-
-static inline unsigned int getBits1(GetBitContext *s) {
- int index;
- uint8 result;
-
- debug(1, "unsigned int getBits1(GetBitContext *s)");
-
- index = s->index;
- result = s->buffer[index >> 3];
-
- debug(1, "index : %d", index);
-
- result >>= (index & 0x07);
- result &= 1;
- index++;
- s->index = index;
-
- return result;
-}
-
-static inline unsigned int getBits(GetBitContext *s, int n) {
- int tmp, reCache, reIndex;
-
- debug(1, "unsigned int getBits(GetBitContext *s, int n)");
-
- reIndex = s->index;
-
- debug(1, "reIndex : %d", reIndex);
-
- reCache = READ_LE_UINT32((const uint8 *)s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
-
- tmp = (reCache) & ((uint32)0xffffffff >> (32 - n));
-
- s->index = reIndex + n;
-
- return tmp;
-}
-
-static inline void skipBits(GetBitContext *s, int n) {
- int reIndex, reCache;
-
- debug(1, "void skipBits(GetBitContext *s, int n)");
-
- reIndex = s->index;
- reCache = 0;
-
- debug(1, "reIndex : %d", reIndex);
-
- reCache = READ_LE_UINT32((const uint8 *)s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
- s->index = reIndex + n;
-}
-
-#define BITS_LEFT(length, gb) ((length) - getBitsCount((gb)))
-
-static int splitRadixPermutation(int i, int n, int inverse) {
- if (n <= 2)
- return i & 1;
-
- int m = n >> 1;
-
- if(!(i & m))
- return splitRadixPermutation(i, m, inverse) * 2;
-
- m >>= 1;
-
- if (inverse == !(i & m))
- return splitRadixPermutation(i, m, inverse) * 4 + 1;
-
- return splitRadixPermutation(i, m, inverse) * 4 - 1;
-}
-
-// sin(2*pi*x/n) for 0<=x<n/4, followed by n/2<=x<3n/4
-float ff_sin_16[8];
-float ff_sin_32[16];
-float ff_sin_64[32];
-float ff_sin_128[64];
-float ff_sin_256[128];
-float ff_sin_512[256];
-float ff_sin_1024[512];
-float ff_sin_2048[1024];
-float ff_sin_4096[2048];
-float ff_sin_8192[4096];
-float ff_sin_16384[8192];
-float ff_sin_32768[16384];
-float ff_sin_65536[32768];
-
-float *ff_sin_tabs[] = {
- NULL, NULL, NULL, NULL,
- ff_sin_16, ff_sin_32, ff_sin_64, ff_sin_128, ff_sin_256, ff_sin_512, ff_sin_1024,
- ff_sin_2048, ff_sin_4096, ff_sin_8192, ff_sin_16384, ff_sin_32768, ff_sin_65536,
-};
-
-// cos(2*pi*x/n) for 0<=x<=n/4, followed by its reverse
-float ff_cos_16[8];
-float ff_cos_32[16];
-float ff_cos_64[32];
-float ff_cos_128[64];
-float ff_cos_256[128];
-float ff_cos_512[256];
-float ff_cos_1024[512];
-float ff_cos_2048[1024];
-float ff_cos_4096[2048];
-float ff_cos_8192[4096];
-float ff_cos_16384[8192];
-float ff_cos_32768[16384];
-float ff_cos_65536[32768];
-
-float *ff_cos_tabs[] = {
- NULL, NULL, NULL, NULL,
- ff_cos_16, ff_cos_32, ff_cos_64, ff_cos_128, ff_cos_256, ff_cos_512, ff_cos_1024,
- ff_cos_2048, ff_cos_4096, ff_cos_8192, ff_cos_16384, ff_cos_32768, ff_cos_65536,
-};
-
-void initCosineTables(int index) {
- int m = 1 << index;
- double freq = 2 * PI / m;
- float *tab = ff_cos_tabs[index];
-
- for (int i = 0; i <= m / 4; i++)
- tab[i] = cos(i * freq);
-
- for (int i = 1; i < m / 4; i++)
- tab[m / 2 - i] = tab[i];
-}
-
-void fftPermute(FFTContext *s, FFTComplex *z) {
- const uint16 *revtab = s->revtab;
- int np = 1 << s->nbits;
-
- if (s->tmpBuf) {
- // TODO: handle split-radix permute in a more optimal way, probably in-place
- for (int j = 0; j < np; j++)
- s->tmpBuf[revtab[j]] = z[j];
- memcpy(z, s->tmpBuf, np * sizeof(FFTComplex));
- return;
- }
-
- // reverse
- for (int j = 0; j < np; j++) {
- int k = revtab[j];
- if (k < j) {
- FFTComplex tmp = z[k];
- z[k] = z[j];
- z[j] = tmp;
- }
- }
-}
-
-#define DECL_FFT(n,n2,n4) \
-static void fft##n(FFTComplex *z) { \
- fft##n2(z); \
- fft##n4(z + n4 * 2); \
- fft##n4(z + n4 * 3); \
- pass(z, ff_cos_##n, n4 / 2); \
-}
-
-#ifndef M_SQRT1_2
-#define M_SQRT1_2 7.0710678118654752440E-1
-#endif
-
-#define sqrthalf (float)M_SQRT1_2
-
-#define BF(x,y,a,b) { \
- x = a - b; \
- y = a + b; \
-}
-
-#define BUTTERFLIES(a0, a1, a2, a3) { \
- BF(t3, t5, t5, t1); \
- BF(a2.re, a0.re, a0.re, t5); \
- BF(a3.im, a1.im, a1.im, t3); \
- BF(t4, t6, t2, t6); \
- BF(a3.re, a1.re, a1.re, t4); \
- BF(a2.im, a0.im, a0.im, t6); \
-}
-
-// force loading all the inputs before storing any.
-// this is slightly slower for small data, but avoids store->load aliasing
-// for addresses separated by large powers of 2.
-#define BUTTERFLIES_BIG(a0, a1, a2, a3) { \
- float r0 = a0.re, i0 = a0.im, r1 = a1.re, i1 = a1.im; \
- BF(t3, t5, t5, t1); \
- BF(a2.re, a0.re, r0, t5); \
- BF(a3.im, a1.im, i1, t3); \
- BF(t4, t6, t2, t6); \
- BF(a3.re, a1.re, r1, t4); \
- BF(a2.im, a0.im, i0, t6); \
-}
-
-#define TRANSFORM(a0, a1, a2, a3, wre, wim) { \
- t1 = a2.re * wre + a2.im * wim; \
- t2 = a2.im * wre - a2.re * wim; \
- t5 = a3.re * wre - a3.im * wim; \
- t6 = a3.im * wre + a3.re * wim; \
- BUTTERFLIES(a0, a1, a2, a3) \
-}
-
-#define TRANSFORM_ZERO(a0, a1, a2, a3) { \
- t1 = a2.re; \
- t2 = a2.im; \
- t5 = a3.re; \
- t6 = a3.im; \
- BUTTERFLIES(a0, a1, a2, a3) \
-}
-
-// z[0...8n-1], w[1...2n-1]
-#define PASS(name) \
-static void name(FFTComplex *z, const float *wre, unsigned int n) { \
- float t1, t2, t3, t4, t5, t6; \
- int o1 = 2 * n; \
- int o2 = 4 * n; \
- int o3 = 6 * n; \
- const float *wim = wre + o1; \
- n--; \
- \
- TRANSFORM_ZERO(z[0], z[o1], z[o2], z[o3]); \
- TRANSFORM(z[1], z[o1 + 1], z[o2 + 1], z[o3 + 1], wre[1], wim[-1]); \
- \
- do { \
- z += 2; \
- wre += 2; \
- wim -= 2; \
- TRANSFORM(z[0], z[o1], z[o2], z[o3], wre[0], wim[0]); \
- TRANSFORM(z[1], z[o1 + 1],z[o2 + 1], z[o3 + 1], wre[1], wim[-1]); \
- } while(--n); \
-}
-
-PASS(pass)
-#undef BUTTERFLIES
-#define BUTTERFLIES BUTTERFLIES_BIG
-PASS(pass_big)
-
-static void fft4(FFTComplex *z) {
- float t1, t2, t3, t4, t5, t6, t7, t8;
-
- BF(t3, t1, z[0].re, z[1].re);
- BF(t8, t6, z[3].re, z[2].re);
- BF(z[2].re, z[0].re, t1, t6);
- BF(t4, t2, z[0].im, z[1].im);
- BF(t7, t5, z[2].im, z[3].im);
- BF(z[3].im, z[1].im, t4, t8);
- BF(z[3].re, z[1].re, t3, t7);
- BF(z[2].im, z[0].im, t2, t5);
-}
-
-static void fft8(FFTComplex *z) {
- float t1, t2, t3, t4, t5, t6, t7, t8;
-
- fft4(z);
-
- BF(t1, z[5].re, z[4].re, -z[5].re);
- BF(t2, z[5].im, z[4].im, -z[5].im);
- BF(t3, z[7].re, z[6].re, -z[7].re);
- BF(t4, z[7].im, z[6].im, -z[7].im);
- BF(t8, t1, t3, t1);
- BF(t7, t2, t2, t4);
- BF(z[4].re, z[0].re, z[0].re, t1);
- BF(z[4].im, z[0].im, z[0].im, t2);
- BF(z[6].re, z[2].re, z[2].re, t7);
- BF(z[6].im, z[2].im, z[2].im, t8);
-
- TRANSFORM(z[1], z[3], z[5], z[7], sqrthalf, sqrthalf);
-}
-
-#undef BF
-
-DECL_FFT(16,8,4)
-DECL_FFT(32,16,8)
-DECL_FFT(64,32,16)
-DECL_FFT(128,64,32)
-DECL_FFT(256,128,64)
-DECL_FFT(512,256,128)
-#define pass pass_big
-DECL_FFT(1024,512,256)
-DECL_FFT(2048,1024,512)
-DECL_FFT(4096,2048,1024)
-DECL_FFT(8192,4096,2048)
-DECL_FFT(16384,8192,4096)
-DECL_FFT(32768,16384,8192)
-DECL_FFT(65536,32768,16384)
-
-void fftCalc(FFTContext *s, FFTComplex *z) {
- static void (* const fftDispatch[])(FFTComplex*) = {
- fft4, fft8, fft16, fft32, fft64, fft128, fft256, fft512, fft1024,
- fft2048, fft4096, fft8192, fft16384, fft32768, fft65536,
- };
-
- fftDispatch[s->nbits - 2](z);
-}
-
-// complex multiplication: p = a * b
-#define CMUL(pre, pim, are, aim, bre, bim) \
-{\
- float _are = (are); \
- float _aim = (aim); \
- float _bre = (bre); \
- float _bim = (bim); \
- (pre) = _are * _bre - _aim * _bim; \
- (pim) = _are * _bim + _aim * _bre; \
-}
-
-/**
- * Compute the middle half of the inverse MDCT of size N = 2^nbits,
- * thus excluding the parts that can be derived by symmetry
- * @param output N/2 samples
- * @param input N/2 samples
- */
-void imdctHalfC(FFTContext *s, float *output, const float *input) {
- const uint16 *revtab = s->revtab;
- const float *tcos = s->tcos;
- const float *tsin = s->tsin;
- FFTComplex *z = (FFTComplex *)output;
-
- int n = 1 << s->mdctBits;
- int n2 = n >> 1;
- int n4 = n >> 2;
- int n8 = n >> 3;
-
- // pre rotation
- const float *in1 = input;
- const float *in2 = input + n2 - 1;
- for (int k = 0; k < n4; k++) {
- int j = revtab[k];
- CMUL(z[j].re, z[j].im, *in2, *in1, tcos[k], tsin[k]);
- in1 += 2;
- in2 -= 2;
- }
-
- fftCalc(s, z);
-
- // post rotation + reordering
- for (int k = 0; k < n8; k++) {
- float r0, i0, r1, i1;
- CMUL(r0, i1, z[n8 - k - 1].im, z[n8 - k - 1].re, tsin[n8 - k - 1], tcos[n8 - k - 1]);
- CMUL(r1, i0, z[n8 + k].im, z[n8 + k].re, tsin[n8 + k], tcos[n8 + k]);
- z[n8 - k - 1].re = r0;
- z[n8 - k - 1].im = i0;
- z[n8 + k].re = r1;
- z[n8 + k].im = i1;
- }
-}
-
-/**
- * Compute inverse MDCT of size N = 2^nbits
- * @param output N samples
- * @param input N/2 samples
- */
-void imdctCalcC(FFTContext *s, float *output, const float *input) {
- int n = 1 << s->mdctBits;
- int n2 = n >> 1;
- int n4 = n >> 2;
-
- imdctHalfC(s, output + n4, input);
-
- for (int k = 0; k < n4; k++) {
- output[k] = -output[n2 - k - 1];
- output[n - k - 1] = output[n2 + k];
- }
-}
-
-/**
- * Compute MDCT of size N = 2^nbits
- * @param input N samples
- * @param out N/2 samples
- */
-void mdctCalcC(FFTContext *s, float *out, const float *input) {
- const uint16 *revtab = s->revtab;
- const float *tcos = s->tcos;
- const float *tsin = s->tsin;
- FFTComplex *x = (FFTComplex *)out;
-
- int n = 1 << s->mdctBits;
- int n2 = n >> 1;
- int n4 = n >> 2;
- int n8 = n >> 3;
- int n3 = 3 * n4;
-
- // pre rotation
- for (int i = 0; i < n8; i++) {
- float re = -input[2 * i + 3 * n4] - input[n3 - 1 - 2 * i];
- float im = -input[n4 + 2 * i] + input[n4 - 1 - 2 * i];
- int j = revtab[i];
- CMUL(x[j].re, x[j].im, re, im, -tcos[i], tsin[i]);
-
- re = input[2 * i] - input[n2 - 1 - 2 * i];
- im = -(input[n2 + 2 * i] + input[n - 1 - 2 * i]);
- j = revtab[n8 + i];
- CMUL(x[j].re, x[j].im, re, im, -tcos[n8 + i], tsin[n8 + i]);
- }
-
- fftCalc(s, x);
-
- // post rotation
- for (int i = 0; i < n8; i++) {
- float r0, i0, r1, i1;
- CMUL(i1, r0, x[n8 - i - 1].re, x[n8 - i - 1].im, -tsin[n8 - i - 1], -tcos[n8 - i - 1]);
- CMUL(i0, r1, x[n8 + i].re, x[n8 + i].im, -tsin[n8 + i], -tcos[n8 + i]);
- x[n8 - i - 1].re = r0;
- x[n8 - i - 1].im = i0;
- x[n8 + i].re = r1;
- x[n8 + i].im = i1;
- }
-}
-
-int fftInit(FFTContext *s, int nbits, int inverse) {
- int i, j, m, n;
- float alpha, c1, s1, s2;
-
- if (nbits < 2 || nbits > 16)
- goto fail;
-
- s->nbits = nbits;
- n = 1 << nbits;
- s->tmpBuf = NULL;
-
- s->exptab = (FFTComplex *)malloc((n / 2) * sizeof(FFTComplex));
- if (!s->exptab)
- goto fail;
-
- s->revtab = (uint16 *)malloc(n * sizeof(uint16));
- if (!s->revtab)
- goto fail;
- s->inverse = inverse;
-
- s2 = inverse ? 1.0 : -1.0;
-
- s->fftPermute = fftPermute;
- s->fftCalc = fftCalc;
- s->imdctCalc = imdctCalcC;
- s->imdctHalf = imdctHalfC;
- s->mdctCalc = mdctCalcC;
- s->splitRadix = 1;
-
- if (s->splitRadix) {
- for (j = 4; j <= nbits; j++)
- initCosineTables(j);
-
- for (i = 0; i < n; i++)
- s->revtab[-splitRadixPermutation(i, n, s->inverse) & (n - 1)] = i;
-
- s->tmpBuf = (FFTComplex *)malloc(n * sizeof(FFTComplex));
- } else {
- for (i = 0; i < n / 2; i++) {
- alpha = 2 * PI * (float)i / (float)n;
- c1 = cos(alpha);
- s1 = sin(alpha) * s2;
- s->exptab[i].re = c1;
- s->exptab[i].im = s1;
- }
-
- //int np = 1 << nbits;
- //int nblocks = np >> 3;
- //int np2 = np >> 1;
-
- // compute bit reverse table
- for (i = 0; i < n; i++) {
- m = 0;
-
- for (j = 0; j < nbits; j++)
- m |= ((i >> j) & 1) << (nbits - j - 1);
-
- s->revtab[i] = m;
- }
- }
-
- return 0;
-
- fail:
- free(&s->revtab);
- free(&s->exptab);
- free(&s->tmpBuf);
- return -1;
-}
-
-/**
- * Sets up a real FFT.
