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https://gitee.com/openharmony/third_party_ffmpeg
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6cb2085278
* qatar/master: (27 commits) ppc: Add ff_ prefix to nonstatic symbols sh4: Add ff_ prefix to nonstatic symbols mpegvideo: Add ff_ prefix to nonstatic functions rtjpeg: Add ff_ prefix to nonstatic symbols rv: Add ff_ prefix to nonstatic symbols vp56: Add ff_ prefix to nonstatic symbols vorbis: Add ff_ prefix to nonstatic symbols msmpeg4: Add ff_ prefix to nonstatic symbols vc1: Add ff_ prefix to nonstatic symbols msmpeg4: Add ff_ prefixes to nonstatic symbols snow: Add ff_ prefix to nonstatic symbols mpeg12: Add ff_ prefix to nonstatic symbols mpeg4: Add ff_ prefixes to nonstatic symbols lagarith: Add ff_ prefix to lag_rac_init libavcodec: Add ff_ prefix to j_rev_dct* dsputil: Add ff_ prefix to inv_zigzag_direct16 libavcodec: Prefix fdct_ifast, fdct_ifast248 dsputil: Add ff_ prefix to the dsputil*_init* functions libavcodec: Add ff_ prefix to some nonstatic symbols vlc/rl: Add ff_ prefix to the nonstatic symbols ... Conflicts: libavcodec/Makefile libavcodec/allcodecs.c libavcodec/dnxhddec.c libavcodec/ffv1.c libavcodec/h263.h libavcodec/h263dec.c libavcodec/h264.c libavcodec/mpegvideo.c libavcodec/mpegvideo_enc.c libavcodec/nuv.c libavcodec/ppc/dsputil_ppc.c libavcodec/proresdsp.c libavcodec/svq3.c libavcodec/version.h libavformat/dv.h libavformat/dvenc.c libavformat/matroskadec.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
997 lines
30 KiB
C
997 lines
30 KiB
C
/*
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* Monkey's Audio lossless audio decoder
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* Copyright (c) 2007 Benjamin Zores <ben@geexbox.org>
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* based upon libdemac from Dave Chapman.
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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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#include "avcodec.h"
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#include "dsputil.h"
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#include "bytestream.h"
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#include "libavutil/audioconvert.h"
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#include "libavutil/avassert.h"
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#include "libavutil/opt.h"
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/**
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* @file
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* Monkey's Audio lossless audio decoder
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*/
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#define MAX_CHANNELS 2
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#define MAX_BYTESPERSAMPLE 3
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#define APE_FRAMECODE_MONO_SILENCE 1
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#define APE_FRAMECODE_STEREO_SILENCE 3
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#define APE_FRAMECODE_PSEUDO_STEREO 4
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#define HISTORY_SIZE 512
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#define PREDICTOR_ORDER 8
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/** Total size of all predictor histories */
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#define PREDICTOR_SIZE 50
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#define YDELAYA (18 + PREDICTOR_ORDER*4)
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#define YDELAYB (18 + PREDICTOR_ORDER*3)
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#define XDELAYA (18 + PREDICTOR_ORDER*2)
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#define XDELAYB (18 + PREDICTOR_ORDER)
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#define YADAPTCOEFFSA 18
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#define XADAPTCOEFFSA 14
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#define YADAPTCOEFFSB 10
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#define XADAPTCOEFFSB 5
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/**
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* Possible compression levels
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* @{
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*/
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enum APECompressionLevel {
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COMPRESSION_LEVEL_FAST = 1000,
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COMPRESSION_LEVEL_NORMAL = 2000,
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COMPRESSION_LEVEL_HIGH = 3000,
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COMPRESSION_LEVEL_EXTRA_HIGH = 4000,
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COMPRESSION_LEVEL_INSANE = 5000
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};
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/** @} */
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#define APE_FILTER_LEVELS 3
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/** Filter orders depending on compression level */
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static const uint16_t ape_filter_orders[5][APE_FILTER_LEVELS] = {
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{ 0, 0, 0 },
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{ 16, 0, 0 },
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{ 64, 0, 0 },
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{ 32, 256, 0 },
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{ 16, 256, 1280 }
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};
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/** Filter fraction bits depending on compression level */
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static const uint8_t ape_filter_fracbits[5][APE_FILTER_LEVELS] = {
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{ 0, 0, 0 },
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{ 11, 0, 0 },
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{ 11, 0, 0 },
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{ 10, 13, 0 },
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{ 11, 13, 15 }
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};
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/** Filters applied to the decoded data */
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typedef struct APEFilter {
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int16_t *coeffs; ///< actual coefficients used in filtering
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int16_t *adaptcoeffs; ///< adaptive filter coefficients used for correcting of actual filter coefficients
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int16_t *historybuffer; ///< filter memory
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int16_t *delay; ///< filtered values
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int avg;
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} APEFilter;
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typedef struct APERice {
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uint32_t k;
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uint32_t ksum;
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} APERice;
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typedef struct APERangecoder {
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uint32_t low; ///< low end of interval
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uint32_t range; ///< length of interval
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uint32_t help; ///< bytes_to_follow resp. intermediate value
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unsigned int buffer; ///< buffer for input/output
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} APERangecoder;
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/** Filter histories */
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typedef struct APEPredictor {
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int32_t *buf;
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int32_t lastA[2];
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int32_t filterA[2];
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int32_t filterB[2];
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int32_t coeffsA[2][4]; ///< adaption coefficients
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int32_t coeffsB[2][5]; ///< adaption coefficients
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int32_t historybuffer[HISTORY_SIZE + PREDICTOR_SIZE];
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} APEPredictor;
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/** Decoder context */
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typedef struct APEContext {
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AVClass *class; ///< class for AVOptions
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AVCodecContext *avctx;
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AVFrame frame;
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DSPContext dsp;
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int channels;
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int samples; ///< samples left to decode in current frame
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int bps;
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int fileversion; ///< codec version, very important in decoding process
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int compression_level; ///< compression levels
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int fset; ///< which filter set to use (calculated from compression level)
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int flags; ///< global decoder flags
