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790 lines
23 KiB
790 lines
23 KiB
/* |
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* FLAC (Free Lossless Audio Codec) decoder |
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* Copyright (c) 2003 Alex Beregszaszi |
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* |
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* This library 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 of the License, or (at your option) any later version. |
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* |
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* This library 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 this library; if not, write to the Free Software |
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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*/ |
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|
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/** |
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* @file flac.c |
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* FLAC (Free Lossless Audio Codec) decoder |
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* @author Alex Beregszaszi |
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* |
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* For more information on the FLAC format, visit: |
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* http://flac.sourceforge.net/ |
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* |
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* This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed |
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* through, starting from the initial 'fLaC' signature; or by passing the |
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* 34-byte streaminfo structure through avctx->extradata[_size] followed |
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* by data starting with the 0xFFF8 marker. |
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*/ |
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|
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#include <limits.h> |
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|
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#include "avcodec.h" |
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#include "golomb.h" |
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|
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#undef NDEBUG |
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#include <assert.h> |
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#define MAX_CHANNELS 8 |
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#define MAX_BLOCKSIZE 65535 |
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#define FLAC_STREAMINFO_SIZE 34 |
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enum decorrelation_type { |
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INDEPENDENT, |
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LEFT_SIDE, |
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RIGHT_SIDE, |
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MID_SIDE, |
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}; |
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typedef struct FLACContext { |
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AVCodecContext *avctx; |
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GetBitContext gb; |
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int min_blocksize, max_blocksize; |
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int min_framesize, max_framesize; |
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int samplerate, channels; |
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int blocksize/*, last_blocksize*/; |
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int bps, curr_bps; |
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enum decorrelation_type decorrelation; |
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int32_t *decoded[MAX_CHANNELS]; |
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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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int allocated_bitstream_size; |
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} FLACContext; |
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#define METADATA_TYPE_STREAMINFO 0 |
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static int sample_rate_table[] = |
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{ 0, 0, 0, 0, |
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8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000, |
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0, 0, 0, 0 }; |
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static int sample_size_table[] = |
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{ 0, 8, 12, 0, 16, 20, 24, 0 }; |
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static int blocksize_table[] = { |
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0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0, |
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256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7 |
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}; |
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static const uint8_t table_crc8[256] = { |
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0x00, 0x07, 0x0e, 0x09, 0x1c, 0x1b, 0x12, 0x15, |
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0x38, 0x3f, 0x36, 0x31, 0x24, 0x23, 0x2a, 0x2d, |
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0x70, 0x77, 0x7e, 0x79, 0x6c, 0x6b, 0x62, 0x65, |
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0x48, 0x4f, 0x46, 0x41, 0x54, 0x53, 0x5a, 0x5d, |
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0xe0, 0xe7, 0xee, 0xe9, 0xfc, 0xfb, 0xf2, 0xf5, |
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0xd8, 0xdf, 0xd6, 0xd1, 0xc4, 0xc3, 0xca, 0xcd, |
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0x90, 0x97, 0x9e, 0x99, 0x8c, 0x8b, 0x82, 0x85, |
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0xa8, 0xaf, 0xa6, 0xa1, 0xb4, 0xb3, 0xba, 0xbd, |
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0xc7, 0xc0, 0xc9, 0xce, 0xdb, 0xdc, 0xd5, 0xd2, |
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0xff, 0xf8, 0xf1, 0xf6, 0xe3, 0xe4, 0xed, 0xea, |
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0xb7, 0xb0, 0xb9, 0xbe, 0xab, 0xac, 0xa5, 0xa2, |
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0x8f, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9d, 0x9a, |
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0x27, 0x20, 0x29, 0x2e, 0x3b, 0x3c, 0x35, 0x32, |
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0x1f, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0d, 0x0a, |
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0x57, 0x50, 0x59, 0x5e, 0x4b, 0x4c, 0x45, 0x42, |
