diff options
Diffstat (limited to 'plugins/wma/libwma/wmadeci.c')
| -rw-r--r-- | plugins/wma/libwma/wmadeci.c | 1480 |
1 files changed, 1480 insertions, 0 deletions
diff --git a/plugins/wma/libwma/wmadeci.c b/plugins/wma/libwma/wmadeci.c new file mode 100644 index 00000000..d7a6f9ae --- /dev/null +++ b/plugins/wma/libwma/wmadeci.c @@ -0,0 +1,1480 @@ +/* + * WMA compatible decoder + * Copyright (c) 2002 The FFmpeg Project. + * + * This library 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 of the License, or (at your option) any later version. + * + * This library 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 this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +/** + * @file wmadec.c + * WMA compatible decoder. + */ + +#include <libasf/asf.h> +#include "wmadec.h" +#include "wmafixed.h" +#include "wmadata.h" + +#define trace(...) { fprintf (stderr, __VA_ARGS__); } +//#define trace(fmt,...) +#define DEBUGF trace + +static void wma_lsp_to_curve_init(WMADecodeContext *s, int frame_len); + +/*declarations of statically allocated variables used to remove malloc calls*/ + +# define IBSS_ATTR + +/*MDCT reconstruction windows*/ +static fixed32 stat0[2048] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR; +static fixed32 stat1[1024] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR; +static fixed32 stat2[ 512] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR; +static fixed32 stat3[ 256] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR; +static fixed32 stat4[ 128] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR; + +/*VLC lookup tables*/ +static uint16_t *runtabarray[2]; +static uint16_t *levtabarray[2]; + +static uint16_t runtab_big[1336] MEM_ALIGN_ATTR; +static uint16_t runtab_small[1072] MEM_ALIGN_ATTR; +static uint16_t levtab_big[1336] MEM_ALIGN_ATTR; +static uint16_t levtab_small[1072] MEM_ALIGN_ATTR; + +#define VLCBUF1SIZE 4598 +#define VLCBUF2SIZE 3574 +#define VLCBUF3SIZE 360 +#define VLCBUF4SIZE 540 + +/*putting these in IRAM actually makes PP slower*/ + +static VLC_TYPE vlcbuf1[VLCBUF1SIZE][2] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR; +static VLC_TYPE vlcbuf2[VLCBUF2SIZE][2] MEM_ALIGN_ATTR; +/* This buffer gets reused for lsp tables */ +static VLC_TYPE vlcbuf3[VLCBUF3SIZE][2] MEM_ALIGN_ATTR; +static VLC_TYPE vlcbuf4[VLCBUF4SIZE][2] MEM_ALIGN_ATTR; + + + + +/** + * Apply MDCT window and add into output. + * + * We ensure that when the windows overlap their squared sum + * is always 1 (MDCT reconstruction rule). + * + * The Vorbis I spec has a great diagram explaining this process. + * See section 1.3.2.3 of http://xiph.org/vorbis/doc/Vorbis_I_spec.html + */ + static void wma_window(WMADecodeContext *s, fixed32 *in, fixed32 *out) + { + //float *in = s->output; + int block_len, bsize, n; + + /* left part */ + + /* previous block was larger, so we'll use the size of the current + * block to set the window size*/ + if (s->block_len_bits <= s->prev_block_len_bits) { + block_len = s->block_len; + bsize = s->frame_len_bits - s->block_len_bits; + + vector_fmul_add_add(out, in, s->windows[bsize], block_len); + + } else { + /*previous block was smaller or the same size, so use it's size to set the window length*/ + block_len = 1 << s->prev_block_len_bits; + /*find the middle of the two overlapped blocks, this will be the first overlapped sample*/ + n = (s->block_len - block_len) / 2; + bsize = s->frame_len_bits - s->prev_block_len_bits; + + vector_fmul_add_add(out+n, in+n, s->windows[bsize], block_len); + + memcpy(out+n+block_len, in+n+block_len, n*sizeof(fixed32)); + } + /* Advance to the end of the current block and prepare to window it for the next block. + * Since the window function needs to be reversed, we do it backwards starting with the + * last sample and moving towards the first + */ + out += s->block_len; + in += s->block_len; + + /* right part */ + if (s->block_len_bits <= s->next_block_len_bits) { + block_len = s->block_len; + bsize = s->frame_len_bits - s->block_len_bits; + + vector_fmul_reverse(out, in, s->windows[bsize], block_len); + + } else { + block_len = 1 << s->next_block_len_bits; + n = (s->block_len - block_len) / 2; + bsize = s->frame_len_bits - s->next_block_len_bits; + + memcpy(out, in, n*sizeof(fixed32)); + + vector_fmul_reverse(out+n, in+n, s->windows[bsize], block_len); + + memset(out+n+block_len, 