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uint32_t CRC; ///< frame CRC
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int frameflags; ///< frame flags
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APEPredictor predictor; ///< predictor used for final reconstruction
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int32_t *decoded_buffer;
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int decoded_size;
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int32_t *decoded[MAX_CHANNELS]; ///< decoded data for each channel
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int blocks_per_loop; ///< maximum number of samples to decode for each call
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int16_t* filterbuf[APE_FILTER_LEVELS]; ///< filter memory
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APERangecoder rc; ///< rangecoder used to decode actual values
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APERice riceX; ///< rice code parameters for the second channel
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APERice riceY; ///< rice code parameters for the first channel
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APEFilter filters[APE_FILTER_LEVELS][2]; ///< filters used for reconstruction
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uint8_t *data; ///< current frame data
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uint8_t *data_end; ///< frame data end
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int data_size; ///< frame data allocated size
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const uint8_t *ptr; ///< current position in frame data
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int error;
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} APEContext;
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// TODO: dsputilize
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static av_cold int ape_decode_close(AVCodecContext *avctx)
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{
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APEContext *s = avctx->priv_data;
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int i;
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for (i = 0; i < APE_FILTER_LEVELS; i++)
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av_freep(&s->filterbuf[i]);
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av_freep(&s->decoded_buffer);
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av_freep(&s->data);
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s->decoded_size = s->data_size = 0;
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return 0;
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}
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static av_cold int ape_decode_init(AVCodecContext *avctx)
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{
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APEContext *s = avctx->priv_data;
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int i;
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if (avctx->extradata_size != 6) {
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av_log(avctx, AV_LOG_ERROR, "Incorrect extradata\n");
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return AVERROR(EINVAL);
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}
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if (avctx->channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "Only mono and stereo is supported\n");
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return AVERROR(EINVAL);
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}
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s->bps = avctx->bits_per_coded_sample;
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switch (s->bps) {
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case 8:
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avctx->sample_fmt = AV_SAMPLE_FMT_U8;
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break;
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case 16:
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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break;
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case 24:
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avctx->sample_fmt = AV_SAMPLE_FMT_S32;
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break;
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default:
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av_log_ask_for_sample(avctx, "Unsupported bits per coded sample %d\n",
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s->bps);
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return AVERROR_PATCHWELCOME;
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}
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s->avctx = avctx;
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s->channels = avctx->channels;
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s->fileversion = AV_RL16(avctx->extradata);
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s->compression_level = AV_RL16(avctx->extradata + 2);
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s->flags = AV_RL16(avctx->extradata + 4);
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av_log(avctx, AV_LOG_DEBUG, "Compression Level: %d - Flags: %d\n",
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s->compression_level, s->flags);
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if (s->compression_level % 1000 || s->compression_level > COMPRESSION_LEVEL_INSANE) {
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av_log(avctx, AV_LOG_ERROR, "Incorrect compression level %d\n",
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s->compression_level);
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return AVERROR_INVALIDDATA;
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}
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s->fset = s->compression_level / 1000 - 1;
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for (i = 0; i < APE_FILTER_LEVELS; i++) {
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if (!ape_filter_orders[s->fset][i])
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break;
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FF_ALLOC_OR_GOTO(avctx, s->filterbuf[i],
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(ape_filter_orders[s->fset][i] * 3 + HISTORY_SIZE) * 4,
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filter_alloc_fail);
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}
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ff_dsputil_init(&s->dsp, avctx);
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avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
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avcodec_get_frame_defaults(&s->frame);
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avctx->coded_frame = &s->frame;
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return 0;
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filter_alloc_fail:
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ape_decode_close(avctx);
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return AVERROR(ENOMEM);
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}
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/**
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* @name APE range decoding functions
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* @{
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*/
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#define CODE_BITS 32
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#define TOP_VALUE ((unsigned int)1 << (CODE_BITS-1))
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#define SHIFT_BITS (CODE_BITS - 9)
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#define EXTRA_BITS ((CODE_BITS-2) % 8 + 1)
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#define BOTTOM_VALUE (TOP_VALUE >> 8)
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/** Start the decoder */
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static inline void range_start_decoding(APEContext *ctx)
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{
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ctx->rc.buffer = bytestream_get_byte(&ctx->ptr);
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ctx->rc.low = ctx->rc.buffer >> (8 - EXTRA_BITS);
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ctx->rc.range = (uint32_t) 1 << EXTRA_BITS;
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}
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/** Perform normalization */
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static inline void range_dec_normalize(APEContext *ctx)
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{
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while (ctx->rc.range <= BOTTOM_VALUE) {
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ctx->rc.buffer <<= 8;
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if(ctx->ptr < ctx->data_end) {
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ctx->rc.buffer += *ctx->ptr;
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ctx->ptr++;
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} else {
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ctx->error = 1;
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}
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ctx->rc.low = (ctx->rc.low << 8) | ((ctx->rc.buffer >> 1) & 0xFF);
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ctx->rc.range <<= 8;
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}
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}
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/**
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* Calculate culmulative frequency for next symbol. Does NO update!