- * @param nbits log2 of the length of the input array
- * @param trans the type of transform
- */
-int rdftInit(RDFTContext *s, int nbits, RDFTransformType trans) {
- int n = 1 << nbits;
- const double theta = (trans == RDFT || trans == IRIDFT ? -1 : 1) * 2 * PI / n;
-
- s->nbits = nbits;
- s->inverse = trans == IRDFT || trans == IRIDFT;
- s->signConvention = trans == RIDFT || trans == IRIDFT ? 1 : -1;
-
- if (nbits < 4 || nbits > 16)
- return -1;
-
- if (fftInit(&s->fft, nbits - 1, trans == IRDFT || trans == RIDFT) < 0)
- return -1;
-
- initCosineTables(nbits);
- s->tcos = ff_cos_tabs[nbits];
- s->tsin = ff_sin_tabs[nbits] + (trans == RDFT || trans == IRIDFT) * (n >> 2);
-
- for (int i = 0; i < n >> 2; i++)
- s->tsin[i] = sin(i*theta);
-
- return 0;
-}
-
-/** Map one real FFT into two parallel real even and odd FFTs. Then interleave
- * the two real FFTs into one complex FFT. Unmangle the results.
- * ref: http://www.engineeringproductivitytools.com/stuff/T0001/PT10.HTM
- */
-void rdftCalc(RDFTContext *s, float *data) {
- FFTComplex ev, od;
-
- const int n = 1 << s->nbits;
- const float k1 = 0.5;
- const float k2 = 0.5 - s->inverse;
- const float *tcos = s->tcos;
- const float *tsin = s->tsin;
-
- if (!s->inverse) {
- fftPermute(&s->fft, (FFTComplex *)data);
- fftCalc(&s->fft, (FFTComplex *)data);
- }
-
- // i=0 is a special case because of packing, the DC term is real, so we
- // are going to throw the N/2 term (also real) in with it.
- ev.re = data[0];
- data[0] = ev.re + data[1];
- data[1] = ev.re - data[1];
-
- int i;
-
- for (i = 1; i < n >> 2; i++) {
- int i1 = i * 2;
- int i2 = n - i1;
-
- // Separate even and odd FFTs
- ev.re = k1 * (data[i1] + data[i2]);
- od.im = -k2 * (data[i1] - data[i2]);
- ev.im = k1 * (data[i1 + 1] - data[i2 + 1]);
- od.re = k2 * (data[i1 + 1] + data[i2 + 1]);
-
- // Apply twiddle factors to the odd FFT and add to the even FFT
- data[i1] = ev.re + od.re * tcos[i] - od.im * tsin[i];
- data[i1 + 1] = ev.im + od.im * tcos[i] + od.re * tsin[i];
- data[i2] = ev.re - od.re * tcos[i] + od.im * tsin[i];
- data[i2 + 1] = -ev.im + od.im * tcos[i] + od.re * tsin[i];
- }
-
- data[i * 2 + 1] = s->signConvention * data[i * 2 + 1];
- if (s->inverse) {
- data[0] *= k1;
- data[1] *= k1;
- fftPermute(&s->fft, (FFTComplex*)data);
- fftCalc(&s->fft, (FFTComplex*)data);
- }
-}
-
-// half mpeg encoding window (full precision)
-const int32 ff_mpa_enwindow[257] = {
- 0, -1, -1, -1, -1, -1, -1, -2,
- -2, -2, -2, -3, -3, -4, -4, -5,
- -5, -6, -7, -7, -8, -9, -10, -11,
- -13, -14, -16, -17, -19, -21, -24, -26,
- -29, -31, -35, -38, -41, -45, -49, -53,
- -58, -63, -68, -73, -79, -85, -91, -97,
- -104, -111, -117, -125, -132, -139, -147, -154,
- -161, -169, -176, -183, -190, -196, -202, -208,
- 213, 218, 222, 225, 227, 228, 228, 227,
- 224, 221, 215, 208, 200, 189, 177, 163,
- 146, 127, 106, 83, 57, 29, -2, -36,
- -72, -111, -153, -197, -244, -294, -347, -401,
- -459, -519, -581, -645, -711, -779, -848, -919,
- -991, -1064, -1137, -1210, -1283, -1356, -1428, -1498,
- -1567, -1634, -1698, -1759, -1817, -1870, -1919, -1962,
- -2001, -2032, -2057, -2075, -2085, -2087, -2080, -2063,
- 2037, 2000, 1952, 1893, 1822, 1739, 1644, 1535,
- 1414, 1280, 1131, 970, 794, 605, 402, 185,
- -45, -288, -545, -814, -1095, -1388, -1692, -2006,
- -2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788,
- -5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597,
- -7910, -8209, -8491, -8755, -8998, -9219, -9416, -9585,
- -9727, -9838, -9916, -9959, -9966, -9935, -9863, -9750,
- -9592, -9389, -9139, -8840, -8492, -8092, -7640, -7134,
- 6574, 5959, 5288, 4561, 3776, 2935, 2037, 1082,
- 70, -998, -2122, -3300, -4533, -5818, -7154, -8540,
- -9975,-11455,-12980,-14548,-16155,-17799,-19478,-21189,
--22929,-24694,-26482,-28289,-30112,-31947,-33791,-35640,
--37489,-39336,-41176,-43006,-44821,-46617,-48390,-50137,
--51853,-53534,-55178,-56778,-58333,-59838,-61289,-62684,
--64019,-65290,-66494,-67629,-68692,-69679,-70590,-71420,
--72169,-72835,-73415,-73908,-74313,-74630,-74856,-74992,
- 75038
-};
-
-void ff_mpa_synth_init(int16 *window) {
- int i;
- int32 v;
-
- // max = 18760, max sum over all 16 coefs : 44736
- for(i = 0; i < 257; i++) {
- v = ff_mpa_enwindow[i];
- v = (v + 2) >> 2;
- window[i] = v;
-
- if ((i & 63) != 0)
- v = -v;
-
- if (i != 0)
- window[512 - i] = v;
- }
-}
-
-static inline uint16 round_sample(int *sum) {
- int sum1;
- sum1 = (*sum) >> 14;
- *sum &= (1 << 14)-1;
- if (sum1 < (-0x7fff - 1))
- sum1 = (-0x7fff - 1);
- if (sum1 > 0x7fff)
- sum1 = 0x7fff;
- return sum1;
-}
-
-static inline int MULH(int a, int b) {
- return ((int64_t)(a) * (int64_t)(b))>>32;
-}
-
-// signed 16x16 -> 32 multiply add accumulate
-#define MACS(rt, ra, rb) rt += (ra) * (rb)
-
-#define MLSS(rt, ra, rb) ((rt) -= (ra) * (rb))
-
-#define SUM8(op, sum, w, p)\
-{\
- op(sum, (w)[0 * 64], (p)[0 * 64]);\
- op(sum, (w)[1 * 64], (p)[1 * 64]);\
- op(sum, (w)[2 * 64], (p)[2 * 64]);\
- op(sum, (w)[3 * 64], (p)[3 * 64]);\
- op(sum, (w)[4 * 64], (p)[4 * 64]);\
- op(sum, (w)[5 * 64], (p)[5 * 64]);\
- op(sum, (w)[6 * 64], (p)[6 * 64]);\
- op(sum, (w)[7 * 64], (p)[7 * 64]);\
-}
-
-#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
-{\
- tmp_s = p[0 * 64];\
- op1(sum1, (w1)[0 * 64], tmp_s);\
- op2(sum2, (w2)[0 * 64], tmp_s);\
- tmp_s = p[1 * 64];\
- op1(sum1, (w1)[1 * 64], tmp_s);\
- op2(sum2, (w2)[1 * 64], tmp_s);\
- tmp_s = p[2 * 64];\
- op1(sum1, (w1)[2 * 64], tmp_s);\
- op2(sum2, (w2)[2 * 64], tmp_s);\
- tmp_s = p[3 * 64];\
- op1(sum1, (w1)[3 * 64], tmp_s);\
- op2(sum2, (w2)[3 * 64], tmp_s);\
- tmp_s = p[4 * 64];\
- op1(sum1, (w1)[4 * 64], tmp_s);\
- op2(sum2, (w2)[4 * 64], tmp_s);\
- tmp_s = p[5 * 64];\
- op1(sum1, (w1)[5 * 64], tmp_s);\
- op2(sum2, (w2)[5 * 64], tmp_s);\
- tmp_s = p[6 * 64];\
- op1(sum1, (w1)[6 * 64], tmp_s);\
- op2(sum2, (w2)[6 * 64], tmp_s);\
- tmp_s = p[7 * 64];\
- op1(sum1, (w1)[7 * 64], tmp_s);\
- op2(sum2, (w2)[7 * 64], tmp_s);\
-}
-
-#define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
-
-// tab[i][j] = 1.0 / (2.0 * cos(pi*(2*k+1) / 2^(6 - j)))
-
-// cos(i*pi/64)
-
-#define COS0_0 FIXHR(0.50060299823519630134/2)
-#define COS0_1 FIXHR(0.50547095989754365998/2)
-#define COS0_2 FIXHR(0.51544730992262454697/2)
-#define COS0_3 FIXHR(0.53104259108978417447/2)
-#define COS0_4 FIXHR(0.55310389603444452782/2)
-#define COS0_5 FIXHR(0.58293496820613387367/2)
-#define COS0_6 FIXHR(0.62250412303566481615/2)
-#define COS0_7 FIXHR(0.67480834145500574602/2)
-#define COS0_8 FIXHR(0.74453627100229844977/2)
-#define COS0_9 FIXHR(0.83934964541552703873/2)
-#define COS0_10 FIXHR(0.97256823786196069369/2)
-#define COS0_11 FIXHR(1.16943993343288495515/4)
-#define COS0_12 FIXHR(1.48416461631416627724/4)
-#define COS0_13 FIXHR(2.05778100995341155085/8)
-#define COS0_14 FIXHR(3.40760841846871878570/8)
-#define COS0_15 FIXHR(10.19000812354805681150/32)
-
-#define COS1_0 FIXHR(0.50241928618815570551/2)
-#define COS1_1 FIXHR(0.52249861493968888062/2)
-#define COS1_2 FIXHR(0.56694403481635770368/2)
-#define COS1_3 FIXHR(0.64682178335999012954/2)
-#define COS1_4 FIXHR(0.78815462345125022473/2)
-#define COS1_5 FIXHR(1.06067768599034747134/4)
-#define COS1_6 FIXHR(1.72244709823833392782/4)
-#define COS1_7 FIXHR(5.10114861868916385802/16)
-
-#define COS2_0 FIXHR(0.50979557910415916894/2)
-#define COS2_1 FIXHR(0.60134488693504528054/2)
-#define COS2_2 FIXHR(0.89997622313641570463/2)
-#define COS2_3 FIXHR(2.56291544774150617881/8)
-
-#define COS3_0 FIXHR(0.54119610014619698439/2)
-#define COS3_1 FIXHR(1.30656296487637652785/4)
-
-#define COS4_0 FIXHR(0.70710678118654752439/2)
-
-/* butterfly operator */
-#define BF(a, b, c, s)\
-{\
- tmp0 = tab[a] + tab[b];\
- tmp1 = tab[a] - tab[b];\
- tab[a] = tmp0;\
- tab[b] = MULH(tmp1<<(s), c);\
-}
-
-#define BF1(a, b, c, d)\
-{\
- BF(a, b, COS4_0, 1);\
- BF(c, d,-COS4_0, 1);\
- tab[c] += tab[d];\
-}
-
-#define BF2(a, b, c, d)\
-{\
- BF(a, b, COS4_0, 1);\
- BF(c, d,-COS4_0, 1);\
- tab[c] += tab[d];\
- tab[a] += tab[c];\
- tab[c] += tab[b];\
- tab[b] += tab[d];\
-}
-
-#define ADD(a, b) tab[a] += tab[b]
-
-// DCT32 without 1/sqrt(2) coef zero scaling.
-static void dct32(int32 *out, int32 *tab) {
- int tmp0, tmp1;
-
- // pass 1
- BF( 0, 31, COS0_0 , 1);
- BF(15, 16, COS0_15, 5);
- // pass 2
- BF( 0, 15, COS1_0 , 1);
- BF(16, 31,-COS1_0 , 1);
- // pass 1
- BF( 7, 24, COS0_7 , 1);
- BF( 8, 23, COS0_8 , 1);
- // pass 2
- BF( 7, 8, COS1_7 , 4);
- BF(23, 24,-COS1_7 , 4);
- // pass 3
- BF( 0, 7, COS2_0 , 1);
- BF( 8, 15,-COS2_0 , 1);
- BF(16, 23, COS2_0 , 1);
- BF(24, 31,-COS2_0 , 1);
- // pass 1
- BF( 3, 28, COS0_3 , 1);
- BF(12, 19, COS0_12, 2);
- // pass 2
- BF( 3, 12, COS1_3 , 1);
- BF(19, 28,-COS1_3 , 1);
- // pass 1
- BF( 4, 27, COS0_4 , 1);
- BF(11, 20, COS0_11, 2);
- // pass 2
- BF( 4, 11, COS1_4 , 1);
- BF(20, 27,-COS1_4 , 1);
- // pass 3
- BF( 3, 4, COS2_3 , 3);
- BF(11, 12,-COS2_3 , 3);
- BF(19, 20, COS2_3 , 3);
- BF(27, 28,-COS2_3 , 3);
- // pass 4
- BF( 0, 3, COS3_0 , 1);
- BF( 4, 7,-COS3_0 , 1);
- BF( 8, 11, COS3_0 , 1);
- BF(12, 15,-COS3_0 , 1);
- BF(16, 19, COS3_0 , 1);
- BF(20, 23,-COS3_0 , 1);
- BF(24, 27, COS3_0 , 1);
- BF(28, 31,-COS3_0 , 1);
-
- // pass 1
- BF( 1, 30, COS0_1 , 1);
- BF(14, 17, COS0_14, 3);
- // pass 2
- BF( 1, 14, COS1_1 , 1);
- BF(17, 30,-COS1_1 , 1);
- // pass 1
- BF( 6, 25, COS0_6 , 1);
- BF( 9, 22, COS0_9 , 1);
- // pass 2
- BF( 6, 9, COS1_6 , 2);
- BF(22, 25,-COS1_6 , 2);
- // pass 3
- BF( 1, 6, COS2_1 , 1);
- BF( 9, 14,-COS2_1 , 1);
- BF(17, 22, COS2_1 , 1);
- BF(25, 30,-COS2_1 , 1);
-
- // pass 1
- BF( 2, 29, COS0_2 , 1);
- BF(13, 18, COS0_13, 3);
- // pass 2
- BF( 2, 13, COS1_2 , 1);
- BF(18, 29,-COS1_2 , 1);
- // pass 1
- BF( 5, 26, COS0_5 , 1);
- BF(10, 21, COS0_10, 1);
- // pass 2
- BF( 5, 10, COS1_5 , 2);
- BF(21, 26,-COS1_5 , 2);
- // pass 3
- BF( 2, 5, COS2_2 , 1);
- BF(10, 13,-COS2_2 , 1);
- BF(18, 21, COS2_2 , 1);
- BF(26, 29,-COS2_2 , 1);
- // pass 4
- BF( 1, 2, COS3_1 , 2);
- BF( 5, 6,-COS3_1 , 2);
- BF( 9, 10, COS3_1 , 2);
- BF(13, 14,-COS3_1 , 2);
- BF(17, 18, COS3_1 , 2);
- BF(21, 22,-COS3_1 , 2);
- BF(25, 26, COS3_1 , 2);
- BF(29, 30,-COS3_1 , 2);
-
- // pass 5
- BF1( 0, 1, 2, 3);
- BF2( 4, 5, 6, 7);
- BF1( 8, 9, 10, 11);
- BF2(12, 13, 14, 15);
- BF1(16, 17, 18, 19);
- BF2(20, 21, 22, 23);
- BF1(24, 25, 26, 27);
- BF2(28, 29, 30, 31);
-
- // pass 6
- ADD( 8, 12);
- ADD(12, 10);
- ADD(10, 14);
- ADD(14, 9);
- ADD( 9, 13);
- ADD(13, 11);
- ADD(11, 15);
-
- out[ 0] = tab[0];
- out[16] = tab[1];
- out[ 8] = tab[2];
- out[24] = tab[3];
- out[ 4] = tab[4];
- out[20] = tab[5];
- out[12] = tab[6];
- out[28] = tab[7];
- out[ 2] = tab[8];
- out[18] = tab[9];
- out[10] = tab[10];
- out[26] = tab[11];
- out[ 6] = tab[12];
- out[22] = tab[13];
- out[14] = tab[14];
- out[30] = tab[15];
-
- ADD(24, 28);
- ADD(28, 26);
- ADD(26, 30);
- ADD(30, 25);
- ADD(25, 29);
- ADD(29, 27);
- ADD(27, 31);
-
- out[ 1] = tab[16] + tab[24];
- out[17] = tab[17] + tab[25];
- out[ 9] = tab[18] + tab[26];
- out[25] = tab[19] + tab[27];
- out[ 5] = tab[20] + tab[28];
- out[21] = tab[21] + tab[29];
- out[13] = tab[22] + tab[30];
- out[29] = tab[23] + tab[31];
- out[ 3] = tab[24] + tab[20];
- out[19] = tab[25] + tab[21];
- out[11] = tab[26] + tab[22];
- out[27] = tab[27] + tab[23];
- out[ 7] = tab[28] + tab[18];
- out[23] = tab[29] + tab[19];
- out[15] = tab[30] + tab[17];
- out[31] = tab[31];
-}
-
-// 32 sub band synthesis filter. Input: 32 sub band samples, Output:
-// 32 samples.