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0x6f, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7d, 0x7a, |
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0x89, 0x8e, 0x87, 0x80, 0x95, 0x92, 0x9b, 0x9c, |
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0xb1, 0xb6, 0xbf, 0xb8, 0xad, 0xaa, 0xa3, 0xa4, |
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0xf9, 0xfe, 0xf7, 0xf0, 0xe5, 0xe2, 0xeb, 0xec, |
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0xc1, 0xc6, 0xcf, 0xc8, 0xdd, 0xda, 0xd3, 0xd4, |
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0x69, 0x6e, 0x67, 0x60, 0x75, 0x72, 0x7b, 0x7c, |
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0x51, 0x56, 0x5f, 0x58, 0x4d, 0x4a, 0x43, 0x44, |
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0x19, 0x1e, 0x17, 0x10, 0x05, 0x02, 0x0b, 0x0c, |
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0x21, 0x26, 0x2f, 0x28, 0x3d, 0x3a, 0x33, 0x34, |
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0x4e, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5c, 0x5b, |
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0x76, 0x71, 0x78, 0x7f, 0x6a, 0x6d, 0x64, 0x63, |
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0x3e, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2c, 0x2b, |
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0x06, 0x01, 0x08, 0x0f, 0x1a, 0x1d, 0x14, 0x13, |
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0xae, 0xa9, 0xa0, 0xa7, 0xb2, 0xb5, 0xbc, 0xbb, |
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0x96, 0x91, 0x98, 0x9f, 0x8a, 0x8d, 0x84, 0x83, |
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0xde, 0xd9, 0xd0, 0xd7, 0xc2, 0xc5, 0xcc, 0xcb, |
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0xe6, 0xe1, 0xe8, 0xef, 0xfa, 0xfd, 0xf4, 0xf3 |
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}; |
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static int64_t get_utf8(GetBitContext *gb) |
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{ |
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uint64_t val; |
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int ones=0, bytes; |
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while(get_bits1(gb)) |
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ones++; |
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if (ones==0) bytes=0; |
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else if(ones==1) return -1; |
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else bytes= ones - 1; |
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val= get_bits(gb, 7-ones); |
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while(bytes--){ |
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const int tmp = get_bits(gb, 8); |
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if((tmp>>6) != 2) |
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return -1; |
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val<<=6; |
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val|= tmp&0x3F; |
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} |
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return val; |
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} |
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static int get_crc8(const uint8_t *buf, int count){ |
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int crc=0; |
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int i; |
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for(i=0; i<count; i++){ |
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crc = table_crc8[crc ^ buf[i]]; |
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} |
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return crc; |
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} |
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static void metadata_streaminfo(FLACContext *s); |
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static void dump_headers(FLACContext *s); |
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static int flac_decode_init(AVCodecContext * avctx) |
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{ |
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FLACContext *s = avctx->priv_data; |
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s->avctx = avctx; |
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|
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/* initialize based on the demuxer-supplied streamdata header */ |
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if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) { |
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init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8); |
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metadata_streaminfo(s); |
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dump_headers(s); |
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} |
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return 0; |
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} |
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static void dump_headers(FLACContext *s) |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d (%d)\n", s->min_blocksize, s->max_blocksize, s->blocksize); |
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av_log(s->avctx, AV_LOG_DEBUG, " Framesize: %d .. %d\n", s->min_framesize, s->max_framesize); |
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av_log(s->avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate); |
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av_log(s->avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels); |
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av_log(s->avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps); |
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} |
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static void allocate_buffers(FLACContext *s){ |
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int i; |
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assert(s->max_blocksize); |
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if(s->max_framesize == 0 && s->max_blocksize){ |
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s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; //FIXME header overhead |
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} |
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for (i = 0; i < s->channels; i++) |
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{ |
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s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t)*s->max_blocksize); |
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} |
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s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize); |
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} |