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* @param ctx decoder context
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* @param tot_f is the total frequency or (code_value)1<<shift
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* @return the culmulative frequency
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*/
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static inline int range_decode_culfreq(APEContext *ctx, int tot_f)
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{
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range_dec_normalize(ctx);
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ctx->rc.help = ctx->rc.range / tot_f;
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return ctx->rc.low / ctx->rc.help;
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}
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/**
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* Decode value with given size in bits
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* @param ctx decoder context
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* @param shift number of bits to decode
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*/
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static inline int range_decode_culshift(APEContext *ctx, int shift)
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{
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range_dec_normalize(ctx);
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ctx->rc.help = ctx->rc.range >> shift;
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return ctx->rc.low / ctx->rc.help;
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}
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/**
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* Update decoding state
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* @param ctx decoder context
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* @param sy_f the interval length (frequency of the symbol)
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* @param lt_f the lower end (frequency sum of < symbols)
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*/
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static inline void range_decode_update(APEContext *ctx, int sy_f, int lt_f)
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{
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ctx->rc.low -= ctx->rc.help * lt_f;
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ctx->rc.range = ctx->rc.help * sy_f;
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}
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/** Decode n bits (n <= 16) without modelling */
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static inline int range_decode_bits(APEContext *ctx, int n)
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{
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int sym = range_decode_culshift(ctx, n);
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range_decode_update(ctx, 1, sym);
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return sym;
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}
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#define MODEL_ELEMENTS 64
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/**
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* Fixed probabilities for symbols in Monkey Audio version 3.97
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*/
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static const uint16_t counts_3970[22] = {
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0, 14824, 28224, 39348, 47855, 53994, 58171, 60926,
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62682, 63786, 64463, 64878, 65126, 65276, 65365, 65419,
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65450, 65469, 65480, 65487, 65491, 65493,
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};
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/**
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* Probability ranges for symbols in Monkey Audio version 3.97
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*/
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static const uint16_t counts_diff_3970[21] = {
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14824, 13400, 11124, 8507, 6139, 4177, 2755, 1756,
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1104, 677, 415, 248, 150, 89, 54, 31,
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19, 11, 7, 4, 2,
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};
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/**
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* Fixed probabilities for symbols in Monkey Audio version 3.98
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*/