-// XXX: optimize by avoiding ring buffer usage
-void ff_mpa_synth_filter(int16 *synth_buf_ptr, int *synth_buf_offset,
- int16 *window, int *dither_state,
- int16 *samples, int incr,
- int32 sb_samples[32])
-{
- int16 *synth_buf;
- const int16 *w, *w2, *p;
- int j, offset;
- int16 *samples2;
- int32 tmp[32];
- int sum, sum2;
- int tmp_s;
-
- offset = *synth_buf_offset;
- synth_buf = synth_buf_ptr + offset;
-
- dct32(tmp, sb_samples);
- for(j = 0; j < 32; j++) {
- // NOTE: can cause a loss in precision if very high amplitude sound
- if (tmp[j] < (-0x7fff - 1))
- synth_buf[j] = (-0x7fff - 1);
- else if (tmp[j] > 0x7fff)
- synth_buf[j] = 0x7fff;
- else
- synth_buf[j] = tmp[j];
- }
-
- // copy to avoid wrap
- memcpy(synth_buf + 512, synth_buf, 32 * sizeof(int16));
-
- samples2 = samples + 31 * incr;
- w = window;
- w2 = window + 31;
-
- sum = *dither_state;
- p = synth_buf + 16;
- SUM8(MACS, sum, w, p);
- p = synth_buf + 48;
- SUM8(MLSS, sum, w + 32, p);
- *samples = round_sample(&sum);
- samples += incr;
- w++;
-
- // we calculate two samples at the same time to avoid one memory
- // access per two sample
- for(j = 1; j < 16; j++) {
- sum2 = 0;
- p = synth_buf + 16 + j;
- SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
- p = synth_buf + 48 - j;
- SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
-
- *samples = round_sample(&sum);
- samples += incr;
- sum += sum2;
- *samples2 = round_sample(&sum);
- samples2 -= incr;
- w++;
- w2--;
- }
-
- p = synth_buf + 32;
- SUM8(MLSS, sum, w + 32, p);
- *samples = round_sample(&sum);
- *dither_state= sum;
-
- offset = (offset - 32) & 511;
- *synth_buf_offset = offset;
-}
-
-/**
- * parses a vlc code, faster then get_vlc()
- * @param bits is the number of bits which will be read at once, must be
- * identical to nb_bits in init_vlc()
- * @param max_depth is the number of times bits bits must be read to completely
- * read the longest vlc code
- * = (max_vlc_length + bits - 1) / bits
- */
-static int getVlc2(GetBitContext *s, int16 (*table)[2], int bits, int maxDepth) {
- int reIndex;
- int reCache;
- int index;
- int code;
- int n;
-
- debug(1, "int getVlc2(GetBitContext *s, int16 (*table)[2], int bits, int maxDepth)");
-
- reIndex = s->index;
- reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
- index = reCache & (0xffffffff >> (32 - bits));
- code = table[index][0];
- n = table[index][1];
-
- debug(1, "reIndex : %d", reIndex);
- debug(1, "reCache : %d", reCache);
- debug(1, "index : %d", index);
- debug(1, "code : %d", code);
- debug(1, "n : %d", n);
-
- if (maxDepth > 1 && n < 0){
- reIndex += bits;
- reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
-
- int nbBits = -n;
-
- index = (reCache & (0xffffffff >> (32 - nbBits))) + code;
- code = table[index][0];
- n = table[index][1];
-
- if(maxDepth > 2 && n < 0) {
- reIndex += nbBits;
- reCache = READ_LE_UINT32(s->buffer + (reIndex >> 3)) >> (reIndex & 0x07);
-
- nbBits = -n;
-
- index = (reCache & (0xffffffff >> (32 - nbBits))) + code;
- code = table[index][0];
- n = table[index][1];
- }
- }
-
- reCache >>= n;
- s->index = reIndex + n;
- return code;
-}
-
-static int allocTable(VLC *vlc, int size, int use_static) {
- int index;
- index = vlc->table_size;
- vlc->table_size += size;
- if (vlc->table_size > vlc->table_allocated) {
- if(use_static)
- error("QDM2 cant do anything, init_vlc() is used with too little memory");
- vlc->table_allocated += (1 << vlc->bits);
- vlc->table = (int16 (*)[2])realloc(vlc->table, sizeof(int16 *) * 2 * vlc->table_allocated);
- if (!vlc->table)
- return -1;
- }
- return index;
-}
-
-#define GET_DATA(v, table, i, wrap, size)\
-{\
- const uint8 *ptr = (const uint8 *)table + i * wrap;\
- switch(size) {\
- case 1:\
- v = *(const uint8 *)ptr;\
- break;\
- case 2:\
- v = *(const uint16 *)ptr;\
- break;\
- default:\
- v = *(const uint32 *)ptr;\
- break;\
- }\
-}
-
-static int build_table(VLC *vlc, int table_nb_bits,
- int nb_codes,
- const void *bits, int bits_wrap, int bits_size,
- const void *codes, int codes_wrap, int codes_size,
- const void *symbols, int symbols_wrap, int symbols_size,
- int code_prefix, int n_prefix, int flags)
-{
- int i, j, k, n, table_size, table_index, nb, n1, index, code_prefix2, symbol;
- uint32 code;
- int16 (*table)[2];
-
- table_size = 1 << table_nb_bits;
- table_index = allocTable(vlc, table_size, flags & 4);
- debug(2, "QDM2 new table index=%d size=%d code_prefix=%x n=%d", table_index, table_size, code_prefix, n_prefix);
- if (table_index < 0)
- return -1;
- table = &vlc->table[table_index];
-
- for(i = 0; i < table_size; i++) {
- table[i][1] = 0; //bits
- table[i][0] = -1; //codes
- }
-
- // first pass: map codes and compute auxillary table sizes
- for(i = 0; i < nb_codes; i++) {
- GET_DATA(n, bits, i, bits_wrap, bits_size);
- GET_DATA(code, codes, i, codes_wrap, codes_size);
- // we accept tables with holes
- if (n <= 0)
- continue;
- if (!symbols)
- symbol = i;
- else
- GET_DATA(symbol, symbols, i, symbols_wrap, symbols_size);
- debug(2, "QDM2 i=%d n=%d code=0x%x", i, n, code);
- // if code matches the prefix, it is in the table
- n -= n_prefix;
- if(flags & 2)
- code_prefix2= code & (n_prefix>=32 ? 0xffffffff : (1 << n_prefix)-1);
- else
- code_prefix2= code >> n;
- if (n > 0 && code_prefix2 == code_prefix) {
- if (n <= table_nb_bits) {
- // no need to add another table
- j = (code << (table_nb_bits - n)) & (table_size - 1);
- nb = 1 << (table_nb_bits - n);
- for(k = 0; k < nb; k++) {
- if(flags & 2)
- j = (code >> n_prefix) + (k<<n);
- debug(2, "QDM2 %4x: code=%d n=%d",j, i, n);
- if (table[j][1] /*bits*/ != 0) {
- error("QDM2 incorrect codes");
- return -1;
- }
- table[j][1] = n; //bits
- table[j][0] = symbol;
- j++;
- }
- } else {
- n -= table_nb_bits;
- j = (code >> ((flags & 2) ? n_prefix : n)) & ((1 << table_nb_bits) - 1);
- debug(2, "QDM2 %4x: n=%d (subtable)", j, n);
- // compute table size
- n1 = -table[j][1]; //bits
- if (n > n1)
- n1 = n;
- table[j][1] = -n1; //bits
- }
- }
- }
-
- // second pass : fill auxillary tables recursively
- for(i = 0;i < table_size; i++) {
- n = table[i][1]; //bits
- if (n < 0) {
- n = -n;
- if (n > table_nb_bits) {
- n = table_nb_bits;
- table[i][1] = -n; //bits
- }
- index = build_table(vlc, n, nb_codes,
- bits, bits_wrap, bits_size,
- codes, codes_wrap, codes_size,
- symbols, symbols_wrap, symbols_size,
- (flags & 2) ? (code_prefix | (i << n_prefix)) : ((code_prefix << table_nb_bits) | i),
- n_prefix + table_nb_bits, flags);
- if (index < 0)
- return -1;
- // note: realloc has been done, so reload tables
- table = &vlc->table[table_index];
- table[i][0] = index; //code
- }
- }
- return table_index;
-}
-
-/* Build VLC decoding tables suitable for use with get_vlc().
-
- 'nb_bits' set thee decoding table size (2^nb_bits) entries. The
- bigger it is, the faster is the decoding. But it should not be too
- big to save memory and L1 cache. '9' is a good compromise.
-
- 'nb_codes' : number of vlcs codes
-
- 'bits' : table which gives the size (in bits) of each vlc code.
-
- 'codes' : table which gives the bit pattern of of each vlc code.
-
- 'symbols' : table which gives the values to be returned from get_vlc().
-
- 'xxx_wrap' : give the number of bytes between each entry of the
- 'bits' or 'codes' tables.
-
- 'xxx_size' : gives the number of bytes of each entry of the 'bits'
- or 'codes' tables.
-
- 'wrap' and 'size' allows to use any memory configuration and types
- (byte/word/long) to store the 'bits', 'codes', and 'symbols' tables.
-
- 'use_static' should be set to 1 for tables, which should be freed
- with av_free_static(), 0 if free_vlc() will be used.
-*/
-void initVlcSparse(VLC *vlc, int nb_bits, int nb_codes,
- const void *bits, int bits_wrap, int bits_size,
- const void *codes, int codes_wrap, int codes_size,
- const void *symbols, int symbols_wrap, int symbols_size) {
- vlc->bits = nb_bits;
-
- if(vlc->table_size && vlc->table_size == vlc->table_allocated) {
- return;
- } else if(vlc->table_size) {
- error("called on a partially initialized table");
- }
-
- debug(2, "QDM2 build table nb_codes=%d", nb_codes);
-
- if (build_table(vlc, nb_bits, nb_codes,
- bits, bits_wrap, bits_size,
- codes, codes_wrap, codes_size,
- symbols, symbols_wrap, symbols_size,
- 0, 0, 4 | 2) < 0) {
- free(&vlc->table);
- return; // Error
- }
-
- if(vlc->table_size != vlc->table_allocated)
- error("QDM2 needed %d had %d", vlc->table_size, vlc->table_allocated);
-}
-
-void QDM2Stream::softclipTableInit(void) {
- uint16 i;
- double dfl = SOFTCLIP_THRESHOLD - 32767;
- float delta = 1.0 / -dfl;
-
- for (i = 0; i < ARRAYSIZE(_softclipTable); i++)
- _softclipTable[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
-}
-
-// random generated table
-void QDM2Stream::rndTableInit(void) {
- uint16 i;
- uint16 j;
- uint32 ldw, hdw;
- // TODO: Replace Code with uint64 less version...
- int64_t tmp64_1;
- int64_t random_seed = 0;
- float delta = 1.0 / 16384.0;
-
- for(i = 0; i < ARRAYSIZE(_noiseTable); i++) {
- random_seed = random_seed * 214013 + 2531011;
- _noiseTable[i] = (delta * (float)(((int32)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
- }
-
- for (i = 0; i < 256; i++) {
- random_seed = 81;
- ldw = i;
- for (j = 0; j < 5; j++) {
- _randomDequantIndex[i][j] = (uint8)((ldw / random_seed) & 0xFF);
- ldw = (uint32)ldw % (uint32)random_seed;
- tmp64_1 = (random_seed * 0x55555556);
- hdw = (uint32)(tmp64_1 >> 32);
- random_seed = (int64_t)(hdw + (ldw >> 31));
- }
- }
-
- for (i = 0; i < 128; i++) {
- random_seed = 25;
- ldw = i;
- for (j = 0; j < 3; j++) {
- _randomDequantType24[i][j] = (uint8)((ldw / random_seed) & 0xFF);
- ldw = (uint32)ldw % (uint32)random_seed;
- tmp64_1 = (random_seed * 0x66666667);
- hdw = (uint32)(tmp64_1 >> 33);
- random_seed = hdw + (ldw >> 31);
- }
- }
-}
-
-void QDM2Stream::initNoiseSamples(void) {
- uint16 i;
- uint32 random_seed = 0;
- float delta = 1.0 / 16384.0;
-
- for (i = 0; i < ARRAYSIZE(_noiseSamples); i++) {
- random_seed = random_seed * 214013 + 2531011;
- _noiseSamples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
- }
-}
-
-static const uint16 qdm2_vlc_offs[18] = {
- 0, 260, 566, 598, 894, 1166, 1230, 1294, 1678, 1950, 2214, 2278, 2310, 2570, 2834, 3124, 3448, 3838
-};
-
-void QDM2Stream::initVlc(void) {
- static int16 qdm2_table[3838][2];
-
- if (!_vlcsInitialized) {
- _vlcTabLevel.table = &qdm2_table[qdm2_vlc_offs[0]];
- _vlcTabLevel.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
- _vlcTabLevel.table_size = 0;
- initVlcSparse(&_vlcTabLevel, 8, 24,
- vlc_tab_level_huffbits, 1, 1,
- vlc_tab_level_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabDiff.table = &qdm2_table[qdm2_vlc_offs[1]];
- _vlcTabDiff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
- _vlcTabDiff.table_size = 0;
- initVlcSparse(&_vlcTabDiff, 8, 37,
- vlc_tab_diff_huffbits, 1, 1,
- vlc_tab_diff_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabRun.table = &qdm2_table[qdm2_vlc_offs[2]];
- _vlcTabRun.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
- _vlcTabRun.table_size = 0;
- initVlcSparse(&_vlcTabRun, 5, 6,
- vlc_tab_run_huffbits, 1, 1,
- vlc_tab_run_huffcodes, 1, 1, NULL, 0, 0);
-
- _fftLevelExpAltVlc.table = &qdm2_table[qdm2_vlc_offs[3]];
- _fftLevelExpAltVlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
- _fftLevelExpAltVlc.table_size = 0;
- initVlcSparse(&_fftLevelExpAltVlc, 8, 28,
- fft_level_exp_alt_huffbits, 1, 1,
- fft_level_exp_alt_huffcodes, 2, 2, NULL, 0, 0);
-
- _fftLevelExpVlc.table = &qdm2_table[qdm2_vlc_offs[4]];
- _fftLevelExpVlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
- _fftLevelExpVlc.table_size = 0;
- initVlcSparse(&_fftLevelExpVlc, 8, 20,
- fft_level_exp_huffbits, 1, 1,
- fft_level_exp_huffcodes, 2, 2, NULL, 0, 0);
-
- _fftStereoExpVlc.table = &qdm2_table[qdm2_vlc_offs[5]];
- _fftStereoExpVlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
- _fftStereoExpVlc.table_size = 0;
- initVlcSparse(&_fftStereoExpVlc, 6, 7,
- fft_stereo_exp_huffbits, 1, 1,
- fft_stereo_exp_huffcodes, 1, 1, NULL, 0, 0);
-
- _fftStereoPhaseVlc.table = &qdm2_table[qdm2_vlc_offs[6]];
- _fftStereoPhaseVlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
- _fftStereoPhaseVlc.table_size = 0;
- initVlcSparse(&_fftStereoPhaseVlc, 6, 9,
- fft_stereo_phase_huffbits, 1, 1,
- fft_stereo_phase_huffcodes, 1, 1, NULL, 0, 0);
-
- _vlcTabToneLevelIdxHi1.table = &qdm2_table[qdm2_vlc_offs[7]];
- _vlcTabToneLevelIdxHi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
- _vlcTabToneLevelIdxHi1.table_size = 0;
- initVlcSparse(&_vlcTabToneLevelIdxHi1, 8, 20,
- vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
- vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabToneLevelIdxMid.table = &qdm2_table[qdm2_vlc_offs[8]];
- _vlcTabToneLevelIdxMid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
- _vlcTabToneLevelIdxMid.table_size = 0;
- initVlcSparse(&_vlcTabToneLevelIdxMid, 8, 24,
- vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
- vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabToneLevelIdxHi2.table = &qdm2_table[qdm2_vlc_offs[9]];
- _vlcTabToneLevelIdxHi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
- _vlcTabToneLevelIdxHi2.table_size = 0;
- initVlcSparse(&_vlcTabToneLevelIdxHi2, 8, 24,
- vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
- vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabType30.table = &qdm2_table[qdm2_vlc_offs[10]];
- _vlcTabType30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
- _vlcTabType30.table_size = 0;
- initVlcSparse(&_vlcTabType30, 6, 9,
- vlc_tab_type30_huffbits, 1, 1,
- vlc_tab_type30_huffcodes, 1, 1, NULL, 0, 0);
-
- _vlcTabType34.table = &qdm2_table[qdm2_vlc_offs[11]];
- _vlcTabType34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
- _vlcTabType34.table_size = 0;
- initVlcSparse(&_vlcTabType34, 5, 10,
- vlc_tab_type34_huffbits, 1, 1,
- vlc_tab_type34_huffcodes, 1, 1, NULL, 0, 0);
-
- _vlcTabFftToneOffset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
- _vlcTabFftToneOffset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
- _vlcTabFftToneOffset[0].table_size = 0;
- initVlcSparse(&_vlcTabFftToneOffset[0], 8, 23,
- vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabFftToneOffset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
- _vlcTabFftToneOffset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
- _vlcTabFftToneOffset[1].table_size = 0;
- initVlcSparse(&_vlcTabFftToneOffset[1], 8, 28,
- vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabFftToneOffset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
- _vlcTabFftToneOffset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
- _vlcTabFftToneOffset[2].table_size = 0;
- initVlcSparse(&_vlcTabFftToneOffset[2], 8, 32,
- vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabFftToneOffset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
- _vlcTabFftToneOffset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
- _vlcTabFftToneOffset[3].table_size = 0;
- initVlcSparse(&_vlcTabFftToneOffset[3], 8, 35,
- vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcTabFftToneOffset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
- _vlcTabFftToneOffset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
- _vlcTabFftToneOffset[4].table_size = 0;
- initVlcSparse(&_vlcTabFftToneOffset[4], 8, 38,
- vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, NULL, 0, 0);
-
- _vlcsInitialized = true;
- }
-}
-
-QDM2Stream::QDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) {
- uint32 tmp;
- int32 tmp_s;
- int tmp_val;
- int i;
-
- debug(1, "QDM2Stream::QDM2Stream() Call");
-
- _stream = stream;
- _compressedData = NULL;
- _subPacket = 0;
- memset(_quantizedCoeffs, 0, sizeof(_quantizedCoeffs));
- memset(_fftLevelExp, 0, sizeof(_fftLevelExp));
- _noiseIdx = 0;
- memset(_fftCoefsMinIndex, 0, sizeof(_fftCoefsMinIndex));
- memset(_fftCoefsMaxIndex, 0, sizeof(_fftCoefsMaxIndex));
- _fftToneStart = 0;
- _fftToneEnd = 0;
- for(i = 0; i < ARRAYSIZE(_subPacketListA); i++) {
- _subPacketListA[i].packet = NULL;
- _subPacketListA[i].next = NULL;
- }
- _subPacketsB = 0;
- for(i = 0; i < ARRAYSIZE(_subPacketListB); i++) {
- _subPacketListB[i].packet = NULL;
- _subPacketListB[i].next = NULL;
- }
- for(i = 0; i < ARRAYSIZE(_subPacketListC); i++) {
- _subPacketListC[i].packet = NULL;
- _subPacketListC[i].next = NULL;
- }
- for(i = 0; i < ARRAYSIZE(_subPacketListD); i++) {
- _subPacketListD[i].packet = NULL;
- _subPacketListD[i].next = NULL;
- }
- memset(_synthBuf, 0, sizeof(_synthBuf));
- memset(_synthBufOffset, 0, sizeof(_synthBufOffset));
- memset(_sbSamples, 0, sizeof(_sbSamples));
- memset(_outputBuffer, 0, sizeof(_outputBuffer));
- _vlcsInitialized = false;
- _superblocktype_2_3 = 0;
- _hasErrors = false;
-
- // Rewind extraData stream from any previous calls...
