From f40bfc9267b13b54e6379dfe7539447662879d24 Mon Sep 17 00:00:00 2001 From: Sean Bartell Date: Sat, 25 Jun 2011 21:32:25 -0400 Subject: Add codecs to librbcodec. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Change-Id: Id7f4717d51ed02d67cb9f9cb3c0ada4a81843f97 Reviewed-on: http://gerrit.rockbox.org/137 Reviewed-by: Nils Wallménius Tested-by: Nils Wallménius --- lib/rbcodec/codecs/libatrac/atrac3.c | 1293 ++++++++++++++++++++++++++++++++++ 1 file changed, 1293 insertions(+) create mode 100644 lib/rbcodec/codecs/libatrac/atrac3.c (limited to 'lib/rbcodec/codecs/libatrac/atrac3.c') diff --git a/lib/rbcodec/codecs/libatrac/atrac3.c b/lib/rbcodec/codecs/libatrac/atrac3.c new file mode 100644 index 0000000000..bb52dd4cf0 --- /dev/null +++ b/lib/rbcodec/codecs/libatrac/atrac3.c @@ -0,0 +1,1293 @@ +/* + * Atrac 3 compatible decoder + * Copyright (c) 2006-2008 Maxim Poliakovski + * Copyright (c) 2006-2008 Benjamin Larsson + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file libavcodec/atrac3.c + * Atrac 3 compatible decoder. + * This decoder handles Sony's ATRAC3 data. + * + * Container formats used to store atrac 3 data: + * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3). + * + * To use this decoder, a calling application must supply the extradata + * bytes provided in the containers above. + */ + +#include +#include +#include + +#include "atrac3.h" +#include "atrac3data.h" +#include "atrac3data_fixed.h" +#include "fixp_math.h" + +#define JOINT_STEREO 0x12 +#define STEREO 0x2 + +#ifdef ROCKBOX +#undef DEBUGF +#define DEBUGF(...) +#endif /* ROCKBOX */ + +/* FFMAX/MIN/SWAP and av_clip were taken from libavutil/common.h */ +#define FFMAX(a,b) ((a) > (b) ? (a) : (b)) +#define FFMIN(a,b) ((a) > (b) ? (b) : (a)) +#define FFSWAP(type,a,b) do{type SWAP_tmp= b; b= a; a= SWAP_tmp;}while(0) + +#if defined(CPU_ARM) && (ARM_ARCH >= 5) + #define QMFWIN_TYPE int16_t /* ARMv5e+ uses 32x16 multiplication */ +#else + #define QMFWIN_TYPE int32_t +#endif + +static VLC spectral_coeff_tab[7] IBSS_ATTR_LARGE_IRAM; +static QMFWIN_TYPE qmf_window[48] IBSS_ATTR MEM_ALIGN_ATTR; +static int32_t atrac3_spectrum [2][1024] IBSS_ATTR MEM_ALIGN_ATTR; +static int32_t atrac3_IMDCT_buf[2][ 512] IBSS_ATTR MEM_ALIGN_ATTR; +static int32_t atrac3_prevFrame[2][1024] IBSS_ATTR MEM_ALIGN_ATTR; +static channel_unit channel_units[2] IBSS_ATTR_LARGE_IRAM; +static VLC_TYPE atrac3_vlc_table[4096][2] IBSS_ATTR_LARGE_IRAM; +static int vlcs_initialized = 0; + + + +/** + * Matrixing within quadrature mirror synthesis filter. + * + * @param p3 output buffer + * @param inlo lower part of spectrum + * @param inhi higher part of spectrum + * @param nIn size of spectrum buffer + */ + +#if defined(CPU_ARM) + extern void + atrac3_iqmf_matrixing(int32_t *p3, + int32_t *inlo, + int32_t *inhi, + unsigned int nIn); +#else + static inline void + atrac3_iqmf_matrixing(int32_t *p3, + int32_t *inlo, + int32_t *inhi, + unsigned int nIn) + { + uint32_t i; + for(i=0; i= 5) + extern void + atrac3_iqmf_dewindowing_armv5e(int32_t *out, + int32_t *in, + int16_t *win, + unsigned int nIn); + static inline void + atrac3_iqmf_dewindowing(int32_t *out, + int32_t *in, + int16_t *win, + unsigned int nIn) + { + atrac3_iqmf_dewindowing_armv5e(out, in, win, nIn); + + } + + +#elif defined(CPU_ARM) + extern void + atrac3_iqmf_dewindowing(int32_t *out, + int32_t *in, + int32_t *win, + unsigned int nIn); + +#elif defined (CPU_COLDFIRE) + #define MULTIPLY_ADD_BLOCK \ + "movem.l (%[win]), %%d0-%%d7 \n\t" \ + "lea.l (8*4, %[win]), %[win] \n\t" \ + "mac.l %%d0, %%a5, (%[in])+, %%a5, %%acc0\n\t" \ + "mac.l %%d1, %%a5, (%[in])+, %%a5, %%acc1\n\t" \ + "mac.l %%d2, %%a5, (%[in])+, %%a5, %%acc0\n\t" \ + "mac.l %%d3, %%a5, (%[in])+, %%a5, %%acc1\n\t" \ + "mac.l %%d4, %%a5, (%[in])+, %%a5, %%acc0\n\t" \ + "mac.l %%d5, %%a5, (%[in])+, %%a5, %%acc1\n\t" \ + "mac.l %%d6, %%a5, (%[in])+, %%a5, %%acc0\n\t" \ + "mac.l %%d7, %%a5, (%[in])+, %%a5, %%acc1\n\t" \ + + + static inline void + atrac3_iqmf_dewindowing(int32_t *out, + int32_t *in, + int32_t *win, + unsigned int nIn) + { + int32_t j; + int32_t *_in, *_win; + for (j = nIn; j != 0; j--, in+=2, out+=2) { + _in = in; + _win = win; + + asm volatile ( + "move.l (%[in])+, %%a5 \n\t" /* preload frist in value */ + MULTIPLY_ADD_BLOCK /* 0.. 