0, n*sizeof(fixed32)); + } + } + + + + +/* XXX: use same run/length optimization as mpeg decoders */ +static void init_coef_vlc(VLC *vlc, + uint16_t **prun_table, uint16_t **plevel_table, + const CoefVLCTable *vlc_table, int tab) +{ + int n = vlc_table->n; + const uint8_t *table_bits = vlc_table->huffbits; + const uint32_t *table_codes = vlc_table->huffcodes; + const uint16_t *levels_table = vlc_table->levels; + uint16_t *run_table, *level_table; + const uint16_t *p; + int i, l, j, level; + + + init_vlc(vlc, VLCBITS, n, table_bits, 1, 1, table_codes, 4, 4, INIT_VLC_USE_NEW_STATIC); + + run_table = runtabarray[tab]; + level_table= levtabarray[tab]; + + p = levels_table; + i = 2; + level = 1; + while (i < n) + { + l = *p++; + for(j=0;j<l;++j) + { + run_table[i] = j; + level_table[i] = level; + ++i; + } + ++level; + } + *prun_table = run_table; + *plevel_table = level_table; +} + +const uint8_t ff_log2_tab[256]={ + 0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4, + 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5, + 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6, + 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6, + 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, + 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, + 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, + 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7 +}; + + +#define av_log2 av_log2_c +static inline av_const int av_log2_c(unsigned int v) +{ + int n = 0; + if (v & 0xffff0000) { + v >>= 16; + n += 16; + } + if (v & 0xff00) { + v >>= 8; + n += 8; + } + n += ff_log2_tab[v]; + + return n; +} + + +int wma_decode_init(WMADecodeContext* s, asf_waveformatex_t *wfx) +{ + + int i, flags2; + fixed32 *window; + uint8_t *extradata; + fixed64 bps1; + fixed32 high_freq; + fixed64 bps; + int sample_rate1; + int coef_vlc_table; + // int filehandle; + #ifdef CPU_COLDFIRE + coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE); + #endif + + /*clear stereo setting to avoid glitches when switching stereo->mono*/ + s->channel_coded[0]=0; + s->channel_coded[1]=0; + s->ms_stereo=0; + + s->sample_rate = wfx->rate; + s->nb_channels = wfx->channels; + s->bit_rate = wfx->bitrate; + s->block_align = wfx->blockalign; + + if (wfx->codec_id == ASF_CODEC_ID_WMAV1) { + s->version = 1; + } else if (wfx->codec_id == ASF_CODEC_ID_WMAV2 ) { + s->version = 2; + } else { + /*one of those other wma flavors that don't have GPLed decoders */ + return -1; + } + + /* extract flag infos */ + flags2 = 0; + extradata = wfx->data; + if (s->version == 1 && wfx->datalen >= 4) { + flags2 = extradata[2] | (extradata[3] << 8); + }else if (s->version == 2 && wfx->datalen >= 6){ + flags2 = extradata[4] | (extradata[5] << 8); + } + s->use_exp_vlc = flags2 & 0x0001; + s->use_bit_reservoir = flags2 & 0x0002; + s->use_variable_block_len = flags2 & 0x0004; + + /* compute MDCT block size */ + if (s->sample_rate <= 16000){ + s->frame_len_bits = 9; + }else if (s->sample_rate <= 22050 || + (s->sample_rate <= 32000 && s->version == 1)){ + s->frame_len_bits = 10; + }else{ + s->frame_len_bits = 11; + } + s->frame_len = 1 << s->frame_len_bits; + if (s-> use_variable_block_len) + { + int nb_max, nb; + nb = ((flags2 >> 3) & 3) + 1; + if ((s->bit_rate / s->nb_channels) >= 32000) + { + nb += 2; + } + nb_max = s->frame_len_bits - BLOCK_MIN_BITS; //max is 11-7 + if (nb > nb_max) + nb = nb_max; + s->nb_block_sizes = nb + 1; + } + else + { + s->nb_block_sizes = 1; + } + + /* init rate dependant parameters */ + s->use_noise_coding = 1; + high_freq = itofix64(s->sample_rate) >> 1; + + + /* if version 2, then the rates are normalized */ + sample_rate1 = s->sample_rate; + if (s->version == 2) + { + if (sample_rate1 >= 44100) + sample_rate1 = 44100; + else if (sample_rate1 >= 22050) + sample_rate1 = 22050; + else if (sample_rate1 >= 16000) + sample_rate1 = 16000; + else if (sample_rate1 >= 11025) + sample_rate1 = 11025; + else if (sample_rate1 >= 8000) + sample_rate1 = 8000; + } + + fixed64 tmp = itofix64(s->bit_rate); + fixed64 tmp2 = itofix64(s->nb_channels * s->sample_rate); + bps = fixdiv64(tmp, tmp2); + fixed64 tim = bps * s->frame_len; + fixed64 tmpi = fixdiv64(tim,itofix64(8)); + s->byte_offset_bits = av_log2(fixtoi64(tmpi+0x8000)) + 2; + + /* compute high frequency value and choose if noise coding should + be activated */ + bps1 = bps; + if (s->nb_channels == 2) + bps1 = fixmul32(bps,0x1999a); + if (sample_rate1 == 44100) + { + if (bps1 >= 0x9c29) + s->use_noise_coding = 0; + else + high_freq = fixmul32(high_freq,0x6666); + } + else if (sample_rate1 == 22050) + { + if (bps1 >= 0x128f6) + s->use_noise_coding = 0; + else if (bps1 >= 0xb852) + high_freq = fixmul32(high_freq,0xb333); + else + high_freq = fixmul32(high_freq,0x999a); + } + else if (sample_rate1 == 16000) + { + if (bps > 0x8000) + high_freq = fixmul32(high_freq,0x8000); + else + high_freq = fixmul32(high_freq,0x4ccd); + } + else if (sample_rate1 == 11025) + { + high_freq = fixmul32(high_freq,0xb333); + } + else if (sample_rate1 == 8000) + { + if (bps <= 0xa000) + { + high_freq = fixmul32(high_freq,0x8000); + } + else if (bps > 0xc000) + { + s->use_noise_coding = 0; + } + else + { + high_freq = fixmul32(high_freq,0xa666); + } + } + else + { + if (bps >= 0xcccd) + { + high_freq = fixmul32(high_freq,0xc000); + } + else if (bps >= 0x999a) + { + high_freq = fixmul32(high_freq,0x999a); + } + else + { + high_freq = fixmul32(high_freq,0x8000); + } + } + + /* compute the scale factor band sizes for each MDCT block size */ + { + int a, b, pos, lpos, k, block_len, i, j, n; + const uint8_t *table; + + if (s->version == 1) + { + s->coefs_start = 3; + } + else + { + s->coefs_start = 0; + } + for(k = 0; k < s->nb_block_sizes; ++k) + { + block_len = s->frame_len >> k; + + if (s->version == 1) + { + lpos = 0; + for(i=0;i<25;++i) + { + a = wma_critical_freqs[i]; + b = s->sample_rate; + pos = ((block_len * 2 * a) + (b >> 1)) / b; + if (pos > block_len) + pos = block_len; + s->exponent_bands[0][i] = pos - lpos; + if (pos >= block_len) + { + ++i; + break; + } + lpos = pos; + } + s->exponent_sizes[0] = i; + } + else + { + /* hardcoded tables */ + table = NULL; + a = s->frame_len_bits - BLOCK_MIN_BITS - k; + if (a < 3) + { + if (s->sample_rate >= 44100) + table = exponent_band_44100[a]; + else if (s->sample_rate >= 32000) + table = exponent_band_32000[a]; + else if (s->sample_rate >= 22050) + table = exponent_band_22050[a]; + } + if (table) + { + n = *table++; + for(i=0;i<n;++i) + s->exponent_bands[k][i] = table[i]; + s->exponent_sizes[k] = n; + } + else + { + j = 0; + lpos = 0; + for(i=0;i<25;++i) + { + a = wma_critical_freqs[i]; + b = s->sample_rate; + pos = ((block_len * 2 * a) + (b << 1)) / (4 * b); + pos <<= 2; + if (pos > block_len) + pos = block_len; + if (pos > lpos) + s->exponent_bands[k][j++] = pos - lpos; + if (pos >= block_len) + break; + lpos = pos; + } + s->exponent_sizes[k] = j; + } + } + + /* max number of coefs */ + s->coefs_end[k] = (s->frame_len - ((s->frame_len * 9) / 100)) >> k; + /* high freq computation */ + + fixed32 tmp1 = high_freq*2; /* high_freq is a fixed32!*/ + fixed32 tmp2=itofix32(s->sample_rate>>1); + s->high_band_start[k] = fixtoi32( fixdiv32(tmp1, tmp2) * (block_len>>1) +0x8000); + + /* + s->high_band_start[k] = (int)((block_len * 2 * high_freq) / + s->sample_rate + 0.5);*/ + + n = s->exponent_sizes[k]; + j = 0; + pos = 0; + for(i=0;i<n;++i) + { + int start, end; + start = pos; + pos += s->exponent_bands[k][i]; + end = pos; + if (start < s->high_band_start[k]) + start = s->high_band_start[k]; + if (end > s->coefs_end[k]) + end = s->coefs_end[k]; + if (end > start) + s->exponent_high_bands[k][j++] = end - start; + } + s->exponent_high_sizes[k] = j; + } + } + + /* ffmpeg uses malloc to only allocate as many window sizes as needed. + * However, we're really only interested in the worst case memory usage. + * In the worst case you can have 5 window sizes, 128 doubling up 2048 + * Smaller windows are handled differently. + * Since we don't have malloc, just statically allocate this + */ + fixed32 *temp[5]; + temp[0] = stat0; + temp[1] = stat1; + temp[2] = stat2; + temp[3] = stat3; + temp[4] = stat4; + + /* init MDCT windows : simple sinus window */ + for(i = 0; i < s->nb_block_sizes; i++) + { + int n, j; + fixed32 alpha; + n = 1 << (s->frame_len_bits - i); + window = temp[i]; + + /* this calculates 0.5/(2*n) */ + alpha = (1<<15)>>(s->frame_len_bits - i+1); + for(j=0;j<n;++j) + { + fixed32 j2 = itofix32(j) + 0x8000; + /*alpha between 0 and pi/2*/ + window[j] = fsincos(fixmul32(j2,alpha)<<16, 0); + } + s->windows[i] = window; + + } + + s->reset_block_lengths = 1; + + if (s->use_noise_coding) /* init the noise generator */ + { + /* LSP