- extraData->seek(0, SEEK_SET);
-
- tmp_s = extraData->readSint32BE();
- debug(1, "QDM2Stream::QDM2Stream() extraSize: %d", tmp_s);
- if ((extraData->size() - extraData->pos()) / 4 + 1 != tmp_s)
- warning("QDM2Stream::QDM2Stream() extraSize mismatch - Expected %d", (extraData->size() - extraData->pos()) / 4 + 1);
- if (tmp_s < 12)
- error("QDM2Stream::QDM2Stream() Insufficient extraData");
-
- tmp = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() extraTag: %d", tmp);
- if (tmp != MKID_BE('frma'))
- warning("QDM2Stream::QDM2Stream() extraTag mismatch");
-
- tmp = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() extraType: %d", tmp);
- if (tmp == MKID_BE('QDMC'))
- warning("QDM2Stream::QDM2Stream() QDMC stream type not supported.");
- else if (tmp != MKID_BE('QDM2'))
- error("QDM2Stream::QDM2Stream() Unsupported stream type");
-
- tmp_s = extraData->readSint32BE();
- debug(1, "QDM2Stream::QDM2Stream() extraSize2: %d", tmp_s);
- if ((extraData->size() - extraData->pos()) + 4 != tmp_s)
- warning("QDM2Stream::QDM2Stream() extraSize2 mismatch - Expected %d", (extraData->size() - extraData->pos()) + 4);
-
- tmp = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() extraTag2: %d", tmp);
- if (tmp != MKID_BE('QDCA'))
- warning("QDM2Stream::QDM2Stream() extraTag2 mismatch");
-
- if (extraData->readUint32BE() != 1)
- warning("QDM2Stream::QDM2Stream() u0 field not 1");
-
- _channels = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() channels: %d", _channels);
-
- _sampleRate = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() sampleRate: %d", _sampleRate);
-
- _bitRate = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() bitRate: %d", _bitRate);
-
- _blockSize = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() blockSize: %d", _blockSize);
-
- _frameSize = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() frameSize: %d", _frameSize);
-
- _packetSize = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() packetSize: %d", _packetSize);
-
- if (extraData->size() - extraData->pos() != 0) {
- tmp_s = extraData->readSint32BE();
- debug(1, "QDM2Stream::QDM2Stream() extraSize3: %d", tmp_s);
- if (extraData->size() + 4 != tmp_s)
- warning("QDM2Stream::QDM2Stream() extraSize3 mismatch - Expected %d", extraData->size() + 4);
-
- tmp = extraData->readUint32BE();
- debug(1, "QDM2Stream::QDM2Stream() extraTag3: %d", tmp);
- if (tmp != MKID_BE('QDCP'))
- warning("QDM2Stream::QDM2Stream() extraTag3 mismatch");
-
- if ((float)extraData->readUint32BE() != 1.0)
- warning("QDM2Stream::QDM2Stream() uf0 field not 1.0");
-
- if (extraData->readUint32BE() != 0)
- warning("QDM2Stream::QDM2Stream() u1 field not 0");
-
- if ((float)extraData->readUint32BE() != 1.0)
- warning("QDM2Stream::QDM2Stream() uf1 field not 1.0");
-
- if ((float)extraData->readUint32BE() != 1.0)
- warning("QDM2Stream::QDM2Stream() uf2 field not 1.0");
-
- if (extraData->readUint32BE() != 27)
- warning("QDM2Stream::QDM2Stream() u2 field not 27");
-
- if (extraData->readUint32BE() != 8)
- warning("QDM2Stream::QDM2Stream() u3 field not 8");
-
- if (extraData->readUint32BE() != 0)
- warning("QDM2Stream::QDM2Stream() u4 field not 0");
- }
-
- _fftOrder = scummvm_log2(_frameSize) + 1;
- _fftFrameSize = 2 * _frameSize; // complex has two floats
-
- // something like max decodable tones
- _groupOrder = scummvm_log2(_blockSize) + 1;
- _sFrameSize = _blockSize / 16; // 16 iterations per super block
-
- _subSampling = _fftOrder - 7;
- _frequencyRange = 255 / (1 << (2 - _subSampling));
-
- switch ((_subSampling * 2 + _channels - 1)) {
- case 0:
- tmp = 40;
- break;
- case 1:
- tmp = 48;
- break;
- case 2:
- tmp = 56;
- break;
- case 3:
- tmp = 72;
- break;
- case 4:
- tmp = 80;
- break;
- case 5:
- tmp = 100;
- break;
- default:
- tmp = _subSampling;
- break;
- }
-
- tmp_val = 0;
- if ((tmp * 1000) < _bitRate) tmp_val = 1;
- if ((tmp * 1440) < _bitRate) tmp_val = 2;
- if ((tmp * 1760) < _bitRate) tmp_val = 3;
- if ((tmp * 2240) < _bitRate) tmp_val = 4;
- _cmTableSelect = tmp_val;
-
- if (_subSampling == 0)
- tmp = 7999;
- else
- tmp = ((-(_subSampling -1)) & 8000) + 20000;
-
- if (tmp < 8000)
- _coeffPerSbSelect = 0;
- else if (tmp <= 16000)
- _coeffPerSbSelect = 1;
- else
- _coeffPerSbSelect = 2;
-
- if (_fftOrder < 7 || _fftOrder > 9)
- error("QDM2Stream::QDM2Stream() Unsupported fft_order: %d", _fftOrder);
-
- rdftInit(&_rdftCtx, _fftOrder, IRDFT);
-
- initVlc();
- ff_mpa_synth_init(ff_mpa_synth_window);
- softclipTableInit();
- rndTableInit();
- initNoiseSamples();
-
- _compressedData = new uint8[_packetSize];
-}
-
-QDM2Stream::~QDM2Stream() {
- delete[] _compressedData;
- delete _stream;
-}
-
-static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) {
- int value = getVlc2(gb, vlc->table, vlc->bits, depth);
-
- // stage-2, 3 bits exponent escape sequence
- if (value-- == 0)
- value = getBits(gb, getBits (gb, 3) + 1);
-
- // stage-3, optional
- if (flag) {
- int tmp = vlc_stage3_values[value];
-
- if ((value & ~3) > 0)
- tmp += getBits(gb, (value >> 2));
- value = tmp;
- }
-
- return value;
-}
-
-static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
-{
- int value = qdm2_get_vlc(gb, vlc, 0, depth);
-
- return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
-}
-
-/**
- * QDM2 checksum
- *
- * @param data pointer to data to be checksum'ed
- * @param length data length
- * @param value checksum value
- *
- * @return 0 if checksum is OK
- */
-static uint16 qdm2_packet_checksum(const uint8 *data, int length, int value) {
- int i;
-
- for (i = 0; i < length; i++)
- value -= data[i];
-
- return (uint16)(value & 0xffff);
-}
-
-/**
- * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
- *
- * @param gb bitreader context
- * @param sub_packet packet under analysis
- */
-static void qdm2_decode_sub_packet_header(GetBitContext *gb, QDM2SubPacket *sub_packet)
-{
- sub_packet->type = getBits (gb, 8);
-
- if (sub_packet->type == 0) {
- sub_packet->size = 0;
- sub_packet->data = NULL;
- } else {
- sub_packet->size = getBits (gb, 8);
-
- if (sub_packet->type & 0x80) {
- sub_packet->size <<= 8;
- sub_packet->size |= getBits (gb, 8);
- sub_packet->type &= 0x7f;
- }
-
- if (sub_packet->type == 0x7f)
- sub_packet->type |= (getBits (gb, 8) << 8);
-
- sub_packet->data = &gb->buffer[getBitsCount(gb) / 8]; // FIXME: this depends on bitreader internal data
- }
-
- debug(1, "QDM2 Subpacket: type=%d size=%d start_offs=%x", sub_packet->type, sub_packet->size, getBitsCount(gb) / 8);
-}
-
-/**
- * Return node pointer to first packet of requested type in list.
- *
- * @param list list of subpackets to be scanned
- * @param type type of searched subpacket
- * @return node pointer for subpacket if found, else NULL
- */
-static QDM2SubPNode* qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, int type)
-{
- while (list != NULL && list->packet != NULL) {
- if (list->packet->type == type)
- return list;
- list = list->next;
- }
- return NULL;
-}
-
-/**
- * Replaces 8 elements with their average value.
- * Called by qdm2_decode_superblock before starting subblock decoding.
- */
-void QDM2Stream::average_quantized_coeffs(void) {
- int i, j, n, ch, sum;
-
- n = coeff_per_sb_for_avg[_coeffPerSbSelect][QDM2_SB_USED(_subSampling) - 1] + 1;
-
- for (ch = 0; ch < _channels; ch++) {
- for (i = 0; i < n; i++) {
- sum = 0;
-
- for (j = 0; j < 8; j++)
- sum += _quantizedCoeffs[ch][i][j];
-
- sum /= 8;
- if (sum > 0)
- sum--;
-
- for (j = 0; j < 8; j++)
- _quantizedCoeffs[ch][i][j] = sum;
- }
- }
-}
-
-/**
- * Build subband samples with noise weighted by q->tone_level.
- * Called by synthfilt_build_sb_samples.
- *
- * @param sb subband index
- */
-void QDM2Stream::build_sb_samples_from_noise(int sb) {
- int ch, j;
-
- FIX_NOISE_IDX(_noiseIdx);
-
- if (!_channels)
- return;
-
- for (ch = 0; ch < _channels; ch++) {
- for (j = 0; j < 64; j++) {
- _sbSamples[ch][j * 2][sb] = (int32)(SB_DITHERING_NOISE(sb, _noiseIdx) * _toneLevel[ch][sb][j] + .5);
- _sbSamples[ch][j * 2 + 1][sb] = (int32)(SB_DITHERING_NOISE(sb, _noiseIdx) * _toneLevel[ch][sb][j] + .5);
- }
- }
-}
-
-/**
- * Called while processing data from subpackets 11 and 12.
- * Used after making changes to coding_method array.