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static const uint16_t counts_3980[22] = {
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0, 19578, 36160, 48417, 56323, 60899, 63265, 64435,
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64971, 65232, 65351, 65416, 65447, 65466, 65476, 65482,
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65485, 65488, 65490, 65491, 65492, 65493,
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};
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/**
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* Probability ranges for symbols in Monkey Audio version 3.98
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*/
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static const uint16_t counts_diff_3980[21] = {
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19578, 16582, 12257, 7906, 4576, 2366, 1170, 536,
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261, 119, 65, 31, 19, 10, 6, 3,
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3, 2, 1, 1, 1,
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};
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/**
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* Decode symbol
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* @param ctx decoder context
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* @param counts probability range start position
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* @param counts_diff probability range widths
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*/
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static inline int range_get_symbol(APEContext *ctx,
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const uint16_t counts[],
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const uint16_t counts_diff[])
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{
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int symbol, cf;
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cf = range_decode_culshift(ctx, 16);
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if(cf > 65492){
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symbol= cf - 65535 + 63;
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range_decode_update(ctx, 1, cf);
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if(cf > 65535)
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ctx->error=1;
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return symbol;
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}
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/* figure out the symbol inefficiently; a binary search would be much better */
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for (symbol = 0; counts[symbol + 1] <= cf; symbol++);
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range_decode_update(ctx, counts_diff[symbol], counts[symbol]);
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return symbol;
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}
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/** @} */ // group rangecoder
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static inline void update_rice(APERice *rice, int x)
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{
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int lim = rice->k ? (1 << (rice->k + 4)) : 0;
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rice->ksum += ((x + 1) / 2) - ((rice->ksum + 16) >> 5);
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if (rice->ksum < lim)
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rice->k--;
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else if (rice->ksum >= (1 << (rice->k + 5)))
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rice->k++;
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}
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static inline int ape_decode_value(APEContext *ctx, APERice *rice)
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{
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int x, overflow;
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if (ctx->fileversion < 3990) {
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int tmpk;
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overflow = range_get_symbol(ctx, counts_3970, counts_diff_3970);
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if (overflow == (MODEL_ELEMENTS - 1)) {
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tmpk = range_decode_bits(ctx, 5);
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overflow = 0;
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} else
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tmpk = (rice->k < 1) ? 0 : rice->k - 1;
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if (tmpk <= 16)
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x = range_decode_bits(ctx, tmpk);