7 */ + MULTIPLY_ADD_BLOCK /* 8..15 */ + MULTIPLY_ADD_BLOCK /* 16..23 */ + MULTIPLY_ADD_BLOCK /* 24..31 */ + MULTIPLY_ADD_BLOCK /* 32..39 */ + /* 40..47 */ + "movem.l (%[win]), %%d0-%%d7 \n\t" + "mac.l %%d0, %%a5, (%[in])+, %%a5, %%acc0 \n\t" + "mac.l %%d1, %%a5, (%[in])+, %%a5, %%acc1 \n\t" + "mac.l %%d2, %%a5, (%[in])+, %%a5, %%acc0 \n\t" + "mac.l %%d3, %%a5, (%[in])+, %%a5, %%acc1 \n\t" + "mac.l %%d4, %%a5, (%[in])+, %%a5, %%acc0 \n\t" + "mac.l %%d5, %%a5, (%[in])+, %%a5, %%acc1 \n\t" + "mac.l %%d6, %%a5, (%[in])+, %%a5, %%acc0 \n\t" + "mac.l %%d7, %%a5, %%acc1 \n\t" + "movclr.l %%acc0, %%d1 \n\t" /* s1 */ + "movclr.l %%acc1, %%d0 \n\t" /* s2 */ + "movem.l %%d0-%%d1, (%[out]) \n\t" + : [in] "+a" (_in), [win] "+a" (_win) + : [out] "a" (out) + : "d0", "d1", "d2", "d3", "d4", "d5", "d6", "d7", "a5", "memory"); + } + } +#else + #define MULTIPLY_ADD_BLOCK(y1, y2, x, c, k) \ + y1 += fixmul31(c[k], x[k]); k++; \ + y2 += fixmul31(c[k], x[k]); k++; \ + y1 += fixmul31(c[k], x[k]); k++; \ + y2 += fixmul31(c[k], x[k]); k++; \ + y1 += fixmul31(c[k], x[k]); k++; \ + y2 += fixmul31(c[k], x[k]); k++; \ + y1 += fixmul31(c[k], x[k]); k++; \ + y2 += fixmul31(c[k], x[k]); k++; + + static inline void + atrac3_iqmf_dewindowing(int32_t *out, + int32_t *in, + int32_t *win, + unsigned int nIn) + { + int32_t i, j, s1, s2; + + for (j = nIn; j != 0; j--, in+=2, out+=2) { + s1 = s2 = i = 0; + + MULTIPLY_ADD_BLOCK(s1, s2, in, win, i); /* 0.. 7 */ + MULTIPLY_ADD_BLOCK(s1, s2, in, win, i); /* 8..15 */ + MULTIPLY_ADD_BLOCK(s1, s2, in, win, i); /* 16..23 */ + MULTIPLY_ADD_BLOCK(s1, s2, in, win, i); /* 24..31 */ + MULTIPLY_ADD_BLOCK(s1, s2, in, win, i); /* 32..39 */ + MULTIPLY_ADD_BLOCK(s1, s2, in, win, i); /* 40..47 */ + + out[0] = s2; + out[1] = s1; + + } + + } +#endif + + +/** + * IMDCT windowing. + * + * @param buffer sample buffer + * @param win window coefficients + */ + +static inline void +atrac3_imdct_windowing(int32_t *buffer, + const int32_t *win) +{ + int32_t i; + /* win[0..127] = win[511..384], win[128..383] = 1 */ + for(i = 0; i<128; i++) { + buffer[ i] = fixmul31(win[i], buffer[ i]); + buffer[511-i] = fixmul31(win[i], buffer[511-i]); + } +} + + +/** + * Quadrature mirror synthesis filter. + * + * @param inlo lower part of spectrum + * @param inhi higher part of spectrum + * @param nIn size of spectrum buffer + * @param pOut out buffer + * @param delayBuf delayBuf buffer + * @param temp temp buffer + */ + +static void iqmf (int32_t *inlo, int32_t *inhi, unsigned int nIn, int32_t *pOut, int32_t *delayBuf, int32_t *temp) +{ + + /* Restore the delay buffer */ + memcpy(temp, delayBuf, 46*sizeof(int32_t)); + + /* loop1: matrixing */ + atrac3_iqmf_matrixing(temp + 46, inlo, inhi, nIn); + + /* loop2: dewindowing */ + atrac3_iqmf_dewindowing(pOut, temp, qmf_window, nIn); + + /* Save the delay buffer */ + memcpy(delayBuf, temp + (nIn << 1), 46*sizeof(int32_t)); +} + + +/** + * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands + * caused by the reverse spectra of the QMF. + * + * @param pInput input + * @param pOutput output + * @param odd_band 1 if the band is an odd band + */ + +static void IMLT(int32_t *pInput, int32_t *pOutput) +{ + /* Apply the imdct. */ + ff_imdct_calc(9, pOutput, pInput); + + /* Windowing. */ + atrac3_imdct_windowing(pOutput, window_lookup); + +} + + +/** + * Atrac 3 indata descrambling, only used for data coming from the rm container + * + * @param in pointer to 8 bit array of indata + * @param bits amount of bits + * @param out pointer to 8 bit array of outdata + */ + +static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){ + int i, off; + uint32_t c; + const uint32_t* buf; + uint32_t* obuf = (uint32_t*) out; + +#if ((defined(TEST) || defined(SIMULATOR)) && !defined(CPU_ARM)) + off = 0; /* no check for memory alignment of inbuffer */ +#else + off = (intptr_t)inbuffer & 3; +#endif /* TEST */ + buf = (const uint32_t*) (inbuffer - off); + + c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8)))); + bytes += 3 + off; + for (i = 0; i < bytes/4; i++) + obuf[i] = c ^ buf[i]; + + return off; +} + + +static void init_atrac3_transforms(void) +{ + int32_t s; + int i; + + /* Generate the mdct window, for details see + * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */ + + /* mdct window had been generated and saved as a lookup table in atrac3data_fixed.h */ + + /* Generate the QMF window. */ + for (i=0 ; i<24; i++) { + s = qmf_48tap_half_fix[i] << 1; + #if defined(CPU_ARM) && (ARM_ARCH >= 5) + qmf_window[i] = qmf_window[47-i] = (int16_t)((s+(1<<15))>>16); + #else + qmf_window[i] = qmf_window[47-i] = s; + #endif + } + +} + + +/** + * Mantissa decoding + * + * @param gb the GetBit context + * @param selector what table is the output values coded with + * @param codingFlag constant length coding or variable length coding + * @param mantissas mantissa output table + * @param numCodes amount of values to get + */ + +static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes) +{ + int numBits, cnt, code, huffSymb; + + if (selector == 1) + numCodes /= 2; + + if (codingFlag != 0) { + /* constant length coding (CLC) */ + numBits = CLCLengthTab[selector]; + + if (selector > 1) { + for (cnt = 0; cnt < numCodes; cnt++) { + if (numBits) + code = get_sbits(gb, numBits); + else + code = 0; + mantissas[cnt] = code; + } + } else { + for (cnt = 0; cnt < numCodes; cnt++) { + if (numBits) + code = get_bits(gb, numBits); /* numBits is always 4 in this case */ + else + code = 0; + mantissas[cnt*2] = seTab_0[code >> 2]; + mantissas[cnt*2+1] = seTab_0[code & 3]; + } + } + } else { + /* variable length coding (VLC) */ + if (selector != 1) { + for (cnt = 0; cnt < numCodes; cnt++) { + huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3); + huffSymb += 1; + code = huffSymb >> 1; + if (huffSymb & 1) + code = -code; + mantissas[cnt] = code; + } + } else { + for (cnt = 0; cnt < numCodes; cnt++) { + huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3); + mantissas[cnt*2] = decTable1[huffSymb*2]; + mantissas[cnt*2+1] = decTable1[huffSymb*2+1]; + } + } + } +} + + +/** + * Requantize the spectrum. + * + * @param *mantissas pointer to mantissas for each spectral line + * @param pOut requantized band spectrum + * @param first first spectral line in subband + * @param last last spectral line in subband + * @param SF scalefactor for all spectral lines of this band + */ + +static void inverseQuantizeSpectrum(int *mantissas, int32_t *pOut, + int32_t first, int32_t last, int32_t SF) +{ + int *pIn = mantissas; + + /* Inverse quantize the coefficients. */ + if((first/256) &1) { + /* Odd band - Reverse coefficients */ + do { + pOut[last--] = fixmul16(*pIn++, SF); + pOut[last--] = fixmul16(*pIn++, SF); + pOut[last--] = fixmul16(*pIn++, SF); + pOut[last--] = fixmul16(*pIn++, SF); + pOut[last--] = fixmul16(*pIn++, SF); + pOut[last--] = fixmul16(*pIn++, SF); + pOut[last--] = fixmul16(*pIn++, SF); + pOut[last--] = fixmul16(*pIn++, SF); + } while (last>first); + } else { + /* Even band - Do not reverse coefficients */ + do { + pOut[first++] = fixmul16(*pIn++, SF); + pOut[first++] = fixmul16(*pIn++, SF); + pOut[first++] = fixmul16(*pIn++, SF); + pOut[first++] = fixmul16(*pIn++, SF); + pOut[first++] = fixmul16(*pIn++, SF); + pOut[first++] = fixmul16(*pIn++, SF); + pOut[first++] = fixmul16(*pIn++, SF); + pOut[first++] = fixmul16(*pIn++, SF); + } while (first> 2] == 0) + continue; + + coded_components = get_bits(gb,3); + + for (k=0; kgBlock; + + for (i=0 ; i<=numBands; i++) + { + numData = get_bits(gb,3); + pGain[i].num_gain_data = numData; + pLevel = pGain[i].levcode; + pLoc = pGain[i].loccode; + + for (cf = 0; cf < numData; cf++){ + pLevel[cf]= get_bits(gb,4); + pLoc [cf]= get_bits(gb,5); + if(cf && pLoc[cf] <= pLoc[cf-1]) + return -1; + } + } + + /* Clear the unused blocks. */ + for (; i<4 ; i++) + pGain[i].num_gain_data = 0; + + return 0; +} + + +/** + * Apply fix (constant) gain and overlap for sample[start...255]. + * + * @param pIn input buffer + * @param pPrev previous buffer to perform overlap against + * @param pOut output buffer + * @param start index to start with (always a multiple of 8) + * @param gain gain to apply + */ + +static void applyFixGain (int32_t *pIn, int32_t *pPrev, int32_t *pOut, + int32_t start, int32_t gain) +{ + int32_t i = start; + + /* start is always a multiple of 8 and therefore allows us to unroll the + * loop to 8 calculation per loop + */ + if (ONE_16 == gain) { + /* gain1 = 1.0 -> no multiplication needed, just adding */ + /* Remark: This path is called >90%. */ + while (i<256) { + pOut[i] = pIn[i] + pPrev[i]; i++; + pOut[i] = pIn[i] + pPrev[i]; i++; + pOut[i] = pIn[i] + pPrev[i]; i++; + pOut[i] = pIn[i] + pPrev[i]; i++; + pOut[i] = pIn[i] + pPrev[i]; i++; + pOut[i] = pIn[i] + pPrev[i]; i++; + pOut[i] = pIn[i] + pPrev[i]; i++; + pOut[i] = pIn[i] + pPrev[i]; i++; + }; + } else { + /* gain1 != 1.0 -> we need to do a multiplication */ + /* Remark: This path is called seldom. */ + while (i<256) { + pOut[i] = fixmul16(pIn[i], gain) + pPrev[i]; i++; + pOut[i] = fixmul16(pIn[i], gain) + pPrev[i]; i++; + pOut[i] = fixmul16(pIn[i], gain) + pPrev[i]; i++; + pOut[i] = fixmul16(pIn[i], gain) + pPrev[i]; i++; + pOut[i] = fixmul16(pIn[i], gain) + pPrev[i]; i++; + pOut[i] = fixmul16(pIn[i], gain) + pPrev[i]; i++; + pOut[i] = fixmul16(pIn[i], gain) + pPrev[i]; i++; + pOut[i] = fixmul16(pIn[i], gain) + pPrev[i]; i++; + }; + } +} + + +/** + * Apply variable gain and overlap. Returns sample index after applying gain, + * resulting sample index is always a multiple of 8. + * + * @param pIn input buffer + * @param pPrev previous buffer to perform overlap against + * @param pOut output buffer + * @param start index to start with (always a multiple of 8) + * @param end end index for first loop (always a multiple of 8) + * @param gain1 current bands gain to apply + * @param gain2 next bands gain to apply + * @param gain_inc stepwise adaption from gain1 to gain2 + */ + +static int applyVariableGain (int32_t *pIn, int32_t *pPrev, int32_t *pOut, + int32_t start, int32_t end, + int32_t gain1, int32_t gain2, int32_t gain_inc) +{ + int32_t i = start; + + /* Apply fix gains until end index is reached */ + do { + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + } while (i < end); + + /* Interpolation is done over next eight samples */ + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + gain2 = fixmul16(gain2, gain_inc); + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + gain2 = fixmul16(gain2, gain_inc); + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + gain2 = fixmul16(gain2, gain_inc); + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + gain2 = fixmul16(gain2, gain_inc); + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + gain2 = fixmul16(gain2, gain_inc); + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + gain2 = fixmul16(gain2, gain_inc); + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + gain2 = fixmul16(gain2, gain_inc); + pOut[i] = fixmul16((fixmul16(pIn[i], gain1) + pPrev[i]), gain2); i++; + gain2 = fixmul16(gain2, gain_inc); + + return i; +} + + +/** + * Apply gain parameters and perform the MDCT overlapping part + * + * @param pIn input buffer + * @param pPrev previous buffer to perform overlap against + * @param pOut output buffer + * @param pGain1 current band gain info + * @param pGain2 next band gain info + */ + +static void gainCompensateAndOverlap (int32_t *pIn, int32_t *pPrev, int32_t *pOut, + gain_info *pGain1, gain_info *pGain2) +{ + /* gain compensation function */ + int32_t gain1, gain2, gain_inc; + int cnt, numdata, nsample, startLoc; + + if (pGain2->num_gain_data == 0) + gain1 = ONE_16; + else + gain1 = (ONE_16<<4)>>(pGain2->levcode[0]); + + if (pGain1->num_gain_data == 0) { + /* Remark: This path is called >90%. */ + /* Apply gain for all samples from 0...255 */ + applyFixGain(pIn, pPrev, pOut, 0, gain1); + } else { + /* Remark: This path is called seldom. */ + numdata = pGain1->num_gain_data; + pGain1->loccode[numdata] = 32; + pGain1->levcode[numdata] = 4; + + nsample = 0; /* starting loop with =0 */ + + for (cnt = 0; cnt < numdata; cnt++) { + startLoc = pGain1->loccode[cnt] * 8; + + gain2 = (ONE_16<<4)>>(pGain1->levcode[cnt]); + gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15]; + + /* Apply variable gain (gain1 -> gain2) to samples */ + nsample = applyVariableGain(pIn, pPrev, pOut, nsample, startLoc, gain1, gain2, gain_inc); + } + /* Apply gain for the residual samples from nsample...255 */ + applyFixGain(pIn, pPrev, pOut, nsample, gain1); + } + + /* Delay for the overlapping part. */ + memcpy(pPrev, &pIn[256], 256*sizeof(int32_t)); +} + + +/** + * Combine the tonal band spectrum and regular band spectrum + * Return position of the last tonal coefficient + + * + * @param pSpectrum output spectrum buffer + * @param numComponents amount of tonal components + * @param pComponent tonal components for this band + */ + +static int addTonalComponents (int32_t *pSpectrum, int numComponents, tonal_component *pComponent) +{ + int cnt, i, lastPos = -1; + int32_t *pOut; + int32_t *pIn; + + for (cnt = 0; cnt < numComponents; cnt++){ + lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos); + pIn = pComponent[cnt].coef; + pOut = &(pSpectrum[pComponent[cnt].pos]); + + for (i=0 ; i>3), (((y) - (x))))) +*/ +#define INTERPOLATE_FP16(x, y, s) ((x) + ((s*((y)-(x)))>>3)) + +static void reverseMatrixing(int32_t *su1, int32_t *su2, int *pPrevCode, int *pCurrCode) +{ + int i, band, nsample, s1, s2; + int32_t c1, c2; + int32_t mc1_l, mc1_r, mc2_l, mc2_r; + + for (i=0,band = 0; band < 4*256; band+=256,i++) { + s1 = pPrevCode[i]; + s2 = pCurrCode[i]; + nsample = 0; + + if (s1 != s2) { + /* Selector value changed, interpolation needed. */ + mc1_l = matrixCoeffs_fix[s1<<1]; + mc1_r = matrixCoeffs_fix[(s1<<1)+1]; + mc2_l = matrixCoeffs_fix[s2<<1]; + mc2_r = matrixCoeffs_fix[(s2<<1)+1]; + + /* Interpolation is done over the first eight samples. */ + for(; nsample < 8; nsample++) { + c1 = su1[band+nsample]; + c2 = su2[band+nsample]; + c2 = fixmul16(c1, INTERPOLATE_FP16(mc1_l, mc2_l, nsample)) + fixmul16(c2, INTERPOLATE_FP16(mc1_r, mc2_r, nsample)); + su1[band+nsample] = c2; + su2[band+nsample] = (c1 << 1) - c2; + } + } + + /* Apply the matrix without interpolation. */ + switch (s2) { + case 0: /* M/S decoding */ + for (; nsample < 256; nsample++) { + c1 = su1[band+nsample]; + c2 = su2[band+nsample]; + su1[band+nsample] = c2 << 1; + su2[band+nsample] = (c1 - c2) << 1; + } + break; + + case 1: + for (; nsample < 256; nsample++) { + c1 = su1[band+nsample]; + c2 = su2[band+nsample]; + su1[band+nsample] = (c1 + c2) << 1; + su2[band+nsample] = -1*(c2 << 1); + } + break; + case 2: + case 3: + for (; nsample < 256; nsample++) { + c1 = su1[band+nsample]; + c2 = su2[band+nsample]; + su1[band+nsample] = c1 + c2; + su2[band+nsample] = c1 - c2; + } + break; + default: + /* assert(0) */; + break; + } + } +} + +static void getChannelWeights (int indx, int flag, int32_t ch[2]){ + /* Read channel weights from table */ + if (flag) { + /* Swap channel weights */ + ch[1] = channelWeights0[indx&7]; + ch[0] = channelWeights1[indx&7]; + } else { + ch[0] = channelWeights0[indx&7]; + ch[1] = channelWeights1[indx&7]; + } +} + +static void channelWeighting (int32_t *su1, int32_t *su2, int *p3) +{ + int band, nsample; + /* w[x][y] y=0 is left y=1 is right */ + int32_t w[2][2]; + + if (p3[1] != 7 || p3[3] != 7){ + getChannelWeights(p3[1], p3[0], w[0]); + getChannelWeights(p3[3], p3[2], w[1]); + + for(band = 1; band < 4; band++) { + /* scale the channels by the weights */ + for(nsample = 0; nsample < 8; nsample++) { + su1[band*256+nsample] = fixmul16(su1[band*256+nsample], INTERPOLATE_FP16(w[0][0], w[0][1], nsample)); + su2[band*256+nsample] = fixmul16(su2[band*256+nsample], INTERPOLATE_FP16(w[1][0], w[1][1], nsample)); + } + + for(; nsample < 256; nsample++) { + su1[band*256+nsample] = fixmul16(su1[band*256+nsample], w[1][0]); + su2[band*256+nsample] = fixmul16(su2[band*256+nsample], w[1][1]); + } + } + } +} + +/** + * Decode a Sound Unit + * + * @param gb the GetBit context + * @param pSnd the channel unit to be used + * @param pOut the decoded samples before IQMF + * @param channelNum channel number + * @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono) + */ + +static int decodeChannelSoundUnit (GetBitContext *gb, channel_unit *pSnd, int32_t *pOut, int channelNum, int codingMode) +{ + int band, result=0, numSubbands, lastTonal, numBands; + if (codingMode == JOINT_STEREO && channelNum == 1) { + if (get_bits(gb,2) != 3) { + DEBUGF("JS mono Sound Unit id != 3.\n"); + return -1; + } + } else { + if (get_bits(gb,6) != 0x28) { + DEBUGF("Sound Unit id != 0x28.\n"); + return -1; + } + } + + /* number of coded QMF bands */ + pSnd->bandsCoded = get_bits(gb,2); + + result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded); + if (result) return result; + + pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded); + if (pSnd->numComponents == -1) return -1; + + numSubbands = decodeSpectrum (gb, pSnd->spectrum); + + /* Merge the decoded spectrum and tonal components. */ + lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components); + + + /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */ + numBands = (subbandTab[numSubbands] - 1) >> 8; + if (lastTonal >= 0) + numBands = FFMAX((lastTonal + 256) >> 8, numBands); + + /* Reconstruct time domain samples. */ + for (band=0; band<4; band++) { + /* Perform the IMDCT step without overlapping. */ + if (band <= numBands) { + IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf); + } else { + memset(pSnd->IMDCT_buf, 0, 512 * sizeof(int32_t)); + } + + /* gain compensation and overlapping */ + gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]), + &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]), + &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band])); + } + + /* Swap the gain control buffers for the next frame. */ + pSnd->gcBlkSwitch ^= 1; + + return 0; +} + +/** + * Frame handling + * + * @param q Atrac3 private context + * @param databuf the input data + */ + +static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf, int off) +{ + int result, i; + int32_t *p1, *p2, *p3, *p4; + uint8_t *ptr1; + + if (q->codingMode == JOINT_STEREO) { + + /* channel coupling mode */ + /* decode Sound Unit 1 */ + init_get_bits(&q->gb,databuf,q->bits_per_frame); + + result = decodeChannelSoundUnit(&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO); + if (result != 0) + return (result); + + /* Framedata of the su2 in the joint-stereo mode is encoded in + * reverse byte order so we need to swap it first. */ + if (databuf == q->decoded_bytes_buffer) { + uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1; + ptr1 = q->decoded_bytes_buffer; + for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) { + FFSWAP(uint8_t,*ptr1,*ptr2); + } + } else { + const uint8_t *ptr2 = databuf+q->bytes_per_frame-1; + for (i = 0; i < q->bytes_per_frame; i++) + q->decoded_bytes_buffer[i] = *ptr2--; + } + + /* Skip the sync codes (0xF8). */ + ptr1 = q->decoded_bytes_buffer; + for (i = 4; *ptr1 == 0xF8; i++, ptr1++) { + if (i >= q->bytes_per_frame) + return -1; + } + + + /* set the bitstream reader at the start of the second Sound Unit*/ + init_get_bits(&q->gb,ptr1,q->bits_per_frame); + + /* Fill the Weighting coeffs delay buffer */ + memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int)); + q->weighting_delay[4] = get_bits1(&q->gb); + q->weighting_delay[5] = get_bits(&q->gb,3); + + for (i = 0; i < 4; i++) { + q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i]; + q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i]; + q->matrix_coeff_index_next[i] = get_bits(&q->gb,2); + } + + /* Decode Sound Unit 2. */ + result = decodeChannelSoundUnit(&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO); + if (result != 0) + return (result); + + /* Reconstruct the channel coefficients. */ + reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now); + + channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay); + + } else { + /* normal stereo mode or mono */ + /* Decode the channel sound units. */ + for (i=0 ; ichannels ; i++) { + + /* Set the bitstream reader at the start of a channel sound unit. */ + init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels)+off, (q->bits_per_frame)/q->channels); + + result = decodeChannelSoundUnit(&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode); + if (result != 0) + return (result); + } + } + + /* Apply the iQMF synthesis filter. */ + p1= q->outSamples; + for (i=0 ; ichannels ; i++) { + p2= p1+256; + p3= p2+256; + p4= p3+256; + iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf); + iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf); + iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf); + p1 +=1024; + } + + return 0; +} + + +/** + * Atrac frame decoding + * + * @param rmctx pointer to the AVCodecContext + */ + +int atrac3_decode_frame(unsigned long block_align, ATRAC3Context *q, + int *data_size, const uint8_t *buf, int buf_size) { + int result = 0, off = 0; + const uint8_t* databuf; + + if ((unsigned)buf_size < block_align) + return buf_size; + + /* Check if we need to descramble and what buffer to pass on. */ + if (q->scrambled_stream) { + off = decode_bytes(buf, q->decoded_bytes_buffer, block_align); + databuf = q->decoded_bytes_buffer; + } else { + databuf = buf; + } + + result = decodeFrame(q, databuf, off); + + if (result != 0) { + DEBUGF("Frame decoding error!