- *
- * @param sb subband index
- * @param channels number of channels
- * @param coding_method q->coding_method[0][0][0]
- */
-void QDM2Stream::fix_coding_method_array(int sb, int channels, sb_int8_array coding_method)
-{
- int j, k;
- int ch;
- int run, case_val;
- int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
-
- for (ch = 0; ch < channels; ch++) {
- for (j = 0; j < 64; ) {
- if((coding_method[ch][sb][j] - 8) > 22) {
- run = 1;
- case_val = 8;
- } else {
- switch (switchtable[coding_method[ch][sb][j]-8]) {
- case 0: run = 10; case_val = 10; break;
- case 1: run = 1; case_val = 16; break;
- case 2: run = 5; case_val = 24; break;
- case 3: run = 3; case_val = 30; break;
- case 4: run = 1; case_val = 30; break;
- case 5: run = 1; case_val = 8; break;
- default: run = 1; case_val = 8; break;
- }
- }
- for (k = 0; k < run; k++)
- if (j + k < 128)
- if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
- if (k > 0) {
- warning("QDM2 Untested Code: not debugged, almost never used");
- memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8));
- memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8));
- }
- j += run;
- }
- }
-}
-
-/**
- * Related to synthesis filter
- * Called by process_subpacket_10
- *
- * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
- */
-void QDM2Stream::fill_tone_level_array(int flag) {
- int i, sb, ch, sb_used;
- int tmp, tab;
-
- // This should never happen
- if (_channels <= 0)
- return;
-
- for (ch = 0; ch < _channels; ch++) {
- for (sb = 0; sb < 30; sb++) {
- for (i = 0; i < 8; i++) {
- if ((tab=coeff_per_sb_for_dequant[_coeffPerSbSelect][sb]) < (last_coeff[_coeffPerSbSelect] - 1))
- tmp = _quantizedCoeffs[ch][tab + 1][i] * dequant_table[_coeffPerSbSelect][tab + 1][sb]+
- _quantizedCoeffs[ch][tab][i] * dequant_table[_coeffPerSbSelect][tab][sb];
- else
- tmp = _quantizedCoeffs[ch][tab][i] * dequant_table[_coeffPerSbSelect][tab][sb];
- if(tmp < 0)
- tmp += 0xff;
- _toneLevelIdxBase[ch][sb][i] = (tmp / 256) & 0xff;
- }
- }
- }
-
- sb_used = QDM2_SB_USED(_subSampling);
-
- if ((_superblocktype_2_3 != 0) && !flag) {
- for (sb = 0; sb < sb_used; sb++) {
- for (ch = 0; ch < _channels; ch++) {
- for (i = 0; i < 64; i++) {
- _toneLevelIdx[ch][sb][i] = _toneLevelIdxBase[ch][sb][i / 8];
- if (_toneLevelIdx[ch][sb][i] < 0)
- _toneLevel[ch][sb][i] = 0;
- else
- _toneLevel[ch][sb][i] = fft_tone_level_table[0][_toneLevelIdx[ch][sb][i] & 0x3f];
- }
- }
- }
- } else {
- tab = _superblocktype_2_3 ? 0 : 1;
- for (sb = 0; sb < sb_used; sb++) {
- if ((sb >= 4) && (sb <= 23)) {
- for (ch = 0; ch < _channels; ch++) {
- for (i = 0; i < 64; i++) {
- tmp = _toneLevelIdxBase[ch][sb][i / 8] -
- _toneLevelIdxHi1[ch][sb / 8][i / 8][i % 8] -
- _toneLevelIdxMid[ch][sb - 4][i / 8] -
- _toneLevelIdxHi2[ch][sb - 4];
- _toneLevelIdx[ch][sb][i] = tmp & 0xff;
- if ((tmp < 0) || (!_superblocktype_2_3 && !tmp))
- _toneLevel[ch][sb][i] = 0;
- else
- _toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
- }
- }
- } else {
- if (sb > 4) {
- for (ch = 0; ch < _channels; ch++) {
- for (i = 0; i < 64; i++) {
- tmp = _toneLevelIdxBase[ch][sb][i / 8] -
- _toneLevelIdxHi1[ch][2][i / 8][i % 8] -
- _toneLevelIdxHi2[ch][sb - 4];
- _toneLevelIdx[ch][sb][i] = tmp & 0xff;
- if ((tmp < 0) || (!_superblocktype_2_3 && !tmp))
- _toneLevel[ch][sb][i] = 0;
- else
- _toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
- }
- }
- } else {
- for (ch = 0; ch < _channels; ch++) {
- for (i = 0; i < 64; i++) {
- tmp = _toneLevelIdx[ch][sb][i] = _toneLevelIdxBase[ch][sb][i / 8];
- if ((tmp < 0) || (!_superblocktype_2_3 && !tmp))
- _toneLevel[ch][sb][i] = 0;
- else
- _toneLevel[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
- }
- }
- }
- }
- }
- }
-}
-
-/**
- * Related to synthesis filter
- * Called by process_subpacket_11
- * c is built with data from subpacket 11
- * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
- *
- * @param tone_level_idx
- * @param tone_level_idx_temp
- * @param coding_method q->coding_method[0][0][0]
- * @param nb_channels number of channels
- * @param c coming from subpacket 11, passed as 8*c
- * @param superblocktype_2_3 flag based on superblock packet type
- * @param cm_table_select q->cm_table_select
- */
-void QDM2Stream::fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
- sb_int8_array coding_method, int nb_channels,
- int c, int superblocktype_2_3, int cm_table_select) {
- int ch, sb, j;
- int tmp, acc, esp_40, comp;
- int add1, add2, add3, add4;
- // TODO : Remove multres 64 bit variable necessity...
- int64_t multres;
-
- // This should never happen
- if (nb_channels <= 0)
- return;
- if (!superblocktype_2_3) {
- warning("QDM2 This case is untested, no samples available");
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++) {
- for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
- add1 = tone_level_idx[ch][sb][j] - 10;
- if (add1 < 0)
- add1 = 0;
- add2 = add3 = add4 = 0;
- if (sb > 1) {
- add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
- if (add2 < 0)
- add2 = 0;
- }
- if (sb > 0) {
- add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
- if (add3 < 0)
- add3 = 0;
- }
- if (sb < 29) {
- add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
- if (add4 < 0)
- add4 = 0;
- }
- tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
- if (tmp < 0)
- tmp = 0;
- tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
- }
- tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
- }
- acc = 0;
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- acc += tone_level_idx_temp[ch][sb][j];
-
- multres = 0x66666667 * (acc * 10);
- esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++) {
- comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
- if (comp < 0)
- comp += 0xff;
- comp /= 256; // signed shift
- switch(sb) {
- case 0:
- if (comp < 30)
- comp = 30;
- comp += 15;
- break;
- case 1:
- if (comp < 24)
- comp = 24;
- comp += 10;
- break;
- case 2:
- case 3:
- case 4:
- if (comp < 16)
- comp = 16;
- }
- if (comp <= 5)
- tmp = 0;
- else if (comp <= 10)
- tmp = 10;
- else if (comp <= 16)
- tmp = 16;
- else if (comp <= 24)
- tmp = -1;
- else
- tmp = 0;
- coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
- }
- for (sb = 0; sb < 30; sb++)
- fix_coding_method_array(sb, nb_channels, coding_method);
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- if (sb >= 10) {
- if (coding_method[ch][sb][j] < 10)
- coding_method[ch][sb][j] = 10;
- } else {
- if (sb >= 2) {
- if (coding_method[ch][sb][j] < 16)
- coding_method[ch][sb][j] = 16;
- } else {
- if (coding_method[ch][sb][j] < 30)
- coding_method[ch][sb][j] = 30;
- }
- }
- } else { // superblocktype_2_3 != 0
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
- }
-}
-
-/**
- *
- * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
- * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
- *
- * @param gb bitreader context
- * @param length packet length in bits
- * @param sb_min lower subband processed (sb_min included)
- * @param sb_max higher subband processed (sb_max excluded)
- */
-void QDM2Stream::synthfilt_build_sb_samples(GetBitContext *gb, int length, int sb_min, int sb_max) {
- int sb, j, k, n, ch, run, channels;
- int joined_stereo, zero_encoding, chs;
- int type34_first;
- float type34_div = 0;
- float type34_predictor;
- float samples[10], sign_bits[16];
-
- if (length == 0) {
- // If no data use noise
- for (sb = sb_min; sb < sb_max; sb++)
- build_sb_samples_from_noise(sb);
-
- return;
- }
-
- for (sb = sb_min; sb < sb_max; sb++) {
- FIX_NOISE_IDX(_noiseIdx);
-
- channels = _channels;
-
- if (_channels <= 1 || sb < 12)
- joined_stereo = 0;
- else if (sb >= 24)
- joined_stereo = 1;
- else
- joined_stereo = (BITS_LEFT(length,gb) >= 1) ? getBits1 (gb) : 0;
-
- if (joined_stereo) {
- if (BITS_LEFT(length,gb) >= 16)
- for (j = 0; j < 16; j++)
- sign_bits[j] = getBits1(gb);
-
- for (j = 0; j < 64; j++)
- if (_codingMethod[1][sb][j] > _codingMethod[0][sb][j])
- _codingMethod[0][sb][j] = _codingMethod[1][sb][j];
-
- fix_coding_method_array(sb, _channels, _codingMethod);
- channels = 1;
- }
-
- for (ch = 0; ch < channels; ch++) {
- zero_encoding = (BITS_LEFT(length,gb) >= 1) ? getBits1(gb) : 0;
- type34_predictor = 0.0;
- type34_first = 1;
-
- for (j = 0; j < 128; ) {
- switch (_codingMethod[ch][sb][j / 2]) {
- case 8:
- if (BITS_LEFT(length,gb) >= 10) {
- if (zero_encoding) {
- for (k = 0; k < 5; k++) {
- if ((j + 2 * k) >= 128)
- break;
- samples[2 * k] = getBits1(gb) ? dequant_1bit[joined_stereo][2 * getBits1(gb)] : 0;
- }
- } else {
- n = getBits(gb, 8);
- for (k = 0; k < 5; k++)
- samples[2 * k] = dequant_1bit[joined_stereo][_randomDequantIndex[n][k]];
- }
- for (k = 0; k < 5; k++)
- samples[2 * k + 1] = SB_DITHERING_NOISE(sb, _noiseIdx);
- } else {
- for (k = 0; k < 10; k++)
- samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx);
- }
- run = 10;
- break;
-
- case 10:
- if (BITS_LEFT(length,gb) >= 1) {
- double f = 0.81;
-
- if (getBits1(gb))
- f = -f;
- f -= _noiseSamples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
- samples[0] = f;
- } else {
- samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx);
- }
- run = 1;
- break;
-
- case 16:
- if (BITS_LEFT(length,gb) >= 10) {
- if (zero_encoding) {
- for (k = 0; k < 5; k++) {
- if ((j + k) >= 128)
- break;
- samples[k] = (getBits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * getBits1(gb)];
- }
- } else {
- n = getBits (gb, 8);
- for (k = 0; k < 5; k++)
- samples[k] = dequant_1bit[joined_stereo][_randomDequantIndex[n][k]];
- }
- } else {
- for (k = 0; k < 5; k++)
- samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx);
- }
- run = 5;
- break;
-
- case 24:
- if (BITS_LEFT(length,gb) >= 7) {
- n = getBits(gb, 7);
- for (k = 0; k < 3; k++)
- samples[k] = (_randomDequantType24[n][k] - 2.0) * 0.5;
- } else {
- for (k = 0; k < 3; k++)
- samples[k] = SB_DITHERING_NOISE(sb, _noiseIdx);
- }
- run = 3;
- break;
-
- case 30:
- if (BITS_LEFT(length,gb) >= 4)
- samples[0] = type30_dequant[qdm2_get_vlc(gb, &_vlcTabType30, 0, 1)];
- else
- samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx);
-
- run = 1;
- break;
-
- case 34:
- if (BITS_LEFT(length,gb) >= 7) {
- if (type34_first) {
- type34_div = (float)(1 << getBits(gb, 2));
- samples[0] = ((float)getBits(gb, 5) - 16.0) / 15.0;
- type34_predictor = samples[0];
- type34_first = 0;
- } else {
- samples[0] = type34_delta[qdm2_get_vlc(gb, &_vlcTabType34, 0, 1)] / type34_div + type34_predictor;
- type34_predictor = samples[0];
- }
- } else {
- samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx);
- }
- run = 1;
- break;
-
- default:
- samples[0] = SB_DITHERING_NOISE(sb, _noiseIdx);
- run = 1;
- break;
- }
-
- if (joined_stereo) {
- float tmp[10][MPA_MAX_CHANNELS];
-
- for (k = 0; k < run; k++) {
- tmp[k][0] = samples[k];
- tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
- }
- for (chs = 0; chs < _channels; chs++)
- for (k = 0; k < run; k++)
- if ((j + k) < 128)
- _sbSamples[chs][j + k][sb] = (int32)(_toneLevel[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
- } else {
- for (k = 0; k < run; k++)
- if ((j + k) < 128)
- _sbSamples[ch][j + k][sb] = (int32)(_toneLevel[ch][sb][(j + k)/2] * samples[k] + .5);
- }
-
- j += run;
- } // j loop
- } // channel loop
- } // subband loop
-}
-
-/**
- * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
- * This is similar to process_subpacket_9, but for a single channel and for element [0]
- * same VLC tables as process_subpacket_9 are used.
- *
- * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
- * @param gb bitreader context
- * @param length packet length in bits
- */
-void QDM2Stream::init_quantized_coeffs_elem0(int8 *quantized_coeffs, GetBitContext *gb, int length) {
- int i, k, run, level, diff;
-
- if (BITS_LEFT(length,gb) < 16)
- return;
- level = qdm2_get_vlc(gb, &_vlcTabLevel, 0, 2);
-
- quantized_coeffs[0] = level;
-
- for (i = 0; i < 7; ) {
- if (BITS_LEFT(length,gb) < 16)
- break;
- run = qdm2_get_vlc(gb, &_vlcTabRun, 0, 1) + 1;
-
- if (BITS_LEFT(length,gb) < 16)
- break;
- diff = qdm2_get_se_vlc(&_vlcTabDiff, gb, 2);
-
- for (k = 1; k <= run; k++)
- quantized_coeffs[i + k] = (level + ((k * diff) / run));
-
- level += diff;
- i += run;
- }
-}
-
-/**
- * Related to synthesis filter, process data from packet 10
- * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
- * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
- *
- * @param gb bitreader context
- * @param length packet length in bits
- */
-void QDM2Stream::init_tone_level_dequantization(GetBitContext *gb, int length) {
- int sb, j, k, n, ch;
-
- for (ch = 0; ch < _channels; ch++) {
- init_quantized_coeffs_elem0(_quantizedCoeffs[ch][0], gb, length);
-
- if (BITS_LEFT(length,gb) < 16) {
- memset(_quantizedCoeffs[ch][0], 0, 8);
- break;
- }
- }
-
- n = _subSampling + 1;
-
- for (sb = 0; sb < n; sb++)
- for (ch = 0; ch < _channels; ch++)
- for (j = 0; j < 8; j++) {
- if (BITS_LEFT(length,gb) < 1)
- break;
- if (getBits1(gb)) {
- for (k=0; k < 8; k++) {
- if (BITS_LEFT(length,gb) < 16)
- break;
- _toneLevelIdxHi1[ch][sb][j][k] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxHi1, 0, 2);
- }
- } else {
- for (k=0; k < 8; k++)
- _toneLevelIdxHi1[ch][sb][j][k] = 0;
- }
- }
-
- n = QDM2_SB_USED(_subSampling) - 4;
-
- for (sb = 0; sb < n; sb++)
- for (ch = 0; ch < _channels; ch++) {
- if (BITS_LEFT(length,gb) < 16)
- break;
- _toneLevelIdxHi2[ch][sb] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxHi2, 0, 2);
- if (sb > 19)
- _toneLevelIdxHi2[ch][sb] -= 16;
- else
- for (j = 0; j < 8; j++)
- _toneLevelIdxMid[ch][sb][j] = -16;
- }
-
- n = QDM2_SB_USED(_subSampling) - 5;
-
- for (sb = 0; sb < n; sb++) {
- for (ch = 0; ch < _channels; ch++) {
- for (j = 0; j < 8; j++) {
- if (BITS_LEFT(length,gb) < 16)
- break;
- _toneLevelIdxMid[ch][sb][j] = qdm2_get_vlc(gb, &_vlcTabToneLevelIdxMid, 0, 2) - 32;
- }
- }
- }
-}
-
-/**
- * Process subpacket 9, init quantized_coeffs with data from it
- *
- * @param node pointer to node with packet
- */
-void QDM2Stream::process_subpacket_9(QDM2SubPNode *node) {
- GetBitContext gb;
- int i, j, k, n, ch, run, level, diff;
-
- initGetBits(&gb, node->packet->data, node->packet->size*8);
-
- n = coeff_per_sb_for_avg[_coeffPerSbSelect][QDM2_SB_USED(_subSampling) - 1] + 1; // same as averagesomething function
-
- for (i = 1; i < n; i++)
- for (ch = 0; ch < _channels; ch++) {
- level = qdm2_get_vlc(&gb, &_vlcTabLevel, 0, 2);
- _quantizedCoeffs[ch][i][0] = level;
-
- for (j = 0; j < (8 - 1); ) {
- run = qdm2_get_vlc(&gb, &_vlcTabRun, 0, 1) + 1;
- diff = qdm2_get_se_vlc(&_vlcTabDiff, &gb, 2);
-
- for (k = 1; k <= run; k++)
- _quantizedCoeffs[ch][i][j + k] = (level + ((k*diff) / run));
-
- level += diff;
- j += run;
- }
- }
-
- for (ch = 0; ch < _channels; ch++)
- for (i = 0; i < 8; i++)
- _quantizedCoeffs[ch][0][i] = 0;
-}
-
-/**
- * Process subpacket 10 if not null, else
- *
- * @param node pointer to node with packet
- * @param length packet length in bits
- */
-void QDM2Stream::process_subpacket_10(QDM2SubPNode *node, int length) {
- GetBitContext gb;
-
- initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
-
- if (length != 0) {
- init_tone_level_dequantization(&gb, length);
- fill_tone_level_array(1);
- } else {
- fill_tone_level_array(0);
- }
-}
-
-/**
- * Process subpacket 11
- *
- * @param node pointer to node with packet
- * @param length packet length in bit
- */
-void QDM2Stream::process_subpacket_11(QDM2SubPNode *node, int length) {
- GetBitContext gb;
-
- initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
- if (length >= 32) {
- int c = getBits (&gb, 13);
-
- if (c > 3)
- fill_coding_method_array(_toneLevelIdx, _toneLevelIdxTemp, _codingMethod,
- _channels, 8*c, _superblocktype_2_3, _cmTableSelect);
- }
-
- synthfilt_build_sb_samples(&gb, length, 0, 8);
-}
-
-/**
- * Process subpacket 12
- *
- * @param node pointer to node with packet
- * @param length packet length in bits
- */
-void QDM2Stream::process_subpacket_12(QDM2SubPNode *node, int length) {
- GetBitContext gb;
-
- initGetBits(&gb, ((node == NULL) ? _emptyBuffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
- synthfilt_build_sb_samples(&gb, length, 8, QDM2_SB_USED(_subSampling));
-}
-
-/*
- * Process new subpackets for synthesis filter
- *
- * @param list list with synthesis filter packets (list D)
- */
-void QDM2Stream::process_synthesis_subpackets(QDM2SubPNode *list) {
- struct QDM2SubPNode *nodes[4];
-
- nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
- if (nodes[0] != NULL)
- process_subpacket_9(nodes[0]);
-
- nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
- if (nodes[1] != NULL)
- process_subpacket_10(nodes[1], nodes[1]->packet->size << 3);
- else
- process_subpacket_10(NULL, 0);
-
- nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
- if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
- process_subpacket_11(nodes[2], (nodes[2]->packet->size << 3));
- else
- process_subpacket_11(NULL, 0);
-
- nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
- if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
- process_subpacket_12(nodes[3], (nodes[3]->packet->size << 3));
- else
- process_subpacket_12(NULL, 0);
-}
-
-/*
- * Decode superblock, fill packet lists.