values are simply 2x the EXP values */ + if (s->use_exp_vlc) + { + s->noise_mult = 0x51f; + /*unlikely, but we may have previoiusly used this table for LSP, + so halve the values if needed*/ + if(noisetable_exp[0] == 0x0a) { + for (i=0;i<NOISE_TAB_SIZE;++i) + noisetable_exp[i] >>= 1; + } + s->noise_table = noisetable_exp; + } + else + { + s->noise_mult = 0xa3d; + /*check that we haven't already doubled this table*/ + if(noisetable_exp[0] == 0x5) { + for (i=0;i<NOISE_TAB_SIZE;++i) + noisetable_exp[i] <<= 1; + } + s->noise_table = noisetable_exp; + } +#if 0 +/*TODO: Rockbox has a dither function. Consider using it for noise coding*/ + +/* We use a lookup table computered in advance, so no need to do this*/ + { + unsigned int seed; + fixed32 norm; + seed = 1; + norm = 0; // PJJ: near as makes any diff to 0! + for (i=0;i<NOISE_TAB_SIZE;++i) + { + seed = seed * 314159 + 1; + s->noise_table[i] = itofix32((int)seed) * norm; + } + } +#endif + + s->hgain_vlc.table = vlcbuf4; + s->hgain_vlc.table_allocated = VLCBUF4SIZE; + init_vlc(&s->hgain_vlc, HGAINVLCBITS, sizeof(hgain_huffbits), + hgain_huffbits, 1, 1, + hgain_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC); + } + + if (s->use_exp_vlc) + { + + s->exp_vlc.table = vlcbuf3; + s->exp_vlc.table_allocated = VLCBUF3SIZE; + + init_vlc(&s->exp_vlc, EXPVLCBITS, sizeof(scale_huffbits), + scale_huffbits, 1, 1, + scale_huffcodes, 4, 4, INIT_VLC_USE_NEW_STATIC); + } + else + { + wma_lsp_to_curve_init(s, s->frame_len); + } + + /* choose the VLC tables for the coefficients */ + coef_vlc_table = 2; + if (s->sample_rate >= 32000) + { + if (bps1 < 0xb852) + coef_vlc_table = 0; + else if (bps1 < 0x128f6) + coef_vlc_table = 1; + } + + /* since the coef2 table is the biggest and that has index 2 in coef_vlcs + it's safe to always assign like this */ + runtabarray[0] = runtab_big; runtabarray[1] = runtab_small; + levtabarray[0] = levtab_big; levtabarray[1] = levtab_small; + + s->coef_vlc[0].table = vlcbuf1; + s->coef_vlc[0].table_allocated = VLCBUF1SIZE; + s->coef_vlc[1].table = vlcbuf2; + s->coef_vlc[1].table_allocated = VLCBUF2SIZE; + + + init_coef_vlc(&s->coef_vlc[0], &s->run_table[0], &s->level_table[0], + &coef_vlcs[coef_vlc_table * 2], 0); + init_coef_vlc(&s->coef_vlc[1], &s->run_table[1], &s->level_table[1], + &coef_vlcs[coef_vlc_table * 2 + 1], 1); + + s->last_superframe_len = 0; + s->last_bitoffset = 0; + + return 0; +} + + +/* compute x^-0.25 with an exponent and mantissa table. We use linear + interpolation to reduce the mantissa table size at a small speed + expense (linear interpolation approximately doubles the number of + bits of precision). */ +static inline fixed32 pow_m1_4(WMADecodeContext *s, fixed32 x) +{ + union { + float f; + unsigned int v; + } u, t; + unsigned int e, m; + fixed32 a, b; + + u.f = fixtof64(x); + e = u.v >> 23; + m = (u.v >> (23 - LSP_POW_BITS)) & ((1 << LSP_POW_BITS) - 1); + /* build interpolation scale: 1 <= t < 2. */ + t.v = ((u.v << LSP_POW_BITS) & ((1 << 23) - 1)) | (127 << 23); + a = ((fixed32*)s->lsp_pow_m_table1)[m]; + b = ((fixed32*)s->lsp_pow_m_table2)[m]; + + /* lsp_pow_e_table contains 32.32 format */ + /* TODO: Since we're unlikely have value that cover the whole + * IEEE754 range, we probably don't need to have all possible exponents */ + + return (lsp_pow_e_table[e] * (a + fixmul32(b, ftofix32(t.f))) >>32); +} + +static void wma_lsp_to_curve_init(WMADecodeContext *s, int frame_len) +{ + fixed32 wdel, a, b, temp2; + int i; + + wdel = fixdiv32(itofix32(1), itofix32(frame_len)); + for (i=0; i<frame_len; ++i) + { + /* TODO: can probably reuse the trig_init values here */ + fsincos((wdel*i)<<15, &temp2); + /* get 3 bits headroom + 1 bit from not doubleing the values */ + s->lsp_cos_table[i] = temp2>>3; + + } + /* NOTE: these two tables are needed to avoid two operations in + pow_m1_4 */ + b = itofix32(1); + int ix = 0; + + s->lsp_pow_m_table1 = &vlcbuf3[0]; + s->lsp_pow_m_table2 = &vlcbuf3[1<<LSP_POW_BITS]; + + /*double check this later*/ + for(i=(1 << LSP_POW_BITS) - 1;i>=0;i--) + { + a = pow_a_table[ix++]<<4; + ((fixed32*)s->lsp_pow_m_table1)[i] = 2 * a - b; + ((fixed32*)s->lsp_pow_m_table2)[i] = b - a; + b = a; + } + +} + +/* NOTE: We use the same code as Vorbis here */ +/* XXX: optimize it further with SSE/3Dnow */ +static void wma_lsp_to_curve(WMADecodeContext *s, + fixed32 *out, + fixed32 *val_max_ptr, + int n, + fixed32 *lsp) +{ + int i, j; + fixed32 p, q, w, v, val_max, temp2; + + val_max = 0; + for(i=0;i<n;++i) + { + /* shift by 2 now to reduce rounding error, + * we can renormalize right before pow_m1_4 + */ + + p = 0x8000<<5; + q = 0x8000<<5; + w = s->lsp_cos_table[i]; + + for (j=1;j<NB_LSP_COEFS;j+=2) + { + /* w is 5.27 format, lsp is in 16.16, temp2 becomes 5.27 format */ + temp2 = ((w - (lsp[j - 1]<<11))); + + /* q is 16.16 format, temp2 is 5.27, q becomes 16.16 */ + q = fixmul32b(q, temp2 )<<4; + p = fixmul32b(p, (w - (lsp[j]<<11)))<<4; + } + + /* 2 in 5.27 format is 0x10000000 */ + p = fixmul32(p, fixmul32b(p, (0x10000000 - w)))<<3; + q = fixmul32(q, fixmul32b(q, (0x10000000 + w)))<<3; + + v = (p + q) >>9; /* p/q end up as 16.16 */ + v = pow_m1_4(s, v); + if (v > val_max) + val_max = v; + out[i] = v; + } + + *val_max_ptr = val_max; +} + +/* decode exponents coded with LSP coefficients (same idea as Vorbis) + * only used for low bitrate (< 16kbps) files + */ +static void decode_exp_lsp(WMADecodeContext *s, int ch) +{ + fixed32 lsp_coefs[NB_LSP_COEFS]; + int val, i; + + for (i = 0; i < NB_LSP_COEFS; ++i) + { + if (i == 0 || i >= 8) + val = get_bits(&s->gb, 3); + else + val = get_bits(&s->gb, 4); + lsp_coefs[i] = lsp_codebook[i][val]; + } + + wma_lsp_to_curve(s, + s->exponents[ch], + &s->max_exponent[ch], + s->block_len, + lsp_coefs); +} + +/* decode exponents coded with VLC codes - used for bitrate >= 32kbps*/ +static int decode_exp_vlc(WMADecodeContext *s, int ch) +{ + int last_exp, n, code; + const uint16_t *ptr, *band_ptr; + fixed32 v, max_scale; + fixed32 *q,*q_end; + + /*accommodate the 60 negative indices */ + const fixed32 *pow_10_to_yover16_ptr = &pow_10_to_yover16[61]; + + band_ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits]; + ptr = band_ptr; + q = s->exponents[ch]; + q_end = q + s->block_len; + max_scale = 0; + + + if (s->version == 1) //wmav1 only + { + last_exp = get_bits(&s->gb, 5) + 10; + + v = pow_10_to_yover16_ptr[last_exp]; + max_scale = v; + n = *ptr++; + switch (n & 3) do { + case 0: *q++ = v; + case 3: *q++ = v; + case 2: *q++ = v; + case 1: *q++ = v; + } while ((n -= 4) > 0); + } else { + last_exp = 36; + } + + while (q < q_end) + { + code = get_vlc2(&s->gb, s->exp_vlc.table, EXPVLCBITS, EXPMAX); + if (code < 0) + { + return -1; + } + /* NOTE: this offset is the same as MPEG4 AAC ! */ + last_exp += code - 60; + + v = pow_10_to_yover16_ptr[last_exp]; + if (v > max_scale) + { + max_scale = v; + } + n = *ptr++; + switch (n & 3) do { + case 0: *q++ = v; + case 3: *q++ = v; + case 2: *q++ = v; + case 1: *q++ = v; + } while ((n -= 4) > 0); + } + + s->max_exponent[ch] = max_scale; + return 0; +} + +/* return 0 if OK. return 1 if last block of frame. return -1 if + unrecorrable error. */ +static int wma_decode_block(WMADecodeContext *s) +{ + int n, v, a, ch, code, bsize; + int coef_nb_bits, total_gain; + int nb_coefs[MAX_CHANNELS]; + fixed32 mdct_norm; + + /*DEBUGF("***decode_block: %d (%d samples of %d in frame)\n", s->block_num, s->block_len, s->frame_len);*/ + + /* compute current block length */ + if (s->use_variable_block_len) + { + n = av_log2(s->nb_block_sizes - 1) + 1; + + if (s->reset_block_lengths) + { + s->reset_block_lengths = 0; + v = get_bits(&s->gb, n); + if (v >= s->nb_block_sizes) + { + return -2; + } + s->prev_block_len_bits = s->frame_len_bits - v; + v = get_bits(&s->gb, n); + if (v >= s->nb_block_sizes) + { + return -3; + } + s->block_len_bits = s->frame_len_bits - v; + } + else + { + /* update block lengths */ + s->prev_block_len_bits = s->block_len_bits; + s->block_len_bits = s->next_block_len_bits; + } + v = get_bits(&s->gb, n); + + if (v >= s->nb_block_sizes) + { + // rb->splash(HZ*4, "v was %d", v); //5, 7 + return -4; //this is it + } + else{ + //rb->splash(HZ, "passed v block (%d)!", v); + } + s->next_block_len_bits = s->frame_len_bits - v; + } + else + { + /* fixed block len */ + s->next_block_len_bits = s->frame_len_bits; + s->prev_block_len_bits = s->frame_len_bits; + s->block_len_bits = s->frame_len_bits; + } + /* now check if the block length is coherent with the frame length */ + s->block_len = 1 << s->block_len_bits; + + if ((s->block_pos + s->block_len) > s->frame_len) + { + return -5; //oddly 32k sample from tracker fails here + } + + if (s->nb_channels == 2) + { + s->ms_stereo = get_bits1(&s->gb); + } + v = 0; + for (ch = 0; ch < s->nb_channels; ++ch) + { + a = get_bits1(&s->gb); + s->channel_coded[ch] = a; + v |= a; + } + /* if no channel coded, no need to go further */ + /* XXX: fix potential framing problems */ + if (!v) + { + goto next; + } + + bsize = s->frame_len_bits - s->block_len_bits; + + /* read total gain and extract corresponding number of bits for + coef escape coding */ + total_gain = 1; + for(;;) + { + a = get_bits(&s->gb, 7); + total_gain += a; + if (a != 127) + { + break; + } + } + + if (total_gain < 15) + coef_nb_bits = 13; + else if (total_gain < 32) + coef_nb_bits = 12; + else if (total_gain < 40) + coef_nb_bits = 11; + else if (total_gain < 45) + coef_nb_bits = 10; + else + coef_nb_bits = 9; + + /* compute number of coefficients */ + n = s->coefs_end[bsize] - s->coefs_start; + + for(ch = 0; ch < s->nb_channels; ++ch) + { + nb_coefs[ch] = n; + } + /* complex coding */ + if (s->use_noise_coding) + { + + for(ch = 0; ch < s->nb_channels; ++ch) + { + if (s->channel_coded[ch]) + { + int i, n, a; + n = s->exponent_high_sizes[bsize]; + for(i=0;i<n;++i) + { + a = get_bits1(&s->gb); + s->high_band_coded[ch][i] = a; + /* if noise coding, the coefficients are not transmitted */ + if (a) + nb_coefs[ch] -= s->exponent_high_bands[bsize][i]; + } + } + } + for(ch = 0; ch < s->nb_channels; ++ch) + { + if (s->channel_coded[ch]) + { + int i, n, val, code; + + n = s->exponent_high_sizes[bsize]; + val = (int)0x80000000; + for(i=0;i<n;++i) + { + if (s->high_band_coded[ch][i]) + { + if (val == (int)0x80000000) + { + val = get_bits(&s->gb, 7) - 19; + } + else + { + //code = get_vlc(&s->gb, &s->hgain_vlc); + code = get_vlc2(&s->gb, s->hgain_vlc.table, HGAINVLCBITS, HGAINMAX); + if (code < 0) + { + return -6; + } + val += code - 18; + } + s->high_band_values[ch][i] = val; + } + } + } + } + } + + /* exponents can be reused in short blocks. */ + if ((s->block_len_bits == s->frame_len_bits) || get_bits1(&s->gb)) + { + for(ch = 0; ch < s->nb_channels; ++ch) + { + if (s->channel_coded[ch]) + { + if (s->use_exp_vlc) + { + if (decode_exp_vlc(s, ch) < 0) + { + return -7; + } + } + else + { + decode_exp_lsp(s, ch); + } + s->exponents_bsize[ch] = bsize; + } + } + } + + /* parse spectral coefficients : just RLE encoding */ + for(ch = 0; ch < s->nb_channels; ++ch) + { + if (s->channel_coded[ch]) + { + VLC *coef_vlc; + int level, run, sign, tindex; + int16_t *ptr, *eptr; + const int16_t *level_table, *run_table; + + /* special VLC tables are used for ms stereo because + there is potentially less energy there */ + tindex = (ch == 1 && s->ms_stereo); + coef_vlc = &s->coef_vlc[tindex]; + run_table = s->run_table[tindex]; + level_table = s->level_table[tindex]; + /* XXX: optimize */ + ptr = &s->coefs1[ch][0]; + eptr = ptr + nb_coefs[ch]; + memset(ptr, 0, s->block_len * sizeof(int16_t)); + + for(;;) + { + code = get_vlc2(&s->gb, coef_vlc->table, VLCBITS, VLCMAX); + + if (code < 0) + { + return -8; + } + if (code == 1) + { + /* EOB */ + break; + } + else if (code == 0) + { + /* escape */ + level = get_bits(&s->gb, coef_nb_bits); + /* NOTE: this is rather suboptimal. reading + block_len_bits would be better */ + run = get_bits(&s->gb, s->frame_len_bits); + } + else + { + /* normal code */ + run = run_table[code]; + level = level_table[code]; + } + sign = get_bits1(&s->gb); + if (!sign) + level = -level; + ptr += run; + if (ptr >= eptr) + { + break; + } + *ptr++ = level; + + + /* NOTE: EOB can be omitted */ + if (ptr >= eptr) + break; + } + } + if (s->version == 1 && s->nb_channels >= 2) + { + align_get_bits(&s->gb); + } + } + + { + int n4 = s->block_len >> 1; + + + mdct_norm = 0x10000>>(s->block_len_bits-1); + + if (s->version == 1) + { + mdct_norm *= fixtoi32(fixsqrt32(itofix32(n4))); + } + } + + + /* finally compute the MDCT coefficients */ + for(ch = 0; ch < s->nb_channels; ++ch) + { + if (s->channel_coded[ch]) + { + int16_t *coefs1; + fixed32 *exponents; + fixed32 *coefs, atemp; + fixed64 mult; + fixed64 mult1; + fixed32 noise, temp1, temp2, mult2; + int i, j, n, n1, last_high_band, esize; + fixed32 exp_power[HIGH_BAND_MAX_SIZE]; + + //total_gain, coefs1, mdctnorm are lossless + + coefs1 = s->coefs1[ch]; + exponents = s->exponents[ch]; + esize = s->exponents_bsize[ch]; + coefs = s->coefs[ch]; + n=0; + + /* + * The calculation of coefs has a shift right by 2 built in. This + * prepares