Libav
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00001 /* 00002 * Atrac 1 compatible decoder 00003 * Copyright (c) 2009 Maxim Poliakovski 00004 * Copyright (c) 2009 Benjamin Larsson 00005 * 00006 * This file is part of FFmpeg. 00007 * 00008 * FFmpeg is free software; you can redistribute it and/or 00009 * modify it under the terms of the GNU Lesser General Public 00010 * License as published by the Free Software Foundation; either 00011 * version 2.1 of the License, or (at your option) any later version. 00012 * 00013 * FFmpeg is distributed in the hope that it will be useful, 00014 * but WITHOUT ANY WARRANTY; without even the implied warranty of 00015 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 00016 * Lesser General Public License for more details. 00017 * 00018 * You should have received a copy of the GNU Lesser General Public 00019 * License along with FFmpeg; if not, write to the Free Software 00020 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 00021 */ 00022 00029 /* Many thanks to Tim Craig for all the help! */ 00030 00031 #include <math.h> 00032 #include <stddef.h> 00033 #include <stdio.h> 00034 00035 #include "avcodec.h" 00036 #include "get_bits.h" 00037 #include "dsputil.h" 00038 #include "fft.h" 00039 00040 #include "atrac.h" 00041 #include "atrac1data.h" 00042 00043 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit 00044 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit 00045 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit 00046 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2 00047 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 00048 #define AT1_MAX_CHANNELS 2 00049 00050 #define AT1_QMF_BANDS 3 00051 #define IDX_LOW_BAND 0 00052 #define IDX_MID_BAND 1 00053 #define IDX_HIGH_BAND 2 00054 00058 typedef struct { 00059 int log2_block_count[AT1_QMF_BANDS]; 00060 int num_bfus; 00061 float* spectrum[2]; 00062 DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES]; 00063 DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES]; 00064 DECLARE_ALIGNED(16, float, fst_qmf_delay)[46]; 00065 DECLARE_ALIGNED(16, float, snd_qmf_delay)[46]; 00066 DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23]; 00067 } AT1SUCtx; 00068 00072 typedef struct { 00073 AT1SUCtx SUs[AT1_MAX_CHANNELS]; 00074 DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES]; 00075 00076 DECLARE_ALIGNED(16, float, low)[256]; 00077 DECLARE_ALIGNED(16, float, mid)[256]; 00078 DECLARE_ALIGNED(16, float, high)[512]; 00079 float* bands[3]; 00080 DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]; 00081 FFTContext mdct_ctx[3]; 00082 int channels; 00083 DSPContext dsp; 00084 } AT1Ctx; 00085 00087 static const uint16_t samples_per_band[3] = {128, 128, 256}; 00088 static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; 00089 00090 00091 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, 00092 int rev_spec) 00093 { 00094 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; 00095 int transf_size = 1 << nbits; 00096 00097 if (rev_spec) { 00098 int i; 00099 for (i = 0; i < transf_size / 2; i++) 00100 FFSWAP(float, spec[i], spec[transf_size - 1 - i]); 00101 } 00102 ff_imdct_half(mdct_context, out, spec); 00103 } 00104 00105 00106 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) 00107 { 00108 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; 00109 unsigned int start_pos, ref_pos = 0, pos = 0; 00110 00111 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { 00112 float *prev_buf; 00113 int j; 00114 00115 band_samples = samples_per_band[band_num]; 00116 log2_block_count = su->log2_block_count[band_num]; 00117 00118 /* number of mdct blocks in the current QMF band: 1 - for long mode */ 00119 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/ 00120 num_blocks = 1 << log2_block_count; 00121 00122 if (num_blocks == 1) { 