\n"); + return -1; + } + + if (q->channels == 1) + *data_size = 1024 * sizeof(int32_t); + else + *data_size = 2048 * sizeof(int32_t); + + return block_align; +} + + +/** + * Atrac3 initialization + * + * @param rmctx pointer to the RMContext + */ +int atrac3_decode_init(ATRAC3Context *q, struct mp3entry *id3) +{ + int i; + uint8_t *edata_ptr = (uint8_t*)&id3->id3v2buf; + +#if defined(CPU_COLDFIRE) + coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE); +#endif + + /* Take data from the RM container. */ + q->sample_rate = id3->frequency; + q->channels = id3->channels; + q->bit_rate = id3->bitrate * 1000; + q->bits_per_frame = id3->bytesperframe * 8; + q->bytes_per_frame = id3->bytesperframe; + + /* Take care of the codec-specific extradata. */ + + if (id3->extradata_size == 14) { + /* Parse the extradata, WAV format */ + DEBUGF("[0-1] %d\n",rm_get_uint16le(&edata_ptr[0])); /* Unknown value always 1 */ + q->samples_per_channel = rm_get_uint32le(&edata_ptr[2]); + q->codingMode = rm_get_uint16le(&edata_ptr[6]); + DEBUGF("[8-9] %d\n",rm_get_uint16le(&edata_ptr[8])); /* Dupe of coding mode */ + q->frame_factor = rm_get_uint16le(&edata_ptr[10]); /* Unknown always 1 */ + DEBUGF("[12-13] %d\n",rm_get_uint16le(&edata_ptr[12])); /* Unknown always 0 */ + + /* setup */ + q->samples_per_frame = 1024 * q->channels; + q->atrac3version = 4; + q->delay = 0x88E; + if (q->codingMode) + q->codingMode = JOINT_STEREO; + else + q->codingMode = STEREO; + q->scrambled_stream = 0; + + if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) { + } else { + DEBUGF("Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor); + return -1; + } + + } else if (id3->extradata_size == 10) { + /* Parse the extradata, RM format. */ + q->atrac3version = rm_get_uint32be(&edata_ptr[0]); + q->samples_per_frame = rm_get_uint16be(&edata_ptr[4]); + q->delay = rm_get_uint16be(&edata_ptr[6]); + q->codingMode = rm_get_uint16be(&edata_ptr[8]); + + q->samples_per_channel = q->samples_per_frame / q->channels; + q->scrambled_stream = 1; + + } else { + DEBUGF("Unknown extradata size %d.\n",id3->extradata_size); + } + /* Check the extradata. */ + + if (q->atrac3version != 4) { + DEBUGF("Version %d != 4.\n",q->atrac3version); + return -1; + } + + if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) { + DEBUGF("Unknown amount of samples per frame %d.\n",q->samples_per_frame); + return -1; + } + + if (q->delay != 0x88E) { + DEBUGF("Unknown amount of delay %x != 0x88E.\n",q->delay); + return -1; + } + + if (q->codingMode == STEREO) { + DEBUGF("Normal stereo detected.\n"); + } else if (q->codingMode == JOINT_STEREO) { + DEBUGF("Joint stereo detected.\n"); + } else { + DEBUGF("Unknown channel coding mode %x!\n",q->codingMode); + return -1; + } + + if (id3->channels <= 0 || id3->channels > 2 ) { + DEBUGF("Channel configuration error!\n"); + return -1; + } + + + if(id3->bytesperframe >= UINT16_MAX/2) + return -1; + + + /* Initialize the VLC tables. */ + if (!vlcs_initialized) { + for (i=0 ; i<7 ; i++) { + spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]]; + spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i]; + init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i], + huff_bits[i], 1, 1, + huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC); + } + + vlcs_initialized = 1; + + } + + init_atrac3_transforms(); + + /* init the joint-stereo decoding data */ + q->weighting_delay[0] = 0; + q->weighting_delay[1] = 7; + q->weighting_delay[2] = 0; + q->weighting_delay[3] = 7; + q->weighting_delay[4] = 0; + q->weighting_delay[5] = 7; + + for (i=0; i<4; i++) { + q->matrix_coeff_index_prev[i] = 3; + q->matrix_coeff_index_now[i] = 3; + q->matrix_coeff_index_next[i] = 3; + } + + /* Link the iram'ed arrays to the decoder's data structure */ + q->pUnits = channel_units; + q->pUnits[0].spectrum = &atrac3_spectrum [0][0]; + q->pUnits[1].spectrum = &atrac3_spectrum [1][0]; + q->pUnits[0].IMDCT_buf = &atrac3_IMDCT_buf[0][0]; + q->pUnits[1].IMDCT_buf = &atrac3_IMDCT_buf[1][0]; + q->pUnits[0].prevFrame = &atrac3_prevFrame[0][0]; + q->pUnits[1].prevFrame = &atrac3_prevFrame[1][0]; + + return 0; +} + -- cgit v1.2.3