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static void metadata_streaminfo(FLACContext *s) |
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{ |
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/* mandatory streaminfo */ |
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s->min_blocksize = get_bits(&s->gb, 16); |
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s->max_blocksize = get_bits(&s->gb, 16); |
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s->min_framesize = get_bits_long(&s->gb, 24); |
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s->max_framesize = get_bits_long(&s->gb, 24); |
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s->samplerate = get_bits_long(&s->gb, 20); |
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s->channels = get_bits(&s->gb, 3) + 1; |
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s->bps = get_bits(&s->gb, 5) + 1; |
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s->avctx->channels = s->channels; |
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s->avctx->sample_rate = s->samplerate; |
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skip_bits(&s->gb, 36); /* total num of samples */ |
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skip_bits(&s->gb, 64); /* md5 sum */ |
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skip_bits(&s->gb, 64); /* md5 sum */ |
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allocate_buffers(s); |
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} |
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static int decode_residuals(FLACContext *s, int channel, int pred_order) |
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{ |
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int i, tmp, partition, method_type, rice_order; |
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int sample = 0, samples; |
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method_type = get_bits(&s->gb, 2); |
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if (method_type != 0){ |
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av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", method_type); |
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return -1; |
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} |
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rice_order = get_bits(&s->gb, 4); |
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samples= s->blocksize >> rice_order; |
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sample= |
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i= pred_order; |
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for (partition = 0; partition < (1 << rice_order); partition++) |
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{ |
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tmp = get_bits(&s->gb, 4); |
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if (tmp == 15) |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "fixed len partition\n"); |
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tmp = get_bits(&s->gb, 5); |
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for (; i < samples; i++, sample++) |
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s->decoded[channel][sample] = get_sbits(&s->gb, tmp); |
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} |
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else |
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{ |
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// av_log(s->avctx, AV_LOG_DEBUG, "rice coded partition k=%d\n", tmp); |
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for (; i < samples; i++, sample++){ |
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s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0); |
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} |
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} |
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i= 0; |
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} |
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// av_log(s->avctx, AV_LOG_DEBUG, "partitions: %d, samples: %d\n", 1 << rice_order, sample); |
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return 0; |
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} |
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static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order) |
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{ |
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int i; |
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// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME FIXED\n"); |
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/* warm up samples */ |
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// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order); |
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for (i = 0; i < pred_order; i++) |
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{ |
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s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps); |
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// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]); |
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} |
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if (decode_residuals(s, channel, pred_order) < 0) |
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return -1; |
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switch(pred_order) |
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{ |
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case 0: |
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break; |
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case 1: |
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for (i = pred_order; i < s->blocksize; i++) |
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s->decoded[channel][i] += s->decoded[channel][i-1]; |
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break; |
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case 2: |
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for (i = pred_order; i < s->blocksize; i++) |
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s->decoded[channel][i] += 2*s->decoded[channel][i-1] |
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- s->decoded[channel][i-2]; |
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break; |
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case 3: |
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for (i = pred_order; i < s->blocksize; i++) |
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s->decoded[channel][i] += 3*s->decoded[channel][i-1] |
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- 3*s->decoded[channel][i-2] |
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+ s->decoded[channel][i-3]; |