00123 /* mdct block size in samples: 128 (long mode, low & mid bands), */ 00124 /* 256 (long mode, high band) and 32 (short mode, all bands) */ 00125 block_size = band_samples >> log2_block_count; 00126 00127 /* calc transform size in bits according to the block_size_mode */ 00128 nbits = mdct_long_nbits[band_num] - log2_block_count; 00129 00130 if (nbits != 5 && nbits != 7 && nbits != 8) 00131 return -1; 00132 } else { 00133 block_size = 32; 00134 nbits = 5; 00135 } 00136 00137 start_pos = 0; 00138 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16]; 00139 for (j=0; j < num_blocks; j++) { 00140 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num); 00141 00142 /* overlap and window */ 00143 q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, 00144 &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16); 00145 00146 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; 00147 start_pos += block_size; 00148 pos += block_size; 00149 } 00150 00151 if (num_blocks == 1) 00152 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); 00153 00154 ref_pos += band_samples; 00155 } 00156 00157 /* Swap buffers so the mdct overlap works */ 00158 FFSWAP(float*, su->spectrum[0], su->spectrum[1]); 00159 00160 return 0; 00161 } 00162 00167 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) 00168 { 00169 int log2_block_count_tmp, i; 00170 00171 for (i = 0; i < 2; i++) { 00172 /* low and mid band */ 00173 log2_block_count_tmp = get_bits(gb, 2); 00174 if (log2_block_count_tmp & 1) 00175 return -1; 00176 log2_block_cnt[i] = 2 - log2_block_count_tmp; 00177 } 00178 00179 /* high band */ 00180 log2_block_count_tmp = get_bits(gb, 2); 00181 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) 00182 return -1; 00183 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; 00184 00185 skip_bits(gb, 2); 00186 return 0; 00187 } 00188 00189 00190 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, 00191 float spec[AT1_SU_SAMPLES]) 00192 { 00193 int bits_used, band_num, bfu_num, i; 00194 uint8_t idwls[AT1_MAX_BFU]; 00195 uint8_t idsfs[AT1_MAX_BFU]; 00196 00197 /* parse the info byte (2nd byte) telling how much BFUs were coded */ 00198 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)]; 00199 00200 /* calc number of consumed bits: 00201 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits) 00202 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */ 00203 bits_used = su->num_bfus * 10 + 32 + 00204 bfu_amount_tab2[get_bits(gb, 2)] + 00205 (bfu_amount_tab3[get_bits(gb, 3)] << 1); 00206 00207 /* get word length index (idwl) for each BFU */ 00208 for (i = 0; i < su->num_bfus; i++) 00209 idwls[i] = get_bits(gb, 4); 00210 00211 /* get scalefactor index (idsf) for each BFU */ 00212 for (i = 0; i < su->num_bfus; i++) 00213 idsfs[i] = get_bits(gb, 6); 00214 00215 /* zero idwl/idsf for empty BFUs */ 00216 for (i = su->num_bfus; i < AT1_MAX_BFU; i++) 00217 idwls[i] = idsfs[i] = 0; 00218 00219 /* read in the spectral data and reconstruct MDCT spectrum of this channel */ 00220 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { 00221 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) { 00222 int pos; 00223 00224 int num_specs = specs_per_bfu[bfu_num]; 00225 int word_len = !!idwls[bfu_num] + idwls[bfu_num]; 00226 float scale_factor = sf_table[idsfs[bfu_num]]; 00227 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ 00228 00229 /* check for bitstream overflow */ 00230 if (bits_used > AT1_SU_MAX_BITS) 00231 return -1; 00232 00233 /* get the position of the 1st spec according to the block size mode */ 00234 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; 00235 00236 if (word_len) { 00237 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); 00238 00239 for (i = 0; i < num_specs; i++) { 00240 /* read in a quantized spec and convert it to 00241 * signed int and then