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  1. /*
  2.  * Copyright (c) 2012
  3.  *      MIPS Technologies, Inc., California.
  4.  *
  5.  * Redistribution and use in source and binary forms, with or without
  6.  * modification, are permitted provided that the following conditions
  7.  * are met:
  8.  * 1. Redistributions of source code must retain the above copyright
  9.  *    notice, this list of conditions and the following disclaimer.
  10.  * 2. Redistributions in binary form must reproduce the above copyright
  11.  *    notice, this list of conditions and the following disclaimer in the
  12.  *    documentation and/or other materials provided with the distribution.
  13.  * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
  14.  *    contributors may be used to endorse or promote products derived from
  15.  *    this software without specific prior written permission.
  16.  *
  17.  * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
  18.  * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
  19.  * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
  20.  * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
  21.  * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
  22.  * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
  23.  * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
  24.  * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
  25.  * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
  26.  * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
  27.  * SUCH DAMAGE.
  28.  *
  29.  * Author:  Stanislav Ocovaj (socovaj@mips.com)
  30.  *
  31.  * AC3 fixed-point decoder for MIPS platforms
  32.  *
  33.  * This file is part of FFmpeg.
  34.  *
  35.  * FFmpeg is free software; you can redistribute it and/or
  36.  * modify it under the terms of the GNU Lesser General Public
  37.  * License as published by the Free Software Foundation; either
  38.  * version 2.1 of the License, or (at your option) any later version.
  39.  *
  40.  * FFmpeg is distributed in the hope that it will be useful,
  41.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  42.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  43.  * Lesser General Public License for more details.
  44.  *
  45.  * You should have received a copy of the GNU Lesser General Public
  46.  * License along with FFmpeg; if not, write to the Free Software
  47.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  48.  */
  49.  
  50. #define FFT_FLOAT 0
  51. #define USE_FIXED 1
  52. #define FFT_FIXED_32 1
  53. #include "ac3dec.h"
  54.  
  55.  
  56. static const int end_freq_inv_tab[8] =
  57. {
  58.     50529027, 44278013, 39403370, 32292987, 27356480, 23729101, 20951060, 18755316
  59. };
  60.  
  61. static void scale_coefs (
  62.     int32_t *dst,
  63.     const int32_t *src,
  64.     int dynrng,
  65.     int len)
  66. {
  67.     int i, shift, round;
  68.     int16_t mul;
  69.     int temp, temp1, temp2, temp3, temp4, temp5, temp6, temp7;
  70.  
  71.     mul = (dynrng & 0x1f) + 0x20;
  72.     shift = 4 - ((dynrng << 23) >> 28);
  73.     if (shift > 0 ) {
  74.       round = 1 << (shift-1);
  75.       for (i=0; i<len; i+=8) {
  76.  
  77.           temp = src[i] * mul;
  78.           temp1 = src[i+1] * mul;
  79.           temp = temp + round;
  80.           temp2 = src[i+2] * mul;
  81.  
  82.           temp1 = temp1 + round;
  83.           dst[i] = temp >> shift;
  84.           temp3 = src[i+3] * mul;
  85.           temp2 = temp2 + round;
  86.  
  87.           dst[i+1] = temp1 >> shift;
  88.           temp4 = src[i + 4] * mul;
  89.           temp3 = temp3 + round;
  90.           dst[i+2] = temp2 >> shift;
  91.  
  92.           temp5 = src[i+5] * mul;
  93.           temp4 = temp4 + round;
  94.           dst[i+3] = temp3 >> shift;
  95.           temp6 = src[i+6] * mul;
  96.  
  97.           dst[i+4] = temp4 >> shift;
  98.           temp5 = temp5 + round;
  99.           temp7 = src[i+7] * mul;
  100.           temp6 = temp6 + round;
  101.  
  102.           dst[i+5] = temp5 >> shift;
  103.           temp7 = temp7 + round;
  104.           dst[i+6] = temp6 >> shift;
  105.           dst[i+7] = temp7 >> shift;
  106.  
  107.       }
  108.     } else {
  109.       shift = -shift;
  110.       for (i=0; i<len; i+=8) {
  111.  
  112.           temp = src[i] * mul;
  113.           temp1 = src[i+1] * mul;
  114.           temp2 = src[i+2] * mul;
  115.  
  116.           dst[i] = temp << shift;
  117.           temp3 = src[i+3] * mul;
  118.  
  119.           dst[i+1] = temp1 << shift;
  120.           temp4 = src[i + 4] * mul;
  121.           dst[i+2] = temp2 << shift;
  122.  
  123.           temp5 = src[i+5] * mul;
  124.           dst[i+3] = temp3 << shift;
  125.           temp6 = src[i+6] * mul;
  126.  
  127.           dst[i+4] = temp4 << shift;
  128.           temp7 = src[i+7] * mul;
  129.  
  130.           dst[i+5] = temp5 << shift;
  131.           dst[i+6] = temp6 << shift;
  132.           dst[i+7] = temp7 << shift;
  133.  
  134.       }
  135.     }
  136. }
  137.  
  138. /**
  139.  * Downmix samples from original signal to stereo or mono (this is for 16-bit samples
  140.  * and fixed point decoder - original (for 32-bit samples) is in ac3dsp.c).
  141.  */
  142. static void ac3_downmix_c_fixed16(int16_t **samples, int16_t (*matrix)[2],
  143.                                   int out_ch, int in_ch, int len)
  144. {
  145.     int i, j;
  146.     int v0, v1;
  147.     if (out_ch == 2) {
  148.         for (i = 0; i < len; i++) {
  149.             v0 = v1 = 0;
  150.             for (j = 0; j < in_ch; j++) {
  151.                 v0 += samples[j][i] * matrix[j][0];
  152.                 v1 += samples[j][i] * matrix[j][1];
  153.             }
  154.             samples[0][i] = (v0+2048)>>12;
  155.             samples[1][i] = (v1+2048)>>12;
  156.         }
  157.     } else if (out_ch == 1) {
  158.         for (i = 0; i < len; i++) {
  159.             v0 = 0;
  160.             for (j = 0; j < in_ch; j++)
  161.                 v0 += samples[j][i] * matrix[j][0];
  162.             samples[0][i] = (v0+2048)>>12;
  163.         }
  164.     }
  165. }
  166.  
  167. #include "eac3dec.c"
  168. #include "ac3dec.c"
  169.  
  170. static const AVOption options[] = {
  171.     { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 6.0, PAR },
  172.     { "heavy_compr", "heavy dynamic range compression enabled", OFFSET(heavy_compression), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, PAR },
  173.     { NULL},
  174. };
  175.  
  176. static const AVClass ac3_decoder_class = {
  177.     .class_name = "Fixed-Point AC-3 Decoder",
  178.     .item_name  = av_default_item_name,
  179.     .option     = options,
  180.     .version    = LIBAVUTIL_VERSION_INT,
  181. };
  182.  
  183. AVCodec ff_ac3_fixed_decoder = {
  184.     .name           = "ac3_fixed",
  185.     .type           = AVMEDIA_TYPE_AUDIO,
  186.     .id             = AV_CODEC_ID_AC3,
  187.     .priv_data_size = sizeof (AC3DecodeContext),
  188.     .init           = ac3_decode_init,
  189.     .close          = ac3_decode_end,
  190.     .decode         = ac3_decode_frame,
  191.     .capabilities   = AV_CODEC_CAP_DR1,
  192.     .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
  193.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
  194.                                                       AV_SAMPLE_FMT_NONE },
  195.     .priv_class     = &ac3_decoder_class,
  196. };
  197.