* qatar/master: rtpdec_asf: Set the no_resync_search option for the chained asf demuxer asfdec: Add an option for not searching for the packet markers cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others cosmetics: Align codec declarations cosmetics: Convert mimic.c to utf-8 avconv: remove an unused function parameter. avconv: remove now pointless variables. avconv: drop support for building without libavfilter. nellymoserenc: fix crash due to memsetting the wrong area. libavformat: Only require first packet to be known for audio/video streams avplay: Don't try to scale timestamps if the tb isn't set Conflicts: Changelog configure ffmpeg.c libavcodec/aacenc.c libavcodec/bmpenc.c libavcodec/dnxhddec.c libavcodec/dnxhdenc.c libavcodec/ffv1.c libavcodec/flacenc.c libavcodec/fraps.c libavcodec/huffyuv.c libavcodec/libopenjpegdec.c libavcodec/mpeg12enc.c libavcodec/mpeg4videodec.c libavcodec/pamenc.c libavcodec/pgssubdec.c libavcodec/pngenc.c libavcodec/qtrleenc.c libavcodec/rawdec.c libavcodec/sgienc.c libavcodec/tiffenc.c libavcodec/v210dec.c libavcodec/wmv2dec.c libavformat/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
		
			
				
	
	
		
			643 lines
		
	
	
		
			19 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			643 lines
		
	
	
		
			19 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * Shorten decoder
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|  * Copyright (c) 2005 Jeff Muizelaar
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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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| /**
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|  * @file
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|  * Shorten decoder
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|  * @author Jeff Muizelaar
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|  *
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|  */
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| 
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| #include <limits.h>
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| #include "avcodec.h"
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| #include "bytestream.h"
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| #include "get_bits.h"
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| #include "golomb.h"
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| 
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| #define MAX_CHANNELS 8
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| #define MAX_BLOCKSIZE 65535
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| 
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| #define OUT_BUFFER_SIZE 16384
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| 
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| #define ULONGSIZE 2
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| 
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| #define WAVE_FORMAT_PCM 0x0001
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| 
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| #define DEFAULT_BLOCK_SIZE 256
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| 
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| #define TYPESIZE 4
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| #define CHANSIZE 0
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| #define LPCQSIZE 2
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| #define ENERGYSIZE 3
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| #define BITSHIFTSIZE 2
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| 
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| #define TYPE_S16HL 3
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| #define TYPE_S16LH 5
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| 
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| #define NWRAP 3
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| #define NSKIPSIZE 1
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| 
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| #define LPCQUANT 5
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| #define V2LPCQOFFSET (1 << LPCQUANT)
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| 
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| #define FNSIZE 2
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| #define FN_DIFF0        0
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| #define FN_DIFF1        1
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| #define FN_DIFF2        2
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| #define FN_DIFF3        3
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| #define FN_QUIT         4
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| #define FN_BLOCKSIZE    5
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| #define FN_BITSHIFT     6
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| #define FN_QLPC         7
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| #define FN_ZERO         8
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| #define FN_VERBATIM     9
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| 
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| /** indicates if the FN_* command is audio or non-audio */
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| static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 };
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| 
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| #define VERBATIM_CKSIZE_SIZE 5
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| #define VERBATIM_BYTE_SIZE 8
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| #define CANONICAL_HEADER_SIZE 44
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| 
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| typedef struct ShortenContext {
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|     AVCodecContext *avctx;
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|     AVFrame frame;
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|     GetBitContext gb;
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| 
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|     int min_framesize, max_framesize;
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|     int channels;
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| 
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|     int32_t *decoded[MAX_CHANNELS];
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|     int32_t *decoded_base[MAX_CHANNELS];
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|     int32_t *offset[MAX_CHANNELS];
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|     int *coeffs;
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|     uint8_t *bitstream;
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|     int bitstream_size;
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|     int bitstream_index;
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|     unsigned int allocated_bitstream_size;
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|     int header_size;
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|     uint8_t header[OUT_BUFFER_SIZE];
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|     int version;
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|     int cur_chan;
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|     int bitshift;
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|     int nmean;
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|     int internal_ftype;
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|     int nwrap;
