mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
3074f03a07
* qatar/master:
get_bits: remove x86 inline asm in A32 bitstream reader
doc: Remove outdated information about our issue tracker
avidec: Factor out the sync fucntionality.
fate-aac: Expand coverage.
ac3dsp: add x86-optimized versions of ac3dsp.extract_exponents().
ac3dsp: simplify extract_exponents() now that it does not need to do clipping.
ac3enc: clip coefficients after MDCT.
ac3enc: add int32_t array clipping function to DSPUtil, including x86 versions.
swscale: for >8bit scaling, read in native bit-depth.
matroskadec: matroska_read_seek after after EBML_STOP leads to failure.
doxygen: fix usage of @file directive in libavutil/{dict,file}.h
doxygen: Help doxygen parser to understand the DECLARE_ALIGNED and offsetof macros
Conflicts:
doc/issue_tracker.txt
libavformat/avidec.c
libavutil/dict.h
libswscale/swscale.c
libswscale/utils.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
139 lines
3.6 KiB
C
139 lines
3.6 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 "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 AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
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ac3_options, LIBAVUTIL_VERSION_INT };
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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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av_cold void ff_ac3_float_mdct_end(AC3MDCTContext *mdct)
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{
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ff_mdct_end(&mdct->fft);
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av_freep(&mdct->window);
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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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av_cold int ff_ac3_float_mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
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int nbits)
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{
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float *window;
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int i, n, n2;
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n = 1 << nbits;
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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(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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mdct->window = window;
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return ff_mdct_init(&mdct->fft, nbits, 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(DSPContext *dsp, float *output, const float *input,
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const float *window, unsigned int len)
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{
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dsp->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 * AC3_MAX_BLOCKS;
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s->ac3dsp.float_to_fixed24(s->fixed_coef_buffer + chan_size,
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s->mdct_coef_buffer + chan_size,
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chan_size * s->channels);
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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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#if CONFIG_AC3_ENCODER
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AVCodec ff_ac3_float_encoder = {
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"ac3_float",
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AVMEDIA_TYPE_AUDIO,
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CODEC_ID_AC3,
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sizeof(AC3EncodeContext),
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ff_ac3_encode_init,
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ff_ac3_float_encode_frame,
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ff_ac3_encode_close,
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NULL,
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,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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};
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#endif
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