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else {
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x = range_decode_bits(ctx, 16);
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x |= (range_decode_bits(ctx, tmpk - 16) << 16);
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}
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x += overflow << tmpk;
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} else {
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int base, pivot;
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pivot = rice->ksum >> 5;
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if (pivot == 0)
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pivot = 1;
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overflow = range_get_symbol(ctx, counts_3980, counts_diff_3980);
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if (overflow == (MODEL_ELEMENTS - 1)) {
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overflow = range_decode_bits(ctx, 16) << 16;
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overflow |= range_decode_bits(ctx, 16);
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}
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if (pivot < 0x10000) {
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base = range_decode_culfreq(ctx, pivot);
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range_decode_update(ctx, 1, base);
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} else {
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int base_hi = pivot, base_lo;
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int bbits = 0;
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while (base_hi & ~0xFFFF) {
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base_hi >>= 1;
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bbits++;
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}
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base_hi = range_decode_culfreq(ctx, base_hi + 1);
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range_decode_update(ctx, 1, base_hi);
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base_lo = range_decode_culfreq(ctx, 1 << bbits);
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range_decode_update(ctx, 1, base_lo);
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base = (base_hi << bbits) + base_lo;
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}
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x = base + overflow * pivot;
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}
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update_rice(rice, x);
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/* Convert to signed */
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if (x & 1)
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return (x >> 1) + 1;
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else
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return -(x >> 1);
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}
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static void entropy_decode(APEContext *ctx, int blockstodecode, int stereo)
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{
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int32_t *decoded0 = ctx->decoded[0];
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int32_t *decoded1 = ctx->decoded[1];
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while (blockstodecode--) {
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*decoded0++ = ape_decode_value(ctx, &ctx->riceY);
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if (stereo)
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*decoded1++ = ape_decode_value(ctx, &ctx->riceX);
|
|
}
|
|
}
|
|
|
|
static int init_entropy_decoder(APEContext *ctx)
|
|
{
|
|
/* Read the CRC */
|
|
if (ctx->data_end - ctx->ptr < 6)
|
|
return AVERROR_INVALIDDATA;
|
|
ctx->CRC = bytestream_get_be32(&ctx->ptr);
|
|
|
|
/* Read the frame flags if they exist */
|
|
ctx->frameflags = 0;
|
|
if ((ctx->fileversion > 3820) && (ctx->CRC & 0x80000000)) {
|
|
ctx->CRC &= ~0x80000000;
|
|
|
|
if (ctx->data_end - ctx->ptr < 6)
|
|
return AVERROR_INVALIDDATA;
|
|
ctx->frameflags = bytestream_get_be32(&ctx->ptr);
|
|
}
|
|
|
|
/* Initialize the rice structs */
|
|
ctx->riceX.k = 10;
|
|
ctx->riceX.ksum = (1 << ctx->riceX.k) * 16;
|
|
ctx->riceY.k = 10;
|
|
ctx->riceY.ksum = (1 << ctx->riceY.k) * 16;
|
|
|
|
/* The first 8 bits of input are ignored. */
|
|
ctx->ptr++;
|
|
|
|
range_start_decoding(ctx);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static const int32_t initial_coeffs[4] = {
|
|
360, 317, -109, 98
|
|
};
|
|
|
|
static void init_predictor_decoder(APEContext *ctx)
|
|
{
|
|
APEPredictor *p = &ctx->predictor;
|
|