samples for the Tremor IMDCT which uses a slightly + * different fixed format then the ffmpeg one. If the old ffmpeg + * imdct is used, each shift storing into coefs should be reduced + * by 1. + * See SVN logs for details. + */ + + + if (s->use_noise_coding) + { + /*This case is only used for low bitrates (typically less then 32kbps)*/ + + /*TODO: mult should be converted to 32 bit to speed up noise coding*/ + + mult = fixdiv64(pow_table[total_gain+20],Fixed32To64(s->max_exponent[ch])); + mult = mult* mdct_norm; + mult1 = mult; + + /* very low freqs : noise */ + for(i = 0;i < s->coefs_start; ++i) + { + *coefs++ = fixmul32( (fixmul32(s->noise_table[s->noise_index], + exponents[i<<bsize>>esize])>>4),Fixed32From64(mult1)) >>2; + s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1); + } + + n1 = s->exponent_high_sizes[bsize]; + + /* compute power of high bands */ + exponents = s->exponents[ch] +(s->high_band_start[bsize]<<bsize); + last_high_band = 0; /* avoid warning */ + for (j=0;j<n1;++j) + { + n = s->exponent_high_bands[s->frame_len_bits - + s->block_len_bits][j]; + if (s->high_band_coded[ch][j]) + { + fixed32 e2, v; + e2 = 0; + for(i = 0;i < n; ++i) + { + /*v is normalized later on so its fixed format is irrelevant*/ + v = exponents[i<<bsize>>esize]>>4; + e2 += fixmul32(v, v)>>3; + } + exp_power[j] = e2/n; /*n is an int...*/ + last_high_band = j; + } + exponents += n<<bsize; + } + + /* main freqs and high freqs */ + exponents = s->exponents[ch] + (s->coefs_start<<bsize); + for(j=-1;j<n1;++j) + { + if (j < 0) + { + n = s->high_band_start[bsize] - + s->coefs_start; + } + else + { + n = s->exponent_high_bands[s->frame_len_bits - + s->block_len_bits][j]; + } + if (j >= 0 && s->high_band_coded[ch][j]) + { + /* use noise with specified power */ + fixed32 tmp = fixdiv32(exp_power[j],exp_power[last_high_band]); + + /*mult1 is 48.16, pow_table is 48.16*/ + mult1 = fixmul32(fixsqrt32(tmp), + pow_table[s->high_band_values[ch][j]+20]) >> 16; + + /*this step has a fairly high degree of error for some reason*/ + mult1 = fixdiv64(mult1,fixmul32(s->max_exponent[ch],s->noise_mult)); + mult1 = mult1*mdct_norm>>PRECISION; + for(i = 0;i < n; ++i) + { + noise = s->noise_table[s->noise_index]; + s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1); + *coefs++ = fixmul32((fixmul32(exponents[i<<bsize>>esize],noise)>>4), + Fixed32From64(mult1)) >>2; + + } + exponents += n<<bsize; + } + else + { + /* coded values + small noise */ + for(i = 0;i < n; ++i) + { + noise = s->noise_table[s->noise_index]; + s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1); + + /*don't forget to renormalize the noise*/ + temp1 = (((int32_t)*coefs1++)<<16) + (noise>>4); + temp2 = fixmul32(exponents[i<<bsize>>esize], mult>>18); + *coefs++ = fixmul32(temp1, temp2); + } + exponents += n<<bsize; + } + } + + /* very high freqs : noise */ + n = s->block_len - s->coefs_end[bsize]; + mult2 = fixmul32(mult>>16,exponents[((-1<<bsize))>>esize]) ; + for (i = 0; i < n; ++i) + { + /*renormalize the noise product and then reduce to 14.18 precison*/ + *coefs++ = fixmul32(s->noise_table[s->noise_index],mult2) >>6; + + s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1); + } + } + else + { + /*Noise coding not used, simply convert from exp to fixed representation*/ + + fixed32 mult3 = (fixed32)(fixdiv64(pow_table[total_gain+20], + Fixed32To64(s->max_exponent[ch]))); + mult3 = fixmul32(mult3, mdct_norm); + + /*zero the first 3 coefficients for WMA V1, does nothing otherwise*/ + for(i=0; i<s->coefs_start; i++) + *coefs++=0; + + n = nb_coefs[ch]; + + /* XXX: optimize more, unrolling this loop in asm + might be a good idea */ + + for(i = 0;i < n; ++i) + { + /*ffmpeg imdct needs 15.17, while tremor 14.18*/ + atemp = (coefs1[i] * mult3)>>2; + *coefs++=fixmul32(atemp,exponents[i<<bsize>>esize]); + } + n = s->block_len - s->coefs_end[bsize]; + memset(coefs, 0, n*sizeof(fixed32)); + } + } + } + + + + if (s->ms_stereo && s->channel_coded[1]) + { + fixed32 a, b; + int i; + /* nominal case for ms stereo: we do it before mdct */ + /* no need to optimize this case because it should almost + never happen */ + if (!s->channel_coded[0]) + { + memset(s->coefs[0], 0, sizeof(fixed32) * s->block_len); + s->channel_coded[0] = 1; + } + + for(i = 0; i < s->block_len; ++i) + { + a = s->coefs[0][i]; + b = s->coefs[1][i]; + s->coefs[0][i] = a + b; + s->coefs[1][i] = a - b; + } + } + + for(ch = 0; ch < s->nb_channels; ++ch) + { + /* BLOCK_MAX_SIZE is 2048 (samples) and MAX_CHANNELS is 2. */ + static uint32_t scratch_buf[BLOCK_MAX_SIZE * MAX_CHANNELS] IBSS_ATTR MEM_ALIGN_ATTR; + if (s->channel_coded[ch]) + { + int n4, index; + + n4 = s->block_len >>1; + + ff_imdct_calc((s->frame_len_bits - bsize + 1), + scratch_buf, + s->coefs[ch]); + + /* add in the frame */ + index = (s->frame_len / 2) + s->block_pos - n4; + wma_window(s, scratch_buf, &(s->frame_out[ch][index])); + + + + /* specific fast case for ms-stereo : add to second + channel if it is not coded */ + if (s->ms_stereo && !s->channel_coded[1]) + { + wma_window(s, scratch_buf, &(s->frame_out[1][index])); + } + } + } +next: + /* update block number */ + ++s->block_num; + s->block_pos += s->block_len; + if (s->block_pos >= s->frame_len) + { + return 1; + } + else + { + return 0; + } +} + +/* decode a frame of frame_len samples */ +static int wma_decode_frame(WMADecodeContext *s) +{ + int ret; + + /* read each block */ + s->block_num = 0; + s->block_pos = 0; + + + for(;;) + { + ret = wma_decode_block(s); + if (ret < 0) + { + + DEBUGF("wma_decode_block failed with code %d\n", ret); + return -1; + } + if (ret) + { + break; + } + } + + return 0; +} + +/* Initialise the superframe decoding */ + +int wma_decode_superframe_init(WMADecodeContext* s, + const uint8_t *buf, /*input*/ + int buf_size) +{ + if (buf_size==0) + { + s->last_superframe_len = 0; + return 0; + } + + s->current_frame = 0; + + init_get_bits(&s->gb, buf, buf_size*8); + + if (s->use_bit_reservoir) + { + /* read super frame header */ + skip_bits(&s->gb, 4); /* super frame index */ + s->nb_frames = get_bits(&s->gb, 4); + + if (s->last_superframe_len == 0) + s->nb_frames --; + else if (s->nb_frames == 0) + s->nb_frames++; + + s->bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3); + } else { + s->nb_frames = 1; + } + + return 1; +} + + +/* Decode a single frame in the current superframe - return -1 if + there was a decoding error, or the number of samples decoded. +*/ + +int wma_decode_superframe_frame(WMADecodeContext* s, + const uint8_t *buf, /*input*/ + int buf_size) +{ + int pos, len, ch; + uint8_t *q; + int done = 0; + + for(ch = 0; ch < s->nb_channels; ch++) + memmove(&(s->frame_out[ch][0]), + &(s->frame_out[ch][s->frame_len]), + s->frame_len * sizeof(fixed32)); + + if ((s->use_bit_reservoir) && (s->current_frame == 0)) + { + if (s->last_superframe_len > 0) + { + /* add s->bit_offset bits to last frame */ + if ((s->last_superframe_len + ((s->bit_offset + 7) >> 3)) > + MAX_CODED_SUPERFRAME_SIZE) + { + DEBUGF("superframe size too large error\n"); + goto fail; + } + q = s->last_superframe + s->last_superframe_len; + len = s->bit_offset; + while (len > 7) + { + *q++ = (get_bits)(&s->gb, 8); + len -= 8; + } + if (len > 0) + { + *q++ = (get_bits)(&s->gb, len) << (8 - len); + } + + /* XXX: s->bit_offset bits into last frame */ + init_get_bits(&s->gb, s->last_superframe, MAX_CODED_SUPERFRAME_SIZE*8); + /* skip unused bits */ + if (s->last_bitoffset > 0) + skip_bits(&s->gb, s->last_bitoffset); + + /* this frame is stored in the last superframe and in the + current one */ + if (wma_decode_frame(s) < 0) + { + goto fail; + } + done = 1; + } + + /* read each frame starting from s->bit_offset */ + pos = s->bit_offset + 4 + 4 + s->byte_offset_bits + 3; + init_get_bits(&s->gb, buf + (pos >> 3), (MAX_CODED_SUPERFRAME_SIZE - (pos >> 3))*8); + len = pos & 7; + if (len > 0) + skip_bits(&s->gb, len); + + s->reset_block_lengths = 1; + } + + /* If we haven't decoded a frame yet, do it now */ + if (!done) + { + if (wma_decode_frame(s) < 0) + { + goto fail; + } + } + + s->current_frame++; + + if ((s->use_bit_reservoir) && (s->current_frame == s->nb_frames)) + { + /* we copy the end of the frame in the last frame buffer */ + pos = get_bits_count(&s->gb) + ((s->bit_offset + 4 + 4 + s->byte_offset_bits + 3) & ~7); + s->last_bitoffset = pos & 7; + pos >>= 3; + len = buf_size - pos; + if (len > MAX_CODED_SUPERFRAME_SIZE || len < 0) + { + DEBUGF("superframe size too large error after decoding\n"); + goto fail; + } + s->last_superframe_len = len; + memcpy(s->last_superframe, buf + pos, len); + } + + return s->frame_len; + +fail: + /* when error, we reset the bit reservoir */ + + s->last_superframe_len = 0; + return -1; +} + |