From 806ccf5d25ebe337103942cdb6b5cf8800de115a Mon Sep 17 00:00:00 2001 From: Eugene Sandulenko Date: Sun, 23 Jan 2011 17:14:43 +0000 Subject: GRAPHICS: Move graphics/video/ to video/. Step 1/2 svn-id: r55473 --- video/codecs/qdm2.cpp | 3287 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 3287 insertions(+) create mode 100644 video/codecs/qdm2.cpp (limited to 'video/codecs/qdm2.cpp') diff --git a/video/codecs/qdm2.cpp b/video/codecs/qdm2.cpp new file mode 100644 index 0000000000..e34c569feb --- /dev/null +++ b/video/codecs/qdm2.cpp @@ -0,0 +1,3287 @@ +/* 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 "common/scummsys.h" +#include "graphics/video/codecs/qdm2.h" + +#ifdef GRAPHICS_QDM2_H + +#include "sound/audiostream.h" +#include "graphics/video/codecs/qdm2data.h" + +#include "common/array.h" +#include "common/stream.h" +#include "common/system.h" + +namespace Graphics { + +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 && _subPacket == 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 _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 + uint32 _superBlockStart; + 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); + + 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; + + 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) { + return s->index; +} + +static inline unsigned int getBits1(GetBitContext *s) { + int index; + uint8 result; + + index = s->index; + result = s->buffer[index >> 3]; + + result >>= (index & 0x07); + result &= 1; + index++; + s->index = index; + + return result; +} + +static inline unsigned int getBits(GetBitContext *s, int n) { + int tmp, reCache, reIndex; + + reIndex = s->index; + + 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; + + reIndex = s->index; + reCache = 0; + + 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<=xrevtab; + 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; + + 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]; + + 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; + int16 (*temp)[2] = NULL; + 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); + temp = (int16 (*)[2])realloc(vlc->table, sizeof(int16 *) * 2 * vlc->table_allocated); + if (!temp) { + free(vlc->table); + vlc->table = NULL; + return -1; + } + vlc->table = temp; + } + 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); + 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); + // 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<> ((flags & 2) ? n_prefix : n)) & ((1 << table_nb_bits) - 1); + // 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"); + } + + 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; + _superBlockStart = 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 + } +} + +/** + * 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) { + 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) { + while ((n = qdm2_get_vlc(gb, &_vlcTabFftToneOffset[local_int_8], 1, 2)) < 2) { + 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) { + 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 + 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? + qdm2_fft_decode_tones(duration, &gb, unknown_flag); + } + } else if (type == 31) { + for (j=0; j < 4; j++) { + 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++) { + 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) { + int i; + + _fft.complex[channel][0].re *= 2.0f; + _fft.complex[channel][0].im = 0.0f; + + 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]; +} + +/** + * @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); + + // If we're in any packet but the first, seek back to the first + if (_subPacket == 0) + _superBlockStart = in->pos(); + else + in->seek(_superBlockStart); + + // select input buffer + if (in->eos() || in->pos() >= in->size()) { + 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; + } + + if (!in->eos()) { + 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"); + + if (!in->eos()) { + // 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"); + } + + for (i = 0; i < frame_size; 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 (decodedSamples < numSamples) { + 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; +} + +Audio::AudioStream *makeQDM2Stream(Common::SeekableReadStream *stream, Common::SeekableReadStream *extraData) { + return new QDM2Stream(stream, extraData); +} + +} // End of namespace Graphics + +#endif -- cgit v1.2.3