/* * The simplest AC-3 encoder * Copyright (c) 2000 Fabrice Bellard * Copyright (c) 2006-2010 Justin Ruggles * Copyright (c) 2006-2010 Prakash Punnoor * * 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 * floating-point AC-3 encoder. */ #define CONFIG_AC3ENC_FLOAT 1 #include "ac3enc.c" #include "kbdwin.h" /** * Finalize MDCT and free allocated memory. */ static av_cold void mdct_end(AC3MDCTContext *mdct) { ff_mdct_end(&mdct->fft); av_freep(&mdct->window); } /** * Initialize MDCT tables. * @param nbits log2(MDCT size) */ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct, int nbits) { float *window; int i, n, n2; n = 1 << nbits; n2 = n >> 1; window = av_malloc(n * sizeof(*window)); if (!window) { av_log(avctx, AV_LOG_ERROR, "Cannot allocate memory.\n"); return AVERROR(ENOMEM); } ff_kbd_window_init(window, 5.0, n2); for (i = 0; i < n2; i++) window[n-1-i] = window[i]; mdct->window = window; return ff_mdct_init(&mdct->fft, nbits, 0, -2.0 / n); } /** * Calculate a 512-point MDCT * @param out 256 output frequency coefficients * @param in 512 windowed input audio samples */ static void mdct512(AC3MDCTContext *mdct, float *out, float *in) { mdct->fft.mdct_calc(&mdct->fft, out, in); } /** * Apply KBD window to input samples prior to MDCT. */ static void apply_window(DSPContext *dsp, float *output, const float *input, const float *window, unsigned int len) { dsp->vector_fmul(output, input, window, len); } /** * Normalize the input samples to use the maximum available precision. */ static int normalize_samples(AC3EncodeContext *s) { /* Normalization is not needed for floating-point samples, so just return 0 */ return 0; } /** * Scale MDCT coefficients from float to 24-bit fixed-point. */ static void scale_coefficients(AC3EncodeContext *s) { s->ac3dsp.float_to_fixed24(s->fixed_coef_buffer, s->mdct_coef_buffer, AC3_MAX_COEFS * AC3_MAX_BLOCKS * s->channels); } AVCodec ff_ac3_encoder = { "ac3", 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_FLT,AV_SAMPLE_FMT_NONE}, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), .channel_layouts = ac3_channel_layouts, };