* qatar/master: (44 commits) replacement Indeo 3 decoder gsm demuxer: do not allocate packet twice. flvenc: use first packet delay as global delay. ac3enc: doxygen update. imc: return error codes instead of 0 for error conditions. imc: return meaningful error codes instead of -1 imc: do not set channel layout for stereo imc: validate channel count imc: check for ff_fft_init() failure imc: check output buffer size before decoding imc: use DSPContext.bswap16_buf() to byte-swap packet data rtsp: add allowed_media_types option libgsm: add flush function to reset the decoder state when seeking libgsm: simplify decoding by using a loop gsm: log error message when packet is too small libgsmdec: do not needlessly set *data_size to 0 gsmdec: do not needlessly set *data_size to 0 gsmdec: add flush function to reset the decoder state when seeking libgsmdec: check output buffer size before decoding gsmdec: log error message when output buffer is too small. ... Conflicts: Changelog ffplay.c libavcodec/indeo3.c libavcodec/mjpeg_parser.c libavcodec/vp3.c libavformat/cutils.c libavformat/id3v2.c libavutil/parseutils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
		
			
				
	
	
		
			161 lines
		
	
	
		
			4.5 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			161 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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 * 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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};
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