|
|
/* Zero the history buffers */
|
|
memset(p->historybuffer, 0, PREDICTOR_SIZE * sizeof(*p->historybuffer));
|
|
p->buf = p->historybuffer;
|
|
|
|
/* Initialize and zero the coefficients */
|
|
memcpy(p->coeffsA[0], initial_coeffs, sizeof(initial_coeffs));
|
|
memcpy(p->coeffsA[1], initial_coeffs, sizeof(initial_coeffs));
|
|
memset(p->coeffsB, 0, sizeof(p->coeffsB));
|
|
|
|
p->filterA[0] = p->filterA[1] = 0;
|
|
p->filterB[0] = p->filterB[1] = 0;
|
|
p->lastA[0] = p->lastA[1] = 0;
|
|
}
|
|
|
|
/** Get inverse sign of integer (-1 for positive, 1 for negative and 0 for zero) */
|
|
static inline int APESIGN(int32_t x) {
|
|
return (x < 0) - (x > 0);
|
|
}
|
|
|
|
static av_always_inline int predictor_update_filter(APEPredictor *p,
|
|
const int decoded, const int filter,
|
|
const int delayA, const int delayB,
|
|
const int adaptA, const int adaptB)
|
|
{
|
|
int32_t predictionA, predictionB, sign;
|
|
|
|
p->buf[delayA] = p->lastA[filter];
|
|
p->buf[adaptA] = APESIGN(p->buf[delayA]);
|
|
p->buf[delayA - 1] = p->buf[delayA] - p->buf[delayA - 1];
|
|
p->buf[adaptA - 1] = APESIGN(p->buf[delayA - 1]);
|
|
|
|
predictionA = p->buf[delayA ] * p->coeffsA[filter][0] +
|
|
p->buf[delayA - 1] * p->coeffsA[filter][1] +
|
|
p->buf[delayA - 2] * p->coeffsA[filter][2] +
|
|
p->buf[delayA - 3] * p->coeffsA[filter][3];
|
|
|
|
/* Apply a scaled first-order filter compression */
|
|
p->buf[delayB] = p->filterA[filter ^ 1] - ((p->filterB[filter] * 31) >> 5);
|
|
p->buf[adaptB] = APESIGN(p->buf[delayB]);
|
|
p->buf[delayB - 1] = p->buf[delayB] - p->buf[delayB - 1];
|
|
p->buf[adaptB - 1] = APESIGN(p->buf[delayB - 1]);
|
|
p->filterB[filter] = p->filterA[filter ^ 1];
|
|
|
|
predictionB = p->buf[delayB ] * p->coeffsB[filter][0] +
|
|
p->buf[delayB - 1] * p->coeffsB[filter][1] +
|
|
p->buf[delayB - 2] * p->coeffsB[filter][2] +
|
|
p->buf[delayB - 3] * p->coeffsB[filter][3] +
|
|
p->buf[delayB - 4] * p->coeffsB[filter][4];
|
|
|
|
p->lastA[filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
|
|
p->filterA[filter] = p->lastA[filter] + ((p->filterA[filter] * 31) >> 5);
|
|
|
|
sign = APESIGN(decoded);
|
|
p->coeffsA[filter][0] += p->buf[adaptA ] * sign;
|
|
p->coeffsA[filter][1] += p->buf[adaptA - 1] * sign;
|
|
p->coeffsA[filter][2] += p->buf[adaptA - 2] * sign;
|
|
p->coeffsA[filter][3] += p->buf[adaptA - 3] * sign;
|
|
p->coeffsB[filter][0] += p->buf[adaptB ] * sign;
|
|
p->coeffsB[filter][1] += p->buf[adaptB - 1] * sign;
|
|
p->coeffsB[filter][2] += p->buf[adaptB - 2] * sign;
|
|
p->coeffsB[filter][3] += p->buf[adaptB - 3] * sign;
|
|
p->coeffsB[filter][4] += p->buf[adaptB - 4] * sign;
|
|
|
|
return p->filterA[filter];
|
|
}
|
|
|
|
static void predictor_decode_stereo(APEContext *ctx, int count)
|
|
{
|
|
APEPredictor *p = &ctx->predictor;
|
|
int32_t *decoded0 = ctx->decoded[0];
|
|
int32_t *decoded1 = ctx->decoded[1];
|
|
|
|
while (count--) {
|
|
/* Predictor Y */
|
|
*decoded0 = predictor_update_filter(p, *decoded0, 0, YDELAYA, YDELAYB,
|
|
YADAPTCOEFFSA, YADAPTCOEFFSB);
|
|
decoded0++;
|
|
*decoded1 = predictor_update_filter(p, *decoded1, 1, XDELAYA, XDELAYB,
|
|
XADAPTCOEFFSA, XADAPTCOEFFSB);
|
|
decoded1++;
|
|
|
|
/* Combined */
|
|
p->buf++;
|
|
|
|
/* Have we filled the history buffer? */
|
|
if (p->buf == p->historybuffer + HISTORY_SIZE) {
|
|
memmove(p->historybuffer, p->buf,
|
|
PREDICTOR_SIZE * sizeof(*p->historybuffer));
|
|
p->buf = p->historybuffer;
|
|
}
|
|
}
|
|
}
|
|
|
|
static void predictor_decode_mono(APEContext *ctx, int count)
|
|
{
|
|
APEPredictor *p = &ctx->predictor;
|
|
int32_t *decoded0 = ctx->decoded[0];
|
|
int32_t predictionA, currentA, A, sign;
|
|
|
|
currentA = p->lastA[0];
|
|
|
|
while (count--) {
|
|
A = *decoded0;
|
|
|
|
p->buf[YDELAYA] = currentA;
|
|
p->buf[YDELAYA - 1] = p->buf[YDELAYA] - p->buf[YDELAYA - 1];
|
|
|
|
predictionA = p->buf[YDELAYA ] * p->coeffsA[0][0] +
|
|
p->buf[YDELAYA - 1] * p->coeffsA[0][1] +
|
|
p->buf[YDELAYA - 2] * p->coeffsA[0][2] +
|
|
p->buf[YDELAYA - 3] * p->coeffsA[0][3];
|
|
|
|
currentA = A + (predictionA >> 10);
|
|
|
|
p->buf[YADAPTCOEFFSA] = APESIGN(p->buf[YDELAYA ]);
|
|
p->buf[YADAPTCOEFFSA - 1] = APESIGN(p->buf[YDELAYA - 1]);
|
|
|
|
sign = APESIGN(A);
|
|
p->coeffsA[0][0] += p->buf[YADAPTCOEFFSA ] * sign;
|
|
p->coeffsA[0][1] += p->buf[YADAPTCOEFFSA - 1] * sign;
|
|
p->coeffsA[0][2] += p->buf[YADAPTCOEFFSA - 2] * sign;
|
|
p->coeffsA[0][3] += p->buf[YADAPTCOEFFSA - 3] * sign;
|
|
|
|
p->buf++;
|
|
|
|
/* Have we filled the history buffer? */
|
|
if (p->buf == p->historybuffer + HISTORY_SIZE) {
|
|
memmove(p->historybuffer, p->buf,
|
|
PREDICTOR_SIZE * sizeof(*p->historybuffer));
|
|
p->buf = p->historybuffer;
|
|
}
|
|
|
|
p->filterA[0] = currentA + ((p->filterA[0] * 31) >> 5);
|
|
*(decoded0++) = p->filterA[0];
|
|
}
|
|
|
|
p->lastA[0] = currentA;
|
|
}
|
|
|
|