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|     int blocksize;
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|     int bitindex;
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|     int32_t lpcqoffset;
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|     int got_header;
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|     int got_quit_command;
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| } ShortenContext;
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| 
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| static av_cold int shorten_decode_init(AVCodecContext * avctx)
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| {
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|     ShortenContext *s = avctx->priv_data;
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|     s->avctx = avctx;
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|     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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| 
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|     avcodec_get_frame_defaults(&s->frame);
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|     avctx->coded_frame = &s->frame;
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| 
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|     return 0;
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| }
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| 
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| static int allocate_buffers(ShortenContext *s)
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| {
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|     int i, chan;
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|     int *coeffs;
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|     void *tmp_ptr;
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| 
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|     for (chan=0; chan<s->channels; chan++) {
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|         if(FFMAX(1, s->nmean) >= UINT_MAX/sizeof(int32_t)){
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|             av_log(s->avctx, AV_LOG_ERROR, "nmean too large\n");
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|             return -1;
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|         }
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|         if(s->blocksize + s->nwrap >= UINT_MAX/sizeof(int32_t) || s->blocksize + s->nwrap <= (unsigned)s->nwrap){
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|             av_log(s->avctx, AV_LOG_ERROR, "s->blocksize + s->nwrap too large\n");
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|             return -1;
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|         }
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| 
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|         tmp_ptr = av_realloc(s->offset[chan], sizeof(int32_t)*FFMAX(1, s->nmean));
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|         if (!tmp_ptr)
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|             return AVERROR(ENOMEM);
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|         s->offset[chan] = tmp_ptr;
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| 
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|         tmp_ptr = av_realloc(s->decoded_base[chan], (s->blocksize + s->nwrap) *
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|                              sizeof(s->decoded_base[0][0]));
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|         if (!tmp_ptr)
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|             return AVERROR(ENOMEM);
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|         s->decoded_base[chan] = tmp_ptr;
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|         for (i=0; i<s->nwrap; i++)
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|             s->decoded_base[chan][i] = 0;
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|         s->decoded[chan] = s->decoded_base[chan] + s->nwrap;
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|     }
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| 
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|     coeffs = av_realloc(s->coeffs, s->nwrap * sizeof(*s->coeffs));
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|     if (!coeffs)
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|         return AVERROR(ENOMEM);
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|     s->coeffs = coeffs;
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| 
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|     return 0;
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| }
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| 
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| 
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| static inline unsigned int get_uint(ShortenContext *s, int k)
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| {
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|     if (s->version != 0)
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|         k = get_ur_golomb_shorten(&s->gb, ULONGSIZE);
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|     return get_ur_golomb_shorten(&s->gb, k);
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| }
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| 
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| 
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| static void fix_bitshift(ShortenContext *s, int32_t *buffer)
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| {
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|     int i;
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| 
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|     if (s->bitshift != 0)
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|         for (i = 0; i < s->blocksize; i++)
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|             buffer[i] <<= s->bitshift;
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| }
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| 
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| 
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| static int init_offset(ShortenContext *s)
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| {
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|     int32_t mean = 0;
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|     int  chan, i;
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|     int nblock = FFMAX(1, s->nmean);
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|     /* initialise offset */
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|     switch (s->internal_ftype)
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|     {
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|         case TYPE_S16HL:
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|         case TYPE_S16LH:
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|             mean = 0;
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|             break;
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|         default:
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|             av_log(s->avctx, AV_LOG_ERROR, "unknown audio type");
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|             return AVERROR_INVALIDDATA;
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|     }
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| 