- *
- */
-void QDM2Stream::qdm2_decode_super_block(void) {
- GetBitContext gb;
- struct QDM2SubPacket header, *packet;
- int i, packet_bytes, sub_packet_size, subPacketsD;
- unsigned int next_index = 0;
-
- memset(_toneLevelIdxHi1, 0, sizeof(_toneLevelIdxHi1));
- memset(_toneLevelIdxMid, 0, sizeof(_toneLevelIdxMid));
- memset(_toneLevelIdxHi2, 0, sizeof(_toneLevelIdxHi2));
-
- _subPacketsB = 0;
- subPacketsD = 0;
-
- average_quantized_coeffs(); // average elements in quantized_coeffs[max_ch][10][8]
-
- initGetBits(&gb, _compressedData, _packetSize*8);
- qdm2_decode_sub_packet_header(&gb, &header);
-
- if (header.type < 2 || header.type >= 8) {
- _hasErrors = true;
- error("QDM2 : bad superblock type");
- return;
- }
-
- _superblocktype_2_3 = (header.type == 2 || header.type == 3);
- packet_bytes = (_packetSize - getBitsCount(&gb) / 8);
-
- initGetBits(&gb, header.data, header.size*8);
-
- if (header.type == 2 || header.type == 4 || header.type == 5) {
- int csum = 257 * getBits(&gb, 8) + 2 * getBits(&gb, 8);
-
- csum = qdm2_packet_checksum(_compressedData, _packetSize, csum);
-
- if (csum != 0) {
- _hasErrors = true;
- error("QDM2 : bad packet checksum");
- return;
- }
- }
-
- _subPacketListB[0].packet = NULL;
- _subPacketListD[0].packet = NULL;
-
- for (i = 0; i < 6; i++)
- if (--_fftLevelExp[i] < 0)
- _fftLevelExp[i] = 0;
-
- for (i = 0; packet_bytes > 0; i++) {
- int j;
-
- _subPacketListA[i].next = NULL;
-
- if (i > 0) {
- _subPacketListA[i - 1].next = &_subPacketListA[i];
-
- // seek to next block
- initGetBits(&gb, header.data, header.size*8);
- skipBits(&gb, next_index*8);
-
- if (next_index >= header.size)
- break;
- }
-
- // decode subpacket
- packet = &_subPackets[i];
- qdm2_decode_sub_packet_header(&gb, packet);
- next_index = packet->size + getBitsCount(&gb) / 8;
- sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
-
- if (packet->type == 0)
- break;
-
- if (sub_packet_size > packet_bytes) {
- if (packet->type != 10 && packet->type != 11 && packet->type != 12)
- break;
- packet->size += packet_bytes - sub_packet_size;
- }
-
- packet_bytes -= sub_packet_size;
-
- // add subpacket to 'all subpackets' list
- _subPacketListA[i].packet = packet;
-
- // add subpacket to related list
- if (packet->type == 8) {
- error("Unsupported packet type 8");
- return;
- } else if (packet->type >= 9 && packet->type <= 12) {
- // packets for MPEG Audio like Synthesis Filter
- QDM2_LIST_ADD(_subPacketListD, subPacketsD, packet);
- } else if (packet->type == 13) {
- for (j = 0; j < 6; j++)
- _fftLevelExp[j] = getBits(&gb, 6);
- } else if (packet->type == 14) {
- for (j = 0; j < 6; j++)
- _fftLevelExp[j] = qdm2_get_vlc(&gb, &_fftLevelExpVlc, 0, 2);
- } else if (packet->type == 15) {
- error("Unsupported packet type 15");
- return;
- } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
- // packets for FFT
- QDM2_LIST_ADD(_subPacketListB, _subPacketsB, packet);
- }
- } // Packet bytes loop
-
-// ****************************************************************
- if (_subPacketListD[0].packet != NULL) {
- process_synthesis_subpackets(_subPacketListD);
- _doSynthFilter = 1;
- } else if (_doSynthFilter) {
- process_subpacket_10(NULL, 0);
- process_subpacket_11(NULL, 0);
- process_subpacket_12(NULL, 0);
- }
-// ****************************************************************
-}
-
-void QDM2Stream::qdm2_fft_init_coefficient(int sub_packet, int offset, int duration,
- int channel, int exp, int phase) {
- if (_fftCoefsMinIndex[duration] < 0)
- _fftCoefsMinIndex[duration] = _fftCoefsIndex;
-
- _fftCoefs[_fftCoefsIndex].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
- _fftCoefs[_fftCoefsIndex].channel = channel;
- _fftCoefs[_fftCoefsIndex].offset = offset;
- _fftCoefs[_fftCoefsIndex].exp = exp;
- _fftCoefs[_fftCoefsIndex].phase = phase;
- _fftCoefsIndex++;
-}
-
-void QDM2Stream::qdm2_fft_decode_tones(int duration, GetBitContext *gb, int b) {
- debug(1, "QDM2Stream::qdm2_fft_decode_tones() duration: %d b:%d", duration, b);
- int channel, stereo, phase, exp;
- int local_int_4, local_int_8, stereo_phase, local_int_10;
- int local_int_14, stereo_exp, local_int_20, local_int_28;
- int n, offset;
-
- local_int_4 = 0;
- local_int_28 = 0;
- local_int_20 = 2;
- local_int_8 = (4 - duration);
- local_int_10 = 1 << (_groupOrder - duration - 1);
- offset = 1;
-
- while (1) {
- if (_superblocktype_2_3) {
- debug(1, "QDM2Stream::qdm2_fft_decode_tones() local_int_8: %d", local_int_8);
- while ((n = qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2)) < 2) {
- debug(1, "QDM2Stream::qdm2_fft_decode_tones() local_int_8: %d", local_int_8);
- offset = 1;
- if (n == 0) {
- local_int_4 += local_int_10;
- local_int_28 += (1 << local_int_8);
- } else {
- local_int_4 += 8*local_int_10;
- local_int_28 += (8 << local_int_8);
- }
- }
- offset += (n - 2);
- } else {
- offset += qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2);
- while (offset >= (local_int_10 - 1)) {
- offset += (1 - (local_int_10 - 1));
- local_int_4 += local_int_10;
- local_int_28 += (1 << local_int_8);
- }
- }
-
- if (local_int_4 >= _blockSize)
- return;
-
- local_int_14 = (offset >> local_int_8);
-
- if (_channels > 1) {
- channel = getBits1(gb);
- stereo = getBits1(gb);
- } else {
- channel = 0;
- stereo = 0;
- }
-
- exp = qdm2_get_vlc(gb, (b ? &_fftLevelExpVlc : &_fftLevelExpAltVlc), 0, 2);
- exp += _fftLevelExp[fft_level_index_table[local_int_14]];
- exp = (exp < 0) ? 0 : exp;
-
- phase = getBits(gb, 3);
- stereo_exp = 0;
- stereo_phase = 0;
-
- if (stereo) {
- stereo_exp = (exp - qdm2_get_vlc(gb, &_fftStereoExpVlc, 0, 1));
- stereo_phase = (phase - qdm2_get_vlc(gb, &_fftStereoPhaseVlc, 0, 1));
- if (stereo_phase < 0)
- stereo_phase += 8;
- }
-
- if (_frequencyRange > (local_int_14 + 1)) {
- int sub_packet = (local_int_20 + local_int_28);
-
- qdm2_fft_init_coefficient(sub_packet, offset, duration, channel, exp, phase);
- if (stereo)
- qdm2_fft_init_coefficient(sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
- }
-
- offset++;
- }
-}
-
-void QDM2Stream::qdm2_decode_fft_packets(void) {
- debug(1, "QDM2Stream::qdm2_decode_fft_packets()");
- int i, j, min, max, value, type, unknown_flag;
- GetBitContext gb;
-
- if (_subPacketListB[0].packet == NULL)
- return;
-
- // reset minimum indexes for FFT coefficients
- _fftCoefsIndex = 0;
- for (i=0; i < 5; i++)
- _fftCoefsMinIndex[i] = -1;
-
- // process subpackets ordered by type, largest type first
- for (i = 0, max = 256; i < _subPacketsB; i++) {
- QDM2SubPacket *packet= NULL;
-
- // find subpacket with largest type less than max
- for (j = 0, min = 0; j < _subPacketsB; j++) {
- value = _subPacketListB[j].packet->type;
- if (value > min && value < max) {
- min = value;
- packet = _subPacketListB[j].packet;
- }
- }
-
- max = min;
-
- // check for errors (?)
- if (!packet)
- return;
-
- if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
- return;
-
- // decode FFT tones
- debug(1, "QDM2Stream::qdm2_decode_fft_packets initGetBits() packet->size*8: %d", packet->size*8);
- initGetBits(&gb, packet->data, packet->size*8);
-
- if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
- unknown_flag = 1;
- else
- unknown_flag = 0;
-
- type = packet->type;
-
- if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
- int duration = _subSampling + 5 - (type & 15);
-
- if (duration >= 0 && duration < 4) { // TODO: Should be <= 4?
- debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #1");
- qdm2_fft_decode_tones(duration, &gb, unknown_flag);
- }
- } else if (type == 31) {
- for (j=0; j < 4; j++) {
- debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #2");
- qdm2_fft_decode_tones(j, &gb, unknown_flag);
- }
- } else if (type == 46) {
- for (j=0; j < 6; j++)
- _fftLevelExp[j] = getBits(&gb, 6);
- for (j=0; j < 4; j++) {
- debug(1, "QDM2Stream::qdm2_decode_fft_packets qdm2_fft_decode_tones() #3");
- qdm2_fft_decode_tones(j, &gb, unknown_flag);
- }
- }
- } // Loop on B packets
-
- // calculate maximum indexes for FFT coefficients
- for (i = 0, j = -1; i < 5; i++)
- if (_fftCoefsMinIndex[i] >= 0) {
- if (j >= 0)
- _fftCoefsMaxIndex[j] = _fftCoefsMinIndex[i];
- j = i;
- }
- if (j >= 0)
- _fftCoefsMaxIndex[j] = _fftCoefsIndex;
-}
-
-void QDM2Stream::qdm2_fft_generate_tone(FFTTone *tone)
-{
- float level, f[6];
- int i;
- QDM2Complex c;
- const double iscale = 2.0 * PI / 512.0;
-
- tone->phase += tone->phase_shift;
-
- // calculate current level (maximum amplitude) of tone
- level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
- c.im = level * sin(tone->phase*iscale);
- c.re = level * cos(tone->phase*iscale);
-
- // generate FFT coefficients for tone
- if (tone->duration >= 3 || tone->cutoff >= 3) {
- tone->complex[0].im += c.im;
- tone->complex[0].re += c.re;
- tone->complex[1].im -= c.im;
- tone->complex[1].re -= c.re;
- } else {
- f[1] = -tone->table[4];
- f[0] = tone->table[3] - tone->table[0];
- f[2] = 1.0 - tone->table[2] - tone->table[3];
- f[3] = tone->table[1] + tone->table[4] - 1.0;
- f[4] = tone->table[0] - tone->table[1];
- f[5] = tone->table[2];
- for (i = 0; i < 2; i++) {
- tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
- tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
- }
- for (i = 0; i < 4; i++) {
- tone->complex[i].re += c.re * f[i+2];
- tone->complex[i].im += c.im * f[i+2];
- }
- }
-
- // copy the tone if it has not yet died out
- if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
- memcpy(&_fftTones[_fftToneEnd], tone, sizeof(FFTTone));
- _fftToneEnd = (_fftToneEnd + 1) % 1000;
- }
-}
-
-void QDM2Stream::qdm2_fft_tone_synthesizer(uint8 sub_packet) {
- int i, j, ch;
- const double iscale = 0.25 * PI;
-
- for (ch = 0; ch < _channels; ch++) {
- memset(_fft.complex[ch], 0, _frameSize * sizeof(QDM2Complex));
- }
-
- // apply FFT tones with duration 4 (1 FFT period)
- if (_fftCoefsMinIndex[4] >= 0)
- for (i = _fftCoefsMinIndex[4]; i < _fftCoefsMaxIndex[4]; i++) {
- float level;
- QDM2Complex c;
-
- if (_fftCoefs[i].sub_packet != sub_packet)
- break;
-
- ch = (_channels == 1) ? 0 : _fftCoefs[i].channel;
- level = (_fftCoefs[i].exp < 0) ? 0.0 : fft_tone_level_table[_superblocktype_2_3 ? 0 : 1][_fftCoefs[i].exp & 63];
-
- c.re = level * cos(_fftCoefs[i].phase * iscale);
- c.im = level * sin(_fftCoefs[i].phase * iscale);
- _fft.complex[ch][_fftCoefs[i].offset + 0].re += c.re;
- _fft.complex[ch][_fftCoefs[i].offset + 0].im += c.im;
- _fft.complex[ch][_fftCoefs[i].offset + 1].re -= c.re;
- _fft.complex[ch][_fftCoefs[i].offset + 1].im -= c.im;
- }
-
- // generate existing FFT tones
- for (i = _fftToneEnd; i != _fftToneStart; ) {
- qdm2_fft_generate_tone(&_fftTones[_fftToneStart]);
- _fftToneStart = (_fftToneStart + 1) % 1000;
- }
-
- // create and generate new FFT tones with duration 0 (long) to 3 (short)
- for (i = 0; i < 4; i++)
- if (_fftCoefsMinIndex[i] >= 0) {
- for (j = _fftCoefsMinIndex[i]; j < _fftCoefsMaxIndex[i]; j++) {
- int offset, four_i;
- FFTTone tone;
-
- if (_fftCoefs[j].sub_packet != sub_packet)
- break;
-
- four_i = (4 - i);
- offset = _fftCoefs[j].offset >> four_i;
- ch = (_channels == 1) ? 0 : _fftCoefs[j].channel;
-
- if (offset < _frequencyRange) {
- if (offset < 2)
- tone.cutoff = offset;
- else
- tone.cutoff = (offset >= 60) ? 3 : 2;
-
- tone.level = (_fftCoefs[j].exp < 0) ? 0.0 : fft_tone_level_table[_superblocktype_2_3 ? 0 : 1][_fftCoefs[j].exp & 63];
- tone.complex = &_fft.complex[ch][offset];
- tone.table = fft_tone_sample_table[i][_fftCoefs[j].offset - (offset << four_i)];
- tone.phase = 64 * _fftCoefs[j].phase - (offset << 8) - 128;
- tone.phase_shift = (2 * _fftCoefs[j].offset + 1) << (7 - four_i);
- tone.duration = i;
- tone.time_index = 0;
-
- qdm2_fft_generate_tone(&tone);
- }
- }
- _fftCoefsMinIndex[i] = j;
- }
-}
-
-void QDM2Stream::qdm2_calculate_fft(int channel) {
- debug(1, "QDM2Stream::qdm2_calculate_fft channel: %d", channel);
- const float gain = (_channels == 1 && _channels == 2) ? 0.5f : 1.0f;
- int i;
-
- _fft.complex[channel][0].re *= 2.0f;
- _fft.complex[channel][0].im = 0.0f;
-
- //debug(1, "QDM2Stream::qdm2_calculate_fft _fft.complex[channel][0].re: %lf", _fft.complex[channel][0].re);
- //debug(1, "QDM2Stream::qdm2_calculate_fft _fft.complex[channel][0].im: %lf", _fft.complex[channel][0].im);
-
- rdftCalc(&_rdftCtx, (float *)_fft.complex[channel]);
-
- // add samples to output buffer
- for (i = 0; i < ((_fftFrameSize + 15) & ~15); i++)
- _outputBuffer[_channels * i + channel] += ((float *) _fft.complex[channel])[i] * gain;
-}
-
-/**
- * @param index subpacket number
- */
-void QDM2Stream::qdm2_synthesis_filter(uint8 index)
-{
- int16 samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
- int i, k, ch, sb_used, sub_sampling, dither_state = 0;
-
- // copy sb_samples
- sb_used = QDM2_SB_USED(_subSampling);
-
- for (ch = 0; ch < _channels; ch++)
- for (i = 0; i < 8; i++)
- for (k = sb_used; k < 32; k++)
- _sbSamples[ch][(8 * index) + i][k] = 0;