inverse quantization 00242 */ 00243 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant; 00244 } 00245 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */ 00246 memset(&spec[pos], 0, num_specs * sizeof(float)); 00247 } 00248 } 00249 } 00250 00251 return 0; 00252 } 00253 00254 00255 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) 00256 { 00257 float temp[256]; 00258 float iqmf_temp[512 + 46]; 00259 00260 /* combine low and middle bands */ 00261 atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); 00262 00263 /* delay the signal of the high band by 23 samples */ 00264 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23); 00265 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256); 00266 00267 /* combine (low + middle) and high bands */ 00268 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); 00269 } 00270 00271 00272 static int atrac1_decode_frame(AVCodecContext *avctx, void *data, 00273 int *data_size, AVPacket *avpkt) 00274 { 00275 const uint8_t *buf = avpkt->data; 00276 int buf_size = avpkt->size; 00277 AT1Ctx *q = avctx->priv_data; 00278 int ch, ret, i; 00279 GetBitContext gb; 00280 float* samples = data; 00281 00282 00283 if (buf_size < 212 * q->channels) { 00284 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n"); 00285 return -1; 00286 } 00287 00288 for (ch = 0; ch < q->channels; ch++) { 00289 AT1SUCtx* su = &q->SUs[ch]; 00290 00291 init_get_bits(&gb, &buf[212 * ch], 212 * 8); 00292 00293 /* parse block_size_mode, 1st byte */ 00294 ret = at1_parse_bsm(&gb, su->log2_block_count); 00295 if (ret < 0) 00296 return ret; 00297 00298 ret = at1_unpack_dequant(&gb, su, q->spec); 00299 if (ret < 0) 00300 return ret; 00301 00302 ret = at1_imdct_block(su, q); 00303 if (ret < 0) 00304 return ret; 00305 at1_subband_synthesis(q, su, q->out_samples[ch]); 00306 } 00307 00308 /* interleave; FIXME, should create/use a DSP function */ 00309 if (q->channels == 1) { 00310 /* mono */ 00311 memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4); 00312 } else { 00313 /* stereo */ 00314 for (i = 0; i < AT1_SU_SAMPLES; i++) { 00315 samples[i * 2] = q->out_samples[0][i]; 00316 samples[i * 2 + 1] = q->out_samples[1][i]; 00317 } 00318 } 00319 00320 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples); 00321 return avctx->block_align; 00322 } 00323 00324 00325 static av_cold int atrac1_decode_init(AVCodecContext *avctx) 00326 { 00327 AT1Ctx *q = avctx->priv_data; 00328 00329 avctx->sample_fmt = SAMPLE_FMT_FLT; 00330 00331 q->channels = avctx->channels; 00332 00333 /* Init the mdct transforms */ 00334 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15)); 00335 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15)); 00336 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)); 00337 00338 ff_init_ff_sine_windows(5); 00339 00340 atrac_generate_tables(); 00341 00342 dsputil_init(&q->dsp, avctx); 00343 00344 q->bands[0] = q->low; 00345 q->bands[1] = q->mid; 00346 q->bands[2] = q->high; 00347 00348 /* Prepare the mdct overlap buffers */ 00349 q->SUs[0].spectrum[0] = q->SUs[0].spec1; 00350 q->SUs[0].spectrum[1] = q->SUs[0].spec2; 00351 q->SUs[1].spectrum[0] = q->SUs[1].spec1; 00352 q->SUs[1].spectrum[1] = q->SUs[1].spec2; 00353 00354 return 0; 00355 } 00356 00357 00358 static av_cold int atrac1_decode_end(AVCodecContext * avctx) { 00359 AT1Ctx *q = avctx->priv_data; 00360 00361 ff_mdct_end(&q->mdct_ctx[0]); 00362 ff_mdct_end(&q->mdct_ctx[1]); 00363 ff_mdct_end(&q->mdct_ctx[2]); 00364 return 0; 00365 } 00366 00367 00368 AVCodec atrac1_decoder = { 00369 .name = "atrac1", 00370 .type = AVMEDIA_TYPE_AUDIO, 00371 .id = CODEC_ID_ATRAC1, 00372 .priv_data_size = sizeof(AT1Ctx), 00373 .init = atrac1_decode_init, 00374 .close = atrac1_decode_end, 00375 .decode = atrac1_decode_frame, 00376 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), 00377 };