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|     for (chan = 0; chan < s->channels; chan++)
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|         for (i = 0; i < nblock; i++)
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|             s->offset[chan][i] = mean;
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|     return 0;
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| }
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| 
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| static int decode_wave_header(AVCodecContext *avctx, const uint8_t *header,
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|                               int header_size)
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| {
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|     int len;
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|     short wave_format;
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|     const uint8_t *end= header + header_size;
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| 
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|     if (bytestream_get_le32(&header) != MKTAG('R','I','F','F')) {
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|         av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n");
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|         return -1;
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|     }
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| 
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|     header += 4; /* chunk size */;
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| 
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|     if (bytestream_get_le32(&header) != MKTAG('W','A','V','E')) {
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|         av_log(avctx, AV_LOG_ERROR, "missing WAVE tag\n");
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|         return -1;
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|     }
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| 
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|     while (bytestream_get_le32(&header) != MKTAG('f','m','t',' ')) {
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|         len = bytestream_get_le32(&header);
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|         if(len<0 || end - header - 8 < len)
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|             return AVERROR_INVALIDDATA;
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|         header += len;
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|     }
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|     len = bytestream_get_le32(&header);
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| 
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|     if (len < 16) {
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|         av_log(avctx, AV_LOG_ERROR, "fmt chunk was too short\n");
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|         return -1;
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|     }
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| 
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|     wave_format = bytestream_get_le16(&header);
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| 
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|     switch (wave_format) {
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|         case WAVE_FORMAT_PCM:
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|             break;
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|         default:
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|             av_log(avctx, AV_LOG_ERROR, "unsupported wave format\n");
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|             return -1;
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|     }
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| 
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|     header += 2;        // skip channels    (already got from shorten header)
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|     avctx->sample_rate = bytestream_get_le32(&header);
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|     header += 4;        // skip bit rate    (represents original uncompressed bit rate)
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|     header += 2;        // skip block align (not needed)
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|     avctx->bits_per_coded_sample = bytestream_get_le16(&header);
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| 
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|     if (avctx->bits_per_coded_sample != 16) {
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|         av_log(avctx, AV_LOG_ERROR, "unsupported number of bits per sample\n");
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|         return -1;
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|     }
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| 
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|     len -= 16;
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|     if (len > 0)
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|         av_log(avctx, AV_LOG_INFO, "%d header bytes unparsed\n", len);
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| 
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|     return 0;
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| }
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| 
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| static void interleave_buffer(int16_t *samples, int nchan, int blocksize,
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|                               int32_t **buffer)
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| {
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|     int i, chan;
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|     for (i=0; i<blocksize; i++)
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|         for (chan=0; chan < nchan; chan++)
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|             *samples++ = av_clip_int16(buffer[chan][i]);
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| }
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| 
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| static const int fixed_coeffs[3][3] = {
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|     { 1,  0,  0 },
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|     { 2, -1,  0 },
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|     { 3, -3,  1 }
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| };
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| 
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| static int decode_subframe_lpc(ShortenContext *s, int command, int channel,
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|                                int residual_size, int32_t coffset)
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| {
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|     int pred_order, sum, qshift, init_sum, i, j;
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|     const int *coeffs;
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| 
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|     if (command == FN_QLPC) {
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|         /* read/validate prediction order */
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|         pred_order = get_ur_golomb_shorten(&s->gb, LPCQSIZE);
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|         if (pred_order > s->nwrap) {