-
- for (ch = 0; ch < _channels; ch++) {
- int16 *samples_ptr = samples + ch;
-
- for (i = 0; i < 8; i++) {
- ff_mpa_synth_filter(_synthBuf[ch], &(_synthBufOffset[ch]),
- ff_mpa_synth_window, &dither_state,
- samples_ptr, _channels,
- _sbSamples[ch][(8 * index) + i]);
- samples_ptr += 32 * _channels;
- }
- }
-
- // add samples to output buffer
- sub_sampling = (4 >> _subSampling);
-
- for (ch = 0; ch < _channels; ch++)
- for (i = 0; i < _sFrameSize; i++)
- _outputBuffer[_channels * i + ch] += (float)(samples[_channels * sub_sampling * i + ch] >> (sizeof(int16)*8-16));
-}
-
-int QDM2Stream::qdm2_decodeFrame(Common::SeekableReadStream *in) {
- debug(1, "QDM2Stream::qdm2_decodeFrame in->pos(): %d in->size(): %d", in->pos(), in->size());
- int ch, i;
- const int frame_size = (_sFrameSize * _channels);
-
- // select input buffer
- if(in->eos() || in->size() == in->pos()) {
- debug(1, "QDM2Stream::qdm2_decodeFrame End of Input Stream");
- return 0;
- }
- if((in->size() - in->pos()) < _packetSize) {
- debug(1, "QDM2Stream::qdm2_decodeFrame Insufficient Packet Data in Input Stream Found: %d Need: %d", in->size() - in->pos(), _packetSize);
- return 0;
- }
-
- in->read(_compressedData, _packetSize);
- debug(1, "QDM2Stream::qdm2_decodeFrame constructed input data");
-
- // copy old block, clear new block of output samples
- memmove(_outputBuffer, &_outputBuffer[frame_size], frame_size * sizeof(float));
- memset(&_outputBuffer[frame_size], 0, frame_size * sizeof(float));
- debug(1, "QDM2Stream::qdm2_decodeFrame cleared outputBuffer");
-
- // decode block of QDM2 compressed data
- debug(1, "QDM2Stream::qdm2_decodeFrame decode block of QDM2 compressed data");
- if (_subPacket == 0) {
- _hasErrors = false; // reset it for a new super block
- debug(1, "QDM2 : Superblock follows");
- qdm2_decode_super_block();
- }
-
- // parse subpackets
- debug(1, "QDM2Stream::qdm2_decodeFrame parse subpackets");
- if (!_hasErrors) {
- if (_subPacket == 2) {
- debug(1, "QDM2Stream::qdm2_decodeFrame qdm2_decode_fft_packets()");
- qdm2_decode_fft_packets();
- }
-
- debug(1, "QDM2Stream::qdm2_decodeFrame qdm2_fft_tone_synthesizer(%d)", _subPacket);
- qdm2_fft_tone_synthesizer(_subPacket);
- }
-
- // sound synthesis stage 1 (FFT)
- debug(1, "QDM2Stream::qdm2_decodeFrame sound synthesis stage 1 (FFT)");
- for (ch = 0; ch < _channels; ch++) {
- qdm2_calculate_fft(ch);
-
- if (!_hasErrors && _subPacketListC[0].packet != NULL) {
- error("QDM2 : has errors, and C list is not empty");
- return 0;
- }
- }
-
- // sound synthesis stage 2 (MPEG audio like synthesis filter)
- debug(1, "QDM2Stream::qdm2_decodeFrame sound synthesis stage 2 (MPEG audio like synthesis filter)");
- if (!_hasErrors && _doSynthFilter)
- qdm2_synthesis_filter(_subPacket);
-
- _subPacket = (_subPacket + 1) % 16;
-
- if(_hasErrors)
- warning("QDM2 Packet error...");
-
- // clip and convert output float[] to 16bit signed samples
- debug(1, "QDM2Stream::qdm2_decodeFrame clip and convert output float[] to 16bit signed samples");
-
-/*
- debugN(1, "Input Data Packet:");
- for(i = 0; i < _packetSize; i++) {
- debugN(1, " %d", _compressedData[i]);
- }
- debugN(1, " Output Data Packet:");
- for(i = 0; i < frame_size; i++) {
- debugN(1, " %d", (int)_outputBuffer[i]);
- }
- debug(1, "");
-*/
-
- for (i = 0; i < frame_size; i++) {
- //debug(1, "QDM2Stream::qdm2_decodeFrame i: %d", i);
- int value = (int)_outputBuffer[i];
-
- if (value > SOFTCLIP_THRESHOLD)
- value = (value > HARDCLIP_THRESHOLD) ? 32767 : _softclipTable[ value - SOFTCLIP_THRESHOLD];
- else if (value < -SOFTCLIP_THRESHOLD)
- value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -_softclipTable[-value - SOFTCLIP_THRESHOLD];
-
- _outputSamples.push_back(value);
- }
- return frame_size;
-}
-
-int QDM2Stream::readBuffer(int16 *buffer, const int numSamples) {
- debug(1, "QDM2Stream::readBuffer numSamples: %d", numSamples);
- int32 decodedSamples = _outputSamples.size();
- int32 i;
-
- //while((int)_outputSamples.size() < numSamples) {
- while(!_stream->eos() && _stream->pos() != _stream->size()) {
- i = qdm2_decodeFrame(_stream);
- if(i == 0)
- break; // Out Of Decode Frames...
- decodedSamples += i;
- }
- if(decodedSamples > numSamples)
- decodedSamples = numSamples;
-
- for(i = 0; i < decodedSamples; i++)
- buffer[i] = _outputSamples.remove_at(0);
-
- return decodedSamples;
-}
-
-AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) {
- return new QDM2Stream(stream, extraData);
-}
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/decoders/qdm2.h b/sound/decoders/qdm2.h
deleted file mode 100644
index 842ede3de0..0000000000
--- a/sound/decoders/qdm2.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/* ScummVM - Graphic Adventure Engine
- *
- * ScummVM is the legal property of its developers, whose names
- * are too numerous to list here. Please refer to the COPYRIGHT
- * file distributed with this source distribution.
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
-
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
-
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
- *
- * $URL$
- * $Id$
- *
- */
-
-// Only compile if Mohawk is enabled or if we're building dynamic modules
-#if defined(ENABLE_MOHAWK) || defined(DYNAMIC_MODULES)
-
-#ifndef SOUND_QDM2_H
-#define SOUND_QDM2_H
-
-namespace Common {
- class SeekableReadStream;
-}
-
-namespace Audio {
- class AudioStream;
-
-/**
- * Create a new AudioStream from the QDM2 data in the given stream.
- *
- * @param stream the SeekableReadStream from which to read the FLAC data
- * @param extraData the QuickTime extra data stream
- * @return a new AudioStream, or NULL, if an error occured
- */
-AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData);
-
-} // End of namespace Audio
-
-#endif // SOUND_QDM2_H
-#endif // Mohawk/Plugins guard
diff --git a/sound/decoders/qdm2data.h b/sound/decoders/qdm2data.h
deleted file mode 100644
index 4c13328dd6..0000000000
--- a/sound/decoders/qdm2data.h
+++ /dev/null
@@ -1,531 +0,0 @@
-/* ScummVM - Graphic Adventure Engine
- *
- * ScummVM is the legal property of its developers, whose names
- * are too numerous to list here. Please refer to the COPYRIGHT
- * file distributed with this source distribution.
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * as published by the Free Software Foundation; either version 2
- * of the License, or (at your option) any later version.
-
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
-
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
- *
- * $URL$
- * $Id$
- *
- */
-
-#ifndef SOUND_QDM2DATA_H
-#define SOUND_QDM2DATA_H
-
-#include "common/scummsys.h"
-
-namespace Audio {
-
-/// VLC TABLES
-
-// values in this table range from -1..23; adjust retrieved value by -1
-static const uint16 vlc_tab_level_huffcodes[24] = {
- 0x037c, 0x0004, 0x003c, 0x004c, 0x003a, 0x002c, 0x001c, 0x001a,
- 0x0024, 0x0014, 0x0001, 0x0002, 0x0000, 0x0003, 0x0007, 0x0005,
- 0x0006, 0x0008, 0x0009, 0x000a, 0x000c, 0x00fc, 0x007c, 0x017c
-};
-
-static const byte vlc_tab_level_huffbits[24] = {
- 10, 6, 7, 7, 6, 6, 6, 6, 6, 5, 4, 4, 4, 3, 3, 3, 3, 4, 4, 5, 7, 8, 9, 10
-};
-
-// values in this table range from -1..36; adjust retrieved value by -1
-static const uint16 vlc_tab_diff_huffcodes[37] = {
- 0x1c57, 0x0004, 0x0000, 0x0001, 0x0003, 0x0002, 0x000f, 0x000e,
- 0x0007, 0x0016, 0x0037, 0x0027, 0x0026, 0x0066, 0x0006, 0x0097,
- 0x0046, 0x01c6, 0x0017, 0x0786, 0x0086, 0x0257, 0x00d7, 0x0357,
- 0x00c6, 0x0386, 0x0186, 0x0000, 0x0157, 0x0c57, 0x0057, 0x0000,
- 0x0b86, 0x0000, 0x1457, 0x0000, 0x0457
-};
-
-static const byte vlc_tab_diff_huffbits[37] = {
- 13, 3, 3, 2, 3, 3, 4, 4, 6, 5, 6, 6, 7, 7, 8, 8,
- 8, 9, 8, 11, 9, 10, 8, 10, 9, 12, 10, 0, 10, 13, 11, 0,
- 12, 0, 13, 0, 13
-};
-
-// values in this table range from -1..5; adjust retrieved value by -1
-static const byte vlc_tab_run_huffcodes[6] = {
- 0x1f, 0x00, 0x01, 0x03, 0x07, 0x0f
-};
-
-static const byte vlc_tab_run_huffbits[6] = {
- 5, 1, 2, 3, 4, 5
-};
-
-// values in this table range from -1..19; adjust retrieved value by -1
-static const uint16 vlc_tab_tone_level_idx_hi1_huffcodes[20] = {
- 0x5714, 0x000c, 0x0002, 0x0001, 0x0000, 0x0004, 0x0034, 0x0054,
- 0x0094, 0x0014, 0x0114, 0x0214, 0x0314, 0x0614, 0x0e14, 0x0f14,
- 0x2714, 0x0714, 0x1714, 0x3714
-};
-
-static const byte vlc_tab_tone_level_idx_hi1_huffbits[20] = {
- 15, 4, 2, 1, 3, 5, 6, 7, 8, 10, 10, 11, 11, 12, 12, 12, 14, 14, 15, 14
-};
-
-// values in this table range from -1..23; adjust retrieved value by -1
-static const uint16 vlc_tab_tone_level_idx_mid_huffcodes[24] = {
- 0x0fea, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
- 0x0000, 0x0000, 0x0000, 0x0000, 0x03ea, 0x00ea, 0x002a, 0x001a,
- 0x0006, 0x0001, 0x0000, 0x0002, 0x000a, 0x006a, 0x01ea, 0x07ea
-};
-
-static const byte vlc_tab_tone_level_idx_mid_huffbits[24] = {
- 12, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12
-};
-
-// values in this table range from -1..23; adjust retrieved value by -1
-static const uint16 vlc_tab_tone_level_idx_hi2_huffcodes[24] = {
- 0x0664, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0064, 0x00e4,
- 0x00a4, 0x0068, 0x0004, 0x0008, 0x0014, 0x0018, 0x0000, 0x0001,
- 0x0002, 0x0003, 0x000c, 0x0028, 0x0024, 0x0164, 0x0000, 0x0264
-};
-
-static const byte vlc_tab_tone_level_idx_hi2_huffbits[24] = {
- 11, 0, 0, 0, 0, 0, 10, 8, 8, 7, 6, 6, 5, 5, 4, 2, 2, 2, 4, 7, 8, 9, 0, 11
-};
-
-// values in this table range from -1..8; adjust retrieved value by -1
-static const byte vlc_tab_type30_huffcodes[9] = {
- 0x3c, 0x06, 0x00, 0x01, 0x03, 0x02, 0x04, 0x0c, 0x1c
-};
-
-static const byte vlc_tab_type30_huffbits[9] = {
- 6, 3, 3, 2, 2, 3, 4, 5, 6
-};
-
-// values in this table range from -1..9; adjust retrieved value by -1
-static const byte vlc_tab_type34_huffcodes[10] = {
- 0x18, 0x00, 0x01, 0x04, 0x05, 0x07, 0x03, 0x02, 0x06, 0x08
-};
-
-static const byte vlc_tab_type34_huffbits[10] = {
- 5, 4, 3, 3, 3, 3, 3, 3, 3, 5
-};
-
-// values in this table range from -1..22; adjust retrieved value by -1
-static const uint16 vlc_tab_fft_tone_offset_0_huffcodes[23] = {
- 0x038e, 0x0001, 0x0000, 0x0022, 0x000a, 0x0006, 0x0012, 0x0002,
- 0x001e, 0x003e, 0x0056, 0x0016, 0x000e, 0x0032, 0x0072, 0x0042,
- 0x008e, 0x004e, 0x00f2, 0x002e, 0x0036, 0x00c2, 0x018e
-};
-
-static const byte vlc_tab_fft_tone_offset_0_huffbits[23] = {
- 10, 1, 2, 6, 4, 5, 6, 7, 6, 6, 7, 7, 8, 7, 8, 8, 9, 7, 8, 6, 6, 8, 10
-};
-
-// values in this table range from -1..27; adjust retrieved value by -1
-static const uint16 vlc_tab_fft_tone_offset_1_huffcodes[28] = {
- 0x07a4, 0x0001, 0x0020, 0x0012, 0x001c, 0x0008, 0x0006, 0x0010,
- 0x0000, 0x0014, 0x0004, 0x0032, 0x0070, 0x000c, 0x0002, 0x003a,
- 0x001a, 0x002c, 0x002a, 0x0022, 0x0024, 0x000a, 0x0064, 0x0030,
- 0x0062, 0x00a4, 0x01a4, 0x03a4
-};
-
-static const byte vlc_tab_fft_tone_offset_1_huffbits[28] = {
- 11, 1, 6, 6, 5, 4, 3, 6, 6, 5, 6, 6, 7, 6, 6, 6,
- 6, 6, 6, 7, 8, 6, 7, 7, 7, 9, 10, 11
-};
-
-// values in this table range from -1..31; adjust retrieved value by -1
-static const uint16 vlc_tab_fft_tone_offset_2_huffcodes[32] = {
- 0x1760, 0x0001, 0x0000, 0x0082, 0x000c, 0x0006, 0x0003, 0x0007,
- 0x0008, 0x0004, 0x0010, 0x0012, 0x0022, 0x001a, 0x0000, 0x0020,
- 0x000a, 0x0040, 0x004a, 0x006a, 0x002a, 0x0042, 0x0002, 0x0060,
- 0x00aa, 0x00e0, 0x00c2, 0x01c2, 0x0160, 0x0360, 0x0760, 0x0f60
-};
-
-static const byte vlc_tab_fft_tone_offset_2_huffbits[32] = {
- 13, 2, 0, 8, 4, 3, 3, 3, 4, 4, 5, 5, 6, 5, 7, 7,
- 7, 7, 7, 7, 8, 8, 8, 9, 8, 8, 9, 9, 10, 11, 13, 12
-};
-
-// values in this table range from -1..34; adjust retrieved value by -1
-static const uint16 vlc_tab_fft_tone_offset_3_huffcodes[35] = {
- 0x33ea, 0x0005, 0x0000, 0x000c, 0x0000, 0x0006, 0x0003, 0x0008,
- 0x0002, 0x0001, 0x0004, 0x0007, 0x001a, 0x000f, 0x001c, 0x002c,
- 0x000a, 0x001d, 0x002d, 0x002a, 0x000d, 0x004c, 0x008c, 0x006a,
- 0x00cd, 0x004d, 0x00ea, 0x020c, 0x030c, 0x010c, 0x01ea, 0x07ea,
- 0x0bea, 0x03ea, 0x13ea
-};
-
-static const byte vlc_tab_fft_tone_offset_3_huffbits[35] = {
- 14, 4, 0, 10, 4, 3, 3, 4, 4, 3, 4, 4, 5, 4, 5, 6,
- 6, 5, 6, 7, 7, 7, 8, 8, 8, 8, 9, 10, 10, 10, 10, 11,
- 12, 13, 14
-};
-
-// values in this table range from -1..37; adjust retrieved value by -1
-static const uint16 vlc_tab_fft_tone_offset_4_huffcodes[38] = {
- 0x5282, 0x0016, 0x0000, 0x0136, 0x0004, 0x0000, 0x0007, 0x000a,
- 0x000e, 0x0003, 0x0001, 0x000d, 0x0006, 0x0009, 0x0012, 0x0005,
- 0x0025, 0x0022, 0x0015, 0x0002, 0x0076, 0x0035, 0x0042, 0x00c2,
- 0x0182, 0x00b6, 0x0036, 0x03c2, 0x0482, 0x01c2, 0x0682, 0x0882,
- 0x0a82, 0x0082, 0x0282, 0x1282, 0x3282, 0x2282
-};
-
-static const byte vlc_tab_fft_tone_offset_4_huffbits[38] = {
- 15, 6, 0, 9, 3, 3, 3, 4, 4, 3, 4, 4, 5, 4, 5, 6,
- 6, 6, 6, 8, 7, 6, 8, 9, 9, 8, 9, 10, 11, 10, 11, 12,
- 12, 12, 14, 15, 14, 14
-};
-
-/// FFT TABLES
-
