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![Michael Niedermayer](/assets/img/avatar_default.png)
* qatar/master: wmaenc: use float planar sample format (e)ac3enc: use planar sample format aacenc: use planar sample format adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt adpcmenc: move 'ch' variable to higher scope adpcmenc: fix 3 instances of variable shadowing adpcm_ima_wav: simplify encoding libvorbis: use planar sample format libmp3lame: use planar sample formats vorbisenc: use float planar sample format ffm: do not write or read the audio sample format parseutils: fix parsing of invalid alpha values doc/RELEASE_NOTES: update for the 9 release. smoothstreamingenc: Add a more verbose error message smoothstreamingenc: Ignore the return value from mkdir smoothstreamingenc: Try writing a manifest when opening the muxer smoothstreamingenc: Move the output_chunk_list and write_manifest functions up smoothstreamingenc: Properly return errors from ism_flush to the caller smoothstreamingenc: Check the output UrlContext before accessing it Conflicts: doc/RELEASE_NOTES libavcodec/aacenc.c libavcodec/ac3enc_template.c libavcodec/wmaenc.c tests/ref/lavf/ffm Merged-by: Michael Niedermayer <michaelni@gmx.at>
169 lines
4.5 KiB
C
169 lines
4.5 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 FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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* floating-point AC-3 encoder.
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*/
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#define CONFIG_AC3ENC_FLOAT 1
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#include "internal.h"
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#include "ac3enc.h"
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#include "eac3enc.h"
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#include "kbdwin.h"
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#if CONFIG_AC3_ENCODER
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#define AC3ENC_TYPE AC3ENC_TYPE_AC3
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#include "ac3enc_opts_template.c"
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static const AVClass ac3enc_class = {
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.class_name = "AC-3 Encoder",
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.item_name = av_default_item_name,
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.option = ac3_options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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#endif
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#include "ac3enc_template.c"
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/**
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* Finalize MDCT and free allocated memory.
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*
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* @param s AC-3 encoder private context
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*/
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av_cold void ff_ac3_float_mdct_end(AC3EncodeContext *s)
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{
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ff_mdct_end(&s->mdct);
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av_freep(&s->mdct_window);
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}
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/**
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* Initialize MDCT tables.
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*
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* @param s AC-3 encoder private context
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* @return 0 on success, negative error code on failure
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*/
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av_cold int ff_ac3_float_mdct_init(AC3EncodeContext *s)
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{
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float *window;
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int i, n, n2;
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n = 1 << 9;
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n2 = n >> 1;
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window = av_malloc(n * sizeof(*window));
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if (!window) {
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av_log(s->avctx, AV_LOG_ERROR, "Cannot allocate memory.\n");
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return AVERROR(ENOMEM);
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}
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ff_kbd_window_init(window, 5.0, n2);
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for (i = 0; i < n2; i++)
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window[n-1-i] = window[i];
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s->mdct_window = window;
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return ff_mdct_init(&s->mdct, 9, 0, -2.0 / n);
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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(void *dsp, float *output,
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const float *input, const float *window,
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unsigned int len)
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{
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AVFloatDSPContext *fdsp = dsp;
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fdsp->vector_fmul(output, input, window, len);
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}
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/*
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* Normalize the input samples.
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* Not needed for the floating-point encoder.
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*/
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static int normalize_samples(AC3EncodeContext *s)
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{
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return 0;
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}
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/*
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* Scale MDCT coefficients from float to 24-bit 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 chan_size = AC3_MAX_COEFS * s->num_blocks;
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int cpl = s->cpl_on;
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s->ac3dsp.float_to_fixed24(s->fixed_coef_buffer + (chan_size * !cpl),
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s->mdct_coef_buffer + (chan_size * !cpl),
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chan_size * (s->channels + cpl));
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}
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static void sum_square_butterfly(AC3EncodeContext *s, float sum[4],
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const float *coef0, const float *coef1,
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int len)
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{
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s->ac3dsp.sum_square_butterfly_float(sum, coef0, coef1, len);
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}
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/*
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* Clip MDCT coefficients to allowable range.
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*/
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static void clip_coefficients(DSPContext *dsp, float *coef, unsigned int len)
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{
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dsp->vector_clipf(coef, coef, COEF_MIN, COEF_MAX, len);
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}
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/*
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* Calculate a single coupling coordinate.
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*/
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static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)
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{
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float coord = 0.125;
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if (energy_cpl > 0)
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coord *= sqrtf(energy_ch / energy_cpl);
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return FFMIN(coord, COEF_MAX);
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}
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#if CONFIG_AC3_ENCODER
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AVCodec ff_ac3_encoder = {
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.name = "ac3",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_AC3,
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.priv_data_size = sizeof(AC3EncodeContext),
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.init = ff_ac3_encode_init,
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.encode2 = ff_ac3_float_encode_frame,
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.close = ff_ac3_encode_close,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
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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 = ff_ac3_channel_layouts,
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.defaults = ac3_defaults,
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};
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#endif
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