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|             av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n", pred_order);
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|             return AVERROR(EINVAL);
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|         }
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|         /* read LPC coefficients */
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|         for (i=0; i<pred_order; i++)
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|             s->coeffs[i] = get_sr_golomb_shorten(&s->gb, LPCQUANT);
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|         coeffs = s->coeffs;
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| 
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|         qshift = LPCQUANT;
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|     } else {
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|         /* fixed LPC coeffs */
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|         pred_order = command;
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|         coeffs     = fixed_coeffs[pred_order-1];
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|         qshift     = 0;
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|     }
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| 
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|     /* subtract offset from previous samples to use in prediction */
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|     if (command == FN_QLPC && coffset)
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|         for (i = -pred_order; i < 0; i++)
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|             s->decoded[channel][i] -= coffset;
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| 
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|     /* decode residual and do LPC prediction */
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|     init_sum = pred_order ? (command == FN_QLPC ? s->lpcqoffset : 0) : coffset;
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|     for (i=0; i < s->blocksize; i++) {
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|         sum = init_sum;
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|         for (j=0; j<pred_order; j++)
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|             sum += coeffs[j] * s->decoded[channel][i-j-1];
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|         s->decoded[channel][i] = get_sr_golomb_shorten(&s->gb, residual_size) + (sum >> qshift);
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|     }
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| 
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|     /* add offset to current samples */
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|     if (command == FN_QLPC && coffset)
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|         for (i = 0; i < s->blocksize; i++)
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|             s->decoded[channel][i] += coffset;
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| 
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|     return 0;
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| }
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| 
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| static int read_header(ShortenContext *s)
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| {
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|     int i, ret;
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|     int maxnlpc = 0;
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|     /* shorten signature */
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|     if (get_bits_long(&s->gb, 32) != AV_RB32("ajkg")) {
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|         av_log(s->avctx, AV_LOG_ERROR, "missing shorten magic 'ajkg'\n");
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|         return -1;
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|     }
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| 
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|     s->lpcqoffset = 0;
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|     s->blocksize = DEFAULT_BLOCK_SIZE;
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|     s->nmean = -1;
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|     s->version = get_bits(&s->gb, 8);
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|     s->internal_ftype = get_uint(s, TYPESIZE);
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| 
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|     s->channels = get_uint(s, CHANSIZE);
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|     if (s->channels > MAX_CHANNELS) {
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|         av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
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|         return -1;
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|     }
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|     s->avctx->channels = s->channels;
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| 
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|     /* get blocksize if version > 0 */
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|     if (s->version > 0) {
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|         int skip_bytes, blocksize;
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| 
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|         blocksize = get_uint(s, av_log2(DEFAULT_BLOCK_SIZE));
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|         if (!blocksize || blocksize > MAX_BLOCKSIZE) {
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|             av_log(s->avctx, AV_LOG_ERROR, "invalid or unsupported block size: %d\n",
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|                    blocksize);
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|             return AVERROR(EINVAL);
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|         }
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|         s->blocksize = blocksize;
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| 
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|         maxnlpc = get_uint(s, LPCQSIZE);
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|         s->nmean = get_uint(s, 0);
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| 
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|         skip_bytes = get_uint(s, NSKIPSIZE);
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|         for (i=0; i<skip_bytes; i++) {
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|             skip_bits(&s->gb, 8);
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|         }
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|     }
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|     s->nwrap = FFMAX(NWRAP, maxnlpc);
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| 
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|     if ((ret = allocate_buffers(s)) < 0)
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|         return ret;
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| 
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|     if ((ret = init_offset(s)) < 0)
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|         return ret;
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| 
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|     if (s->version > 1)
