/* * 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 Libav. * * Libav 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. * * Libav 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 Libav; 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 CONFIG_FFT_FLOAT 0 #undef CONFIG_AC3ENC_FLOAT #include "ac3enc.c" /** * Finalize MDCT and free allocated memory. */ static av_cold void mdct_end(AC3MDCTContext *mdct) { ff_mdct_end(&mdct->fft); } /** * Initialize MDCT tables. * @param nbits log2(MDCT size) */ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct, int nbits) { int ret = ff_mdct_init(&mdct->fft, nbits, 0, -1.0); mdct->window = ff_ac3_window; return ret; } /** * Apply KBD window to input samples prior to MDCT. */ static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input, const int16_t *window, unsigned int len) { dsp->apply_window_int16(output, input, window, len); } /** * Calculate the log2() of the maximum absolute value in an array. * @param tab input array * @param n number of values in the array * @return log2(max(abs(tab[]))) */ static int log2_tab(AC3EncodeContext *s, int16_t *src, int len) { int v = s->ac3dsp.ac3_max_msb_abs_int16(src, len); return av_log2(v); } /** * Normalize the input samples to use the maximum available precision. * This assumes signed 16-bit input samples. * * @return exponent shift */ static int normalize_samples(AC3EncodeContext *s) { int v = 14 - log2_tab(s, s->windowed_samples, AC3_WINDOW_SIZE); 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 < AC3_MAX_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]); } } } AVCodec ff_ac3_fixed_encoder = { "ac3_fixed", AVMEDIA_TYPE_AUDIO, CODEC_ID_AC3, sizeof(AC3EncodeContext), ac3_encode_init, ac3_encode_frame, ac3_encode_close, NULL, .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), .priv_class = &ac3enc_class, .channel_layouts = ac3_channel_layouts, };