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Channel coupling is an optional AC-3 feature that increases quality by combining high frequency information from multiple channels into a single channel. The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies.
126 lines
3.3 KiB
C
126 lines
3.3 KiB
C
/*
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* The simplest AC-3 encoder
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* Copyright (c) 2000 Fabrice Bellard
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* Copyright (c) 2006-2010 Justin Ruggles <justin.ruggles@gmail.com>
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* Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de>
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* fixed-point AC-3 encoder.
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*/
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#define CONFIG_FFT_FLOAT 0
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#undef CONFIG_AC3ENC_FLOAT
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#include "ac3enc.c"
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/**
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* Finalize MDCT and free allocated memory.
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*/
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static av_cold void mdct_end(AC3MDCTContext *mdct)
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{
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ff_mdct_end(&mdct->fft);
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}
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/**
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* Initialize MDCT tables.
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* @param nbits log2(MDCT size)
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*/
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static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
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int nbits)
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{
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int ret = ff_mdct_init(&mdct->fft, nbits, 0, -1.0);
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mdct->window = ff_ac3_window;
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return ret;
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}
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/**
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* Apply KBD window to input samples prior to MDCT.
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*/
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static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input,
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const int16_t *window, unsigned int len)
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{
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dsp->apply_window_int16(output, input, window, len);
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}
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/**
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* Calculate the log2() of the maximum absolute value in an array.
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* @param tab input array
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* @param n number of values in the array
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* @return log2(max(abs(tab[])))
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*/
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static int log2_tab(AC3EncodeContext *s, int16_t *src, int len)
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{
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int v = s->ac3dsp.ac3_max_msb_abs_int16(src, len);
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return av_log2(v);
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}
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/**
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* Normalize the input samples to use the maximum available precision.
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* This assumes signed 16-bit input samples.
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*
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* @return exponent shift
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*/
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static int normalize_samples(AC3EncodeContext *s)
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{
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int v = 14 - log2_tab(s, s->windowed_samples, AC3_WINDOW_SIZE);
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if (v > 0)
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s->ac3dsp.ac3_lshift_int16(s->windowed_samples, AC3_WINDOW_SIZE, v);
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/* +6 to right-shift from 31-bit to 25-bit */
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return v + 6;
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}
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/**
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* Scale MDCT coefficients to 25-bit signed fixed-point.
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*/
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static void scale_coefficients(AC3EncodeContext *s)
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{
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int blk, ch;
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for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
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AC3Block *block = &s->blocks[blk];
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for (ch = 1; ch <= s->channels; ch++) {
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s->ac3dsp.ac3_rshift_int32(block->mdct_coef[ch], AC3_MAX_COEFS,
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block->coeff_shift[ch]);
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}
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}
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}
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AVCodec ff_ac3_fixed_encoder = {
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"ac3_fixed",
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AVMEDIA_TYPE_AUDIO,
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CODEC_ID_AC3,
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sizeof(AC3EncodeContext),
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ac3_encode_init,
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ac3_encode_frame,
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ac3_encode_close,
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NULL,
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
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.priv_class = &ac3enc_class,
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.channel_layouts = ac3_channel_layouts,
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};
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