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break; |
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case 4: |
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for (i = pred_order; i < s->blocksize; i++) |
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s->decoded[channel][i] += 4*s->decoded[channel][i-1] |
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- 6*s->decoded[channel][i-2] |
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+ 4*s->decoded[channel][i-3] |
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- s->decoded[channel][i-4]; |
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break; |
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default: |
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av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order); |
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return -1; |
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} |
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return 0; |
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} |
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static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order) |
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{ |
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int sum, i, j; |
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int coeff_prec, qlevel; |
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int coeffs[pred_order]; |
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// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME LPC\n"); |
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/* warm up samples */ |
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// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order); |
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for (i = 0; i < pred_order; i++) |
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{ |
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s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps); |
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// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]); |
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} |
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coeff_prec = get_bits(&s->gb, 4) + 1; |
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if (coeff_prec == 16) |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "invalid coeff precision\n"); |
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return -1; |
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} |
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// av_log(s->avctx, AV_LOG_DEBUG, " qlp coeff prec: %d\n", coeff_prec); |
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qlevel = get_sbits(&s->gb, 5); |
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// av_log(s->avctx, AV_LOG_DEBUG, " quant level: %d\n", qlevel); |
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if(qlevel < 0){ |
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av_log(s->avctx, AV_LOG_DEBUG, "qlevel %d not supported, maybe buggy stream\n", qlevel); |
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return -1; |
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} |
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for (i = 0; i < pred_order; i++) |
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{ |
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coeffs[i] = get_sbits(&s->gb, coeff_prec); |
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// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, coeffs[i]); |
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} |
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if (decode_residuals(s, channel, pred_order) < 0) |
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return -1; |
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for (i = pred_order; i < s->blocksize; i++) |
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{ |
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sum = 0; |
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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] += sum >> qlevel; |
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} |
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return 0; |
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} |
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static inline int decode_subframe(FLACContext *s, int channel) |
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{ |
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int type, wasted = 0; |
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int i, tmp; |
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s->curr_bps = s->bps; |
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if(channel == 0){ |
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if(s->decorrelation == RIGHT_SIDE) |
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s->curr_bps++; |
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}else{ |
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if(s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE) |
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s->curr_bps++; |
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} |
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if (get_bits1(&s->gb)) |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "invalid subframe padding\n"); |
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return -1; |
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} |
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type = get_bits(&s->gb, 6); |
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// wasted = get_bits1(&s->gb); |
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|
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// if (wasted) |
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// { |
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// while (!get_bits1(&s->gb)) |
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// wasted++; |
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// if (wasted) |
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// wasted++; |
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// s->curr_bps -= wasted; |
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// } |
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#if 0 |
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wasted= 16 - av_log2(show_bits(&s->gb, 17)); |
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skip_bits(&s->gb, wasted+1); |
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s->curr_bps -= wasted; |
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#else |
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if (get_bits1(&s->gb)) |
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{ |
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wasted = 1; |