static void do_init_filter(APEFilter *f, int16_t *buf, int order)
|
|
{
|
|
f->coeffs = buf;
|
|
f->historybuffer = buf + order;
|
|
f->delay = f->historybuffer + order * 2;
|
|
f->adaptcoeffs = f->historybuffer + order;
|
|
|
|
memset(f->historybuffer, 0, (order * 2) * sizeof(*f->historybuffer));
|
|
memset(f->coeffs, 0, order * sizeof(*f->coeffs));
|
|
f->avg = 0;
|
|
}
|
|
|
|
static void init_filter(APEContext *ctx, APEFilter *f, int16_t *buf, int order)
|
|
{
|
|
do_init_filter(&f[0], buf, order);
|
|
do_init_filter(&f[1], buf + order * 3 + HISTORY_SIZE, order);
|
|
}
|
|
|
|
static void do_apply_filter(APEContext *ctx, int version, APEFilter *f,
|
|
int32_t *data, int count, int order, int fracbits)
|
|
{
|
|
int res;
|
|
int absres;
|
|
|
|
while (count--) {
|
|
/* round fixedpoint scalar product */
|
|
res = ctx->dsp.scalarproduct_and_madd_int16(f->coeffs, f->delay - order,
|
|
f->adaptcoeffs - order,
|
|
order, APESIGN(*data));
|
|
res = (res + (1 << (fracbits - 1))) >> fracbits;
|
|
res += *data;
|
|
*data++ = res;
|
|
|
|
/* Update the output history */
|
|
*f->delay++ = av_clip_int16(res);
|
|
|
|
if (version < 3980) {
|
|
/* Version ??? to < 3.98 files (untested) */
|
|
f->adaptcoeffs[0] = (res == 0) ? 0 : ((res >> 28) & 8) - 4;
|
|
f->adaptcoeffs[-4] >>= 1;
|
|
f->adaptcoeffs[-8] >>= 1;
|
|
} else {
|
|
/* Version 3.98 and later files */
|
|
|
|
/* Update the adaption coefficients */
|
|
absres = FFABS(res);
|
|
if (absres)
|
|
*f->adaptcoeffs = ((res & (-1<<31)) ^ (-1<<30)) >>
|
|
(25 + (absres <= f->avg*3) + (absres <= f->avg*4/3));
|
|
else
|
|
*f->adaptcoeffs = 0;
|
|
|
|
f->avg += (absres - f->avg) / 16;
|
|
|
|
f->adaptcoeffs[-1] >>= 1;
|
|
f->adaptcoeffs[-2] >>= 1;
|
|
f->adaptcoeffs[-8] >>= 1;
|
|
}
|
|
|
|
f->adaptcoeffs++;
|
|
|
|
/* Have we filled the history buffer? */
|
|
if (f->delay == f->historybuffer + HISTORY_SIZE + (order * 2)) {
|
|
memmove(f->historybuffer, f->delay - (order * 2),
|
|
(order * 2) * sizeof(*f->historybuffer));
|
|
f->delay = f->historybuffer + order * 2;
|
|
f->adaptcoeffs = f->historybuffer + order;
|
|
}
|
|
}
|
|
}
|
|
|
|
static void apply_filter(APEContext *ctx, APEFilter *f,
|
|
int32_t *data0, int32_t *data1,
|
|
int count, int order, int fracbits)
|
|
{
|
|
do_apply_filter(ctx, ctx->fileversion, &f[0], data0, count, order, fracbits);
|
|
if (data1)
|
|
do_apply_filter(ctx, ctx->fileversion, &f[1], data1, count, order, fracbits);
|
|
}
|
|
|
|
static void ape_apply_filters(APEContext *ctx, int32_t *decoded0,
|
|
int32_t *decoded1, int count)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < APE_FILTER_LEVELS; i++) {
|
|
if (!ape_filter_orders[ctx->fset][i])
|
|
break;
|
|
apply_filter(ctx, ctx->filters[i], decoded0, decoded1, count,
|
|
ape_filter_orders[ctx->fset][i],
|
|
ape_filter_fracbits[ctx->fset][i]);
|
|
}
|
|
}
|
|
|
|
static int init_frame_decoder(APEContext *ctx)
|
|
{
|
|
int i, ret;
|
|
if ((ret = init_entropy_decoder(ctx)) < 0)
|
|
return ret;
|
|
init_predictor_decoder(ctx);
|
|
|
|
for (i = 0; i < APE_FILTER_LEVELS; i++) {
|
|
if (!ape_filter_orders[ctx->fset][i])
|
|
break;
|
|
init_filter(ctx, ctx->filters[i], ctx->filterbuf[i],
|
|
ape_filter_orders[ctx->fset][i]);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static void ape_unpack_mono(APEContext *ctx, int count)
|
|
{
|
|
if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
|
|
/* We are pure silence, so we're done. */
|
|
av_log(ctx->avctx, AV_LOG_DEBUG, "pure silence mono\n");
|
|
return;
|
|
}
|
|
|
|
entropy_decode(ctx, count, 0);
|
|
ape_apply_filters(ctx, ctx->decoded[0], NULL, count);
|
|
|
|
/* Now apply the predictor decoding */
|
|
predictor_decode_mono(ctx, count);
|
|
|
|
/* Pseudo-stereo - just copy left channel to right channel */
|
|
if (ctx->channels == 2) {
|
|
memcpy(ctx->decoded[1], ctx->decoded[0], count * sizeof(*ctx->decoded[1]));
|
|
}
|
|
}
|
|
|
|
static void ape_unpack_stereo(APEContext *ctx, int count)
|
|
{
|
|
int32_t left, right;
|
|
int32_t *decoded0 = ctx->decoded[0];
|
|
int32_t *decoded1 = ctx->decoded[1];
|
|
|
|
if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
|
|
/* We are pure silence, so we're done. */
|
|
av_log(ctx->avctx, AV_LOG_DEBUG, "pure silence stereo\n");
|
|
return;
|
|
}
|
|
|
|
entropy_decode(ctx, count, 1);
|
|
ape_apply_filters(ctx, decoded0, decoded1, count);
|
|
|
|
/* Now apply the predictor decoding */
|
|
predictor_decode_stereo(ctx, count);
|
|
|
|
/* Decorrelate and scale to output depth */
|
|
while (count--) {
|
|
left = *decoded1 - (*decoded0 / 2);
|
|
right = left + *decoded0;
|
|
|
|
*(decoded0++) = left;
|
|
*(decoded1++) = right;
|
|
}
|
|
}
|
|
|
|
static int ape_decode_frame(AVCodecContext *avctx, void *data,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
const uint8_t *buf = avpkt->data;
|
|
APEContext *s = avctx->priv_data;
|
|
uint8_t *sample8;