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|         s->lpcqoffset = V2LPCQOFFSET;
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| 
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|     if (get_ur_golomb_shorten(&s->gb, FNSIZE) != FN_VERBATIM) {
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|         av_log(s->avctx, AV_LOG_ERROR, "missing verbatim section at beginning of stream\n");
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|         return -1;
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|     }
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| 
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|     s->header_size = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE);
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|     if (s->header_size >= OUT_BUFFER_SIZE || s->header_size < CANONICAL_HEADER_SIZE) {
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|         av_log(s->avctx, AV_LOG_ERROR, "header is wrong size: %d\n", s->header_size);
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|         return -1;
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|     }
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| 
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|     for (i=0; i<s->header_size; i++)
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|         s->header[i] = (char)get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
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| 
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|     if (decode_wave_header(s->avctx, s->header, s->header_size) < 0)
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|         return -1;
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| 
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|     s->cur_chan = 0;
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|     s->bitshift = 0;
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| 
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|     s->got_header = 1;
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| 
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|     return 0;
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| }
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| 
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| static int shorten_decode_frame(AVCodecContext *avctx, void *data,
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|                                 int *got_frame_ptr, AVPacket *avpkt)
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| {
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|     const uint8_t *buf = avpkt->data;
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|     int buf_size = avpkt->size;
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|     ShortenContext *s = avctx->priv_data;
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|     int i, input_buf_size = 0;
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|     int ret;
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| 
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|     /* allocate internal bitstream buffer */
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|     if(s->max_framesize == 0){
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|         void *tmp_ptr;
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|         s->max_framesize= 1024; // should hopefully be enough for the first header
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|         tmp_ptr = av_fast_realloc(s->bitstream, &s->allocated_bitstream_size,
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|                                   s->max_framesize);
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|         if (!tmp_ptr) {
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|             av_log(avctx, AV_LOG_ERROR, "error allocating bitstream buffer\n");
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|             return AVERROR(ENOMEM);
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|         }
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|         s->bitstream = tmp_ptr;
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|     }
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| 
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|     /* append current packet data to bitstream buffer */
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|     if(1 && s->max_framesize){//FIXME truncated
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|         buf_size= FFMIN(buf_size, s->max_framesize - s->bitstream_size);
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|         input_buf_size= buf_size;
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| 
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|         if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
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|             memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
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|             s->bitstream_index=0;
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|         }
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|         if (buf)
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|             memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size);
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|         buf= &s->bitstream[s->bitstream_index];
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|         buf_size += s->bitstream_size;
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|         s->bitstream_size= buf_size;
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| 
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|         /* do not decode until buffer has at least max_framesize bytes or
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|            the end of the file has been reached */
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|         if (buf_size < s->max_framesize && avpkt->data) {
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|             *got_frame_ptr = 0;
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|             return input_buf_size;
 | |
|         }
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|     }
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|     /* init and position bitstream reader */
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|     init_get_bits(&s->gb, buf, buf_size*8);
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|     skip_bits(&s->gb, s->bitindex);
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| 
 | |
|     /* process header or next subblock */
 | |
|     if (!s->got_header) {
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|         if ((ret = read_header(s)) < 0)
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|             return ret;
 | |
|         *got_frame_ptr = 0;
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|         goto finish_frame;
 | |
|     }
 | |
| 
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|     /* if quit command was read previously, don't decode anything */
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|     if (s->got_quit_command) {
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|         *got_frame_ptr = 0;
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|         return avpkt->size;
 | |
|     }
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| 
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|     s->cur_chan = 0;
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|     while (s->cur_chan < s->channels) {
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|         int cmd;