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while (!get_bits1(&s->gb)) |
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wasted++; |
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s->curr_bps -= wasted; |
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av_log(s->avctx, AV_LOG_DEBUG, "%d wasted bits\n", wasted); |
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} |
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#endif |
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//FIXME use av_log2 for types |
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if (type == 0) |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "coding type: constant\n"); |
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tmp = get_sbits(&s->gb, s->curr_bps); |
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for (i = 0; i < s->blocksize; i++) |
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s->decoded[channel][i] = tmp; |
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} |
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else if (type == 1) |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "coding type: verbatim\n"); |
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for (i = 0; i < s->blocksize; i++) |
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s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps); |
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} |
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else if ((type >= 8) && (type <= 12)) |
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{ |
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// av_log(s->avctx, AV_LOG_DEBUG, "coding type: fixed\n"); |
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if (decode_subframe_fixed(s, channel, type & ~0x8) < 0) |
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return -1; |
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} |
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else if (type >= 32) |
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{ |
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// av_log(s->avctx, AV_LOG_DEBUG, "coding type: lpc\n"); |
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if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0) |
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return -1; |
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} |
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else |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "invalid coding type\n"); |
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return -1; |
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} |
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|
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if (wasted) |
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{ |
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int i; |
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for (i = 0; i < s->blocksize; i++) |
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s->decoded[channel][i] <<= wasted; |
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} |
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|
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return 0; |
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} |
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|
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static int decode_frame(FLACContext *s) |
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{ |
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int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8; |
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int decorrelation, bps, blocksize, samplerate; |
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|
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blocksize_code = get_bits(&s->gb, 4); |
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|
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sample_rate_code = get_bits(&s->gb, 4); |
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|
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assignment = get_bits(&s->gb, 4); /* channel assignment */ |
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if (assignment < 8 && s->channels == assignment+1) |
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decorrelation = INDEPENDENT; |
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else if (assignment >=8 && assignment < 11 && s->channels == 2) |
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decorrelation = LEFT_SIDE + assignment - 8; |
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else |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels); |
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return -1; |
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} |
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|
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sample_size_code = get_bits(&s->gb, 3); |
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if(sample_size_code == 0) |
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bps= s->bps; |
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else if((sample_size_code != 3) && (sample_size_code != 7)) |
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bps = sample_size_table[sample_size_code]; |
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else |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "invalid sample size code (%d)\n", sample_size_code); |
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return -1; |
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} |
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|
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if (get_bits1(&s->gb)) |
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{ |
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av_log(s->avctx, AV_LOG_DEBUG, "broken stream, invalid padding\n"); |
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return -1; |
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} |
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|
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if(get_utf8(&s->gb) < 0){ |
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av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n"); |
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return -1; |
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} |
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#if 0 |
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if (/*((blocksize_code == 6) || (blocksize_code == 7)) &&*/ |
|
(s->min_blocksize != s->max_blocksize)){ |
|
}else{ |
|
} |
|
#endif |
|
|
|
if (blocksize_code == 0) |
|
blocksize = s->min_blocksize; |
|
else if (blocksize_code == 6) |
|
blocksize = get_bits(&s->gb, 8)+1; |
|
else if (blocksize_code == 7) |
|
blocksize = get_bits(&s->gb, 16)+1; |