|
|
int16_t *sample16;
|
|
int32_t *sample24;
|
|
int i, ret;
|
|
int blockstodecode;
|
|
int bytes_used = 0;
|
|
|
|
/* this should never be negative, but bad things will happen if it is, so
|
|
check it just to make sure. */
|
|
av_assert0(s->samples >= 0);
|
|
|
|
if(!s->samples){
|
|
uint32_t nblocks, offset;
|
|
int buf_size;
|
|
|
|
if (!avpkt->size) {
|
|
*got_frame_ptr = 0;
|
|
return 0;
|
|
}
|
|
if (avpkt->size < 8) {
|
|
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
buf_size = avpkt->size & ~3;
|
|
if (buf_size != avpkt->size) {
|
|
av_log(avctx, AV_LOG_WARNING, "packet size is not a multiple of 4. "
|
|
"extra bytes at the end will be skipped.\n");
|
|
}
|
|
|
|
av_fast_malloc(&s->data, &s->data_size, buf_size);
|
|
if (!s->data)
|
|
return AVERROR(ENOMEM);
|
|
s->dsp.bswap_buf((uint32_t*)s->data, (const uint32_t*)buf, buf_size >> 2);
|
|
s->ptr = s->data;
|
|
s->data_end = s->data + buf_size;
|
|
|
|
nblocks = bytestream_get_be32(&s->ptr);
|
|
offset = bytestream_get_be32(&s->ptr);
|
|
if (offset > 3) {
|
|
av_log(avctx, AV_LOG_ERROR, "Incorrect offset passed\n");
|
|
s->data = NULL;
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
if (s->data_end - s->ptr < offset) {
|
|
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
s->ptr += offset;
|
|
|
|
if (!nblocks || nblocks > INT_MAX) {
|
|
av_log(avctx, AV_LOG_ERROR, "Invalid sample count: %u.\n", nblocks);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
s->samples = nblocks;
|
|
|
|
/* Initialize the frame decoder */
|
|
if (init_frame_decoder(s) < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "Error reading frame header\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
bytes_used = avpkt->size;
|
|
}
|
|
|
|
if (!s->data) {
|
|
*got_frame_ptr = 0;
|
|
return avpkt->size;
|
|
}
|
|
|
|
blockstodecode = FFMIN(s->blocks_per_loop, s->samples);
|
|
|
|
/* reallocate decoded sample buffer if needed */
|
|
av_fast_malloc(&s->decoded_buffer, &s->decoded_size,
|
|
2 * FFALIGN(blockstodecode, 8) * sizeof(*s->decoded_buffer));
|
|
if (!s->decoded_buffer)
|
|
return AVERROR(ENOMEM);
|
|
memset(s->decoded_buffer, 0, s->decoded_size);
|
|
s->decoded[0] = s->decoded_buffer;
|
|
s->decoded[1] = s->decoded_buffer + FFALIGN(blockstodecode, 8);
|
|
|
|
/* get output buffer */
|
|
s->frame.nb_samples = blockstodecode;
|
|
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
|
|
return ret;
|
|
}
|
|
|
|
s->error=0;
|
|
|
|
if ((s->channels == 1) || (s->frameflags & APE_FRAMECODE_PSEUDO_STEREO))
|
|
ape_unpack_mono(s, blockstodecode);
|
|
else
|
|
ape_unpack_stereo(s, blockstodecode);
|
|
emms_c();
|
|
|
|
if (s->error) {
|
|
s->samples=0;
|
|
av_log(avctx, AV_LOG_ERROR, "Error decoding frame\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
switch (s->bps) {
|
|
case 8:
|
|
sample8 = (uint8_t *)s->frame.data[0];
|
|
for (i = 0; i < blockstodecode; i++) {
|
|
*sample8++ = (s->decoded[0][i] + 0x80) & 0xff;
|
|
if (s->channels == 2)
|
|
*sample8++ = (s->decoded[1][i] + 0x80) & 0xff;
|
|
}
|
|
break;
|
|
case 16:
|
|
sample16 = (int16_t *)s->frame.data[0];
|
|
for (i = 0; i < blockstodecode; i++) {
|
|
*sample16++ = s->decoded[0][i];
|
|
if (s->channels == 2)
|
|
*sample16++ = s->decoded[1][i];
|
|
}
|
|
break;
|
|
case 24:
|
|
sample24 = (int32_t *)s->frame.data[0];
|
|
for (i = 0; i < blockstodecode; i++) {
|
|
*sample24++ = s->decoded[0][i] << 8;
|
|
if (s->channels == 2)
|
|
*sample24++ = s->decoded[1][i] << 8;
|
|
}
|
|
break;
|
|
}
|
|
|
|
s->samples -= blockstodecode;
|
|
|
|
*got_frame_ptr = 1;
|
|
*(AVFrame *)data = s->frame;
|
|
|
|
return bytes_used;
|
|
}
|
|
|
|
static void ape_flush(AVCodecContext *avctx)
|
|
{
|
|
APEContext *s = avctx->priv_data;
|
|
s->samples= 0;
|
|
}
|
|
|
|
#define OFFSET(x) offsetof(APEContext, x)
|
|
#define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
|
|
static const AVOption options[] = {
|
|
{ "max_samples", "maximum number of samples decoded per call", OFFSET(blocks_per_loop), AV_OPT_TYPE_INT, { 4608 }, 1, INT_MAX, PAR, "max_samples" },
|
|
{ "all", "no maximum. decode all samples for each packet at once", 0, AV_OPT_TYPE_CONST, { INT_MAX }, INT_MIN, INT_MAX, PAR, "max_samples" },
|
|
{ NULL},
|
|
};
|
|
|
|
static const AVClass ape_decoder_class = {
|
|
.class_name = "APE decoder",
|
|
.item_name = av_default_item_name,
|
|
.option = options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVCodec ff_ape_decoder = {
|
|
.name = "ape",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = CODEC_ID_APE,
|
|
.priv_data_size = sizeof(APEContext),
|
|
.init = ape_decode_init,
|
|
.close = ape_decode_close,
|
|
.decode = ape_decode_frame,
|
|
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1,
|
|
.flush = ape_flush,
|
|
.long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
|
|
.priv_class = &ape_decoder_class,
|
|
};
|