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|         int len;
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| 
 | |
|         if (get_bits_left(&s->gb) < 3+FNSIZE) {
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|             *got_frame_ptr = 0;
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|             break;
 | |
|         }
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| 
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|         cmd = get_ur_golomb_shorten(&s->gb, FNSIZE);
 | |
| 
 | |
|         if (cmd > FN_VERBATIM) {
 | |
|             av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd);
 | |
|             *got_frame_ptr = 0;
 | |
|             break;
 | |
|         }
 | |
| 
 | |
|         if (!is_audio_command[cmd]) {
 | |
|             /* process non-audio command */
 | |
|             switch (cmd) {
 | |
|                 case FN_VERBATIM:
 | |
|                     len = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE);
 | |
|                     while (len--) {
 | |
|                         get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
 | |
|                     }
 | |
|                     break;
 | |
|                 case FN_BITSHIFT:
 | |
|                     s->bitshift = get_ur_golomb_shorten(&s->gb, BITSHIFTSIZE);
 | |
|                     break;
 | |
|                 case FN_BLOCKSIZE: {
 | |
|                     int blocksize = get_uint(s, av_log2(s->blocksize));
 | |
|                     if (blocksize > s->blocksize) {
 | |
|                         av_log(avctx, AV_LOG_ERROR, "Increasing block size is not supported\n");
 | |
|                         return AVERROR_PATCHWELCOME;
 | |
|                     }
 | |
|                     if (!blocksize || blocksize > MAX_BLOCKSIZE) {
 | |
|                         av_log(avctx, AV_LOG_ERROR, "invalid or unsupported "
 | |
|                                "block size: %d\n", blocksize);
 | |
|                         return AVERROR(EINVAL);
 | |
|                     }
 | |
|                     s->blocksize = blocksize;
 | |
|                     break;
 | |
|                 }
 | |
|                 case FN_QUIT:
 | |
|                     s->got_quit_command = 1;
 | |
|                     break;
 | |
|             }
 | |
|             if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) {
 | |
|                 *got_frame_ptr = 0;
 | |
|                 break;
 | |
|             }
 | |
|         } else {
 | |
|             /* process audio command */
 | |
|             int residual_size = 0;
 | |
|             int channel = s->cur_chan;
 | |
|             int32_t coffset;
 | |
| 
 | |
|             /* get Rice code for residual decoding */
 | |
|             if (cmd != FN_ZERO) {
 | |
|                 residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE);
 | |
|                 /* this is a hack as version 0 differed in defintion of get_sr_golomb_shorten */
 | |
|                 if (s->version == 0)
 | |
|                     residual_size--;
 | |
|             }
 | |
| 
 | |
|             /* calculate sample offset using means from previous blocks */
 | |
|             if (s->nmean == 0)
 | |
|                 coffset = s->offset[channel][0];
 | |
|             else {
 | |
|                 int32_t sum = (s->version < 2) ? 0 : s->nmean / 2;
 | |
|                 for (i=0; i<s->nmean; i++)
 | |
|                     sum += s->offset[channel][i];
 | |
|                 coffset = sum / s->nmean;
 | |
|                 if (s->version >= 2)
 | |
|                     coffset >>= FFMIN(1, s->bitshift);
 | |
|             }
 | |
| 
 | |
|             /* decode samples for this channel */
 | |
|             if (cmd == FN_ZERO) {
 | |
|                 for (i=0; i<s->blocksize; i++)
 | |
|                     s->decoded[channel][i] = 0;
 | |
|             } else {
 | |
|                 if ((ret = decode_subframe_lpc(s, cmd, channel, residual_size, coffset)) < 0)
 | |
|                     return ret;
 | |
|             }
 | |
| 
 | |
|             /* update means with info from the current block */
 | |
|             if (s->nmean > 0) {
 | |
|                 int32_t sum = (s->version < 2) ? 0 : s->blocksize / 2;
 | |
|                 for (i=0; i<s->blocksize; i++)
 | |
|                     sum += s->decoded[channel][i];
 | |
| 
 | |
|                 for (i=1; i<s->nmean; i++)
 | |
|                     s->offset[channel][i-1] = s->offset[channel][i];
 | |
| 
 | |
|                 if (s->version < 2)
 | |
|                     s->offset[channel][s->nmean - 1] = sum / s->blocksize;
 | |
|                 else
 | |
|                     s->offset[channel][s->nmean - 1] = (sum / s->blocksize) << s->bitshift;
 | |
|             }
 | |
| 
 | |
|             /* copy wrap samples for use with next block */
 | |
|             for (i=-s->nwrap; i<0; i++)
 | |
|                 s->decoded[channel][i] = s->decoded[channel][i + s->blocksize];
 | |
| 
 | |
|             /* shift samples to add in unused zero bits which were removed
 | |
|                during encoding */
 | |
|             fix_bitshift(s, s->decoded[channel]);
 | |
| 
 | |
|             /* if this is the last channel in the block, output the samples */
 | |
|             s->cur_chan++;
 | |
|             if (s->cur_chan == s->channels) {
 | |
|                 /* get output buffer */
 | |
|                 s->frame.nb_samples = s->blocksize;
 | |
|                 if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
 | |
|                     av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
 | |
|                     return ret;
 | |
|                 }
 | |
|                 /* interleave output */
 | |
|                 interleave_buffer((int16_t *)s->frame.data[0], s->channels,
 | |
|                                   s->blocksize, s->decoded);
 | |
| 
 | |
|                 *got_frame_ptr   = 1;
 | |
|                 *(AVFrame *)data = s->frame;
 | |
|             }
 | |
|         }
 | |
|     }
 | |
|     if (s->cur_chan < s->channels)
 | |
|         *got_frame_ptr = 0;
 | |
| 
 | |
| finish_frame:
 | |
|     s->bitindex = get_bits_count(&s->gb) - 8*((get_bits_count(&s->gb))/8);
 | |
|     i= (get_bits_count(&s->gb))/8;
 | |
|     if (i > buf_size) {
 | |
|         av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
 | |
|         s->bitstream_size=0;
 | |
|         s->bitstream_index=0;
 | |
|         return -1;
 | |
|     }
 | |
|     if (s->bitstream_size) {
 | |
|         s->bitstream_index += i;
 | |
|         s->bitstream_size  -= i;
 | |
|         return input_buf_size;
 | |
|     } else
 | |
|         return i;
 | |
| }
 | |
| 
 | |
| static av_cold int shorten_decode_close(AVCodecContext *avctx)
 | |
| {
 | |
|     ShortenContext *s = avctx->priv_data;
 | |
|     int i;
 | |
| 
 | |
|     for (i = 0; i < s->channels; i++) {
 | |
|         s->decoded[i] = NULL;
 | |
|         av_freep(&s->decoded_base[i]);
 | |
|         av_freep(&s->offset[i]);
 | |
|     }
 | |
|     av_freep(&s->bitstream);
 | |
|     av_freep(&s->coeffs);
 | |
| 
 | |
|     return 0;
 | |
| }
 | |
| 
 | |
| AVCodec ff_shorten_decoder = {
 | |
|     .name           = "shorten",
 | |
|     .type           = AVMEDIA_TYPE_AUDIO,
 | |
|     .id             = CODEC_ID_SHORTEN,
 | |
|     .priv_data_size = sizeof(ShortenContext),
 | |
|     .init           = shorten_decode_init,
 | |
|     .close          = shorten_decode_close,
 | |
|     .decode         = shorten_decode_frame,
 | |
|     .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,
 | |
|     .long_name      = NULL_IF_CONFIG_SMALL("Shorten"),
 | |
| };
 |