|
else |
|
blocksize = blocksize_table[blocksize_code]; |
|
|
|
if(blocksize > s->max_blocksize){ |
|
av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize, s->max_blocksize); |
|
return -1; |
|
} |
|
|
|
if (sample_rate_code == 0){ |
|
samplerate= s->samplerate; |
|
}else if ((sample_rate_code > 3) && (sample_rate_code < 12)) |
|
samplerate = sample_rate_table[sample_rate_code]; |
|
else if (sample_rate_code == 12) |
|
samplerate = get_bits(&s->gb, 8) * 1000; |
|
else if (sample_rate_code == 13) |
|
samplerate = get_bits(&s->gb, 16); |
|
else if (sample_rate_code == 14) |
|
samplerate = get_bits(&s->gb, 16) * 10; |
|
else{ |
|
av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n", sample_rate_code); |
|
return -1; |
|
} |
|
|
|
skip_bits(&s->gb, 8); |
|
crc8= get_crc8(s->gb.buffer, get_bits_count(&s->gb)/8); |
|
if(crc8){ |
|
av_log(s->avctx, AV_LOG_ERROR, "header crc missmatch crc=%2X\n", crc8); |
|
return -1; |
|
} |
|
|
|
s->blocksize = blocksize; |
|
s->samplerate = samplerate; |
|
s->bps = bps; |
|
s->decorrelation= decorrelation; |
|
|
|
// dump_headers(s); |
|
|
|
/* subframes */ |
|
for (i = 0; i < s->channels; i++) |
|
{ |
|
// av_log(s->avctx, AV_LOG_DEBUG, "decoded: %x residual: %x\n", s->decoded[i], s->residual[i]); |
|
if (decode_subframe(s, i) < 0) |
|
return -1; |
|
} |
|
|
|
align_get_bits(&s->gb); |
|
|
|
/* frame footer */ |
|
skip_bits(&s->gb, 16); /* data crc */ |
|
|
|
return 0; |
|
} |
|
|
|
static int flac_decode_frame(AVCodecContext *avctx, |
|
void *data, int *data_size, |
|
uint8_t *buf, int buf_size) |
|
{ |
|
FLACContext *s = avctx->priv_data; |
|
int metadata_last, metadata_type, metadata_size; |
|
int tmp = 0, i, j = 0, input_buf_size = 0; |
|
int16_t *samples = data; |
|
|
|
if(s->max_framesize == 0){ |
|
s->max_framesize= 8192; // should hopefully be enough for the first header |
|
s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize); |
|
} |
|
|
|
if(1 && s->max_framesize){//FIXME truncated |
|
buf_size= FFMIN(buf_size, s->max_framesize - s->bitstream_size); |
|
input_buf_size= buf_size; |
|
|
|
if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){ |
|
// printf("memmove\n"); |
|
memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size); |
|
s->bitstream_index=0; |
|
} |
|
memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size); |
|
buf= &s->bitstream[s->bitstream_index]; |
|
buf_size += s->bitstream_size; |
|
s->bitstream_size= buf_size; |
|
|
|
if(buf_size < s->max_framesize){ |
|
// printf("wanna more data ...\n"); |
|
return input_buf_size; |
|
} |
|
} |
|
|
|
init_get_bits(&s->gb, buf, buf_size*8); |
|
|
|
/* fLaC signature (be) */ |
|
if (show_bits_long(&s->gb, 32) == bswap_32(ff_get_fourcc("fLaC"))) |
|
{ |
|
skip_bits(&s->gb, 32); |
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "STREAM HEADER\n"); |
|
do { |
|
metadata_last = get_bits(&s->gb, 1); |
|
metadata_type = get_bits(&s->gb, 7); |
|
metadata_size = get_bits_long(&s->gb, 24); |
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " metadata block: flag = %d, type = %d, size = %d\n", |
|
metadata_last, metadata_type, |
|
metadata_size); |
|
if(metadata_size){ |
|
switch(metadata_type) |
|
{ |
|
case METADATA_TYPE_STREAMINFO: |
|
metadata_streaminfo(s); |
|
dump_headers(s); |
|
break; |
|
default: |
|
for(i=0; i<metadata_size; i++) |
|
skip_bits(&s->gb, 8); |
|
} |
|
} |
|
} while(!metadata_last); |
|
} |
|
else |
|
{ |
|
|
|
tmp = show_bits(&s->gb, 16); |
|
if(tmp != 0xFFF8){ |
|
av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n"); |
|
while(get_bits_count(&s->gb)/8+2 < buf_size && show_bits(&s->gb, 16) != 0xFFF8) |
|
skip_bits(&s->gb, 8); |
|
goto end; // we may not have enough bits left to decode a frame, so try next time |
|
} |
|
skip_bits(&s->gb, 16); |
|
if (decode_frame(s) < 0){ |
|
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n"); |
|
s->bitstream_size=0; |
|
s->bitstream_index=0; |
|
return -1; |
|
} |
|
} |
|
|
|
|
|
#if 0 |
|
/* fix the channel order here */ |
|
if (s->order == MID_SIDE) |
|
{ |
|
short *left = samples; |
|
short *right = samples + s->blocksize; |
|
for (i = 0; i < s->blocksize; i += 2) |
|
{ |
|
uint32_t x = s->decoded[0][i]; |
|
uint32_t y = s->decoded[0][i+1]; |
|
|
|
right[i] = x - (y / 2); |
|
left[i] = right[i] + y; |
|
} |
|
*data_size = 2 * s->blocksize; |
|
} |
|
else |
|
{ |
|
for (i = 0; i < s->channels; i++) |
|
{ |
|
switch(s->order) |
|
{ |
|
case INDEPENDENT: |
|
for (j = 0; j < s->blocksize; j++) |
|
samples[(s->blocksize*i)+j] = s->decoded[i][j]; |
|
break; |
|
case LEFT_SIDE: |
|
case RIGHT_SIDE: |
|
if (i == 0) |
|
for (j = 0; j < s->blocksize; j++) |
|
samples[(s->blocksize*i)+j] = s->decoded[0][j]; |
|
else |
|
for (j = 0; j < s->blocksize; j++) |
|
samples[(s->blocksize*i)+j] = s->decoded[0][j] - s->decoded[i][j]; |
|
break; |
|
// case MID_SIDE: |
|
// av_log(s->avctx, AV_LOG_DEBUG, "mid-side unsupported\n"); |
|
} |
|
*data_size += s->blocksize; |
|
} |
|
} |
|
#else |
|
switch(s->decorrelation) |
|
{ |
|
case INDEPENDENT: |
|
for (j = 0; j < s->blocksize; j++) |
|
{ |
|
for (i = 0; i < s->channels; i++) |
|
*(samples++) = s->decoded[i][j]; |
|
} |
|
break; |
|
case LEFT_SIDE: |
|
assert(s->channels == 2); |
|
for (i = 0; i < s->blocksize; i++) |
|
{ |
|
*(samples++) = s->decoded[0][i]; |
|
*(samples++) = s->decoded[0][i] - s->decoded[1][i]; |
|
} |
|
break; |
|
case RIGHT_SIDE: |
|
assert(s->channels == 2); |
|
for (i = 0; i < s->blocksize; i++) |
|
{ |
|
*(samples++) = s->decoded[0][i] + s->decoded[1][i]; |
|
*(samples++) = s->decoded[1][i]; |
|
} |
|
break; |
|
case MID_SIDE: |
|
assert(s->channels == 2); |
|
for (i = 0; i < s->blocksize; i++) |
|
{ |
|
int mid, side; |
|
mid = s->decoded[0][i]; |
|
side = s->decoded[1][i]; |
|
|
|
#if 1 //needs to be checked but IMHO it should be binary identical |
|
mid -= side>>1; |
|
*(samples++) = mid + side; |
|
*(samples++) = mid; |
|
#else |
|
|
|
mid <<= 1; |
|
if (side & 1) |
|
mid++; |
|
*(samples++) = (mid + side) >> 1; |
|
*(samples++) = (mid - side) >> 1; |
|
#endif |
|
} |
|
break; |
|
} |
|
#endif |
|
|
|
*data_size = (int8_t *)samples - (int8_t *)data; |
|
// av_log(s->avctx, AV_LOG_DEBUG, "data size: %d\n", *data_size); |
|
|
|
// s->last_blocksize = s->blocksize; |
|
end: |
|
i= (get_bits_count(&s->gb)+7)/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 int flac_decode_close(AVCodecContext *avctx) |
|
{ |
|
FLACContext *s = avctx->priv_data; |
|
int i; |
|
|
|
for (i = 0; i < s->channels; i++) |
|
{ |
|
av_freep(&s->decoded[i]); |
|
} |
|
av_freep(&s->bitstream); |
|
|
|
return 0; |
|
} |
|
|
|
static void flac_flush(AVCodecContext *avctx){ |
|
FLACContext *s = avctx->priv_data; |
|
|
|
s->bitstream_size= |
|
s->bitstream_index= 0; |
|
} |
|
|
|
AVCodec flac_decoder = { |
|
"flac", |
|
CODEC_TYPE_AUDIO, |
|
CODEC_ID_FLAC, |
|
sizeof(FLACContext), |
|
flac_decode_init, |
|
NULL, |
|
flac_decode_close, |
|
flac_decode_frame, |
|
.flush= flac_flush, |
|
};
|
|
|