/* * The simplest AC-3 encoder * Copyright (c) 2000 Fabrice Bellard * Copyright (c) 2006-2010 Justin Ruggles <justin.ruggles@gmail.com> * Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de> * * 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 * fixed-point AC-3 encoder. */ #define FFT_FLOAT 0 #undef CONFIG_AC3ENC_FLOAT #include "internal.h" #include "audiodsp.h" #include "ac3enc.h" #include "eac3enc.h" #define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED #include "ac3enc_opts_template.c" static const AVClass ac3enc_class = { .class_name = "Fixed-Point AC-3 Encoder", .item_name = av_default_item_name, .option = ac3_options, .version = LIBAVUTIL_VERSION_INT, }; /* * Normalize the input samples to use the maximum available precision. * This assumes signed 16-bit input samples. */ static int normalize_samples(AC3EncodeContext *s) { int v = s->ac3dsp.ac3_max_msb_abs_int16(s->windowed_samples, AC3_WINDOW_SIZE); v = 14 - av_log2(v); if (v > 0) s->ac3dsp.ac3_lshift_int16(s->windowed_samples, AC3_WINDOW_SIZE, v); /* +6 to right-shift from 31-bit to 25-bit */ return v + 6; } /* * Scale MDCT coefficients to 25-bit signed fixed-point. */ static void scale_coefficients(AC3EncodeContext *s) { int blk, ch; for (blk = 0; blk < s->num_blocks; blk++) { AC3Block *block = &s->blocks[blk]; for (ch = 1; ch <= s->channels; ch++) { s->ac3dsp.ac3_rshift_int32(block->mdct_coef[ch], AC3_MAX_COEFS, block->coeff_shift[ch]); } } } static void sum_square_butterfly(AC3EncodeContext *s, int64_t sum[4], const int32_t *coef0, const int32_t *coef1, int len) { s->ac3dsp.sum_square_butterfly_int32(sum, coef0, coef1, len); } /* * Clip MDCT coefficients to allowable range. */ static void clip_coefficients(AudioDSPContext *adsp, int32_t *coef, unsigned int len) { adsp->vector_clip_int32(coef, coef, COEF_MIN, COEF_MAX, len); } /* * Calculate a single coupling coordinate. */ static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl) { if (energy_cpl <= COEF_MAX) { return 1048576; } else { uint64_t coord = energy_ch / (energy_cpl >> 24); uint32_t coord32 = FFMIN(coord, 1073741824); coord32 = ff_sqrt(coord32) << 9; return FFMIN(coord32, COEF_MAX); } } #include "ac3enc_template.c" /** * Finalize MDCT and free allocated memory. * * @param s AC-3 encoder private context */ av_cold void ff_ac3_fixed_mdct_end(AC3EncodeContext *s) { ff_mdct_end(&s->mdct); } /** * Initialize MDCT tables. * * @param s AC-3 encoder private context * @return 0 on success, negative error code on failure */ av_cold int ff_ac3_fixed_mdct_init(AC3EncodeContext *s) { int ret = ff_mdct_init(&s->mdct, 9, 0, -1.0); s->mdct_window = ff_ac3_window; return ret; } static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx) { AC3EncodeContext *s = avctx->priv_data; s->fixed_point = 1; return ff_ac3_encode_init(avctx); } AVCodec ff_ac3_fixed_encoder = { .name = "ac3_fixed", .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_AC3, .priv_data_size = sizeof(AC3EncodeContext), .init = ac3_fixed_encode_init, .encode2 = ff_ac3_fixed_encode_frame, .close = ff_ac3_encode_close, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE }, .priv_class = &ac3enc_class, .channel_layouts = ff_ac3_channel_layouts, .defaults = ac3_defaults, };