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79ae084e9b
* qatar/master: (58 commits) amrnbdec: check frame size before decoding. cscd: use negative error values to indicate decode_init() failures. h264: prevent overreads in intra PCM decoding. FATE: do not decode audio in the nuv test. dxa: set audio stream time base using the sample rate psx-str: do not allow seeking by bytes asfdec: Do not set AVCodecContext.frame_size vqf: set packet parameters after av_new_packet() mpegaudiodec: use DSPUtil.butterflies_float(). FATE: add mp3 test for sample that exhibited false overreads fate: add cdxl test for bit line plane arrangement vmnc: return error on decode_init() failure. libvorbis: add/update error messages libvorbis: use AVFifoBuffer for output packet buffer libvorbis: remove unneeded e_o_s check libvorbis: check return values for functions that can return errors libvorbis: use float input instead of s16 libvorbis: do not flush libvorbis analysis if dsp state was not initialized libvorbis: use VBR by default, with default quality of 3 libvorbis: fix use of minrate/maxrate AVOptions ... Conflicts: Changelog doc/APIchanges libavcodec/avcodec.h libavcodec/dpxenc.c libavcodec/libvorbis.c libavcodec/vmnc.c libavformat/asfdec.c libavformat/id3v2enc.c libavformat/internal.h libavformat/mp3enc.c libavformat/utils.c libavformat/version.h libswscale/utils.c tests/fate/video.mak tests/ref/fate/nuv tests/ref/fate/prores-alpha tests/ref/lavf/ffm tests/ref/vsynth1/prores tests/ref/vsynth2/prores Merged-by: Michael Niedermayer <michaelni@gmx.at>
162 lines
4.6 KiB
C
162 lines
4.6 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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* 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.h"
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#include "eac3enc.h"
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#define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED
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#include "ac3enc_opts_template.c"
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static const AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
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ac3fixed_options, LIBAVUTIL_VERSION_INT };
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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 AC3_NAME(mdct_end)(AC3EncodeContext *s)
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{
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ff_mdct_end(&s->mdct);
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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 AC3_NAME(mdct_init)(AC3EncodeContext *s)
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{
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int ret = ff_mdct_init(&s->mdct, 9, 0, -1.0);
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s->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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* 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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static int normalize_samples(AC3EncodeContext *s)
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{
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int v = s->ac3dsp.ac3_max_msb_abs_int16(s->windowed_samples, AC3_WINDOW_SIZE);
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v = 14 - av_log2(v);
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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 < s->num_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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static void sum_square_butterfly(AC3EncodeContext *s, int64_t sum[4],
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const int32_t *coef0, const int32_t *coef1,
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int len)
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{
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s->ac3dsp.sum_square_butterfly_int32(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, int32_t *coef, unsigned int len)
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{
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dsp->vector_clip_int32(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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if (energy_cpl <= COEF_MAX) {
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return 1048576;
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} else {
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uint64_t coord = energy_ch / (energy_cpl >> 24);
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uint32_t coord32 = FFMIN(coord, 1073741824);
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coord32 = ff_sqrt(coord32) << 9;
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return FFMIN(coord32, COEF_MAX);
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}
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}
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static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx)
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{
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AC3EncodeContext *s = avctx->priv_data;
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s->fixed_point = 1;
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return ff_ac3_encode_init(avctx);
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}
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AVCodec ff_ac3_fixed_encoder = {
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.name = "ac3_fixed",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_AC3,
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.priv_data_size = sizeof(AC3EncodeContext),
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.init = ac3_fixed_encode_init,
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.encode = ff_ac3_fixed_encode_frame,
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.close = ff_ac3_encode_close,
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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 = ff_ac3_channel_layouts,
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.defaults = ac3_defaults,
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};
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