-// values in this table range from -1..27; adjust retrieved value by -1
-static const uint16 fft_level_exp_alt_huffcodes[28] = {
- 0x1ec6, 0x0006, 0x00c2, 0x0142, 0x0242, 0x0246, 0x00c6, 0x0046,
- 0x0042, 0x0146, 0x00a2, 0x0062, 0x0026, 0x0016, 0x000e, 0x0005,
- 0x0004, 0x0003, 0x0000, 0x0001, 0x000a, 0x0012, 0x0002, 0x0022,
- 0x01c6, 0x02c6, 0x06c6, 0x0ec6
-};
-
-static const byte fft_level_exp_alt_huffbits[28] = {
- 13, 7, 8, 9, 10, 10, 10, 10, 10, 9, 8, 7, 6, 5, 4, 3,
- 3, 2, 3, 3, 4, 5, 7, 8, 9, 11, 12, 13
-};
-
-// values in this table range from -1..19; adjust retrieved value by -1
-static const uint16 fft_level_exp_huffcodes[20] = {
- 0x0f24, 0x0001, 0x0002, 0x0000, 0x0006, 0x0005, 0x0007, 0x000c,
- 0x000b, 0x0014, 0x0013, 0x0004, 0x0003, 0x0023, 0x0064, 0x00a4,
- 0x0024, 0x0124, 0x0324, 0x0724
-};
-
-static const byte fft_level_exp_huffbits[20] = {
- 12, 3, 3, 3, 3, 3, 3, 4, 4, 5, 5, 6, 6, 6, 7, 8, 9, 10, 11, 12
-};
-
-// values in this table range from -1..6; adjust retrieved value by -1
-static const byte fft_stereo_exp_huffcodes[7] = {
- 0x3e, 0x01, 0x00, 0x02, 0x06, 0x0e, 0x1e
-};
-
-static const byte fft_stereo_exp_huffbits[7] = {
- 6, 1, 2, 3, 4, 5, 6
-};
-
-// values in this table range from -1..8; adjust retrieved value by -1
-static const byte fft_stereo_phase_huffcodes[9] = {
- 0x35, 0x02, 0x00, 0x01, 0x0d, 0x15, 0x05, 0x09, 0x03
-};
-
-static const byte fft_stereo_phase_huffbits[9] = {
- 6, 2, 2, 4, 4, 6, 5, 4, 2
-};
-
-static const int fft_cutoff_index_table[4][2] = {
- { 1, 2 }, {-1, 0 }, {-1,-2 }, { 0, 0 }
-};
-
-static const int16 fft_level_index_table[256] = {
- 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1,
- 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
- 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
- 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
- 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
- 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
- 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
- 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
-};
-
-static const byte last_coeff[3] = {
- 4, 7, 10
-};
-
-static const byte coeff_per_sb_for_avg[3][30] = {
- { 0, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3 },
- { 0, 1, 2, 2, 3, 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6 },
- { 0, 1, 2, 3, 4, 4, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9, 9, 9 }
-};
-
-static const uint32 dequant_table[3][10][30] = {
- { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 256, 256, 205, 154, 102, 51, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 51, 102, 154, 205, 256, 238, 219, 201, 183, 165, 146, 128, 110, 91, 73, 55, 37, 18, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 18, 37, 55, 73, 91, 110, 128, 146, 165, 183, 201, 219, 238, 256, 228, 199, 171, 142, 114, 85, 57, 28 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
- { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 85, 171, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 85, 171, 256, 219, 183, 146, 110, 73, 37, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 37, 73, 110, 146, 183, 219, 256, 228, 199, 171, 142, 114, 85, 57, 28, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 28, 57, 85, 114, 142, 171, 199, 228, 256, 213, 171, 128, 85, 43 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
- { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 256, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 85, 171, 256, 192, 128, 64, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 64, 128, 192, 256, 205, 154, 102, 51, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 51, 102, 154, 205, 256, 213, 171, 128, 85, 43, 0, 0, 0, 0, 0, 0 },
- { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 43, 85, 128, 171, 213, 256, 213, 171, 128, 85, 43 } }
-};
-
-static const byte coeff_per_sb_for_dequant[3][30] = {
- { 0, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3 },
- { 0, 1, 2, 2, 2, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6 },
- { 0, 1, 2, 3, 4, 4, 5, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9 }
-};
-
-// first index is subband, 2nd index is 0, 1 or 3 (2 is unused)
-static const int8 tone_level_idx_offset_table[30][4] = {
- { -50, -50, 0, -50 },
- { -50, -50, 0, -50 },
- { -50, -9, 0, -19 },
- { -16, -6, 0, -12 },
- { -11, -4, 0, -8 },
- { -8, -3, 0, -6 },
- { -7, -3, 0, -5 },
- { -6, -2, 0, -4 },
- { -5, -2, 0, -3 },
- { -4, -1, 0, -3 },
- { -4, -1, 0, -2 },
- { -3, -1, 0, -2 },
- { -3, -1, 0, -2 },
- { -3, -1, 0, -2 },
- { -2, -1, 0, -1 },
- { -2, -1, 0, -1 },
- { -2, -1, 0, -1 },
- { -2, 0, 0, -1 },
- { -2, 0, 0, -1 },
- { -1, 0, 0, -1 },
- { -1, 0, 0, -1 },
- { -1, 0, 0, -1 },
- { -1, 0, 0, -1 },
- { -1, 0, 0, -1 },
- { -1, 0, 0, -1 },
- { -1, 0, 0, -1 },
- { -1, 0, 0, 0 },
- { -1, 0, 0, 0 },
- { -1, 0, 0, 0 },
- { -1, 0, 0, 0 }
-};
-
-/* all my samples have 1st index 0 or 1 */
-/* second index is subband, only indexes 0-29 seem to be used */
-static const int8 coding_method_table[5][30] = {
- { 34, 30, 24, 24, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10,
- 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10
- },
- { 34, 30, 24, 24, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10,
- 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10
- },
- { 34, 30, 30, 30, 24, 24, 16, 16, 16, 16, 16, 16, 10, 10, 10,
- 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10
- },
- { 34, 34, 30, 30, 24, 24, 24, 24, 16, 16, 16, 16, 16, 16, 16,
- 16, 16, 16, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10, 10
- },
- { 34, 34, 30, 30, 30, 30, 30, 30, 24, 24, 24, 24, 24, 24, 24,
- 24, 24, 24, 24, 24, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16
- },
-};
-
-static const int vlc_stage3_values[60] = {
- 0, 1, 2, 3, 4, 6, 8, 10, 12, 16, 20, 24,
- 28, 36, 44, 52, 60, 76, 92, 108, 124, 156, 188, 220,
- 252, 316, 380, 444, 508, 636, 764, 892, 1020, 1276, 1532, 1788,
- 2044, 2556, 3068, 3580, 4092, 5116, 6140, 7164, 8188, 10236, 12284, 14332,
- 16380, 20476, 24572, 28668, 32764, 40956, 49148, 57340, 65532, 81916, 98300,114684
-};
-
-static const float fft_tone_sample_table[4][16][5] = {
- { { .0100000000f,-.0037037037f,-.0020000000f,-.0069444444f,-.0018416207f },
- { .0416666667f, .0000000000f, .0000000000f,-.0208333333f,-.0123456791f },
- { .1250000000f, .0558035709f, .0330687836f,-.0164473690f,-.0097465888f },
- { .1562500000f, .0625000000f, .0370370370f,-.0062500000f,-.0037037037f },
- { .1996007860f, .0781250000f, .0462962948f, .0022727272f, .0013468013f },
- { .2000000000f, .0625000000f, .0370370373f, .0208333333f, .0074074073f },
- { .2127659619f, .0555555556f, .0329218097f, .0208333333f, .0123456791f },
- { .2173913121f, .0473484844f, .0280583613f, .0347222239f, .0205761325f },
- { .2173913121f, .0347222239f, .0205761325f, .0473484844f, .0280583613f },
- { .2127659619f, .0208333333f, .0123456791f, .0555555556f, .0329218097f },
- { .2000000000f, .0208333333f, .0074074073f, .0625000000f, .0370370370f },
- { .1996007860f, .0022727272f, .0013468013f, .0781250000f, .0462962948f },
- { .1562500000f,-.0062500000f,-.0037037037f, .0625000000f, .0370370370f },
- { .1250000000f,-.0164473690f,-.0097465888f, .0558035709f, .0330687836f },
- { .0416666667f,-.0208333333f,-.0123456791f, .0000000000f, .0000000000f },
- { .0100000000f,-.0069444444f,-.0018416207f,-.0037037037f,-.0020000000f } },
-
- { { .0050000000f,-.0200000000f, .0125000000f,-.3030303030f, .0020000000f },
- { .1041666642f, .0400000000f,-.0250000000f, .0333333333f,-.0200000000f },
- { .1250000000f, .0100000000f, .0142857144f,-.0500000007f,-.0200000000f },
- { .1562500000f,-.0006250000f,-.00049382716f,-.000625000f,-.00049382716f },
- { .1562500000f,-.0006250000f,-.00049382716f,-.000625000f,-.00049382716f },
- { .1250000000f,-.0500000000f,-.0200000000f, .0100000000f, .0142857144f },
- { .1041666667f, .0333333333f,-.0200000000f, .0400000000f,-.0250000000f },
- { .0050000000f,-.3030303030f, .0020000001f,-.0200000000f, .0125000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } },
-
- { { .1428571492f, .1250000000f,-.0285714287f,-.0357142873f, .0208333333f },
- { .1818181818f, .0588235296f, .0333333333f, .0212765951f, .0100000000f },
- { .1818181818f, .0212765951f, .0100000000f, .0588235296f, .0333333333f },
- { .1428571492f,-.0357142873f, .0208333333f, .1250000000f,-.0285714287f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } },
-
- { { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f },
- { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } }
-};
-
-static const float fft_tone_level_table[2][64] = { {
-// pow ~ (i > 46) ? 0 : (((((i & 1) ? 431 : 304) << (i >> 1))) / 1024.0);
- 0.17677669f, 0.42677650f, 0.60355347f, 0.85355347f,
- 1.20710683f, 1.68359375f, 2.37500000f, 3.36718750f,
- 4.75000000f, 6.73437500f, 9.50000000f, 13.4687500f,
- 19.0000000f, 26.9375000f, 38.0000000f, 53.8750000f,
- 76.0000000f, 107.750000f, 152.000000f, 215.500000f,
- 304.000000f, 431.000000f, 608.000000f, 862.000000f,
- 1216.00000f, 1724.00000f, 2432.00000f, 3448.00000f,
- 4864.00000f, 6896.00000f, 9728.00000f, 13792.0000f,
- 19456.0000f, 27584.0000f, 38912.0000f, 55168.0000f,
- 77824.0000f, 110336.000f, 155648.000f, 220672.000f,
- 311296.000f, 441344.000f, 622592.000f, 882688.000f,
- 1245184.00f, 1765376.00f, 2490368.00f, 0.00000000f,
- 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
- 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
- 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
- 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
- }, {
-// pow = (i > 45) ? 0 : ((((i & 1) ? 431 : 304) << (i >> 1)) / 512.0);
- 0.59375000f, 0.84179688f, 1.18750000f, 1.68359375f,
- 2.37500000f, 3.36718750f, 4.75000000f, 6.73437500f,
- 9.50000000f, 13.4687500f, 19.0000000f, 26.9375000f,
- 38.0000000f, 53.8750000f, 76.0000000f, 107.750000f,
- 152.000000f, 215.500000f, 304.000000f, 431.000000f,
- 608.000000f, 862.000000f, 1216.00000f, 1724.00000f,
- 2432.00000f, 3448.00000f, 4864.00000f, 6896.00000f,
- 9728.00000f, 13792.0000f, 19456.0000f, 27584.0000f,
- 38912.0000f, 55168.0000f, 77824.0000f, 110336.000f,
- 155648.000f, 220672.000f, 311296.000f, 441344.000f,
- 622592.000f, 882688.000f, 1245184.00f, 1765376.00f,
- 2490368.00f, 3530752.00f, 0.00000000f, 0.00000000f,
- 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
- 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
- 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
- 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f
-} };
-
-static const float fft_tone_envelope_table[4][31] = {
- { .009607375f, .038060248f, .084265202f, .146446645f, .222214907f, .308658302f,
- .402454883f, .500000060f, .597545207f, .691341758f, .777785182f, .853553414f,
- .915734828f, .961939812f, .990392685f, 1.00000000f, .990392625f, .961939752f,
- .915734768f, .853553295f, .777785063f, .691341639f, .597545087f, .500000000f,
- .402454853f, .308658272f, .222214878f, .146446615f, .084265172f, .038060218f,
- .009607345f },
- { .038060248f, .146446645f, .308658302f, .500000060f, .691341758f, .853553414f,
- .961939812f, 1.00000000f, .961939752f, .853553295f, .691341639f, .500000000f,
- .308658272f, .146446615f, .038060218f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f },
- { .146446645f, .500000060f, .853553414f, 1.00000000f, .853553295f, .500000000f,
- .146446615f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f },
- { .500000060f, 1.00000000f, .500000000f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f,
- .000000000f }
-};
-
-static const float sb_noise_attenuation[32] = {
- 0.0f, 0.0f, 0.3f, 0.4f, 0.5f, 0.7f, 1.0f, 1.0f,
- 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
- 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
- 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
-};
-
-static const byte fft_subpackets[32] = {
- 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 0,
- 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 0, 0
-};
-
-// first index is joined_stereo, second index is 0 or 2 (1 is unused)
-static const float dequant_1bit[2][3] = {
- {-0.920000f, 0.000000f, 0.920000f },
- {-0.890000f, 0.000000f, 0.890000f }
-};
-
-static const float type30_dequant[8] = {
- -1.0f,-0.625f,-0.291666656732559f,0.0f,
- 0.25f,0.5f,0.75f,1.0f,
-};
-
-static const float type34_delta[10] = { // FIXME: covers 8 entries..
- -1.0f,-0.60947573184967f,-0.333333343267441f,-0.138071194291115f,0.0f,
- 0.138071194291115f,0.333333343267441f,0.60947573184967f,1.0f,0.0f,
-};
-
-} // End of namespace Audio
-
-#endif
diff --git a/sound/module.mk b/sound/module.mk
index cd1ff0df8e..df593d8e1f 100644
--- a/sound/module.mk
+++ b/sound/module.mk
@@ -18,7 +18,6 @@ MODULE_OBJS := \
decoders/flac.o \
decoders/iff_sound.o \
decoders/mp3.o \
- decoders/qdm2.o \
decoders/raw.o \
decoders/vag.o \
decoders/voc.o \
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-23 12:18:19 +0000